ffplay [options] [‘input_file’]
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.
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 multiplies, 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.
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.
stream_type is one of following: ’v’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data, and ’t’ for attachments. If stream_index is given, then it matches stream number stream_index of this type. Otherwise, it matches all streams of this type.
If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.
Matches the stream by a format-specific ID.
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 name filter_name. Use the ‘-filters’ option to get a list of all filters.
Show version.
Show available formats.
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.
Set the logging level used by the library. Adding "repeat+" indicates that repeated log output should not be compressed to the first line and the "Last message repeated n times" line will be omitted. "repeat" can also be used alone. If "repeat" is used alone, and with no prior loglevel set, the default loglevel will be used. If multiple loglevel parameters are given, using ’repeat’ will not change the loglevel. loglevel is a number or a string 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 and assert failure. This is not currently used for anything.
Only show fatal errors. These are errors after which the process absolutely cannot continue after.
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.
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting
the environment variable AV_LOG_FORCE_COLOR.
The use of the environment variable NO_COLOR is deprecated and
will be dropped in a following FFmpeg version.
Dump full command line and console output to a file named
program-YYYYMMDD-HHMMSS.log in the current
directory.
This file can be useful for bug reports.
It also implies -loglevel verbose.
Setting the environment variable FFREPORT to any value has the
same effect. If the value is a ’:’-separated key=value sequence, these
options will affect the report; options 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 option is recognized:
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 %
Errors in parsing the environment variable are not fatal, and will not appear in the report.
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:
Benchmark all available OpenCL devices and show the results. This option
is only available when FFmpeg has been compiled with --enable-opencl.
Set OpenCL environment options. This option is only available when
FFmpeg has been compiled with --enable-opencl.
options must be a list of key=value option pairs separated by ’:’. See the “OpenCL Options” section in the ffmpeg-utils manual for the list of supported options.
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.
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.
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.
Disable audio.
Disable video.
Seek to a given position in seconds.
play <duration> seconds of audio/video
Seek by bytes.
Disable graphical display.
Force format.
Set window title (default is the input filename).
Loops movie playback <number> times. 0 means forever.
Set the show mode to use. Available values for mode are:
show video
show audio waves
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.
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_file.
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 on by default, to
explicitly disable it you need to specify -nostats.
Work around bugs.
Non-spec-compliant optimizations.
Generate pts.
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful if you are streaming with the RTSP protocol.
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.
Set the thread count.
Select the desired audio stream number, counting from 0. The number refers to the list of all the input audio streams. If it is greater than the number of audio streams minus one, then the last one is selected, if it is negative the audio playback is disabled.
Select the desired video stream number, counting from 0. The number refers to the list of all the input video streams. If it is greater than the number of video streams minus one, then the last one is selected, if it is negative the video playback is disabled.
Select the desired subtitle stream number, counting from 0. The number refers to the list of all the input subtitle streams. If it is greater than the number of subtitle streams minus one, then the last one is selected, if it is negative the subtitle rendering is disabled.
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.
Quit.
Toggle full screen.
Pause.
Cycle audio channel in the curret program.
Cycle video channel.
Cycle subtitle channel in the current program.
Cycle program.
Show audio waves.
Seek backward/forward 10 seconds.
Seek backward/forward 1 minute.
Seek backward/forward 10 minutes.
Seek to percentage in file corresponding to fraction of width.
This section documents the syntax and formats employed by the FFmpeg libraries and tools.
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.
' itself cannot be quoted,
so you may need to close the quote and escape it.
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.
Crime d'Amour containing the ' special
character:
Crime d\'Amour |
' needs to be escaped
when quoting it:
'Crime d'\''Amour' |
' this string starts and ends with whitespaces ' |
' The string '\'string\'' is a string ' |
\ you can use either escaping or quoting:
'c:\foo' can be written as c:\\foo |
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.
There are two accepted syntaxes for expressing time duration.
[-][HH:]MM:SS[.m...] |
HH expresses the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.
or
[-]S+[.m...] |
S expresses the number of seconds, with the optional decimal part m.
In both expressions, the optional ‘-’ indicates negative duration.
The following examples are all valid time duration:
55 seconds
12 hours, 03 minutes and 45 seconds
23.189 seconds
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
704x576
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
2048x1080
1998x1080
2048x858
4096x2160
3996x2160
4096x1716
640x360
240x160
400x240
432x240
480x320
960x540
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
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.
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
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
FL+FR+LFE
FL+FR+FC
FL+FR+BC
FL+FR+FC+BC
FL+FR+BL+BR
FL+FR+SL+SR
FL+FR+FC+LFE
FL+FR+FC+BL+BR
FL+FR+FC+SL+SR
FL+FR+FC+LFE+BC
FL+FR+FC+LFE+BL+BR
FL+FR+FC+LFE+SL+SR
FL+FR+FC+BC+SL+SR
FL+FR+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC
FL+FR+FC+LFE+BC+SL+SR
FL+FR+FC+LFE+BL+BR+BC
FL+FR+LFE+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+SL+SR
FL+FR+FC+FLC+FRC+SL+SR
FL+FR+FC+LFE+BL+BR+SL+SR
FL+FR+FC+LFE+BL+BR+FLC+FRC
FL+FR+FC+LFE+FLC+FRC+SL+SR
FL+FR+FC+BL+BR+BC+SL+SR
DL+DR
A custom channel layout can be specified as a sequence of terms, separated by ’+’ or ’|’. Each term can be:
av_get_default_channel_layout)
AV_CH_* macros in ‘libavutil/channel_layout.h’.
Starting from libavutil version 53 the trailing character "c" to specify a number of channels will be required, while a channel layout mask could also be specified as a decimal number (if and only if not followed by "c").
See also the function av_get_channel_layout defined in
‘libavutil/channel_layout.h’.
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.
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".
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.
Allow to load the value of the internal variable with number var, which was previously stored with st(var, expr). The function returns the loaded value.
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 maximum 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.
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)).
Allow to 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
When FFmpeg is configured with --enable-opencl, it is possible
to set the options for the global OpenCL context.
The list of supported options follows:
Set build options used to compile the registered kernels.
See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
Select the index of the platform to run OpenCL code.
The specified index must be one of the indexes in the device list
which can be obtained with ffmpeg -opencl_bench or av_opencl_get_device_list().
Select the index of the device used to run OpenCL code.
The specifed index must be one of the indexes in the device list which
can be obtained with ffmpeg -opencl_bench or av_opencl_get_device_list().
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 unsensical 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.
Use loop filter.
Use fixed qscale.
Use gmc.
Always try a mb with mv=<0,0>.
Use internal 2pass ratecontrol in first pass mode.
Use internal 2pass ratecontrol in second pass mode.
Only decode/encode grayscale.
Do not draw edges.
Set error[?] variables during encoding.
Normalize adaptive quantization.
Use interlaced DCT.
Force low delay.
Place global headers in extradata instead of every keyframe.
Use only bitexact stuff (except (I)DCT).
Apply H263 advanced intra coding / mpeg4 ac prediction.
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead.
Apply interlaced motion estimation.
Use closed gop.
Set motion estimation method.
Possible values:
zero motion estimation (fastest)
full motion estimation (slowest)
EPZS motion estimation (default)
esa motion estimation (alias for full)
tesa motion estimation
dia motion estimation (alias for epzs)
log motion estimation
phods motion estimation
X1 motion estimation
hex motion estimation
umh motion estimation
iter motion estimation
Set extradata size.
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 size. Default value is 12.
Set audio sampling rate (in Hz).
Set number of audio channels.
Set cutoff bandwidth.
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 ratecontrol method.
Set strategy to choose between I/P/B-frames.
Set RTP payload size in bytes.
Workaround not auto detected encoder bugs.
Possible values:
some old lavc generated msmpeg4v3 files (no autodetection)
Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
illegal vlc bug (autodetected per fourcc)
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
Set single coefficient elimination threshold for luminance (negative values also consider DC coefficient).
Set single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)
Specify how strictly to follow the standards.
Possible values:
strictly conform to a 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
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
Use MPEG quantizers instead of H.263.
How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).
Set experimental quantizer modulation.
Set experimental quantizer modulation.
Set rate control equation. When computing the expression, besides the standard functions defined in the section ’Expression Evaluation’, the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
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).
Currently useless.
Set QP factor between P and I frames.
Set QP offset between P and I frames.
Set initial complexity for 1-pass encoding.
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:
floating point AAN IDCT
Set error concealment strategy.
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
Set prediction method.
Possible values:
Set sample aspect ratio.
Print specific debug info.
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
motion vector
error recognition
memory management control operations (H.264)
visualize quantization parameter (QP), lower QP are tinted greener
visualize block types
picture buffer allocations
threading operations
Visualize motion vectors (MVs).
Possible values:
forward predicted MVs of P-frames
forward predicted MVs of B-frames
backward predicted MVs of B-frames
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
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
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
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
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.
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
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).
Set intra quant bias.
Set inter quant bias.
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length coder
deflate-based 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 min lagrange factor (VBR).
Set max lagrange factor (VBR).
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.
Deprecated, use mpegvideo private options instead.
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.
Deprecated, use mpegvideo private options instead.
Deprecated, use mpegvideo private options instead.
Possible values:
detect a good number of threads
Set motion estimation threshold.
Set macroblock threshold.
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.
Possible values:
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.
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
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
Increase the quantizer for macroblocks close to borders.
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.
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.
Set value multiplied by qscale for each frame and added to scene_change_score.
Adjust sensitivity of b_frame_strategy 1.
Set GOP timecode frame start number, in non drop frame format.
Set desired number of audio channels.
Possible values:
Possible values:
Set the log level offset.
Number of slices, used in parallelized encoding.
Select multithreading type.
Possible values:
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.
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 -codecs of the ff* tools will display the list of
enabled decoders.
A description of some of the currently available video decoders follows.
Raw video decoder.
This decoder decodes rawvideo streams.
Specify the assumed field type of the input video.
the video is assumed to be progressive (default)
bottom-field-first is assumed
top-field-first is assumed
Internal wave synthetizer.
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 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 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 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.
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 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 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 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.
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.
Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when reading from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal
numbers (without 0x prefix) separated by comas, for example 0d00ee,
ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a, 7b7b7b, d1d1d1,
7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b.
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.
List of teletext page numbers to decode. You may use the special * string to match all pages. Pages that do not match the specified list are dropped. Default value is *.
Discards the top teletext line. Default value is 1.
Specifies the format of the decoded subtitles. The teletext decoder is capable of decoding the teletext pages to bitmaps or to simple text, you should use "bitmap" for teletext pages, because certain graphics and colors cannot be expressed in simple text. You might use "text" for teletext based subtitles if your application can handle simple text based subtitles. Default value is bitmap.
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 charactes. Default value is 1.
Sets the display duration of the decoded teletext pages or subtitles in miliseconds. Default value is 30000 which is 30 seconds.
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an opaque (black) background.
Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available encoders using the configure option --list-encoders.
You can disable all the encoders with the configure option
--disable-encoders and selectively enable / disable single encoders
with the options --enable-encoder=ENCODER /
--disable-encoder=ENCODER.
The option -codecs of the ff* tools will display the list of
enabled encoders.
A description of some of the currently available audio encoders follows.
Advanced Audio Coding (AAC) encoder.
This encoder is an experimental FFmpeg-native AAC encoder. Currently only the low complexity (AAC-LC) profile is supported. To use this encoder, you must set ‘strict’ option to ‘experimental’ or lower.
As this encoder is experimental, unexpected behavior may exist from time to time. For a more stable AAC encoder, see libvo-aacenc. However, be warned that it has a worse quality reported by some users.
See also libfdk_aac and libfaac.
Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode.
Set quality for variable bit rate (VBR) mode. This option is valid only using
the ffmpeg command-line tool. For library interface users, use
‘global_quality’.
Set stereo encoding mode. Possible values:
Automatically selected by the encoder.
Disable middle/side encoding. This is the default.
Force middle/side encoding.
Set AAC encoder coding method. Possible values:
FAAC-inspired method.
This method is a simplified reimplementation of the method used in FAAC, which sets thresholds proportional to the band energies, and then decreases all the thresholds with quantizer steps to find the appropriate quantization with distortion below threshold band by band.
The quality of this method is comparable to the two loop searching method descibed below, but somewhat a little better and slower.
Average noise to mask ratio (ANMR) trellis-based solution.
This has a theoretic best quality out of all the coding methods, but at the cost of the slowest speed.
Two loop searching (TLS) method.
This method first sets quantizers depending on band thresholds and then tries to find an optimal combination by adding or subtracting a specific value from all quantizers and adjusting some individual quantizer a little.
This method produces similar quality with the FAAC method and is the default.
Constant quantizer method.
This method sets a constant quantizer for all bands. This is the fastest of all the methods, yet produces the worst quality.
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The ac3_fixed encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option -acodec ac3_fixed in order to use it.
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available documents.
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
The metadata values set at initialization will be used for every frame in the stream. (default)
Metadata values can be changed before encoding each frame.
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:
Apply -3dB gain
Apply -4.5dB gain (default)
Apply -6dB gain
Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:
Apply -3dB gain
Apply -6dB gain (default)
Silence Surround Channel(s)
Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the room_type option is not the default value, the mixing_level
option must not be -1.
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the mixing_level
option and the room_type option have the default values.
Not Indicated (default)
Large Room
Small Room
Copyright Indicator. Specifies whether a copyright exists for this audio.
No Copyright Exists (default)
Copyright Exists
Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital 100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of -31dB will result in no volume level change, relative to the source volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31 being the default.
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean the encoder will actually apply Dolby Surround processing.
Not Indicated (default)
Not Dolby Surround Encoded
Dolby Surround Encoded
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
Not Original Source
Original Source (default)
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the center_mixlev
and surround_mixlev options if it supports the Alternate Bit Stream
Syntax.
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
Not Indicated (default)
Lt/Rt Downmix Preferred
Lo/Ro Downmix Preferred
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lt/Rt mode.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.
Not Indicated (default)
Dolby Surround EX Off
Dolby Surround EX On
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually apply Dolby Headphone processing.
Not Indicated (default)
Dolby Headphone Off
Dolby Headphone On
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
Standard A/D Converter (default)
HDCD A/D Converter
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3 feature that increases quality by selectively encoding the left/right channels as mid/side. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes.
These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point.
Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.
Selected by Encoder (default)
Disable Channel Coupling
Enable Channel Coupling
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled.
Selected by Encoder (default)
libfaac AAC (Advanced Audio Coding) encoder wrapper.
Requires the presence of the libfaac headers and library during
configuration. You need to explicitly configure the build with
--enable-libfaac --enable-nonfree.
This encoder is considered to be of higher quality with respect to the the native experimental FFmpeg AAC encoder.
For more information see the libfaac project at http://www.audiocoding.com/faac.html/.
The following shared FFmpeg codec options are recognized.
The following options are supported by the libfaac wrapper. The
faac-equivalent of the options are listed in parentheses.
Set bit rate in bits/s for ABR (Average Bit Rate) mode. If the bit rate
is not explicitly specified, it is automatically set to a suitable
value depending on the selected profile. faac bitrate is
expressed in kilobits/s.
Note that libfaac does not support CBR (Constant Bit Rate) but only ABR (Average Bit Rate).
If VBR mode is enabled this option is ignored.
Set audio sampling rate (in Hz).
Set the number of audio channels.
Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically computed by the library. Default value is 0.
Set audio profile.
The following profiles are recognized:
Main AAC (Main)
Low Complexity AAC (LC)
Scalable Sample Rate (SSR)
Long Term Prediction (LTP)
If not specified it is set to ‘aac_low’.
Set constant quality VBR (Variable Bit Rate) mode.
Set quality in VBR mode as an integer number of lambda units.
Only relevant when VBR mode is enabled with flags +qscale. The
value is converted to QP units by dividing it by FF_QP2LAMBDA,
and used to set the quality value used by libfaac. A reasonable range
for the option value in QP units is [10-500], the higher the value the
higher the quality.
Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating point value in QP units.
The value sets the quality value used by libfaac. A reasonable range for the option value is [10-500], the higher the value the higher the quality.
This option is valid only using the ffmpeg command-line
tool. For library interface users, use ‘global_quality’.
ffmpeg to convert an audio file to ABR 128 kbps AAC in an M4A (MP4)
container:
ffmpeg -i input.wav -codec:a libfaac -b:a 128k -output.m4a |
ffmpeg to convert an audio file to VBR AAC, using the
LTP AAC profile:
ffmpeg -i input.wav -c:a libfaac -profile:a aac_ltp -q:a 100 output.m4a |
libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project.
Requires the presence of the libfdk-aac headers and library during
configuration. You need to explicitly configure the build with
--enable-libfdk-aac. The library is also incompatible with GPL,
so if you allow the use of GPL, you should configure with
--enable-gpl --enable-nonfree --enable-libfdk-aac.
This encoder is considered to be of higher quality with respect to both the native experimental FFmpeg AAC encoder and libfaac.
VBR encoding, enabled through the ‘vbr’ or ‘flags +qscale’ options, is experimental and only works with some combinations of parameters.
For more information see the fdk-aac project at http://sourceforge.net/p/opencore-amr/fdk-aac/.
The following options are mapped on the shared FFmpeg codec options.
Set bit rate in bits/s. If the bitrate is not explicitly specified, it is automatically set to a suitable value depending on the selected profile.
In case VBR mode is enabled the option is ignored.
Set audio sampling rate (in Hz).
Set the number of audio channels.
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the ‘vbr’ value is positive.
Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically computed by the library. Default value is 0.
Set audio profile.
The following profiles are recognized:
Low Complexity AAC (LC)
High Efficiency AAC (HE-AAC)
High Efficiency AAC version 2 (HE-AACv2)
Low Delay AAC (LD)
Enhanced Low Delay AAC (ELD)
If not specified it is set to ‘aac_low’.
The following are private options of the libfdk_aac encoder.
Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the required processing power.
Default value is 1.
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.
Default value is 0.
Set SBR/PS signaling style.
It can assume one of the following values:
choose signaling implicitly (explicit hierarchical by default, implicit if global header is disabled)
implicit backwards compatible signaling
explicit SBR, implicit PS signaling
explicit hierarchical signaling
Default value is ‘default’.
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
Default value is 0.
Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band configuration buffers within LATM/LOAS transport layer.
Must be a 16-bits non-negative integer.
Default value is 0.
Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is highest quality. A value of 0 will disable VBR, and CBR (Constant Bit Rate) is enabled.
Currently only the ‘aac_low’ profile supports VBR encoding.
VBR modes 1-5 correspond to roughly the following average bit rates:
32 kbps/channel
40 kbps/channel
48-56 kbps/channel
64 kbps/channel
about 80-96 kbps/channel
Default value is 0.
ffmpeg to convert an audio file to VBR AAC in an M4A (MP4)
container:
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a |
ffmpeg to convert an audio file to CBR 64k kbps AAC, using the
High-Efficiency AAC profile:
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a |
LAME (Lame Ain’t an MP3 Encoder) MP3 encoder wrapper.
Requires the presence of the libmp3lame headers and library during
configuration. You need to explicitly configure the build with
--enable-libmp3lame.
See libshine for a fixed-point MP3 encoder, although with a lower quality.
The following options are supported by the libmp3lame wrapper. The
lame-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR or ABR. LAME bitrate is
expressed in kilobits/s.
Set constant quality setting for VBR. This option is valid only
using the ffmpeg command-line tool. For library interface
users, use ‘global_quality’.
Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality but slowest, and 9 meaning fastest while producing the worst quality.
Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but can be overriden by use ‘--nores’ option.
Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default value is 1.
Enable the encoder to use ABR when set to 1. The lame
‘--abr’ sets the target bitrate, while this options only
tells FFmpeg to use ABR still relies on ‘b’ to set bitrate.
OpenCORE Adaptive Multi-Rate Narrowband encoder.
Requires the presence of the libopencore-amrnb headers and library during
configuration. You need to explicitly configure the build with
--enable-libopencore-amrnb --enable-version3.
This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting ‘strict’ to ‘unofficial’ or lower.
Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
Shine Fixed-Point MP3 encoder wrapper.
Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g. armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it is not on par with LAME and other production-grade encoders quality-wise. Also, according to the project’s homepage, this encoder may not be free of bugs as the code was written a long time ago and the project was dead for at least 5 years.
This encoder only supports stereo and mono input. This is also CBR-only.
The original project (last updated in early 2007) is at http://sourceforge.net/projects/libshine-fxp/. We only support the updated fork by the Savonet/Liquidsoap project at https://github.com/savonet/shine.
Requires the presence of the libshine headers and library during
configuration. You need to explicitly configure the build with
--enable-libshine.
See also libmp3lame.
The following options are supported by the libshine wrapper. The
shineenc-equivalent of the options are listed in parentheses.
Set bitrate expressed in bits/s for CBR. shineenc ‘-b’ option
is expressed in kilobits/s.
TwoLAME MP2 encoder wrapper.
Requires the presence of the libtwolame headers and library during
configuration. You need to explicitly configure the build with
--enable-libtwolame.
The following options are supported by the libtwolame wrapper. The
twolame-equivalent options follow the FFmpeg ones and are in
parentheses.
Set bitrate expressed in bits/s for CBR. twolame ‘b’
option is expressed in kilobits/s. Default value is 128k.
Set quality for experimental VBR support. Maximum value range is
from -50 to 50, useful range is from -10 to 10. The higher the
value, the better the quality. This option is valid only using the
ffmpeg command-line tool. For library interface users,
use ‘global_quality’.
Set the mode of the resulting audio. Possible values:
Choose mode automatically based on the input. This is the default.
Stereo
Joint stereo
Dual channel
Mono
Set psychoacoustic model to use in encoding. The argument must be an integer between -1 and 4, inclusive. The higher the value, the better the quality. The default value is 3.
Enable energy levels extensions when set to 1. The default value is 0 (disabled).
Enable CRC error protection when set to 1. The default value is 0 (disabled).
Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).
Set MPEG audio original flag when set to 1. The default value is 0 (disabled).
VisualOn AAC encoder.
Requires the presence of the libvo-aacenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-aacenc --enable-version3.
This encoder is considered to be worse than the native experimental FFmpeg AAC encoder, according to multiple sources.
The VisualOn AAC encoder only support encoding AAC-LC and up to 2 channels. It is also CBR-only.
Set bit rate in bits/s.
VisualOn Adaptive Multi-Rate Wideband encoder.
Requires the presence of the libvo-amrwbenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-amrwbenc --enable-version3.
This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting ‘strict’ to ‘unofficial’ or lower.
Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during
configuration. You need to explicitly configure the build with
--enable-libopus.
Most libopus options are modeled after the opusenc utility from
opus-tools. The following is an option mapping chart describing options
supported by the libopus wrapper, and their opusenc-equivalent
in parentheses.
Set the bit rate in bits/s. FFmpeg’s ‘b’ option is
expressed in bits/s, while opusenc’s ‘bitrate’ in
kilobits/s.
Set VBR mode. The FFmpeg ‘vbr’ option has the following
valid arguments, with the their opusenc equivalent options
in parentheses:
Use constant bit rate encoding.
Use variable bit rate encoding (the default).
Use constrained variable bit rate encoding.
Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is 10.
Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is 20ms.
Set expected packet loss percentage. The default is 0.
Set intended application type. Valid options are listed below:
Favor improved speech intelligibility.
Favor faithfulness to the input (the default).
Restrict to only the lowest delay modes.
Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000, 12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband respectively. The default is 0 (cutoff disabled).
libvorbis encoder wrapper.
Requires the presence of the libvorbisenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvorbis.
The following options are supported by the libvorbis wrapper. The
oggenc-equivalent of the options are listed in parentheses.
To get a more accurate and extensive documentation of the libvorbis
options, consult the libvorbisenc’s and oggenc’s documentations.
See http://xiph.org/vorbis/,
http://wiki.xiph.org/Vorbis-tools, and oggenc(1).
Set bitrate expressed in bits/s for ABR. oggenc ‘-b’ is
expressed in kilobits/s.
Set constant quality setting for VBR. The value should be a float number in the range of -1.0 to 10.0. The higher the value, the better the quality. The default value is ‘3.0’.
This option is valid only using the ffmpeg command-line tool.
For library interface users, use ‘global_quality’.
Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc’s
related option is expressed in kHz. The default value is ‘0’ (cutoff
disabled).
Set minimum bitrate expressed in bits/s. oggenc ‘-m’ is
expressed in kilobits/s.
Set maximum bitrate expressed in bits/s. oggenc ‘-M’ is
expressed in kilobits/s. This only has effect on ABR mode.
Set noise floor bias for impulse blocks. The value is a float number from -15.0 to 0.0. A negative bias instructs the encoder to pay special attention to the crispness of transients in the encoded audio. The tradeoff for better transient response is a higher bitrate.
A wrapper providing WavPack encoding through libwavpack.
Only lossless mode using 32-bit integer samples is supported currently. The ‘compression_level’ option can be used to control speed vs. compression tradeoff, with the values mapped to libwavpack as follows:
Fast mode - corresponding to the wavpack ‘-f’ option.
Normal (default) settings.
High quality - corresponding to the wavpack ‘-h’ option.
Very high quality - corresponding to the wavpack ‘-hh’ option.
Same as 3, but with extra processing enabled - corresponding to the wavpack ‘-x’ option. I.e. 4 is the same as ‘-x2’ and 8 is the same as ‘-x6’.
A description of some of the currently available video encoders follows.
libtheora Theora encoder wrapper.
Requires the presence of the libtheora headers and library during
configuration. You need to explicitly configure the build with
--enable-libtheora.
For more informations about the libtheora project see http://www.theora.org/.
The following global options are mapped to internal libtheora options which affect the quality and the bitrate of the encoded stream.
Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate) mode is enabled this option is ignored.
Used to enable constant quality mode (VBR) encoding through the
‘qscale’ flag, and to enable the pass1 and pass2
modes.
Set the GOP size.
Set the global quality as an integer in lambda units.
Only relevant when VBR mode is enabled with flags +qscale. The
value is converted to QP units by dividing it by FF_QP2LAMBDA,
clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value corresponds
to a higher quality.
Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating point value in QP units.
The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native libtheora range [0-63].
This option is valid only using the ffmpeg command-line
tool. For library interface users, use ‘global_quality’.
ffmpeg:
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg |
ffmpeg to convert a CBR 1000 kbps Theora video stream:
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg |
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with --enable-libvpx.
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
g_threads
g_profile
rc_target_bitrate
kf_max_dist
kf_min_dist
rc_min_quantizer
rc_max_quantizer
rc_buf_sz
(bufsize * 1000 / vb)
rc_buf_optimal_sz
(bufsize * 1000 / vb * 5 / 6)
rc_buf_initial_sz
(rc_init_occupancy * 1000 / vb)
rc_undershoot_pct
rc_dropframe_thresh
rc_2pass_vbr_bias_pct
rc_2pass_vbr_maxsection_pct
(maxrate * 100 / vb)
rc_2pass_vbr_minsection_pct
(minrate * 100 / vb)
VPX_CBR
(minrate == maxrate == vb)
VPX_CQ, VP8E_SET_CQ_LEVEL
VPX_DL_BEST_QUALITY
VPX_DL_GOOD_QUALITY
VPX_DL_REALTIME
VP8E_SET_CPUUSED
VP8E_SET_NOISE_SENSITIVITY
VP8E_SET_STATIC_THRESHOLD
VP8E_SET_TOKEN_PARTITIONS
VP8E_SET_MAX_INTRA_BITRATE_PCT
VPX_EFLAG_FORCE_KF
VP8E_SET_ENABLEAUTOALTREF
VP8E_SET_ARNR_MAXFRAMES
VP8E_SET_ARNR_TYPE
VP8E_SET_ARNR_STRENGTH
g_lag_in_frames
g_error_resilient
For more information about libvpx see: http://www.webmproject.org/
x264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libx264 headers and library
during configuration. You need to explicitly configure the build with
--enable-libx264.
libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy optimizations for detail retention (adaptive quantization, psy-RD, psy-trellis).
Many libx264 encoder options are mapped to FFmpeg global codec
options, while unique encoder options are provided through private
options. Additionally the ‘x264opts’ and ‘x264-params’
private options allows to pass a list of key=value tuples as accepted
by the libx264 x264_param_parse function.
The x264 project website is at http://www.videolan.org/developers/x264.html.
The following options are supported by the libx264 wrapper. The
x264-equivalent options or values are listed in parentheses
for easy migration.
To reduce the duplication of documentation, only the private options and some others requiring special attention are documented here. For the documentation of the undocumented generic options, see the Codec Options chapter.
To get a more accurate and extensive documentation of the libx264
options, invoke the command x264 --full-help or consult
the libx264 documentation.
Set bitrate in bits/s. Note that FFmpeg’s ‘b’ option is
expressed in bits/s, while x264’s ‘bitrate’ is in
kilobits/s.
Set motion estimation method. Possible values in the decreasing order of speed:
Diamond search with radius 1 (fastest). ‘epzs’ is an alias for ‘dia’.
Hexagonal search with radius 2.
Uneven multi-hexagon search.
Exhaustive search.
Hadamard exhaustive search (slowest).
Set entropy encoder. Possible values:
Enable CABAC.
Enable CAVLC and disable CABAC. It generates the same effect as
x264’s ‘--no-cabac’ option.
Set full pixel motion estimation comparation algorithm. Possible values:
Enable chroma in motion estimation.
Ignore chroma in motion estimation. It generates the same effect as
x264’s ‘--no-chroma-me’ option.
Set multithreading technique. Possible values:
Slice-based multithreading. It generates the same effect as
x264’s ‘--sliced-threads’ option.
Frame-based multithreading.
Set encoding flags. It can be used to disable closed GOP and enable
open GOP by setting it to -cgop. The result is similar to
the behavior of x264’s ‘--open-gop’ option.
Set the encoding preset.
Set tuning of the encoding params.
Set profile restrictions.
Enable fast settings when encoding first pass, when set to 1. When set
to 0, it has the same effect of x264’s
‘--slow-firstpass’ option.
Set the quality for constant quality mode.
In CRF mode, prevents VBV from lowering quality beyond this point.
Set constant quantization rate control method parameter.
Set AQ method. Possible values:
Disabled.
Variance AQ (complexity mask).
Auto-variance AQ (experimental).
Set AQ strength, reduce blocking and blurring in flat and textured areas.
Use psychovisual optimizations when set to 1. When set to 0, it has the
same effect as x264’s ‘--no-psy’ option.
Set strength of psychovisual optimization, in psy-rd:psy-trellis format.
Set number of frames to look ahead for frametype and ratecontrol.
Enable weighted prediction for B-frames when set to 1. When set to 0,
it has the same effect as x264’s ‘--no-weightb’ option.
Set weighted prediction method for P-frames. Possible values:
Disabled
Enable only weighted refs
Enable both weighted refs and duplicates
Enable calculation and printing SSIM stats after the encoding.
Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.
Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting "bluray-compat=1 force-cfr=1".
Set the influence on how often B-frames are used.
Set method for keeping of some B-frames as references. Possible values:
Disabled.
Strictly hierarchical pyramid.
Non-strict (not Blu-ray compatible).
Enable the use of one reference per partition, as opposed to one
reference per macroblock when set to 1. When set to 0, it has the
same effect as x264’s ‘--no-mixed-refs’ option.
Enable adaptive spatial transform (high profile 8x8 transform)
when set to 1. When set to 0, it has the same effect as
x264’s ‘--no-8x8dct’ option.
Enable early SKIP detection on P-frames when set to 1. When set
to 0, it has the same effect as x264’s
‘--no-fast-pskip’ option.
Enable use of access unit delimiters when set to 1.
Enable use macroblock tree ratecontrol when set to 1. When set
to 0, it has the same effect as x264’s
‘--no-mbtree’ option.
Set loop filter parameters, in alpha:beta form.
Set fluctuations reduction in QP (before curve compression).
Set partitions to consider as a comma-separated list of. Possible values in the list:
8x8 P-frame partition.
4x4 P-frame partition.
4x4 B-frame partition.
8x8 I-frame partition.
4x4 I-frame partition. (Enabling ‘p4x4’ requires ‘p8x8’ to be enabled. Enabling ‘i8x8’ requires adaptive spatial transform (‘8x8dct’ option) to be enabled.)
Do not consider any partitions.
Consider every partition.
Set direct MV prediction mode. Possible values:
Disable MV prediction.
Enable spatial predicting.
Enable temporal predicting.
Automatically decided.
Set the limit of the size of each slice in bytes. If not specified but RTP payload size (‘ps’) is specified, that is used.
Set the file name for multi-pass stats.
Set signal HRD information (requires ‘vbv-bufsize’ to be set). Possible values:
Disable HRD information signaling.
Variable bit rate.
Constant bit rate (not allowed in MP4 container).
Set any x264 option, see x264 --fullhelp for a list.
Argument is a list of key=value couples separated by ":". In filter and psy-rd options that use ":" as a separator themselves, use "," instead. They accept it as well since long ago but this is kept undocumented for some reason.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv |
Override the x264 configuration using a :-separated list of key=value parameters.
This option is functionally the same as the ‘x264opts’, but is duplicated for compability with the Libav fork.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\ cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\ no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT |
Encoding ffpresets for common usages are provided so they can be used with the general presets system (e.g. passing the ‘pre’ option).
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and library
during configuration. You need to explicitly configure the build with
--enable-libxvid --enable-gpl.
The native mpeg4 encoder supports the MPEG-4 Part 2 format, so
users can encode to this format without this library.
The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. See the Codec Options chapter for their documentation. The other shared options which are not listed have no effect for the libxvid encoder.
Set specific encoding flags. Possible values:
Use four motion vector by macroblock.
Enable high quality AC prediction.
Only encode grayscale.
Enable the use of global motion compensation (GMC).
Enable quarter-pixel motion compensation.
Enable closed GOP.
Place global headers in extradata instead of every keyframe.
Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:
Use no motion estimation (default).
Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. ‘x1’ and ‘log’ are aliases for ‘phods’.
Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma planes.
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.
Set macroblock decision algorithm. Possible values in the increasing order of quality:
Use macroblock comparing function algorithm (default).
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern.
Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).
Enable variance adaptive quantization when set to 1. Default is 0 (disabled).
When combined with ‘lumi_aq’, the resulting quality will not be better than any of the two specified individually. In other words, the resulting quality will be the worse one of the two effects.
Set structural similarity (SSIM) displaying method. Possible values:
Disable displaying of SSIM information.
Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is:
Average SSIM: %f |
For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232).
Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to stdout. The format of per-frame information is:
SSIM: avg: %1.3f min: %1.3f max: %1.3f |
For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the dot (e.g. 0.932).
Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most accurate result and 4 computes the fastest.
PNG image encoder.
Set physical density of pixels, in dots per inch, unset by default
Set physical density of pixels, in dots per meter, unset by default
Apple ProRes encoder.
FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
The used encoder can be choosen with the -vcodec option.
Select the ProRes profile to encode
Select quantization matrix.
If set to auto, the matrix matching the profile will be picked. If not set, the matrix providing the highest quality, default, will be picked.
How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits per macroblock, the maximum is 8000.
Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all situations.
Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by the Apple encoder.
Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha plane coding.
In the default mode of operation the encoder has to honor frame constraints (i.e. not produc frames with size bigger than requested) while still making output picture as good as possible. A frame containing a lot of small details is harder to compress and the encoder would spend more time searching for appropriate quantizers for each slice.
Setting a higher ‘bits_per_mb’ limit will improve the speed.
For the fastest encoding speed set the ‘qscale’ parameter (4 is the recommended value) and do not set a size constraint.
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.
Below is a description of the currently available bitstream filters.
Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration bitstream filter.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS header.
This is required for example when copying an AAC stream from a raw ADTS AAC container to a FLV or a MOV/MP4 file.
Remove zero padding at the end of a packet.
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered. It accepts the values:
add extradata to all key packets, but only if local_header is set in the ‘flags2’ codec context field
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 |
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 ("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 |
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 |
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 allow to detect 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 packet size.
Set format flags.
Possible values:
Ignore index.
Generate PTS.
Do not fill in missing values that can be exactly calculated.
Disable AVParsers, this needs +nofillin too.
Ignore DTS.
Discard corrupted frames.
Try to interleave output packets by DTS.
Do not merge side data.
Enable RTP MP4A-LATM payload.
Reduce the latency introduced by optional buffering
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 allow to detect 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.
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.
Use wallclock as timestamps.
Shift timestamps to make them non-negative. A value of 1 enables shifting, a value of 0 disables it, the default value of -1 enables shifting when required by the target format.
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.
Also note that this affects only leading negative timestamps, and not non-monotonic negative timestamps.
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 1 enables it, and has the effect of reducing the latency; 0 disables it and may slightly increase performance in some cases.
Format stream specifiers allow selection of one or more streams that match specific properties.
Possible forms of stream specifiers are:
Matches the stream with this index.
stream_type is one of following: ’v’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data, and ’t’ for attachments. If stream_index is given, then it matches the stream number stream_index of this type. Otherwise, it matches all streams of this type.
If stream_index is given, then it matches the stream with number stream_index in the program with the id program_id. Otherwise, it matches all streams in the program.
Matches the stream by a format-specific ID.
The exact semantics of stream specifiers is defined by the
avformat_match_stream_specifier() function declared in the
‘libavformat/avformat.h’ header.
Demuxers 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 -formats of the ff* tools will display the list of
enabled demuxers.
The description of some of the currently available demuxers follows.
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".
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.
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 packet 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.
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 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 to its default -1.
To make FFmpeg recognize the format automatically, this directive must appears 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.
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, it is equivalent to 1 if the format was automatically probed and 0 otherwise.
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams.
Allocate the streams according to the onMetaData array content.
The Game Music Emu library is a collection of video game music file emulators.
See http://code.google.com/p/game-music-emu/ for more information.
Some files have multiple tracks. The demuxer will pick the first track by default. The ‘track_index’ option can be used to select a different track. Track indexes start at 0. The demuxer exports the number of tracks as tracks meta data entry.
For very large files, the ‘max_size’ option may have to be adjusted.
Play media from Internet services using the quvi project.
The demuxer accepts a ‘format’ option to request a specific quality. It is by default set to best.
See http://quvi.sourceforge.net/ for more information.
FFmpeg needs to be built with --enable-libquvi for this demuxer to be
enabled.
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:
Set the frame rate for the video stream. It defaults to 25.
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.
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.
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.
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 |
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv |
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv |
MPEG-2 transport stream demuxer.
Overrides 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.
Raw video demuxer.
This demuxer allows 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:
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 |
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.
JSON captions used for TED Talks.
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 |
Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option --list-muxers.
You can disable all the muxers with the configure option
--disable-muxers and selectively enable / disable single muxers
with the options --enable-muxer=MUXER /
--disable-muxer=MUXER.
The option -formats of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
Audio Interchange File Format muxer.
It accepts the following options:
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file ‘out.crc’:
ffmpeg -i INPUT -f crc out.crc |
You can print the CRC to stdout with the command:
ffmpeg -i INPUT -f crc - |
You can select the output format of each frame with ffmpeg by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - |
See also the framecrc muxer.
Per-packet CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a line for each audio and video packet of the form:
stream_index, packet_dts, packet_pts, packet_duration, packet_size, 0xCRC |
CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.
For example to compute the CRC of the audio and video frames in ‘INPUT’, converted to raw audio and video packets, and store it in the file ‘out.crc’:
ffmpeg -i INPUT -f framecrc out.crc |
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framecrc - |
With ffmpeg, you can select the output format to which the
audio and video frames are encoded before computing the CRC for each
packet by specifying the audio and video codec. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - |
See also the crc muxer.
Per-packet MD5 testing format.
This muxer computes and prints the MD5 hash for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash.
The output of the muxer consists of a line for each audio and video packet of the form:
stream_index, packet_dts, packet_pts, packet_duration, packet_size, MD5 |
MD5 is a hexadecimal number representing the computed MD5 hash for the packet.
For example to compute the MD5 of the audio and video frames in ‘INPUT’, converted to raw audio and video packets, and store it in the file ‘out.md5’:
ffmpeg -i INPUT -f framemd5 out.md5 |
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framemd5 - |
See also the md5 muxer.
Animated GIF muxer.
It accepts the following options:
Set the number of times to loop the output. Use -1 for no loop, 0
for looping indefinitely (default).
Force the delay (expressed in centiseconds) after the last frame. Each frame
ends with a delay until the next frame. The default is -1, which is a
special value to tell the muxer to re-use the previous delay. In case of a
loop, you might want to customize this value to mark a pause for instance.
For example, to encode a gif looping 10 times, with a 5 seconds delay between the loops:
ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif |
Note 1: if you wish to extract the frames in separate GIF files, you need to force the image2 muxer:
ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif" |
Note 2: the GIF format has a very small time base: the delay between two frames can not be smaller than one centi second.
Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming specification.
It creates a playlist file and numbered segment files. The output filename specifies the playlist filename; the segment filenames receive the same basename as the playlist, a sequential number and a .ts extension.
ffmpeg -i in.nut out.m3u8 |
Set the segment length in seconds.
Set the maximum number of playlist entries.
Set the number after which index wraps.
Start the sequence from number.
ICO file muxer.
Microsoft’s icon file format (ICO) has some strict limitations that should be noted:
BMP Bit Depth FFmpeg Pixel Format 1bit pal8 4bit pal8 8bit pal8 16bit rgb555le 24bit bgr24 32bit bgra |
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character ’%’ can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.
The pattern may contain a suffix which is used to automatically determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ..., ‘img-010.bmp’, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form ‘img%-1.jpg’, ‘img%-2.jpg’, ..., ‘img%-10.jpg’, etc.
The following example shows how to use ffmpeg for creating a
sequence of files ‘img-001.jpeg’, ‘img-002.jpeg’, ...,
taking one image every second from the input video:
ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' |
Note that with ffmpeg, if the format is not specified with the
-f option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' |
Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to create a single image file ‘img.jpeg’ from the input video you can employ the command:
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg |
Start the sequence from number. Default value is 1. Must be a non-negative number.
If number is nonzero, the filename will always be interpreted as just a filename, not a pattern, and this file will be continuously overwritten with new images.
The image muxer supports the .Y.U.V image file format. This format is special in that that each image frame consists of three files, for each of the YUV420P components. To read or write this image file format, specify the name of the ’.Y’ file. The muxer will automatically open the ’.U’ and ’.V’ files as required.
Matroska container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
Name provided to a single track
Specifies the language of the track in the Matroska languages form
Stereo 3D video layout of two views in a single video track
video is not stereo
Both views are arranged side by side, Left-eye view is on the left
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
Each view is constituted by a row based interleaving, Right-eye view is first row
Each view is constituted by a row based interleaving, Left-eye view is first row
Both views are arranged in a column based interleaving manner, Right-eye view is first column
Both views are arranged in a column based interleaving manner, Left-eye view is first column
All frames are in anaglyph format viewable through red-cyan filters
Both views are arranged side by side, Right-eye view is on the left
All frames are in anaglyph format viewable through green-magenta filters
Both eyes laced in one Block, Left-eye view is first
Both eyes laced in one Block, Right-eye view is first
For example a 3D WebM clip can be created using the following command line:
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm |
This muxer supports the following options:
By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases – e.g. streaming where seeking is possible but slow – it is useful to put the index at the beginning of the file.
If this option is set to a non-zero value, the muxer will reserve a given amount of space in the file header and then try to write the cues there when the muxing finishes. If the available space does not suffice, muxing will fail. A safe size for most use cases should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this option will have no effect if it is not.
MD5 testing format.
This muxer computes and prints the MD5 hash of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash.
The output of the muxer consists of a single line of the form: MD5=MD5, where MD5 is a hexadecimal number representing the computed MD5 hash.
For example to compute the MD5 hash of the input converted to raw audio and video, and store it in the file ‘out.md5’:
ffmpeg -i INPUT -f md5 out.md5 |
You can print the MD5 to stdout with the command:
ffmpeg -i INPUT -f md5 - |
See also the framemd5 muxer.
MOV/MP4/ISMV (Smooth Streaming) muxer.
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
file has all the metadata about all packets stored in one location
(written at the end of the file, it can be moved to the start for
better playback by adding faststart to the movflags, or
using the qt-faststart tool). A fragmented
file consists of a number of fragments, where packets and metadata
about these packets are stored together. Writing a fragmented
file has the advantage that the file is decodable even if the
writing is interrupted (while a normal MOV/MP4 is undecodable if
it is not properly finished), and it requires less memory when writing
very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.
Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments:
Reserves space for the moov atom at the beginning of the file instead of placing the moov atom at the end. If the space reserved is insufficient, muxing will fail.
Start a new fragment at each video keyframe.
Create fragments that are duration microseconds long.
Create fragments that contain up to size bytes of payload data.
Allow the caller to manually choose when to cut fragments, by
calling av_write_frame(ctx, NULL) to write a fragment with
the packets written so far. (This is only useful with other
applications integrating libavformat, not from ffmpeg.)
Don’t create fragments that are shorter than duration microseconds long.
If more than one condition is specified, fragments are cut when
one of the specified conditions is fulfilled. The exception to this is
-min_frag_duration, which has to be fulfilled for any of the other
conditions to apply.
Additionally, the way the output file is written can be adjusted through a few other options:
Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration.
Files written with this option set do not work in QuickTime. This option is implicitly set when writing ismv (Smooth Streaming) files.
Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default.
Add RTP hinting tracks to the output file.
Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example:
ffmpeg -re <normal input/transcoding options> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) |
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
id3v2_version option controls which one is used. The legacy ID3v1 tag is
not written by default, but may be enabled with the write_id3v1 option.
For seekable output the muxer also writes a Xing frame at the beginning, which contains the number of frames in the file. It is useful for computing duration of VBR files.
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures are supplied to the muxer in form of a video stream with a single packet. There can be any number of those streams, each will correspond to a single APIC frame. The stream metadata tags title and comment map to APIC description and picture type respectively. See http://id3.org/id3v2.4.0-frames for allowed picture types.
Note that the APIC frames must be written at the beginning, so the muxer will buffer the audio frames until it gets all the pictures. It is therefore advised to provide the pictures as soon as possible to avoid excessive buffering.
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 |
To attach a picture to an mp3 file select both the audio and the picture stream
with map:
ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1 -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 |
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
Set the original_network_id (default 0x0001). This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path Original_Network_ID, Transport_Stream_ID.
Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB.
Set the service_id (default 0x0001) also known as program in DVB.
Set the first PID for PMT (default 0x1000, max 0x1f00).
Set the first PID for data packets (default 0x0100, max 0x0f00).
Enable m2ts mode if set to 1. Default value is -1 which disables m2ts mode.
Set muxrate.
Set minimum PES packet payload in bytes.
Set flags (see below).
Preserve original timestamps, if value is set to 1. Default value is -1, which results in shifting timestamps so that they start from 0.
Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively). This option allows updating stream structure so that standard consumer may detect the change. To do so, reopen output AVFormatContext (in case of API usage) or restart ffmpeg instance, cyclically changing tables_version value:
ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 ... ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111 ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111 ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111 ... |
Option mpegts_flags may take a set of such flags:
Reemit PAT/PMT before writing the next packet.
Use LATM packetization for AAC.
The recognized metadata settings in mpegts muxer are service_provider
and service_name. If they are not set the default for
service_provider is "FFmpeg" and the default for
service_name is "Service01".
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
|
Null muxer.
This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes.
For example to benchmark decoding with ffmpeg you can use the
command:
ffmpeg -benchmark -i INPUT -f null out.null |
Note that the above command does not read or write the ‘out.null’
file, but specifying the output file is required by the ffmpeg
syntax.
Alternatively you can write the command as:
ffmpeg -benchmark -i INPUT -f null - |
Ogg container muxer.
Preferred page duration, in microseconds. The muxer will attempt to create pages that are approximately duration microseconds long. This allows the user to compromise between seek granularity and container overhead. The default is 1 second. A value of 0 will fill all segments, making pages as large as possible. A value of 1 will effectively use 1 packet-per-page in most situations, giving a small seek granularity at the cost of additional container overhead.
Basic stream segmenter.
The segmenter muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to image2.
stream_segment is a variant of the muxer used to write to
streaming output formats, i.e. which do not require global headers,
and is recommended for outputting e.g. to MPEG transport stream segments.
ssegment is a shorter alias for stream_segment.
Every segment starts with a keyframe of the selected reference stream, which is set through the ‘reference_stream’ option.
Note that if you want accurate splitting for a video file, you need to make the input key frames correspond to the exact splitting times expected by the segmenter, or the segment muxer will start the new segment with the key frame found next after the specified start time.
The segment muxer works best with a single constant frame rate video.
Optionally it can generate a list of the created segments, by setting the option segment_list. The list type is specified by the segment_list_type option. The entry filenames in the segment list are set by default to the basename of the corresponding segment files.
The segment muxer supports the following options:
Set the reference stream, as specified by the string specifier.
If specifier is set to auto, the reference is choosen
automatically. Otherwise it must be a stream specifier (see the “Stream
specifiers” chapter in the ffmpeg manual) which specifies the
reference stream. The default value is auto.
Override the inner container format, by default it is guessed by the filename extension.
Generate also a listfile named name. If not specified no listfile is generated.
Set flags affecting the segment list generation.
It currently supports the following flags:
Allow caching (only affects M3U8 list files).
Allow live-friendly file generation.
Update the list file so that it contains at most the last size segments. If 0 the list file will contain all the segments. Default value is 0.
Set prefix to prepend to the name of each entry filename. By default no prefix is applied.
Specify the format for the segment list file.
The following values are recognized:
Generate a flat list for the created segments, one segment per line.
Generate a list for the created segments, one segment per line, each line matching the format (comma-separated values):
segment_filename,segment_start_time,segment_end_time |
segment_filename is the name of the output file generated by the muxer according to the provided pattern. CSV escaping (according to RFC4180) is applied if required.
segment_start_time and segment_end_time specify the segment start and end time expressed in seconds.
A list file with the suffix ".csv" or ".ext" will
auto-select this format.
‘ext’ is deprecated in favor or ‘csv’.
Generate an ffconcat file for the created segments. The resulting file can be read using the FFmpeg concat demuxer.
A list file with the suffix ".ffcat" or ".ffconcat" will
auto-select this format.
Generate an extended M3U8 file, version 3, compliant with http://tools.ietf.org/id/draft-pantos-http-live-streaming.
A list file with the suffix ".m3u8" will auto-select this format.
If not specified the type is guessed from the list file name suffix.
Set segment duration to time, the value must be a duration specification. Default value is "2". See also the ‘segment_times’ option.
Note that splitting may not be accurate, unless you force the reference stream key-frames at the given time. See the introductory notice and the examples below.
Specify the accuracy time when selecting the start time for a segment, expressed as a duration specification. Default value is "0".
When delta is specified a key-frame will start a new segment if its PTS satisfies the relation:
PTS >= start_time - time_delta |
This option is useful when splitting video content, which is always split at GOP boundaries, in case a key frame is found just before the specified split time.
In particular may be used in combination with the ‘ffmpeg’ option force_key_frames. The key frame times specified by force_key_frames may not be set accurately because of rounding issues, with the consequence that a key frame time may result set just before the specified time. For constant frame rate videos a value of 1/2*frame_rate should address the worst case mismatch between the specified time and the time set by force_key_frames.
Specify a list of split points. times contains a list of comma separated duration specifications, in increasing order. See also the ‘segment_time’ option.
Specify a list of split video frame numbers. frames contains a list of comma separated integer numbers, in increasing order.
This option specifies to start a new segment whenever a reference stream key frame is found and the sequential number (starting from 0) of the frame is greater or equal to the next value in the list.
Wrap around segment index once it reaches limit.
Set the sequence number of the first segment. Defaults to 0.
Reset timestamps at the begin of each segment, so that each segment
will start with near-zero timestamps. It is meant to ease the playback
of the generated segments. May not work with some combinations of
muxers/codecs. It is set to 0 by default.
Specify timestamp offset to apply to the output packet timestamps. The argument must be a time duration specification, and defaults to 0.
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut |
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut |
ffmpeg ‘force_key_frames’
option to force key frames in the input at the specified location, together
with the segment option ‘segment_time_delta’ to account for
possible roundings operated when setting key frame times.
ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \ -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut |
In order to force key frames on the input file, transcoding is required.
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut |
libx264
and libfaac encoders:
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts |
ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ -segment_list_flags +live -segment_time 10 out%03d.mkv |
The tee muxer can be used to write the same data to several files or any other kind of muxer. It can be used, for example, to both stream a video to the network and save it to disk at the same time.
It is different from specifying several outputs to the ffmpeg
command-line tool because the audio and video data will be encoded only once
with the tee muxer; encoding can be a very expensive process. It is not
useful when using the libavformat API directly because it is then possible
to feed the same packets to several muxers directly.
The slave outputs are specified in the file name given to the muxer, separated by ’|’. If any of the slave name contains the ’|’ separator, leading or trailing spaces or any special character, it must be escaped (see the “Quoting and escaping” section in the ffmpeg-utils manual).
Muxer options can be specified for each slave by prepending them as a list of key=value pairs separated by ’:’, between square brackets. If the options values contain a special character or the ’:’ separator, they must be escaped; note that this is a second level escaping.
The following special options are also recognized:
Specify the format name. Useful if it cannot be guessed from the output name suffix.
Specify a list of bitstream filters to apply to the specified
output. It is possible to specify to which streams a given bitstream
filter applies, by appending a stream specifier to the option
separated by /. If the stream specifier is not specified, the
bistream filters will be applied to all streams in the output.
Several bitstream filters can be specified, separated by ",".
Select the streams that should be mapped to the slave output, specified by a stream specifier. If not specified, this defaults to all the input streams.
Some examples follow.
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/" |
ffmpeg to encode the input, and send the output
to three different destinations. The dump_extra bitstream
filter is used to add extradata information to all the output video
keyframes packets, as requested by the MPEG-TS format. The select
option is applied to ‘out.aac’ in order to make it contain only
audio packets.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac -strict experimental
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
|
Note: some codecs may need different options depending on the output format; the auto-detection of this can not work with the tee muxer. The main example is the ‘global_header’ flag.
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:
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 |
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.
A description of the currently available protocols follows.
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 |
Caching wrapper for input stream.
Cache the input stream to temporary file. It brings seeking capability to live streams.
cache:URL |
Physical concatenation protocol.
Allow to read and seek from many resource 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.
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 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 access protocol.
Allow to read from or read to a file.
For example to read from a file ‘input.mpeg’ with ffmpeg
use the command:
ffmpeg -i file:input.mpeg output.mpeg |
The ff* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.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.
FTP (File Transfer Protocol).
Allow to 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 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.
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 protocol.
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 (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 custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.
Force a content type.
Override User-Agent header. If not specified the protocol will use a string describing the libavformat build.
Use persistent connections if set to 1. By default it is 0.
Set custom HTTP post data.
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.
Set MIME type.
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 0.
If the server supports ICY metadata, this contains the ICY specific HTTP reply headers, separated with 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.
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.
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 |
MMS (Microsoft Media Server) protocol over TCP.
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
mmsh://server[:port][/app][/playpath] |
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.
UNIX pipe access protocol.
Allow to 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.
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 AVOptions):
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/ |
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.
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.
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.
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.
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.
Secure File Transfer Protocol via libssh
Allow to 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.
Example: Play a file stored on remote server.
ffplay sftp://user:password@server_address:22/home/user/resource.mpeg |
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" |
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.
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.
This is a deprecated option. Instead, ‘localrtpport’ should be used.
Important notes:
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 RTSP server).
The required syntax for a RTSP url is:
rtsp://hostname[:port]/path |
The following options (set on the ffmpeg/ffplay command
line, or set in code via AVOptions or in avformat_open_input),
are supported:
Flags for rtsp_transport:
Use UDP as lower transport protocol.
Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
Use UDP multicast as lower transport protocol.
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.
Flags for rtsp_flags:
Accept packets only from negotiated peer address and port.
Act as a server, listening for an incoming connection.
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.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 |
To watch a stream tunneled over HTTP:
ffplay -rtsp_transport http rtsp://server/video.mp4 |
To send a stream in realtime to a RTSP server, for others to watch:
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp |
To receive a stream in realtime:
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp output |
Socket IO timeout in micro seconds.
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.
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] |
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] |
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.
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.
Trasmission Control Protocol.
The required syntax for a TCP url is:
tcp://hostname:port[?options] |
Listen for an incoming connection
In read mode: if no data arrived in more than this time interval, raise error. In write mode: if socket cannot be written in more than this time interval, raise error. This also sets timeout on TCP connection establishing.
ffmpeg -i input -f format tcp://hostname:port?listen ffplay tcp://hostname:port |
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 AVOptions):
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 GnuTLS, 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.
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 |
User Datagram Protocol.
The required syntax for a 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 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 socket buffer size in bytes. This is used both for the receiving and the sending buffer size.
Override the local UDP port to bind with.
Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.
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 to the multicast group from one of the specified sender IP addresses.
Ignore packets sent to the multicast group from the specified sender IP addresses.
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.
In read mode: if no data arrived in more than this time interval, raise error.
Some usage examples of the UDP protocol with ffmpeg follow.
To stream over UDP to a remote endpoint:
ffmpeg -i input -f format udp://hostname:port |
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 |
To receive over UDP from a remote endpoint:
ffmpeg -i udp://[multicast-address]:port |
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 AVOptions):
Timeout in ms.
Create the Unix socket in listening mode.
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 are configured elements in FFmpeg which allow to access 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 "-formats" of the ff* tools will display the list of supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
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
BSD video input device.
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.
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.
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 same name (starts at 0, defaults to 0).
Set audio device number for devices with 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
$ ffmpeg -list_devices true -f dshow -i dummy |
$ ffmpeg -f dshow -i video="Camera" |
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" |
$ ffmpeg -f dshow -i video="Camera":audio="Microphone" |
$ ffmpeg -list_options true -f dshow -i video="Camera" |
Linux DV 1394 input device.
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.
To record from the framebuffer device ‘/dev/fb0’ with
ffmpeg:
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi |
You can take a single screenshot image with the command:
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg |
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
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.
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 maxiumum 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 it’s 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.
ffplay -f iec61883 -i auto |
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg |
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/
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’.
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.
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.
ffplay:
ffplay -f lavfi -graph "color=c=pink [out0]" dummy |
ffplay -f lavfi color=c=pink |
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 |
ffplay:
ffplay -f lavfi "amovie=test.wav" |
ffplay:
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" |
IIDC1394 input device, based on libdc1394 and libraw1394.
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 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.
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’.
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.
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
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.
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.
Record a stream from default device:
ffmpeg -f pulse -i default /tmp/pulse.wav |
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 |
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:
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 |
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg |
For more information about Video4Linux, check http://linuxtv.org/.
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 to select the input format, when several are available.
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.
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.
X11 video input device.
This device allows 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 dpyinfo 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 |
Specify whether to draw the mouse pointer. A value of 0 specify
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 |
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.
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 |
Set the video frame size. Default value is vga.
Output devices are configured elements in FFmpeg that can write multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices are enabled by default. You can list all available ones using the configure option "–list-outdevs".
You can disable all the output devices using the configure option "–disable-outdevs", and selectively enable an output device using the option "–enable-outdev=OUTDEV", or you can disable a particular input device using the option "–disable-outdev=OUTDEV".
The option "-formats" of the ff* tools will display the list of enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
ffmpeg -i INPUT -f alsa default |
ffmpeg -i INPUT -f alsa hw:1,7 |
CACA output device.
This output device allows to show a video stream in CACA window. Only one CACA window is allowed per application, so you can have only one instance of this output device in an application.
To enable this output device you need to configure FFmpeg with
--enable-libcaca.
libcaca is a graphics library that outputs text instead of pixels.
For more information about libcaca, check: http://caca.zoy.org/wiki/libcaca
Set the CACA window title, if not specified default to the filename specified for the output device.
Set the CACA window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video.
Set display driver.
Set dithering algorithm. Dithering is necessary
because the picture being rendered has usually far more colours than
the available palette.
The accepted values are listed with -list_dither algorithms.
Set antialias method. Antialiasing smoothens the rendered
image and avoids the commonly seen staircase effect.
The accepted values are listed with -list_dither antialiases.
Set which characters are going to be used when rendering text.
The accepted values are listed with -list_dither charsets.
Set color to be used when rendering text.
The accepted values are listed with -list_dither colors.
If set to ‘true’, print a list of available drivers and exit.
List available dither options related to the argument.
The argument must be one of algorithms, antialiases,
charsets, colors.
ffmpeg output is an
CACA window, forcing its size to 80x25:
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - |
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - |
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - |
Linux framebuffer output 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.
Set x/y coordinate of top left corner. Default is 0.
Play a file on framebuffer device ‘/dev/fb0’. Required pixel format depends on current framebuffer settings.
ffmpeg -re -i INPUT -vcodec rawvideo -pix_fmt bgra -f fbdev /dev/fb0 |
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
OSS (Open Sound System) output device.
PulseAudio output device.
To enable this output device you need to configure FFmpeg with --enable-libpulse.
More information about PulseAudio can be found on http://www.pulseaudio.org
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 set to the specified output name.
Specify the device to use. Default device is used when not provided.
List of output devices can be obtained with command pactl list sinks.
Control the size and duration of the PulseAudio buffer. A small buffer gives more control, but requires more frequent updates.
‘buffer_size’ specifies size in bytes while ‘buffer_duration’ specifies duration in milliseconds.
When both options are provided then the highest value is used (duration is recalculated to bytes using stream parameters). If they are set to 0 (which is default), the device will use the default PulseAudio duration value. By default PulseAudio set buffer duration to around 2 seconds.
Play a file on default device on default server:
ffmpeg -i INPUT -f pulse "stream name" |
SDL (Simple DirectMedia Layer) output device.
This output device allows to show a video stream in an SDL window. Only one SDL window is allowed per application, so you can have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system when configuring your build.
For more information about SDL, check: http://www.libsdl.org/
Set the SDL window title, if not specified default to the filename specified for the output device.
Set the name of the iconified SDL window, if not specified it is set to the same value of window_title.
Set the SDL window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
Set fullscreen mode when non-zero value is provided. Default value is zero.
The window created by the device can be controlled through the following interactive commands.
Quit the device immediately.
The following command shows the ffmpeg output is an
SDL window, forcing its size to the qcif format:
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" |
sndio audio output device.
XV (XVideo) output device.
This output device allows to show a video stream in a X Window System window.
Specify the hardware display name, which determines the display and communications domain to be used.
The display name or DISPLAY environment variable can be a string in the format hostname[:number[.screen_number]].
hostname specifies the name of the host machine on which the display is physically attached. number specifies the number of the display server on that host machine. screen_number specifies the screen to be used on that server.
If unspecified, it defaults to the value of the DISPLAY environment variable.
For example, dual-headed:0.1 would specify screen 1 of display
0 on the machine named “dual-headed”.
Check the X11 specification for more detailed information about the display name format.
Set the created window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video.
Set the X and Y window offsets for the created window. They are both set to 0 by default. The values may be ignored by the window manager.
Set the window title, if not specified default to the filename specified for the output device.
For more information about XVideo see http://www.x.org/.
ffmpeg at the
same time:
ffmpeg -i INPUT OUTPUT -f xv display |
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated |
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 (ffmpeg-utils)channel layout syntax 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 volumn reduction A value of 1.0 prevents cliping.
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, 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 if set to 1, default value is 0.
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.
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 an integer 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.
The video scaler supports the following named options.
Options may be set by specifying -option value in the
FFmpeg tools. For programmatic use, they can be set explicitly in the
SwsContext options or through the ‘libavutil/opt.h’ API.
Set the scaler flags. This is also used to set the scaling algorithm. Only a single algorithm should be selected.
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.
Select sinc rescaling algorithm.
Select lanczos rescaling algorithm.
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).
Select source range.
Select destination 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
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, 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 in output the top half of the video is mirrored onto the bottom half.
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.
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.
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 the 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", a filter with no output pads is called a "sink".
A filtergraph can be represented using a textual representation, which is
recognized by the ‘-filter’/‘-vf’ and ‘-filter_complex’
options in ffmpeg and ‘-vf’ in ffplay, and by the
avfilter_graph_parse()/avfilter_graph_parse2() 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=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. 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 the following forms:
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’.
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 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 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.
Follows a BNF description for the filtergraph syntax:
NAME ::= sequence of alphanumeric characters and '_' LINKLABEL ::= "[" NAME "]" LINKLABELS ::= LINKLABEL [LINKLABELS] FILTER_ARGUMENTS ::= sequence of chars (eventually quoted) FILTER ::= [LINKLABELS] NAME ["=" FILTER_ARGUMENTS] [LINKLABELS] FILTERCHAIN ::= FILTER [,FILTERCHAIN] FILTERGRAPH ::= [sws_flags=flags;] FILTERCHAIN [;FILTERGRAPH] |
Some filter arguments require the use of special characters, typically
: to separate key=value pairs in a named options list. In this
case the user should perform a first level escaping when specifying
the filter arguments. For example, consider the following literal
string to be embedded in the drawtext filter arguments:
this is a 'string': may contain one, or more, special characters |
Since : is special for the filter arguments syntax, it needs to
be escaped, so you get:
text=this is a \'string\'\: may contain one, or more, special characters |
A second level of escaping is required when embedding the filter arguments 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 |
Finally an additional level of escaping may be 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" |
Sometimes, it might be more convenient to employ quoting in place of escaping. For example the string:
Caesar: tu quoque, Brute, fili mi |
Can be quoted in the filter arguments as:
text='Caesar: tu quoque, Brute, fili mi' |
And finally inserted in a filtergraph like:
drawtext=text=\'Caesar: tu quoque\, Brute\, fili mi\' |
See the “Quoting and escaping” section in the ffmpeg-utils manual for more information about the escaping and quoting rules adopted by FFmpeg.
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
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 |
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.
Convert the input audio format to the specified formats.
This filter is deprecated. Use aformat instead.
The filter accepts a string of the form: "sample_format:channel_layout".
sample_format specifies the sample format, and can be a string or the corresponding numeric value defined in ‘libavutil/samplefmt.h’. Use ’p’ suffix for a planar sample format.
channel_layout specifies the channel layout, and can be a string or the corresponding number value defined in ‘libavutil/channel_layout.h’.
The special parameter "auto", signifies that the filter will automatically select the output format depending on the output filter.
aconvert=fltp:stereo |
aconvert=u8:auto |
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 ’|’. At least one delay greater than 0 should be provided. Unused delays will be silently ignored. If number of given delays is smaller than number of channels all remaining channels will not be delayed.
adelay=1500|0|500 |
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 loudnesses of reflected signals separated by ’|’.
Allowed range for each decay is (0 - 1.0].
Default is 0.5.
aecho=0.8:0.88:60:0.4 |
aecho=0.8:0.88:6:0.4 |
aecho=0.8:0.9:1000:0.3 |
aecho=0.8:0.9:1000|1800:0.3|0.25 |
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.
This filter accepts the following options:
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.
aeval=val(ch)/2:c=same |
eval=val(0)|-val(1) |
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 time for starting to apply the fade effect. Default is 0. The accepted syntax is:
[-]HH[:MM[:SS[.m...]]] [-]S+[.m...] |
See also the function av_parse_time().
If set this option is used instead of start_sample one.
Specify the duration for which the fade effect has to last. Default is 0. The accepted syntax is:
[-]HH[:MM[:SS[.m...]]] [-]S+[.m...] |
See also the function av_parse_time().
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.
If set this option is used instead of nb_samples one.
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
afade=t=in:ss=0:d=15 |
afade=t=out:st=875:d=25 |
Set output format constraints for the input audio. The framework will negotiate the most appropriate format to minimize conversions.
The filter accepts the following named parameters:
A ’|’-separated list of requested sample formats.
A ’|’-separated list of requested sample rates.
A ’|’-separated list of requested channel layouts.
See (ffmpeg-utils)channel layout syntax for the required syntax.
If a parameter is omitted, all values are allowed.
For example to force the output to either unsigned 8-bit or signed 16-bit stereo:
aformat=sample_fmts=u8|s16:channel_layouts=stereo |
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
Specify the band-width of a filter in width_type units.
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.
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge |
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 |
Mixes multiple audio inputs into a single output.
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.
The filter accepts the following named parameters:
Number of inputs. If unspecified, it defaults to 2.
How to determine the end-of-stream.
Duration of longest input. (default)
Duration of shortest input.
Duration of first input.
Transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
Pass the audio source unchanged to the output.
Pad the end of a audio stream with silence, this can be used together with -shortest to extend audio streams to the same length as the video stream.
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:
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 ffmpeg-resampler manual for the complete list of supported options.
aresample=44100 |
aresample=async=1000 |
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 signal 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 |
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.
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.
A description of each shown parameter follows:
sequential number of the input frame, starting from 0
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.
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)
sample format
channel layout
sample rate for the audio frame
number of samples (per channel) in the frame
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.
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.
The filter accepts the following option:
Short window length in seconds, used for peak and trough RMS measurement.
Default is 0.05 (50 miliseconds). Allowed range is [0.1 - 10].
A description of each shown parameter follows:
Mean amplitude displacement from zero.
Minimal sample level.
Maximal sample level.
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.
Forward two audio streams and control the order the buffers are forwarded.
The filter accepts the following options:
Set the expression deciding which stream should be forwarded next: if the result is negative, the first stream is forwarded; if the result is positive or zero, the second stream is forwarded. It can use the following variables:
number of buffers forwarded so far on each stream
number of samples forwarded so far on each stream
current timestamp of each stream
The default value is t1-t2, which means to always forward the stream
that has a smaller timestamp.
Stress-test amerge by randomly sending buffers on the wrong
input, while avoiding too much of a desynchronization:
amovie=file.ogg [a] ; amovie=file.mp3 [b] ; [a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ; [a2] [b2] amerge |
Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed.
This filter is not built by default, please use aresample to do squeezing/stretching.
The filter accepts the following named parameters:
Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence.
Minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. Default value is 0.1. If you get non-perfect sync with this filter, try setting this parameter to 0.
Maximum compensation in samples per second. Relevant only with compensate=1. Default value 500.
Assume the first pts should be this value. The time base 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.
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, 2.0] range.
atempo=0.8 |
atempo=1.25 |
Trim the input so that the output contains one continuous subpart of the input.
This filter accepts the following options:
Specify time of the start of the kept section, 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.
Specify maximum duration of the output.
Number of the first sample that should be passed to output.
Number of the first sample that should be dropped.
‘start’, ‘end’, ‘duration’ are expressed as time duration specifications, check the "Time duration" section in the ffmpeg-utils 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 that 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:
ffmpeg -i INPUT -af atrim=60:120 |
ffmpeg -i INPUT -af atrim=end_sample=1000 |
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
Specify the band-width of a filter in width_type units.
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
Specify the band-width of a filter in width_type units.
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
Determine how steep is the filter’s shelf transition.
Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator and denominator coefficients respectively.
Remap input channels to new locations.
This filter accepts the following named parameters:
Channel layout of the output stream.
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.
If no mapping is present, the filter will implicitly map input channels to output channels preserving index.
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:channel_layout=5.1' out.wav |
Split each channel in input audio stream into a separate output stream.
This filter accepts the following named parameters:
Channel layout of the input stream. Default is "stereo".
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.
To 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 |
Compress or expand audio dynamic range.
A description of the accepted options follows.
Set 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.
Typical value for attack is 0.3 seconds and for decay 0.8
seconds.
Set list of points for transfer function, specified in dB relative to maximum
possible signal amplitude.
Each key points list need to be defined using the following syntax:
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.
Set amount for which the points at where adjacent line segments on the
transfer function meet will be rounded. Defaults is 0.01.
Set additional gain in dB to be applied at all points on the transfer function
and allows easy adjustment of the overall gain.
Default is 0.
Set 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. Default is 0.
Set delay in seconds. Default is 0. The input audio
is analysed 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.
compand=.3 .3:1 1:-90/-60 -60/-40 -40/-30 -20/-20:6:0:-90:0.2 |
compand=.1 .1:.2 .2:-900/-900 -50.1/-900 -50/-50:.01:0:-90:.1 |
compand=.1 .1:.1 .1:-45.1/-45.1 -45/-900 0/-900:.01:45:-90:.1 |
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.
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
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.
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
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.
Join multiple input streams into one multi-channel stream.
The filter accepts the following named parameters:
Number of input streams. Defaults to 2.
Desired output channel layout. 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 those 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.
E.g. to join 3 inputs (with properly set channel layouts)
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT |
To 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 |
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.
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 function
av_parse_time() for the accepted format, also check the "Time duration"
section in the ffmpeg-utils manual.
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.
ladspa=file=amp |
vcf_notch
plugin from VCF library:
ladspa=f=vcf:p=vcf_notch:c=help |
Computer Music Toolkit (CMT)
plugin library:
ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12 |
ladspa=file=tap_reverb:tap_reverb |
ladspa=file=cmt:noise_source_white:c=c0=.2 |
C* Click - Metronome from the
C* Audio Plugin Suite (CAPS) library:
ladspa=file=caps:Click:c=c1=20' |
C* Eq10X2 - Stereo 10-band equaliser effect:
ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2 |
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.
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
Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The default is 0.707q and gives a Butterworth response.
Mix 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 remap efficiently 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.
For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:
pan=1: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.
The channel remapping will be effective if, and only if:
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 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.
Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly.
Detect silence in an audio stream.
This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration.
The printed times and duration are expressed in seconds.
The filter accepts the following options:
Set silence duration until notification (default is 2 seconds).
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.
silencedetect=n=-50dB:d=5 |
ffmpeg to detect silence with 0.0001 noise
tolerance in ‘silence.mp3’:
ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null - |
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
Determine how steep is the filter’s shelf transition.
Adjust the input audio volume.
The filter accepts the following options:
Expresses how the audio volume will be increased or decreased.
Output values are clipped to the maximum value.
The output audio volume is given by the relation:
output_volume = volume * input_volume |
Default value for volume is 1.0.
Set the mathematical precision.
This determines which input sample formats will be allowed, which affects the precision of the volume scaling.
8-bit fixed-point; limits input sample format to U8, S16, and S32.
32-bit floating-point; limits input sample format to FLT. (default)
64-bit floating-point; limits input sample format to DBL.
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 |
volume=volume=6dB:precision=fixed |
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.
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:
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.
Below is a description of the currently available audio sources.
Buffer audio frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in ‘libavfilter/asrc_abuffer.h’.
It accepts the following named parameters:
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 corresponging 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.
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 |
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 function
av_parse_time() for the accepted format.
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
aevalsrc=0 |
aevalsrc="sin(440*2*PI*t):s=8000" |
aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC" |
aevalsrc="-2+random(0)" |
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" |
aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" |
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:
Specify 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.
Specify the sample rate, and defaults to 44100.
Set the number of samples per requested frames.
anullsrc=r=48000:cl=4 |
anullsrc=r=48000:cl=mono |
All the parameters need to be explicitly defined.
Synthesize a voice utterance using the libflite library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libflite.
Note that the flite library is 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.
flite=textfile=speech.txt |
slt voice:
flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt |
ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt |
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.speech.cs.cmu.edu/flite/
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, default is 1024.
sine |
sine=220:4:d=5 sine=f=220:b=4:d=5 sine=frequency=220:beep_factor=4:duration=5 |
Below is a description of the currently available audio sinks.
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.
Null audio sink, do absolutely nothing with the input audio. It is mainly useful as a template and to be employed in analysis / debugging tools.
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.
Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.
Add or replace the alpha component of the primary input with the grayscale value of a second input. This is intended for use with alphaextract to allow the transmission or storage of frame sequences that have alpha in a format that doesn’t support an alpha channel.
For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with alphaextract, you might use:
movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] |
Since this filter is designed for reconstruction, it operates on frame sequences without considering timestamps, and terminates when either input reaches end of stream. This will cause problems if your encoding pipeline drops frames. If you’re trying to apply an image as an overlay to a video stream, consider the overlay filter instead.
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.
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.
Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the detected black interval expressed in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
The filter accepts the following options:
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 |
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.
The filter accepts the following options:
Set the percentage of the pixels that have to be below the threshold, defaults
to 98.
Set the threshold below which a pixel value is considered black, defaults to
32.
Blend two video frames into each other.
It takes two input streams and outputs one stream, the first input is the "top" layer and second input is "bottom" layer. Output terminates when shortest input terminates.
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 depending on the currently filtered plane. It is the
ratio between the corresponding luma plane number of pixels and the current
plane ones. E.g. for YUV4:2:0 the values are 1,1 for the luma plane, and
0.5,0.5 for chroma planes.
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).
Force termination when the shortest input terminates. Default is 0.
Continue applying the last bottom frame after the end of the stream. A value of
0 disable the filter after the last frame of the bottom layer is reached.
Default is 1.
blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))' |
blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)' |
blend=all_expr='if(gte(N*SW+X,W),A,B)' |
blend=all_expr='if(gte(Y-N*SH,0),A,B)' |
blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)' |
Apply boxblur algorithm to the input video.
The filter accepts the following options:
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
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.
boxblur=luma_radius=2:luma_power=1 boxblur=2:1 |
boxblur=2:1:cr=0:ar=0 |
boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1 |
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.
colorbalance=rs=.3 |
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.
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].
colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3 |
colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131 |
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
BT.601
SMPTE-240M
FCC
For example to convert from BT.601 to SMPTE-240M, use the command:
colormatrix=bt601:smpte240m |
Copy the input source unchanged to the output. Mainly useful for testing purposes.
Crop the input video to given dimensions.
The filter accepts the following options:
Width of the output video. It defaults to iw.
This expression is evaluated only once during the filter
configuration.
Height of the output video. It defaults to ih.
This expression is evaluated only once during the filter
configuration.
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.
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.
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
same as in_w and in_h
the output (cropped) width and height
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 input frame, starting from 0
the position in the file of the input frame, NAN if unknown
timestamp expressed in seconds, 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.
crop=100:100:12:34 |
Using named options, the example above becomes:
crop=w=100:h=100:x=12:y=34 |
crop=100:100 |
crop=2/3*in_w:2/3*in_h |
crop=out_w=in_h crop=in_h |
crop=in_w-100:in_h-100:100:100 |
crop=in_w-2*10:in_h-2*20 |
crop=in_w/2:in_h/2:in_w/2:in_h/2 |
crop=in_w:1/PHI*in_w |
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) |
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)" |
crop=in_w/2:in_h/2:y:10+10*sin(n/10) |
Auto-detect crop size.
Calculate necessary cropping parameters and prints the recommended parameters through the logging system. The detected dimensions correspond to the non-black area of the input video.
The filter accepts the following options:
Set higher black value threshold, which can be optionally specified from nothing (0) to everything (255). An intensity value greater to the set value is considered non-black. Default value is 24.
Set the value for which the width/height should be divisible by. 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. Default value is 16.
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 return the largest area encountered during playback.
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.
If there is no key point defined in x=0, the filter will automatically
insert a (0;0) point. In the same way, if there is no key point defined
in x=1, the filter will automatically insert a (1;1) point.
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 (.asv) to import the settings from.
To avoid some filtergraph syntax conflicts, each key points list need to be
defined using the following syntax: x0/y0 x1/y1 x2/y2 ....
curves=blue='0.5/0.58' |
curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':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)
curves=preset=vintage |
curves=vintage |
curves=psfile='MyCurvesPresets/purple.asv':green='0.45/0.53' |
Denoise frames using 2D DCT (frequency domain filtering).
This filter is not designed for real time and can be extremely slow.
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. Each block is of size
16x16. 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 allow to process the whole input width or height, a warning will be displayed and according borders won’t be denoised.
Default value is 15.
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.
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)' |
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.
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).
This filter accepts the following options:
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.
Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 4.
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.
delogo=x=0:y=0:w=100:h=77:band=10 |
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.
If set to 1, specify using OpenCL capabilities, only available if
FFmpeg was configured with --enable-opencl. Default value is 0.
Draw a colored box on the input image.
This filter accepts the following options:
The expressions which specify the top left corner coordinates of the box. Default to 0.
The expressions which specify the width and height of the box, if 0 they are interpreted as the input width and height. Default 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. Default value is 3.
See below for the list of accepted constants.
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.
drawbox |
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 |
drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max |
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 |
Draw a grid on the input image.
This filter accepts the following options:
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.
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.
drawgrid=width=100:height=100:thickness=2:color=red@0.5 |
drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5 |
Draw text string or text from specified file on top of video using the libfreetype library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libfreetype.
The description of the accepted parameters follows.
Used to draw a box around text using background color. Value should be either 1 (enable) or 0 (disable). The default value of box 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".
Select how the text is expanded. Can be either none,
strftime (deprecated) or
normal (default). See the Text expansion section
below for details.
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".
The font file to be used for drawing text. Path must be included. This parameter is mandatory.
The font size to be used for drawing text. The default value of fontsize is 16.
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 "render".
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. 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).
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
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
each other, so you can for example specify y=x/dar.
If libavfilter was built with --enable-fontconfig, then
‘fontfile’ can be a fontconfig pattern or omitted.
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.
Sequence of the form %{...} are expanded. The text between the
braces is a function name, possibly followed by arguments separated by ’:’.
If the arguments contain special characters or delimiters (’:’ or ’}’),
they should be escaped.
Note that they probably must also be escaped as the value for the ‘text’ option in the filter argument string and as the filter argument in the filtergraph description, and possibly also for the shell, that makes up to four levels of escaping; using a text file avoids these problems.
The following functions are available:
expr, eThe 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.
gmtimeThe time at which the filter is running, expressed in UTC. It can accept an argument: a strftime() format string.
localtimeThe time at which the filter is running, expressed in the local time zone. It can accept an argument: a strftime() format string.
metadataFrame metadata. It must take one argument specifying metadata key.
n, frame_numThe frame number, starting from 0.
pict_typeA 1 character description of the current picture type.
ptsThe timestamp of the current frame, in seconds, with microsecond accuracy.
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" |
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.
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2" |
drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" |
drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" |
drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" |
drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'" |
drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg' |
drawtext='fontfile=FreeSans.ttf:text=%{localtime:%a %b %d %Y}'
|
For more information about libfreetype, check: http://www.freetype.org/.
For more information about fontconfig, check: http://freedesktop.org/software/fontconfig/fontconfig-user.html.
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 choosen 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.
Example:
edgedetect=low=0.1:high=0.4 |
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.
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 |
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.
Apply fade-in/out effect to input video.
This filter accepts the following options:
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 frame for starting to apply the fade effect. Default is 0.
The number of frames 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’. 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.
Specify the color of the fade. Default is "black".
fade=in:0:30 |
The command above is equivalent to:
fade=t=in:s=0:n=30 |
fade=out:155:45 fade=type=out:start_frame=155:nb_frames=45 |
fade=in:0:25, fade=out:975:25 |
fade=in:5:20:color=yellow |
fade=in:0:25:alpha=1 |
fade=t=in:st=5.5:d=0.5 |
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).
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 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).
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.
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:
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
|
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 |
Transform the field order of the input video.
This filter accepts the following options:
Output field order. Valid values are tff for top field first or bff for bottom field first.
Default value is ‘tff’.
Transformation is achieved by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. This method is consistent with most broadcast field order converters.
If the input video is not flagged as being interlaced, or it is already flagged as being of the required output field order then this filter does not alter the incoming video.
This filter is very useful when converting to or from PAL DV material, which is bottom field first.
For example:
ffmpeg -i in.vob -vf "fieldorder=bff" out.dv |
Buffer input images and send them when they are requested.
This filter is mainly useful when auto-inserted by the libavfilter framework.
The filter does not take parameters.
Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is supported for the input to the next filter.
This filter accepts the following parameters:
A ’|’-separated list of pixel format names, for example "pix_fmts=yuv420p|monow|rgb24".
format=pix_fmts=yuv420p |
Convert the input video to any of the formats in the list
format=pix_fmts=yuv420p|yuv444p|yuv410p |
Convert the video to specified constant frame rate by duplicating or dropping frames as necessary.
This filter accepts the following named parameters:
Desired output frame rate. The default is 25.
Rounding method.
Possible values are:
zero round towards 0
round away from 0
round towards -infinity
round towards +infinity
round to nearest
The default is near.
Assume the first PTS should be the given value, in seconds. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame’s expected PTS, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with duplicates of the first frame if a video stream starts after the audio stream or to trim any frames with a negative PTS.
Alternatively, the options can be specified as a flat string: fps[:round].
See also the setpts filter.
fps=fps=25 |
fps=fps=film:round=near |
Select one frame every N-th frame.
This filter accepts the following option:
Select frame after every step frames.
Allowed values are positive integers higher than 0. Default value is 1.
Apply a frei0r effect to the input video.
To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with --enable-frei0r.
This filter accepts the following options:
The name to the frei0r effect to load. If the environment variable
FREI0R_PATH is defined, the frei0r effect is searched in each one of the
directories specified by the colon separated list in FREIOR_PATH,
otherwise in the standard frei0r paths, which are in this order:
‘HOME/.frei0r-1/lib/’, ‘/usr/local/lib/frei0r-1/’,
‘/usr/lib/frei0r-1/’.
A ’|’-separated list of parameters to pass to the frei0r effect.
A frei0r effect parameter can be a boolean (whose values are specified with "y" and "n"), a double, a color (specified by the syntax R/G/B, (R, G, and B being float numbers from 0.0 to 1.0) or by a color description specified in the "Color" section in the ffmpeg-utils manual), a position (specified by the syntax X/Y, X and Y being float numbers) and a string.
The number and kind of parameters depend on the loaded effect. If an effect parameter is not specified the default value is set.
frei0r=filter_name=distort0r:filter_params=0.5|0.01 |
frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 |
frei0r=perspective:0.2/0.2|0.8/0.2 |
For more information see: http://frei0r.dyne.org
The filter accepts the following options:
Set the luminance expression.
Set the chrominance blue expression.
Set the chrominance red expression.
Set the alpha expression.
Set the red expression.
Set the green expression.
Set the blue expression.
The colorspace is selected according to the specified options. If one of the ‘lum_expr’, ‘cb_expr’, or ‘cr_expr’ options is specified, the filter will automatically select a YCbCr colorspace. If one of the ‘red_expr’, ‘green_expr’, or ‘blue_expr’ options is specified, it will select an RGB colorspace.
If one of the chrominance expression is not defined, it falls back on the other one. If no alpha expression is specified it will evaluate to opaque value. If none of chrominance expressions are specified, they will evaluate to the luminance expression.
The expressions can use the following variables and functions:
The sequential number of the filtered frame, starting from 0.
The coordinates of the current sample.
The width and height of the image.
Width and height scale depending on the currently filtered plane. It is the
ratio between the corresponding luma plane number of pixels and the current
plane ones. E.g. for YUV4:2:0 the values are 1,1 for the luma plane, and
0.5,0.5 for chroma planes.
Time of the current frame, expressed in seconds.
Return the value of the pixel at location (x,y) of the current plane.
Return the value of the pixel at location (x,y) of the luminance plane.
Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if there is no such plane.
Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there is no such plane.
Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is no such component.
Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there is no such plane.
For functions, if x and y are outside the area, the value will be automatically clipped to the closer edge.
geq=p(W-X\,Y) |
PI/3 and a
wavelength of 100 pixels:
geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128 |
nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128 |
format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2' |
geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)' |
Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8bit color depth. Interpolate the gradients that should go where the bands are, and dither them.
This filter is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands.
This filter accepts the following options:
The maximum amount by which the filter will change any one pixel. Also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 64, default value is 1.2, out-of-range values will be clipped to the valid range.
The neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32, default value is 16, out-of-range values will be clipped to the valid range.
Alternatively, the options can be specified as a flat string: strength[:radius]
3.5 strength and radius of 8:
gradfun=3.5:8 |
gradfun=radius=8 |
Apply a Hald CLUT to a video stream.
First input is the video stream to process, and second one is the Hald CLUT. The Hald CLUT input can be a simple picture or a complete video stream.
The filter accepts the following options:
Force termination when the shortest input terminates. Default is 0.
Continue applying the last CLUT after the end of the stream. A value of
0 disable the filter after the last frame of the CLUT is reached.
Default is 1.
haldclut also has the same interpolation options as lut3d (both
filters share the same internals).
More information about the Hald CLUT can be found on Eskil Steenberg’s website (Hald CLUT author) at http://www.quelsolaar.com/technology/clut.html.
Generate an identity Hald CLUT stream altered with various effects:
ffmpeg -f lavfi -i haldclutsrc=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut |
Note: make sure you use a lossless codec.
Then use it with haldclut to apply it on some random stream:
ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv |
The Hald CLUT will be applied to the 10 first seconds (duration of
‘clut.nut’), then the latest picture of that CLUT stream will be applied
to the remaining frames of the mandelbrot stream.
A Hald CLUT is supposed to be a squared image of Level*Level*Level by
Level*Level*Level pixels. For a given Hald CLUT, FFmpeg will select the
biggest possible square starting at the top left of the picture. The remaining
padding pixels (bottom or right) will be ignored. This area can be used to add
a preview of the Hald CLUT.
Typically, the following generated Hald CLUT will be supported by the
haldclut filter:
ffmpeg -f lavfi -i haldclutsrc=8 -vf " pad=iw+320 [padded_clut]; smptebars=s=320x256, split [a][b]; [padded_clut][a] overlay=W-320:h, curves=color_negative [main]; [main][b] overlay=W-320" -frames:v 1 clut.png |
It contains the original and a preview of the effect of the CLUT: SMPTE color bars are displayed on the right-top, and below the same color bars processed by the color changes.
Then, the effect of this Hald CLUT can be visualized with:
ffplay input.mkv -vf "movie=clut.png, [in] haldclut" |
Flip the input video horizontally.
For example to horizontally flip the input video with ffmpeg:
ffmpeg -i in.avi -vf "hflip" out.avi |
This filter applies a global color histogram equalization on a per-frame basis.
It can be used to correct video that has a compressed range of pixel intensities. The filter redistributes the pixel intensities to equalize their distribution across the intensity range. It may be viewed as an "automatically adjusting contrast filter". This filter is useful only for correcting degraded or poorly captured source video.
The filter accepts the following options:
Determine the amount of equalization to be applied. As the strength is reduced, the distribution of pixel intensities more-and-more approaches that of the input frame. The value must be a float number in the range [0,1] and defaults to 0.200.
Set the maximum intensity that can generated and scale the output values appropriately. The strength should be set as desired and then the intensity can be limited if needed to avoid washing-out. The value must be a float number in the range [0,1] and defaults to 0.210.
Set the antibanding level. If enabled the filter will randomly vary
the luminance of output pixels by a small amount to avoid banding of
the histogram. Possible values are none, weak or
strong. It defaults to none.
Compute and draw a color distribution histogram for the input video.
The computed histogram is a representation of distribution of color components in an image.
The filter accepts the following options:
Set histogram mode.
It accepts the following values:
standard histogram that display color components distribution in an image. Displays color graph for each color component. Shows distribution of the Y, U, V, A or R, G, B components, depending on input format, in current frame. Bellow each graph is color component scale meter.
chroma values in vectorscope, if brighter more such chroma values are distributed in an image. Displays chroma values (U/V color placement) in two dimensional graph (which is called a vectorscope). It can be used to read of the hue and saturation of the current frame. At a same time it is a histogram. The whiter a pixel in the vectorscope, the more pixels of the input frame correspond to that pixel (that is the more pixels have this chroma value). The V component is displayed on the horizontal (X) axis, with the leftmost side being V = 0 and the rightmost side being V = 255. The U component is displayed on the vertical (Y) axis, with the top representing U = 0 and the bottom representing U = 255.
The position of a white pixel in the graph corresponds to the chroma value of a pixel of the input clip. So the graph can be used to read of the hue (color flavor) and the saturation (the dominance of the hue in the color). As the hue of a color changes, it moves around the square. At the center of the square, the saturation is zero, which means that the corresponding pixel has no color. If you increase the amount of a specific color, while leaving the other colors unchanged, the saturation increases, and you move towards the edge of the square.
chroma values in vectorscope, similar as color but actual chroma values
are displayed.
per row/column color component graph. In row mode graph in the left side represents color component value 0 and right side represents value = 255. In column mode top side represents color component value = 0 and bottom side represents value = 255.
Default value is levels.
Set height of level in levels. Default value is 200.
Allowed range is [50, 2048].
Set height of color scale in levels. Default value is 12.
Allowed range is [0, 40].
Set step for waveform mode. Smaller values are useful to find out how much
of same luminance values across input rows/columns are distributed.
Default value is 10. Allowed range is [1, 255].
Set mode for waveform. Can be either row, or column.
Default is row.
Set mirroring mode for waveform. 0 means unmirrored, 1
means mirrored. In mirrored mode, higher values will be represented on the left
side for row mode and at the top for column mode. Default is
0 (unmirrored).
Set display mode for waveform and levels.
It accepts the following values:
Display separate graph for the color components side by side in
row waveform mode or one below other in column waveform mode
for waveform histogram mode. For levels histogram mode
per color component graphs are placed one bellow other.
This display mode in waveform histogram mode makes it easy to spot
color casts in the highlights and shadows of an image, by comparing the
contours of the top and the bottom of each waveform.
Since whites, grays, and blacks are characterized by
exactly equal amounts of red, green, and blue, neutral areas of the
picture should display three waveforms of roughly equal width/height.
If not, the correction is easy to make by making adjustments to level the
three waveforms.
Presents information that’s identical to that in the parade, except
that the graphs representing color components are superimposed directly
over one another.
This display mode in waveform histogram mode can make it easier to spot
the relative differences or similarities in overlapping areas of the color
components that are supposed to be identical, such as neutral whites, grays,
or blacks.
Default is parade.
Set mode for levels. Can be either linear, or logarithmic.
Default is linear.
ffplay -i input -vf histogram |
High precision/quality 3d denoise filter. This filter aims to reduce image noise producing smooth images and making still images really still. It should enhance compressibility.
It accepts the following optional parameters:
a non-negative float number which specifies spatial luma strength, defaults to 4.0
a non-negative float number which specifies spatial chroma strength, defaults to 3.0*luma_spatial/4.0
a float number which specifies luma temporal strength, defaults to 6.0*luma_spatial/4.0
a float number which specifies chroma temporal strength, defaults to luma_tmp*chroma_spatial/luma_spatial
Modify the hue and/or the saturation of the input.
This filter accepts the following options:
Specify the hue angle as a number of degrees. It accepts an expression, and defaults to "0".
Specify the saturation in the [-10,10] range. It accepts an expression and defaults to "1".
Specify the hue angle as a number of radians. It accepts an expression, and defaults to "0".
Specify the brightness in the [-10,10] range. It accepts an expression and defaults to "0".
‘h’ and ‘H’ are mutually exclusive, and can’t be specified at the same time.
The ‘b’, ‘h’, ‘H’ and ‘s’ option values are expressions containing the following constants:
frame count of the input frame starting from 0
presentation timestamp of the input frame expressed in time base units
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
time base of the input video
hue=h=90:s=1 |
hue=H=PI/2:s=1 |
hue="H=2*PI*t: s=sin(2*PI*t)+1" |
hue="s=min(t/3\,1)" |
The general fade-in expression can be written as:
hue="s=min(0\, max((t-START)/DURATION\, 1))" |
hue="s=max(0\, min(1\, (8-t)/3))" |
The general fade-out expression can be written as:
hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" |
This filter supports the following commands:
Modify the hue and/or the saturation and/or brightness of the input video. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
Detect video interlacing type.
This filter tries to detect if the input is interlaced or progressive, top or bottom field first.
The filter accepts the following options:
Set interlacing threshold.
Set progressive threshold.
Deinterleave or interleave fields.
This filter allows to process interlaced images fields without deinterlacing them. Deinterleaving splits the input frame into 2 fields (so called half pictures). Odd lines are moved to the top half of the output image, even lines to the bottom half. You can process (filter) them independently and then re-interleave them.
The filter accepts the following options:
Available values for luma_mode, chroma_mode and alpha_mode are:
Do nothing.
Deinterleave fields, placing one above the other.
Interleave fields. Reverse the effect of deinterleaving.
Default value is none.
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.
Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.
Original Original New Frame
Frame 'j' Frame 'j+1' (tff)
========== =========== ==================
Line 0 --------------------> Frame 'j' Line 0
Line 1 Line 1 ----> Frame 'j+1' Line 1
Line 2 ---------------------> Frame 'j' Line 2
Line 3 Line 3 ----> Frame 'j+1' Line 3
... ... ...
New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on
|
It accepts the following optional parameters:
determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.
Enable (default) or disable the vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.
Deinterlace input video by applying Donald Graft’s adaptive kernel deinterling. Work on interlaced parts of a video to produce progressive frames.
The description of the accepted parameters follows.
Set the threshold which affects the filter’s tolerance when determining if a pixel line must be processed. It must be an integer in the range [0,255] and defaults to 10. A value of 0 will result in applying the process on every pixels.
Paint pixels exceeding the threshold value to white if set to 1. Default is 0.
Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.
Enable additional sharpening if set to 1. Default is 0.
Enable twoway sharpening if set to 1. Default is 0.
kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0 |
kerndeint=sharp=1 |
kerndeint=map=1 |
Apply a 3D LUT to an input video.
The filter accepts the following options:
Set the 3D LUT file name.
Currently supported formats:
AfterEffects
Iridas
DaVinci
Pandora
Select interpolation mode.
Available values are:
Use values from the nearest defined point.
Interpolate values using the 8 points defining a cube.
Interpolate values using a tetrahedron.
Compute a look-up table for binding each pixel component input value to an output value, and apply it to input video.
lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.
These filters accept the following options:
set first pixel component expression
set second pixel component expression
set third pixel component expression
set fourth pixel component expression, corresponds to the alpha component
set red component expression
set green component expression
set blue component expression
alpha component expression
set Y/luminance component expression
set U/Cb component expression
set V/Cr component expression
Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values.
The exact component associated to each of the c* options depends on the format in input.
The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.
The expressions can contain the following constants and functions:
the input width and height
input value for the pixel component
the input value clipped in the minval-maxval range
maximum value for the pixel component
minimum value for the pixel component
the negated value for the pixel component value clipped in the minval-maxval range , it corresponds to the expression "maxval-clipval+minval"
the computed value in val clipped in the minval-maxval range
the computed gamma correction value of the pixel component value clipped in the minval-maxval range, corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"
All expressions default to "val".
lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" |
The above is the same as:
lutrgb="r=negval:g=negval:b=negval" lutyuv="y=negval:u=negval:v=negval" |
lutyuv=y=negval |
lutyuv="u=128:v=128" |
lutyuv="y=2*val" |
lutrgb="g=0:b=0" |
format=rgba,lutrgb=a="maxval-minval/2" |
lutyuv=y=gammaval(0.5) |
lutyuv=y='bitand(val, 128+64+32)' |
Merge color channel components from several video streams.
The filter accepts up to 4 input streams, and merge selected input planes to the output video.
This filter accepts the following options:
Set input to output plane mapping. Default is 0.
The mappings is specified as a bitmap. It should be specified as a hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. ’Aa’ describes the mapping for the first plane of the output stream. ’A’ sets the number of the input stream to use (from 0 to 3), and ’a’ the plane number of the corresponding input to use (from 0 to 3). The rest of the mappings is similar, ’Bb’ describes the mapping for the output stream second plane, ’Cc’ describes the mapping for the output stream third plane and ’Dd’ describes the mapping for the output stream fourth plane.
Set output pixel format. Default is yuva444p.
[a0][a1][a2]mergeplanes=0x001020:yuv444p |
[a0][a1]mergeplanes=0x00010210:yuva444p |
format=yuva444p,mergeplanes=0x03010200:yuva444p |
format=yuv420p,mergeplanes=0x000201:yuv420p |
format=rgb24,mergeplanes=0x000102:yuv444p |
Apply motion-compensation deinterlacing.
It needs one field per frame as input and must thus be used together with yadif=1/3 or equivalent.
This filter accepts the following options:
Set the deinterlacing mode.
It accepts one of the following values:
use iterative motion estimation
like ‘slow’, but use multiple reference frames.
Default value is ‘fast’.
Set the picture field parity assumed for the input video. It must be one of the following values:
assume top field first
assume bottom field first
Default value is ‘bff’.
Set per-block quantization parameter (QP) used by the internal encoder.
Higher values should result in a smoother motion vector field but less optimal individual vectors. Default value is 1.
Apply an MPlayer filter to the input video.
This filter provides a wrapper around some of the filters of MPlayer/MEncoder.
This wrapper is considered experimental. Some of the wrapped filters may not work properly and we may drop support for them, as they will be implemented natively into FFmpeg. Thus you should avoid depending on them when writing portable scripts.
The filter accepts the parameters: filter_name[:=]filter_params
filter_name is the name of a supported MPlayer filter, filter_params is a string containing the parameters accepted by the named filter.
The list of the currently supported filters follows:
The parameter syntax and behavior for the listed filters are the same of the corresponding MPlayer filters. For detailed instructions check the "VIDEO FILTERS" section in the MPlayer manual.
mp=eq2=1.0:2:0.5 |
See also mplayer(1), http://www.mplayerhq.hu/.
Drop frames that do not differ greatly from the previous frame in order to reduce frame rate.
The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly.
A description of the accepted options follows.
Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped unregarding the number of previous sequentially dropped frames.
Default value is 0.
Set the dropping threshold values.
Values for ‘hi’ and ‘lo’ are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block.
A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of ‘hi’, and if no more than ‘frac’ blocks (1 meaning the whole image) differ by more than a threshold of ‘lo’.
Default value for ‘hi’ is 64*12, default value for ‘lo’ is 64*5, and default value for ‘frac’ is 0.33.
Negate input video.
This filter accepts an integer in input, if non-zero it negates the alpha component (if available). The default value in input is 0.
Force libavfilter not to use any of the specified pixel formats for the input to the next filter.
This filter accepts the following parameters:
A ’|’-separated list of pixel format names, for example "pix_fmts=yuv420p|monow|rgb24".
noformat=pix_fmts=yuv420p,vflip |
noformat=yuv420p|yuv444p|yuv410p |
Add noise on video input frame.
The filter accepts the following options:
Set noise seed for specific pixel component or all pixel components in case
of all_seed. Default value is 123457.
Set noise strength for specific pixel component or all pixel components in case
all_strength. Default value is 0. Allowed range is [0, 100].
Set pixel component flags or set flags for all components if all_flags. Available values for component flags are:
averaged temporal noise (smoother)
mix random noise with a (semi)regular pattern
temporal noise (noise pattern changes between frames)
uniform noise (gaussian otherwise)
Add temporal and uniform noise to input video:
noise=alls=20:allf=t+u |
Pass the video source unchanged to the output.
Apply video transform using libopencv.
To enable this filter install libopencv library and headers and
configure FFmpeg with --enable-libopencv.
This filter accepts the following parameters:
The name of the libopencv filter to apply.
The parameters to pass to the libopencv filter. If not specified the default values are assumed.
Refer to the official libopencv documentation for more precise information: http://opencv.willowgarage.com/documentation/c/image_filtering.html
Follows the list of supported libopencv filters.
Dilate an image by using a specific structuring element.
This filter corresponds to the libopencv function cvDilate.
It accepts the parameters: struct_el|nb_iterations.
struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape
cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element, and can be one of the values "rect", "cross", "ellipse", "custom".
If the value for shape is "custom", it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead.
The default value for struct_el is "3x3+0x0/rect".
nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.
Follow some example:
# use the default values ocv=dilate # dilate using a structuring element with a 5x5 cross, iterate two times ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2 # read the shape from the file diamond.shape, iterate two times # the file diamond.shape may contain a pattern of characters like this: # * # *** # ***** # *** # * # the specified cols and rows are ignored (but not the anchor point coordinates) ocv=dilate:0x0+2x2/custom=diamond.shape|2 |
Erode an image by using a specific structuring element.
This filter corresponds to the libopencv function cvErode.
The filter accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter.
Smooth the input video.
The filter takes the following parameters: type|param1|param2|param3|param4.
type is the type of smooth filter to apply, and can be one of the following values: "blur", "blur_no_scale", "median", "gaussian", "bilateral". The default value is "gaussian".
param1, param2, param3, and param4 are parameters whose meanings depend on smooth type. param1 and param2 accept integer positive values or 0, param3 and param4 accept float values.
The default value for param1 is 3, the default value for the other parameters is 0.
These parameters correspond to the parameters assigned to the
libopencv function cvSmooth.
Overlay one video on top of another.
It takes two inputs and one output, the first input is the "main" video on which the second input is overlayed.
This filter accepts the following parameters:
A description of the accepted options follows.
Set the expression for the x and y coordinates of the overlayed video on the main video. Default value is "0" for both expressions. In case the expression is invalid, it is set to a huge value (meaning that the overlay will not be displayed within the output visible area).
Set when the expressions for ‘x’, and ‘y’ 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 ‘frame’.
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
Set the format for the output video.
It accepts the following values:
force YUV420 output
force YUV444 output
force RGB output
Default value is ‘yuv420’.
If set to 1, force the filter to accept inputs in the RGB color space. Default value is 0. This option is deprecated, use ‘format’ instead.
If set to 1, force the filter to draw the last overlay frame over the main input until the end of the stream. A value of 0 disables this behavior. Default value is 1.
The ‘x’, and ‘y’ expressions can contain the following parameters.
main input width and height
overlay input width and height
the computed values for x and y. They are evaluated for each new frame.
horizontal and vertical chroma subsample values of the output format. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the number of input frame, starting from 0
the position in the file of the input frame, NAN if unknown
timestamp expressed in seconds, NAN if the input timestamp is unknown
Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate to NAN when ‘eval’ is set to ‘init’.
Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as it does the example for the movie filter.
You can chain together more overlays but you should test the efficiency of such approach.
This filter supports the following commands:
Modify the x and y of the overlay input. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
overlay=main_w-overlay_w-10:main_h-overlay_h-10 |
Using named options the example above becomes:
overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10 |
ffmpeg tool with the -filter_complex option:
ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output |
ffmpeg tool:
ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output |
WxH
must specify the size of the main input to the overlay filter:
color=color=red@.3:size=WxH [over]; [in][over] overlay [out] |
ffplay tool:
ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' |
The above command is the same as:
ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' |
overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0 |
ffmpeg -i left.avi -i right.avi -filter_complex " nullsrc=size=200x100 [background]; [0:v] setpts=PTS-STARTPTS, scale=100x100 [left]; [1:v] setpts=PTS-STARTPTS, scale=100x100 [right]; [background][left] overlay=shortest=1 [background+left]; [background+left][right] overlay=shortest=1:x=100 [left+right] " |
nullsrc=s=200x200 [bg]; testsrc=s=100x100, split=4 [in0][in1][in2][in3]; [in0] lutrgb=r=0, [bg] overlay=0:0 [mid0]; [in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1]; [in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2]; [in3] null, [mid2] overlay=100:100 [out0] |
Apply Overcomplete Wavelet denoiser.
The filter accepts the following options:
Set depth.
Larger depth values will denoise lower frequency components more, but slow down filtering.
Must be an int in the range 8-16, default is 8.
Set luma strength.
Must be a double value in the range 0-1000, default is 1.0.
Set chroma strength.
Must be a double value in the range 0-1000, default is 1.0.
Add paddings to the input image, and place the original input at the given coordinates x, y.
This filter accepts the following parameters:
Specify an expression for the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output.
The width expression can reference the value set by the height expression, and vice versa.
The default value of width and height is 0.
Specify an expression for the offsets where to place the input image in the padded area with respect to the top/left border of the output image.
The x expression can reference the value set by the y expression, and vice versa.
The default value of x and y is 0.
Specify the color of the padded area. For the syntax of this option, check the "Color" section in the ffmpeg-utils manual.
The default value of color is "black".
The value for the width, height, x, and y options are expressions containing the following constants:
the input video width and height
same as in_w and in_h
the output width and height, that is the size of the padded area as specified by the width and height expressions
same as out_w and out_h
x and y offsets as specified by the x and y expressions, or NAN if not yet specified
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.
pad=640:480:0:40:violet |
The example above is equivalent to the following command:
pad=width=640:height=480:x=0:y=40:color=violet |
pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" |
pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" |
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" |
(ih * X / ih) * sar = output_dar X = output_dar / sar |
Thus the previous example needs to be modified to:
pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" |
pad="2*iw:2*ih:ow-iw:oh-ih" |
Correct perspective of video not recorded perpendicular to the screen.
A description of the accepted parameters follows.
Set coordinates expression for top left, top right, bottom left and bottom right corners.
Default values are 0:0:W:0:0:H:W:H with which perspective will remain unchanged.
The expressions can use the following variables:
the width and height of video frame.
Set interpolation for perspective correction.
It accepts the following values:
Default value is ‘linear’.
Delay interlaced video by one field time so that the field order changes.
The intended use is to fix PAL movies that have been captured with the opposite field order to the film-to-video transfer.
A description of the accepted parameters follows.
Set phase mode.
It accepts the following values:
Capture field order top-first, transfer bottom-first. Filter will delay the bottom field.
Capture field order bottom-first, transfer top-first. Filter will delay the top field.
Capture and transfer with the same field order. This mode only exists for the documentation of the other options to refer to, but if you actually select it, the filter will faithfully do nothing.
Capture field order determined automatically by field flags, transfer opposite. Filter selects among ‘t’ and ‘b’ modes on a frame by frame basis using field flags. If no field information is available, then this works just like ‘u’.
Capture unknown or varying, transfer opposite. Filter selects among ‘t’ and ‘b’ on a frame by frame basis by analyzing the images and selecting the alternative that produces best match between the fields.
Capture top-first, transfer unknown or varying. Filter selects among ‘t’ and ‘p’ using image analysis.
Capture bottom-first, transfer unknown or varying. Filter selects among ‘b’ and ‘p’ using image analysis.
Capture determined by field flags, transfer unknown or varying. Filter selects among ‘t’, ‘b’ and ‘p’ using field flags and image analysis. If no field information is available, then this works just like ‘U’. This is the default mode.
Both capture and transfer unknown or varying. Filter selects among ‘t’, ‘b’ and ‘p’ using image analysis only.
Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video.
For example:
format=monow, pixdesctest |
can be used to test the monowhite pixel format descriptor definition.
Enable the specified chain of postprocessing subfilters using libpostproc. This
library should be automatically selected with a GPL build (--enable-gpl).
Subfilters must be separated by ’/’ and can be disabled by prepending a ’-’.
Each subfilter and some options have a short and a long name that can be used
interchangeably, i.e. dr/dering are the same.
The filters accept the following options:
Set postprocessing subfilters string.
All subfilters share common options to determine their scope:
Honor the quality commands for this subfilter.
Do chrominance filtering, too (default).
Do luminance filtering only (no chrominance).
Do chrominance filtering only (no luminance).
These options can be appended after the subfilter name, separated by a ’|’.
Available subfilters are:
Horizontal deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Vertical deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Accurate horizontal deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
Accurate vertical deblocking filter
Difference factor where higher values mean more deblocking (default: 32).
Flatness threshold where lower values mean more deblocking (default: 39).
The horizontal and vertical deblocking filters share the difference and flatness values so you cannot set different horizontal and vertical thresholds.
Experimental horizontal deblocking filter
Experimental vertical deblocking filter
Deringing filter
larger -> stronger filtering
larger -> stronger filtering
larger -> stronger filtering
Stretch luminance to 0-255.
Linear blend deinterlacing filter that deinterlaces the given block by
filtering all lines with a (1 2 1) filter.
Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating every second line.
Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating every second line.
Median deinterlacing filter that deinterlaces the given block by applying a median filter to every second line.
FFmpeg deinterlacing filter that deinterlaces the given block by filtering every
second line with a (-1 4 2 4 -1) filter.
Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given
block by filtering all lines with a (-1 2 6 2 -1) filter.
Overrides the quantizer table from the input with the constant quantizer you specify.
Quantizer to use
Default pp filter combination (hb|a,vb|a,dr|a)
Fast pp filter combination (h1|a,v1|a,dr|a)
High quality pp filter combination (ha|a|128|7,va|a,dr|a)
pp=hb/vb/dr/al |
pp=de/-al |
pp=default/tmpnoise|1|2|3 |
pp=hb|y/vb|a |
Obtain the average, maximum and minimum PSNR (Peak Signal to Noise Ratio) between two input videos.
This filter takes in input two input videos, the first input is considered the "main" source and is passed unchanged to the output. The second input is used as a "reference" video for computing the PSNR.
Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also it assumes that both inputs have the same number of frames, which are compared one by one.
The obtained average PSNR is printed through the logging system.
The filter stores the accumulated MSE (mean squared error) of each frame, and at the end of the processing it is averaged across all frames equally, and the following formula is applied to obtain the PSNR:
PSNR = 10*log10(MAX^2/MSE) |
Where MAX is the average of the maximum values of each component of the image.
The description of the accepted parameters follows.
If specified the filter will use the named file to save the PSNR of each individual frame.
The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value for each compared couple of frames.
A description of each shown parameter follows:
sequential number of the input frame, starting from 1
Mean Square Error pixel-by-pixel average difference of the compared frames, averaged over all the image components.
Mean Square Error pixel-by-pixel average difference of the compared frames for the component specified by the suffix.
Peak Signal to Noise ratio of the compared frames for the component specified by the suffix.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main]; [main][ref] psnr="stats_file=stats.log" [out] |
On this example the input file being processed is compared with the reference file ‘ref_movie.mpg’. The PSNR of each individual frame is stored in ‘stats.log’.
Pulldown reversal (inverse telecine) filter, capable of handling mixed hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps progressive content.
The pullup filter is designed to take advantage of future context in making its decisions. This filter is stateless in the sense that it does not lock onto a pattern to follow, but it instead looks forward to the following fields in order to identify matches and rebuild progressive frames.
To produce content with an even framerate, insert the fps filter after
pullup, use fps=24000/1001 if the input frame rate is 29.97fps,
fps=24 for 30fps and the (rare) telecined 25fps input.
The filter accepts the following options:
These options set the amount of "junk" to ignore at the left, right, top, and bottom of the image, respectively. Left and right are in units of 8 pixels, while top and bottom are in units of 2 lines. The default is 8 pixels on each side.
Set the strict breaks. Setting this option to 1 will reduce the chances of
filter generating an occasional mismatched frame, but it may also cause an
excessive number of frames to be dropped during high motion sequences.
Conversely, setting it to -1 will make filter match fields more easily.
This may help processing of video where there is slight blurring between
the fields, but may also cause there to be interlaced frames in the output.
Default value is 0.
Set the metric plane to use. It accepts the following values:
Use luma plane.
Use chroma blue plane.
Use chroma red plane.
This option may be set to use chroma plane instead of the default luma plane for doing filter’s computations. This may improve accuracy on very clean source material, but more likely will decrease accuracy, especially if there is chroma noise (rainbow effect) or any grayscale video. The main purpose of setting ‘mp’ to a chroma plane is to reduce CPU load and make pullup usable in realtime on slow machines.
For best results (without duplicated frames in the output file) it is necessary to change the output frame rate. For example, to inverse telecine NTSC input:
ffmpeg -i input -vf pullup -r 24000/1001 ... |
Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by filling in the pixels that comprise the logo with neighboring pixels.
The filter accepts the following options:
Set the filter bitmap file, which can be any image format supported by libavformat. The width and height of the image file must match those of the video stream being processed.
Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest, you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black frame with the logo visible, and then using a threshold filter followed by the erode filter once or twice.
If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much, but it will increase the amount of blurring needed to cover over the image and will destroy more information than necessary, and extra pixels will slow things down on a large logo.
Rotate video by an arbitrary angle expressed in radians.
The filter accepts the following options:
A description of the optional parameters follows.
Set an expression for the angle by which to rotate the input video clockwise, expressed as a number of radians. A negative value will result in a counter-clockwise rotation. By default it is set to "0".
This expression is evaluated for each frame.
Set the output width expression, default value is "iw". This expression is evaluated just once during configuration.
Set the output height expression, default value is "ih". This expression is evaluated just once during configuration.
Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is 1.
Set the color used to fill the output area not covered by the rotated image. For the generalsyntax of this option, check the "Color" section in the ffmpeg-utils manual. If the special value "none" is selected then no background is printed (useful for example if the background is never shown).
Default value is "black".
The expressions for the angle and the output size can contain the following constants and functions:
sequential number of the input frame, starting from 0. It is always NAN before the first frame is filtered.
time in seconds of the input frame, it is set to 0 when the filter is configured. It is always NAN before the first frame is filtered.
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the input video width and heigth
the output width and heigth, that is the size of the padded area as specified by the width and height expressions
the minimal width/height required for completely containing the input video rotated by a radians.
These are only available when computing the ‘out_w’ and ‘out_h’ expressions.
rotate=PI/6 |
rotate=-PI/6 |
rotate=PI/3+2*PI*t/T |
rotate=A*sin(2*PI/T*t) |
rotate='2*PI*t:ow=hypot(iw,ih):oh=ow' |
rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none |
The filter supports the following commands:
Set the angle expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
Apply Shape Adaptive Blur.
The filter accepts the following options:
Set luma blur filter strength, must be a value in range 0.1-4.0, default value is 1.0. A greater value will result in a more blurred image, and in slower processing.
Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is 1.0.
Set luma maximum difference between pixels to still be considered, must be a value in the 0.1-100.0 range, default value is 1.0.
Set chroma blur filter strength, must be a value in range 0.1-4.0. A greater value will result in a more blurred image, and in slower processing.
Set chroma pre-filter radius, must be a value in the 0.1-2.0 range.
Set chroma maximum difference between pixels to still be considered, must be a value in the 0.1-100.0 range.
Each chroma option value, if not explicitly specified, is set to the corresponding luma option value.
Scale (resize) the input video, using the libswscale library.
The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.
If the input image format is different from the format requested by the next filter, the scale filter will convert the input to the requested format.
The filter accepts the following options, or any of the options supported by the libswscale scaler.
See (ffmpeg-scaler)scaler_options for the complete list of scaler options.
Set the output video dimension expression. Default value is the input dimension.
If the value is 0, the input width is used for the output.
If one of the values is -1, the scale filter will use a value that maintains the aspect ratio of the input image, calculated from the other specified dimension. If both of them are -1, the input size is used
See below for the list of accepted constants for use in the dimension expression.
Set the interlacing mode. It accepts the following values:
Force interlaced aware scaling.
Do not apply interlaced scaling.
Select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not.
Default value is ‘0’.
Set libswscale scaling flags. See (ffmpeg-scaler)sws_flags for the complete list of values. If not explictly specified the filter applies the default flags.
Set the video size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
Set in/output YCbCr color space type.
This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder.
If not specified, the color space type depends on the pixel format.
Possible values:
Choose automatically.
Format conforming to International Telecommunication Union (ITU) Recommendation BT.709.
Set color space conforming to the United States Federal Communications Commission (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a).
Set color space conforming to:
Set color space conforming to SMPTE ST 240:1999.
Set in/output YCbCr sample range.
This allows the autodetected value to be overridden as well as allows forcing a specific value used for the output and encoder. If not specified, the range depends on the pixel format. Possible values:
Choose automatically.
Set full range (0-255 in case of 8-bit luma).
Set "MPEG" range (16-235 in case of 8-bit luma).
Enable decreasing or increasing output video width or height if necessary to keep the original aspect ratio. Possible values:
Scale the video as specified and disable this feature.
The output video dimensions will automatically be decreased if needed.
The output video dimensions will automatically be increased if needed.
One useful instance of this option is that when you know a specific device’s maximum allowed resolution, you can use this to limit the output video to that, while retaining the aspect ratio. For example, device A allows 1280x720 playback, and your video is 1920x800. Using this option (set it to decrease) and specifying 1280x720 to the command line makes the output 1280x533.
Please note that this is a different thing than specifying -1 for ‘w’ or ‘h’, you still need to specify the output resolution for this option to work.
The values of the ‘w’ and ‘h’ options are expressions containing the following constants:
the input width and height
same as in_w and in_h
the output (scaled) width and height
same as out_w and out_h
same as iw / ih
input sample aspect ratio
input display aspect ratio. Calculated from (iw / ih) * sar.
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
scale=w=200:h=100 |
This is equivalent to:
scale=200:100 |
or:
scale=200x100 |
scale=qcif |
which can also be written as:
scale=size=qcif |
scale=w=2*iw:h=2*ih |
scale=2*in_w:2*in_h |
scale=2*iw:2*ih:interl=1 |
scale=w=iw/2:h=ih/2 |
scale=3/2*iw:ow |
scale=iw:1/PHI*iw scale=ih*PHI:ih |
scale=w=3/2*oh:h=3/5*ih |
scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" |
scale=w='min(500\, iw*3/2):h=-1' |
The separatefields takes a frame-based video input and splits
each frame into its components fields, producing a new half height clip
with twice the frame rate and twice the frame count.
This filter use field-dominance information in frame to decide which
of each pair of fields to place first in the output.
If it gets it wrong use setfield filter before separatefields filter.
The setdar filter sets the Display Aspect Ratio for the filter
output video.
This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following equation:
DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR |
Keep in mind that the setdar filter does not modify the pixel
dimensions of the video frame. Also the display aspect ratio set by
this filter may be changed by later filters in the filterchain,
e.g. in case of scaling or if another "setdar" or a "setsar" filter is
applied.
The setsar filter sets the Sample (aka Pixel) Aspect Ratio for
the filter output video.
Note that as a consequence of the application of this filter, the output display aspect ratio will change according to the equation above.
Keep in mind that the sample aspect ratio set by the setsar
filter may be changed by later filters in the filterchain, e.g. if
another "setsar" or a "setdar" filter is applied.
The filters accept the following options:
setdar only), sar (setsar only)’Set the aspect ratio used by the filter.
The parameter can be a floating point number string, an expression, or
a string of the form num:den, where num and
den are the numerator and denominator of the aspect ratio. If
the parameter is not specified, it is assumed the value "0".
In case the form "num:den" is used, the : character
should be escaped.
Set the maximum integer value to use for expressing numerator and
denominator when reducing the expressed aspect ratio to a rational.
Default value is 100.
The parameter sar is an expression containing the following constants:
the corresponding mathematical approximated values for e (euler number), pi (greek PI), phi (golden ratio)
the input width and height
same as w / h
input sample aspect ratio
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.
setdar=dar=1.77777 setdar=dar=16/9 setdar=dar=1.77777 |
setsar=sar=10/11 |
setdar=ratio=16/9:max=1000 |
Force field for the output video frame.
The setfield filter marks the interlace type field for the
output frames. It does not change the input frame, but only sets the
corresponding property, which affects how the frame is treated by
following filters (e.g. fieldorder or yadif).
The filter accepts the following options:
Available values are:
Keep the same field property.
Mark the frame as bottom-field-first.
Mark the frame as top-field-first.
Mark the frame as progressive.
Show a line containing various information for each input video frame. The input video is not modified.
The shown line contains a sequence of key/value pairs of the form key:value.
A description of each shown parameter follows:
sequential number of the input frame, starting from 0
Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.
Presentation TimeStamp of the input frame, expressed as a number of seconds
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic video)
pixel format name
sample aspect ratio of the input frame, expressed in the form num/den
size of the input frame. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first)
1 if the frame is a key frame, 0 otherwise
picture type of the input frame ("I" for an I-frame, "P" for a
P-frame, "B" for a B-frame, "?" for unknown type).
Check also the documentation of the AVPictureType enum and of
the av_get_picture_type_char function defined in
‘libavutil/avutil.h’.
Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame
Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]"
Blur the input video without impacting the outlines.
The filter accepts the following options:
Set the luma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
Set the luma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is 1.0.
Set the luma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is 0.
Set the chroma radius. The option value must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
Set the chroma strength. The option value must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image. Default value is 1.0.
Set the chroma threshold used as a coefficient to determine whether a pixel should be blurred or not. The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges. Default value is 0.
If a chroma option is not explicitly set, the corresponding luma value is set.
Convert between different stereoscopic image formats.
The filters accept the following options:
Set stereoscopic image format of input.
Available values for input image formats are:
side by side parallel (left eye left, right eye right)
side by side crosseye (right eye left, left eye right)
side by side parallel with half width resolution (left eye left, right eye right)
side by side crosseye with half width resolution (right eye left, left eye right)
above-below (left eye above, right eye below)
above-below (right eye above, left eye below)
above-below with half height resolution (left eye above, right eye below)
above-below with half height resolution (right eye above, left eye below)
alternating frames (left eye first, right eye second)
alternating frames (right eye first, left eye second)
Default value is ‘sbsl’.
Set stereoscopic image format of output.
Available values for output image formats are all the input formats as well as:
anaglyph red/blue gray (red filter on left eye, blue filter on right eye)
anaglyph red/green gray (red filter on left eye, green filter on right eye)
anaglyph red/cyan gray (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan half colored (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan color (red filter on left eye, cyan filter on right eye)
anaglyph red/cyan color optimized with the least squares projection of dubois (red filter on left eye, cyan filter on right eye)
anaglyph green/magenta gray (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta half colored (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta colored (green filter on left eye, magenta filter on right eye)
anaglyph green/magenta color optimized with the least squares projection of dubois (green filter on left eye, magenta filter on right eye)
anaglyph yellow/blue gray (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue half colored (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue colored (yellow filter on left eye, blue filter on right eye)
anaglyph yellow/blue color optimized with the least squares projection of dubois (yellow filter on left eye, blue filter on right eye)
interleaved rows (left eye has top row, right eye starts on next row)
interleaved rows (right eye has top row, left eye starts on next row)
mono output (left eye only)
mono output (right eye only)
Default value is ‘arcd’.
stereo3d=sbsl:aybd |
stereo3d=abl:sbsr |
Apply a simple postprocessing filter that compresses and decompresses the image
at several (or - in the case of ‘quality’ level 6 - all) shifts
and average the results.
The filter accepts the following options:
Set quality. This option defines the number of levels for averaging. It accepts
an integer in the range 0-6. If set to 0, the filter will have no
effect. A value of 6 means the higher quality. For each increment of
that value the speed drops by a factor of approximately 2. Default value is
3.
Force a constant quantization parameter. If not set, the filter will use the QP from the video stream (if available).
Set thresholding mode. Available modes are:
Set hard thresholding (default).
Set soft thresholding (better de-ringing effect, but likely blurrier).
Enable the use of the QP from the B-Frames if set to 1. Using this
option may cause flicker since the B-Frames have often larger QP. Default is
0 (not enabled).
Draw subtitles on top of input video using the libass library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libass. This filter also requires a build with libavcodec and
libavformat to convert the passed subtitles file to ASS (Advanced Substation
Alpha) subtitles format.
The filter accepts the following options:
Set the filename of the subtitle file to read. It must be specified.
Specify the size of the original video, the video for which the ASS file was composed. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been changed.
Set subtitles input character encoding. subtitles filter only. Only
useful if not UTF-8.
If the first key is not specified, it is assumed that the first value specifies the ‘filename’.
For example, to render the file ‘sub.srt’ on top of the input video, use the command:
subtitles=sub.srt |
which is equivalent to:
subtitles=filename=sub.srt |
Scale the input by 2x and smooth using the Super2xSaI (Scale and Interpolate) pixel art scaling algorithm.
Useful for enlarging pixel art images without reducing sharpness.
Swap U & V plane.
Apply telecine process to the video.
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.
Some typical patterns:
NTSC output (30i):
27.5p: 32222
24p: 23 (classic)
24p: 2332 (preferred)
20p: 33
18p: 334
16p: 3444
PAL output (25i):
27.5p: 12222
24p: 222222222223 ("Euro pulldown")
16.67p: 33
16p: 33333334
|
Select the most representative frame in a given sequence of consecutive frames.
The filter accepts the following options:
Set the frames batch size to analyze; in a set of n frames, the filter
will pick one of them, and then handle the next batch of n frames until
the end. Default is 100.
Since the filter keeps track of the whole frames sequence, a bigger n value will result in a higher memory usage, so a high value is not recommended.
thumbnail=50 |
ffmpeg:
ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png |
Tile several successive frames together.
The filter accepts the following options:
Set the grid size (i.e. the number of lines and columns). For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
Set the maximum number of frames to render in the given area. It must be less
than or equal to wxh. The default value is 0, meaning all
the area will be used.
Set the outer border margin in pixels.
Set the inner border thickness (i.e. the number of pixels between frames). For more advanced padding options (such as having different values for the edges), refer to the pad video filter.
Specify the color of the unused areaFor the syntax of this option, check the "Color" section in the ffmpeg-utils manual. The default value of color is "black".
ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png |
The ‘-vsync 0’ is necessary to prevent ffmpeg from
duplicating each output frame to accomodate the originally detected frame
rate.
5 pictures in an area of 3x2 frames,
with 7 pixels between them, and 2 pixels of initial margin, using
mixed flat and named options:
tile=3x2:nb_frames=5:padding=7:margin=2 |
Perform various types of temporal field interlacing.
Frames are counted starting from 1, so the first input frame is considered odd.
The filter accepts the following options:
Specify the mode of the interlacing. This option can also be specified as a value alone. See below for a list of values for this option.
Available values are:
Move odd frames into the upper field, even into the lower field, generating a double height frame at half frame rate.
Only output even frames, odd frames are dropped, generating a frame with unchanged height at half frame rate.
Only output odd frames, even frames are dropped, generating a frame with unchanged height at half frame rate.
Expand each frame to full height, but pad alternate lines with black, generating a frame with double height at the same input frame rate.
Interleave the upper field from odd frames with the lower field from even frames, generating a frame with unchanged height at half frame rate.
Interleave the lower field from odd frames with the upper field from even frames, generating a frame with unchanged height at half frame rate.
Double frame rate with unchanged height. Frames are inserted each containing the second temporal field from the previous input frame and the first temporal field from the next input frame. This mode relies on the top_field_first flag. Useful for interlaced video displays with no field synchronisation.
Numeric values are deprecated but are accepted for backward compatibility reasons.
Default mode is merge.
Specify flags influencing the filter process.
Available value for flags is:
Enable vertical low-pass filtering in the filter. Vertical low-pass filtering is required when creating an interlaced destination from a progressive source which contains high-frequency vertical detail. Filtering will reduce interlace ’twitter’ and Moire patterning.
Vertical low-pass filtering can only be enabled for ‘mode’ interleave_top and interleave_bottom.
Transpose rows with columns in the input video and optionally flip it.
This filter accepts the following options:
Specify the transposition direction.
Can assume the following values:
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:
L.R L.l . . -> . . l.r R.r |
Rotate by 90 degrees clockwise, that is:
L.R l.L . . -> . . l.r r.R |
Rotate by 90 degrees counterclockwise, that is:
L.R R.r . . -> . . l.r L.l |
Rotate by 90 degrees clockwise and vertically flip, that is:
L.R r.R . . -> . . l.r l.L |
For values between 4-7, the transposition is only done if the input
video geometry is portrait and not landscape. These values are
deprecated, the passthrough option should be used instead.
Numerical values are deprecated, and should be dropped in favor of symbolic constants.
Do not apply the transposition if the input geometry matches the one specified by the specified value. It accepts the following values:
Always apply transposition.
Preserve portrait geometry (when height >= width).
Preserve landscape geometry (when width >= height).
Default value is none.
For example to rotate by 90 degrees clockwise and preserve portrait layout:
transpose=dir=1:passthrough=portrait |
The command above can also be specified as:
transpose=1:portrait |
Trim the input so that the output contains one continuous subpart of the input.
This filter accepts the following options:
Specify time of the start of the kept section, i.e. the frame with the timestamp start will be the first frame in the output.
Specify time of the first frame that will be dropped, i.e. the frame immediately preceding the one with the timestamp end will be the last frame in the output.
Same as start, except this option sets the start timestamp in timebase units instead of seconds.
Same as end, except this option sets the end timestamp in timebase units instead of seconds.
Specify maximum duration of the output.
Number of the first frame that should be passed to output.
Number of the first frame that should be dropped.
‘start’, ‘end’, ‘duration’ are expressed as time duration specifications, check the "Time duration" section in the ffmpeg-utils manual.
Note that the first two sets of the start/end options and the ‘duration’ option look at the frame timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that this filter does not modify the timestamps. If you wish that the output timestamps start at zero, insert a setpts filter after the trim filter.
If multiple start or end options are set, this filter tries to be greedy and keep all the frames that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple trim filters.
The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time.
Examples:
ffmpeg -i INPUT -vf trim=60:120 |
ffmpeg -i INPUT -vf trim=duration=1 |
Sharpen or blur the input video.
It accepts the following parameters:
Set the luma matrix horizontal size. It must be an odd integer between 3 and 63, default value is 5.
Set the luma matrix vertical size. It must be an odd integer between 3 and 63, default value is 5.
Set the luma effect strength. It can be a float number, reasonable values lay between -1.5 and 1.5.
Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.
Default value is 1.0.
Set the chroma matrix horizontal size. It must be an odd integer between 3 and 63, default value is 5.
Set the chroma matrix vertical size. It must be an odd integer between 3 and 63, default value is 5.
Set the chroma effect strength. It can be a float number, reasonable values lay between -1.5 and 1.5.
Negative values will blur the input video, while positive values will sharpen it, a value of zero will disable the effect.
Default value is 0.0.
If set to 1, specify using OpenCL capabilities, only available if
FFmpeg was configured with --enable-opencl. Default value is 0.
All parameters are optional and default to the equivalent of the string ’5:5:1.0:5:5:0.0’.
unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5 |
unsharp=7:7:-2:7:7:-2 |
Analyze video stabilization/deshaking. Perform pass 1 of 2, see vidstabtransform for pass 2.
This filter generates a file with relative translation and rotation transform information about subsequent frames, which is then used by the vidstabtransform filter.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libvidstab.
This filter accepts the following options:
Set the path to the file used to write the transforms information. Default value is ‘transforms.trf’.
Set how shaky the video is and how quick the camera is. It accepts an integer in the range 1-10, a value of 1 means little shakiness, a value of 10 means strong shakiness. Default value is 5.
Set the accuracy of the detection process. It must be a value in the range 1-15. A value of 1 means low accuracy, a value of 15 means high accuracy. Default value is 9.
Set stepsize of the search process. The region around minimum is scanned with 1 pixel resolution. Default value is 6.
Set minimum contrast. Below this value a local measurement field is discarded. Must be a floating point value in the range 0-1. Default value is 0.3.
Set reference frame number for tripod mode.
If enabled, the motion of the frames is compared to a reference frame in the filtered stream, identified by the specified number. The idea is to compensate all movements in a more-or-less static scene and keep the camera view absolutely still.
If set to 0, it is disabled. The frames are counted starting from 1.
Show fields and transforms in the resulting frames. It accepts an integer in the range 0-2. Default value is 0, which disables any visualization.
vidstabdetect |
vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf" |
vidstabdetect=show=1 |
ffmpeg:
ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi |
Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass 1.
Read a file with transform information for each frame and apply/compensate them. Together with the vidstabdetect filter this can be used to deshake videos. See also http://public.hronopik.de/vid.stab. It is important to also use the unsharp filter, see below.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libvidstab.
This filter accepts the following options:
path to the file used to read the transforms (default: ‘transforms.trf’)
number of frames (value*2 + 1) used for lowpass filtering the camera movements (default: 10). For example a number of 10 means that 21 frames are used (10 in the past and 10 in the future) to smoothen the motion in the video. A larger values leads to a smoother video, but limits the acceleration of the camera (pan/tilt movements).
maximal number of pixels to translate frames (default: -1 no limit)
maximal angle in radians (degree*PI/180) to rotate frames (default: -1 no limit)
How to deal with borders that may be visible due to movement compensation. Available values are:
keep image information from previous frame (default)
fill the border black
keep transforms normal (default)
invert transforms
consider transforms as
absolute
relative to previous frame (default)
percentage to zoom (default: 0)
zoom in
zoom out
set optimal zooming to avoid borders
disabled
optimal static zoom value is determined (only very strong movements will lead to visible borders) (default)
optimal adaptive zoom value is determined (no borders will be visible)
Note that the value given at zoom is added to the one calculated here.
type of interpolation
Available values are:
no interpolation
linear only horizontal
linear in both directions (default)
cubic in both directions (slow)
virtual tripod mode means that the video is stabilized such that the
camera stays stationary. Use also tripod option of
vidstabdetect.
off (default)
virtual tripod mode: equivalent to relative=0:smoothing=0
ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg |
vidstabtransform=zoom=5:input="mytransforms.trf" |
vidstabtransform=smoothing=30 |
Flip the input video vertically.
For example, to vertically flip a video with ffmpeg:
ffmpeg -i in.avi -vf "vflip" out.avi |
Make or reverse a natural vignetting effect.
The filter accepts the following options:
Set lens angle expression as a number of radians.
The value is clipped in the [0,PI/2] range.
Default value: "PI/5"
Set center coordinates expressions. Respectively "w/2" and "h/2"
by default.
Set forward/backward mode.
Available modes are:
The larger the distance from the central point, the darker the image becomes.
The larger the distance from the central point, the brighter the image becomes. This can be used to reverse a vignette effect, though there is no automatic detection to extract the lens ‘angle’ and other settings (yet). It can also be used to create a burning effect.
Default value is ‘forward’.
Set evaluation mode for the expressions (‘angle’, ‘x0’, ‘y0’).
It accepts the following values:
Evaluate expressions only once during the filter initialization.
Evaluate expressions for each incoming frame. This is way slower than the ‘init’ mode since it requires all the scalers to be re-computed, but it allows advanced dynamic expressions.
Default value is ‘init’.
Set dithering to reduce the circular banding effects. Default is 1
(enabled).
Set vignette aspect. This setting allows to adjust the shape of the vignette. Setting this value to the SAR of the input will make a rectangular vignetting following the dimensions of the video.
Default is 1/1.
The ‘alpha’, ‘x0’ and ‘y0’ expressions can contain the following parameters.
input width and height
the number of input frame, starting from 0
the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in TB units, NAN if undefined
frame rate of the input video, NAN if the input frame rate is unknown
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
time base of the input video
vignette=PI/4 |
vignette='PI/4+random(1)*PI/50':eval=frame |
Deinterlace the input video ("w3fdif" stands for "Weston 3 Field Deinterlacing Filter").
Based on the process described by Martin Weston for BBC R&D, and implemented based on the de-interlace algorithm written by Jim Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses filter coefficients calculated by BBC R&D.
There are two sets of filter coefficients, so called "simple": and "complex". Which set of filter coefficients is used can be set by passing an optional parameter:
Set the interlacing filter coefficients. Accepts one of the following values:
Simple filter coefficient set.
More-complex filter coefficient set.
Default value is ‘complex’.
Specify which frames to deinterlace. Accept one of the following values:
Deinterlace all frames,
Only deinterlace frames marked as interlaced.
Default value is ‘all’.
Deinterlace the input video ("yadif" means "yet another deinterlacing filter").
This filter accepts the following options:
The interlacing mode to adopt, accepts one of the following values:
output 1 frame for each frame
output 1 frame for each field
like send_frame but skip spatial interlacing check
like send_field but skip spatial interlacing check
Default value is send_frame.
The picture field parity assumed for the input interlaced video, accepts one of the following values:
assume top field first
assume bottom field first
enable automatic detection
Default value is auto.
If interlacing is unknown or decoder does not export this information,
top field first will be assumed.
Specify which frames to deinterlace. Accept one of the following values:
deinterlace all frames
only deinterlace frames marked as interlaced
Default value is all.
Below is a description of the currently available video sources.
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in ‘libavfilter/vsrc_buffer.h’.
This source accepts the following options:
Specify the size (width and height) of the buffered video frames. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
Input video width.
Input video height.
A string representing the pixel format of the buffered video frames. It may be a number corresponding to a pixel format, or a pixel format name.
Specify the timebase assumed by the timestamps of the buffered frames.
Specify the frame rate expected for the video stream.
Specify the sample aspect ratio assumed by the video frames.
Specify the optional parameters to be used for the scale filter which is automatically inserted when an input change is detected in the input size or format.
For example:
buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1 |
will instruct the source to accept video frames with size 320x240 and with format "yuv410p", assuming 1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). Since the pixel format with name "yuv410p" corresponds to the number 6 (check the enum AVPixelFormat definition in ‘libavutil/pixfmt.h’), this example corresponds to:
buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1 |
Alternatively, the options can be specified as a flat string, but this syntax is deprecated:
width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]
Create a pattern generated by an elementary cellular automaton.
The initial state of the cellular automaton can be defined through the ‘filename’, and ‘pattern’ options. If such options are not specified an initial state is created randomly.
At each new frame a new row in the video is filled with the result of the cellular automaton next generation. The behavior when the whole frame is filled is defined by the ‘scroll’ option.
This source accepts the following options:
Read the initial cellular automaton state, i.e. the starting row, from the specified file. In the file, each non-whitespace character is considered an alive cell, a newline will terminate the row, and further characters in the file will be ignored.
Read the initial cellular automaton state, i.e. the starting row, from the specified string.
Each non-whitespace character in the string is considered an alive cell, a newline will terminate the row, and further characters in the string will be ignored.
Set the video rate, that is the number of frames generated per second. Default is 25.
Set the random fill ratio for the initial cellular automaton row. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI.
This option is ignored when a file or a pattern is specified.
Set the seed for filling randomly the initial row, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.
Set the size of the output video. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
If ‘filename’ or ‘pattern’ is specified, the size is set by default to the width of the specified initial state row, and the height is set to width * PHI.
If ‘size’ is set, it must contain the width of the specified pattern string, and the specified pattern will be centered in the larger row.
If a filename or a pattern string is not specified, the size value defaults to "320x518" (used for a randomly generated initial state).
If set to 1, scroll the output upward when all the rows in the output have been already filled. If set to 0, the new generated row will be written over the top row just after the bottom row is filled. Defaults to 1.
If set to 1, completely fill the output with generated rows before outputting the first frame. This is the default behavior, for disabling set the value to 0.
If set to 1, stitch the left and right row edges together. This is the default behavior, for disabling set the value to 0.
cellauto=f=pattern:s=200x400 |
cellauto=ratio=2/3:s=200x200 |
cellauto=p=@:s=100x400:full=0:rule=18 |
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18 |
Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and start_y.
This source accepts the following options:
Set the terminal pts value. Default value is 400.
Set the terminal scale value. Must be a floating point value. Default value is 0.3.
Set the inner coloring mode, that is the algorithm used to draw the Mandelbrot fractal internal region.
It shall assume one of the following values:
Set black mode.
Show time until convergence.
Set color based on point closest to the origin of the iterations.
Set period mode.
Default value is mincol.
Set the bailout value. Default value is 10.0.
Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.
Set outer coloring mode. It shall assume one of following values:
Set iteration cound mode.
set normalized iteration count mode.
Default value is normalized_iteration_count.
Set frame rate, expressed as number of frames per second. Default value is "25".
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. Default value is "640x480".
Set the initial scale value. Default value is 3.0.
Set the initial x position. Must be a floating point value between -100 and 100. Default value is -0.743643887037158704752191506114774.
Set the initial y position. Must be a floating point value between -100 and 100. Default value is -0.131825904205311970493132056385139.
Generate various test patterns, as generated by the MPlayer test filter.
The size of the generated video is fixed, and is 256x256. This source is useful in particular for testing encoding features.
This source accepts the following options:
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
Set the video duration of the sourced video. The accepted syntax is:
[-]HH:MM:SS[.m...] [-]S+[.m...] |
See also the function av_parse_time().
If not specified, or the expressed duration is negative, the video is supposed to be generated forever.
Set the number or the name of the test to perform. Supported tests are:
Default value is "all", which will cycle through the list of all tests.
For example the following:
testsrc=t=dc_luma |
will generate a "dc_luma" test pattern.
Provide a frei0r source.
To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with --enable-frei0r.
This source accepts the following options:
The size of the video to generate. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
Framerate of the generated video, may be a string of the form num/den or a frame rate abbreviation.
The name to the frei0r source to load. For more information regarding frei0r and how to set the parameters read the section frei0r in the description of the video filters.
A ’|’-separated list of parameters to pass to the frei0r source.
For example, to generate a frei0r partik0l source with size 200x200 and frame rate 10 which is overlayed on the overlay filter main input:
frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay |
Generate a life pattern.
This source is based on a generalization of John Conway’s life game.
The sourced input represents a life grid, each pixel represents a cell which can be in one of two possible states, alive or dead. Every cell interacts with its eight neighbours, which are the cells that are horizontally, vertically, or diagonally adjacent.
At each interaction the grid evolves according to the adopted rule, which specifies the number of neighbor alive cells which will make a cell stay alive or born. The ‘rule’ option allows to specify the rule to adopt.
This source accepts the following options:
Set the file from which to read the initial grid state. In the file, each non-whitespace character is considered an alive cell, and newline is used to delimit the end of each row.
If this option is not specified, the initial grid is generated randomly.
Set the video rate, that is the number of frames generated per second. Default is 25.
Set the random fill ratio for the initial random grid. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI. It is ignored when a file is specified.
Set the seed for filling the initial random grid, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
Set the life rule.
A rule can be specified with a code of the kind "SNS/BNB", where NS and NB are sequences of numbers in the range 0-8, NS specifies the number of alive neighbor cells which make a live cell stay alive, and NB the number of alive neighbor cells which make a dead cell to become alive (i.e. to "born"). "s" and "b" can be used in place of "S" and "B", respectively.
Alternatively a rule can be specified by an 18-bits integer. The 9
high order bits are used to encode the next cell state if it is alive
for each number of neighbor alive cells, the low order bits specify
the rule for "borning" new cells. Higher order bits encode for an
higher number of neighbor cells.
For example the number 6153 = (12<<9)+9 specifies a stay alive
rule of 12 and a born rule of 9, which corresponds to "S23/B03".
Default value is "S23/B3", which is the original Conway’s game of life rule, and will keep a cell alive if it has 2 or 3 neighbor alive cells, and will born a new cell if there are three alive cells around a dead cell.
Set the size of the output video. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual.
If ‘filename’ is specified, the size is set by default to the same size of the input file. If ‘size’ is set, it must contain the size specified in the input file, and the initial grid defined in that file is centered in the larger resulting area.
If a filename is not specified, the size value defaults to "320x240" (used for a randomly generated initial grid).
If set to 1, stitch the left and right grid edges together, and the top and bottom edges also. Defaults to 1.
Set cell mold speed. If set, a dead cell will go from ‘death_color’ to ‘mold_color’ with a step of ‘mold’. ‘mold’ can have a value from 0 to 255.
Set the color of living (or new born) cells.
Set the color of dead cells. If ‘mold’ is set, this is the first color used to represent a dead cell.
Set mold color, for definitely dead and moldy cells.
For the syntax of these 3 color options, check the "Color" section in the ffmpeg-utils manual.
life=f=pattern:s=300x300 |
life=ratio=2/3:s=200x200 |
life=rule=S14/B34 |
ffplay:
ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 |
The color source provides an uniformly colored input.
The haldclutsrc source provides an identity Hald CLUT. See also
haldclut filter.
The nullsrc source returns unprocessed video frames. It is
mainly useful to be employed in analysis / debugging tools, or as the
source for filters which ignore the input data.
The rgbtestsrc source generates an RGB test pattern useful for
detecting RGB vs BGR issues. You should see a red, green and blue
stripe from top to bottom.
The smptebars source generates a color bars pattern, based on
the SMPTE Engineering Guideline EG 1-1990.
The smptehdbars source generates a color bars pattern, based on
the SMPTE RP 219-2002.
The testsrc source generates a test video pattern, showing a
color pattern, a scrolling gradient and a timestamp. This is mainly
intended for testing purposes.
The sources accept the following options:
Specify the color of the source, only available in the color
source. For the syntax of this option, check the "Color" section in the
ffmpeg-utils manual.
Specify the level of the Hald CLUT, only available in the haldclutsrc
source. A level of N generates a picture of N*N*N by N*N*N
pixels to be used as identity matrix for 3D lookup tables. Each component is
coded on a 1/(N*N) scale.
Specify the size of the sourced video. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. The default value is "320x240".
This option is not available with the haldclutsrc filter.
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
Set the sample aspect ratio of the sourced video.
Set the video duration of the sourced video. The accepted syntax is:
[-]HH[:MM[:SS[.m...]]] [-]S+[.m...] |
See also the function av_parse_time().
If not specified, or the expressed duration is negative, the video is supposed to be generated forever.
Set the number of decimals to show in the timestamp, only available in the
testsrc source.
The displayed timestamp value will correspond to the original timestamp value multiplied by the power of 10 of the specified value. Default value is 0.
For example the following:
testsrc=duration=5.3:size=qcif:rate=10 |
will generate a video with a duration of 5.3 seconds, with size 176x144 and a frame rate of 10 frames per second.
The following graph description will generate a red source with an opacity of 0.2, with size "qcif" and a frame rate of 10 frames per second.
color=c=red@0.2:s=qcif:r=10 |
If the input content is to be ignored, nullsrc can be used. The
following command generates noise in the luminance plane by employing
the geq filter:
nullsrc=s=256x256, geq=random(1)*255:128:128 |
The color source supports the following commands:
Set the color of the created image. Accepts the same syntax of the corresponding ‘color’ option.
Below is a description of the currently available video sinks.
Buffer video frames, and make them available to the end of the filter graph.
This sink is mainly intended for a programmatic use, in particular through the interface defined in ‘libavfilter/buffersink.h’ or the options system.
It accepts a pointer to an AVBufferSinkContext structure, which
defines the incoming buffers’ formats, to be passed as the opaque
parameter to avfilter_init_filter for initialization.
Null video sink, do absolutely nothing with the input video. It is mainly useful as a template and to be employed in analysis / debugging tools.
Below is a description of the currently available multimedia filters.
Convert input audio to a video output, representing the audio vector scope.
The filter is used to measure the difference between channels of stereo audio stream. A monoaural signal, consisting of identical left and right signal, results in straight vertical line. Any stereo separation is visible as a deviation from this line, creating a Lissajous figure. If the straight (or deviation from it) but horizontal line appears this indicates that the left and right channels are out of phase.
The filter accepts the following options:
Set the vectorscope mode.
Available values are:
Lissajous rotated by 45 degrees.
Same as above but not rotated.
Default value is ‘lissajous’.
Set the video size for the output. For the syntax of this option, check the "Video size"
section in the ffmpeg-utils manual. Default value is 400x400.
Set the output frame rate. Default value is 25.
Specify the red, green and blue contrast. Default values are 40, 160 and 80.
Allowed range is [0, 255].
Specify the red, green and blue fade. Default values are 15, 10 and 5.
Allowed range is [0, 255].
Set the zoom factor. Default value is 1. Allowed range is [1, 10].
ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'
|
Concatenate audio and video streams, joining them together one after the other.
The filter works on segments of synchronized video and audio streams. All segments must have the same number of streams of each type, and that will also be the number of streams at output.
The filter accepts the following options:
Set the number of segments. Default is 2.
Set the number of output video streams, that is also the number of video streams in each segment. Default is 1.
Set the number of output audio streams, that is also the number of video streams in each segment. Default is 0.
Activate unsafe mode: do not fail if segments have a different format.
The filter has v+a outputs: first v video outputs, then a audio outputs.
There are nx(v+a) inputs: first the inputs for the first segment, in the same order as the outputs, then the inputs for the second segment, etc.
Related streams do not always have exactly the same duration, for various reasons including codec frame size or sloppy authoring. For that reason, related synchronized streams (e.g. a video and its audio track) should be concatenated at once. The concat filter will use the duration of the longest stream in each segment (except the last one), and if necessary pad shorter audio streams with silence.
For this filter to work correctly, all segments must start at timestamp 0.
All corresponding streams must have the same parameters in all segments; the filtering system will automatically select a common pixel format for video streams, and a common sample format, sample rate and channel layout for audio streams, but other settings, such as resolution, must be converted explicitly by the user.
Different frame rates are acceptable but will result in variable frame rate at output; be sure to configure the output file to handle it.
ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \ '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] concat=n=3:v=1:a=2 [v] [a1] [a2]' \ -map '[v]' -map '[a1]' -map '[a2]' output.mkv |
movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ; movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ; [v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa] |
Note that a desync will happen at the stitch if the audio and video streams do not have exactly the same duration in the first file.
EBU R128 scanner filter. This filter takes an audio stream as input and outputs
it unchanged. By default, it logs a message at a frequency of 10Hz with the
Momentary loudness (identified by M), Short-term loudness (S),
Integrated loudness (I) and Loudness Range (LRA).
The filter also has a video output (see the video option) with a real time graph to observe the loudness evolution. The graphic contains the logged message mentioned above, so it is not printed anymore when this option is set, unless the verbose logging is set. The main graphing area contains the short-term loudness (3 seconds of analysis), and the gauge on the right is for the momentary loudness (400 milliseconds).
More information about the Loudness Recommendation EBU R128 on http://tech.ebu.ch/loudness.
The filter accepts the following options:
Activate the video output. The audio stream is passed unchanged whether this
option is set or no. The video stream will be the first output stream if
activated. Default is 0.
Set the video size. This option is for video only. For the syntax of this
option, check the "Video size" section in the ffmpeg-utils manual. Default
and minimum resolution is 640x480.
Set the EBU scale meter. Default is 9. Common values are 9 and
18, respectively for EBU scale meter +9 and EBU scale meter +18. Any
other integer value between this range is allowed.
Set metadata injection. If set to 1, the audio input will be segmented
into 100ms output frames, each of them containing various loudness information
in metadata. All the metadata keys are prefixed with lavfi.r128..
Default is 0.
Force the frame logging level.
Available values are:
information logging level
verbose logging level
By default, the logging level is set to info. If the ‘video’ or the ‘metadata’ options are set, it switches to verbose.
ffplay, with a EBU scale meter +18:
ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" |
ffmpeg:
ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null - |
Temporally interleave frames from several inputs.
interleave works with video inputs, ainterleave with audio.
These filters read frames from several inputs and send the oldest queued frame to the output.
Input streams must have a well defined, monotonically increasing frame timestamp values.
In order to submit one frame to output, these filters need to enqueue at least one frame for each input, so they cannot work in case one input is not yet terminated and will not receive incoming frames.
For example consider the case when one input is a select filter
which always drop input frames. The interleave filter will keep
reading from that input, but it will never be able to send new frames
to output until the input will send an end-of-stream signal.
Also, depending on inputs synchronization, the filters will drop frames in case one input receives more frames than the other ones, and the queue is already filled.
These filters accept the following options:
Set the number of different inputs, it is 2 by default.
ffmpeg:
ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi |
select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave |
Set read/write permissions for the output frames.
These filters are mainly aimed at developers to test direct path in the following filter in the filtergraph.
The filters accept the following options:
Select the permissions mode.
It accepts the following values:
Do nothing. This is the default.
Set all the output frames read-only.
Set all the output frames directly writable.
Make the frame read-only if writable, and writable if read-only.
Set each output frame read-only or writable randomly.
Set the seed for the random mode, must be an integer included between
0 and UINT32_MAX. If not specified, or if explicitly set to
-1, the filter will try to use a good random seed on a best effort
basis.
Note: in case of auto-inserted filter between the permission filter and the following one, the permission might not be received as expected in that following filter. Inserting a format or aformat filter before the perms/aperms filter can avoid this problem.
Select frames to pass in output.
This filter accepts the following options:
Set expression, which is evaluated for each input frame.
If the expression is evaluated to zero, the frame is discarded.
If the evaluation result is negative or NaN, the frame is sent to the
first output; otherwise it is sent to the output with index
ceil(val)-1, assuming that the input index starts from 0.
For example a value of 1.2 corresponds to the output with index
ceil(1.2)-1 = 2-1 = 1, that is the second output.
Set the number of outputs. The output to which to send the selected frame is based on the result of the evaluation. Default value is 1.
The expression can contain the following constants:
the sequential number of the filtered frame, starting from 0
the sequential number of the selected frame, starting from 0
the sequential number of the last selected frame, NAN if undefined
timebase of the input timestamps
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units, NAN if undefined
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
the PTS of the previously filtered video frame, NAN if undefined
the PTS of the last previously filtered video frame, NAN if undefined
the PTS of the last previously selected video frame, NAN if undefined
the PTS of the first video frame in the video, NAN if undefined
the time of the first video frame in the video, NAN if undefined
the type of the filtered frame, can assume one of the following values:
the frame interlace type, can assume one of the following values:
the frame is progressive (not interlaced)
the frame is top-field-first
the frame is bottom-field-first
the number of selected samples before the current frame
the number of samples in the current frame
the input sample rate
1 if the filtered frame is a key-frame, 0 otherwise
the position in the file of the filtered frame, -1 if the information is not available (e.g. for synthetic video)
value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one (see the example below)
The default value of the select expression is "1".
select |
The example above is the same as:
select=1 |
select=0 |
select='eq(pict_type\,I)' |
select='not(mod(n\,100))' |
select=between(t\,10\,20) |
select=between(t\,10\,20)*eq(pict_type\,I) |
select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' |
aselect='gt(samples_n\,100)' |
ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png |
Comparing scene against a value between 0.3 and 0.5 is generally a sane choice.
select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h |
Send commands to filters in the filtergraph.
These filters read commands to be sent to other filters in the filtergraph.
sendcmd must be inserted between two video filters,
asendcmd must be inserted between two audio filters, but apart
from that they act the same way.
The specification of commands can be provided in the filter arguments with the commands option, or in a file specified by the filename option.
These filters accept the following options:
Set the commands to be read and sent to the other filters.
Set the filename of the commands to be read and sent to the other filters.
A commands description consists of a sequence of interval specifications, comprising a list of commands to be executed when a particular event related to that interval occurs. The occurring event is typically the current frame time entering or leaving a given time interval.
An interval is specified by the following syntax:
START[-END] COMMANDS; |
The time interval is specified by the START and END times. END is optional and defaults to the maximum time.
The current frame time is considered within the specified interval if it is included in the interval [START, END), that is when the time is greater or equal to START and is lesser than END.
COMMANDS consists of a sequence of one or more command specifications, separated by ",", relating to that interval. The syntax of a command specification is given by:
[FLAGS] TARGET COMMAND ARG |
FLAGS is optional and specifies the type of events relating to the time interval which enable sending the specified command, and must be a non-null sequence of identifier flags separated by "+" or "|" and enclosed between "[" and "]".
The following flags are recognized:
The command is sent when the current frame timestamp enters the specified interval. In other words, the command is sent when the previous frame timestamp was not in the given interval, and the current is.
The command is sent when the current frame timestamp leaves the specified interval. In other words, the command is sent when the previous frame timestamp was in the given interval, and the current is not.
If FLAGS is not specified, a default value of [enter] is
assumed.
TARGET specifies the target of the command, usually the name of the filter class or a specific filter instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional list of argument for the given COMMAND.
Between one interval specification and another, whitespaces, or
sequences of characters starting with # until the end of line,
are ignored and can be used to annotate comments.
A simplified BNF description of the commands specification syntax follows:
COMMAND_FLAG ::= "enter" | "leave" COMMAND_FLAGS ::= COMMAND_FLAG [(+|"|")COMMAND_FLAG] COMMAND ::= ["[" COMMAND_FLAGS "]"] TARGET COMMAND [ARG] COMMANDS ::= COMMAND [,COMMANDS] INTERVAL ::= START[-END] COMMANDS INTERVALS ::= INTERVAL[;INTERVALS] |
asendcmd=c='4.0 atempo tempo 1.5',atempo |
# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';
# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue s 0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue s 1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';
# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s exp(25-t)
|
A filtergraph allowing to read and process the above command list stored in a file ‘test.cmd’, can be specified with:
sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue |
Change the PTS (presentation timestamp) of the input frames.
setpts works on video frames, asetpts on audio frames.
This filter accepts the following options:
The expression which is evaluated for each frame to construct its timestamp.
The expression is evaluated through the eval API and can contain the following constants:
frame rate, only defined for constant frame-rate video
the presentation timestamp in input
the count of the input frame for video or the number of consumed samples, not including the current frame for audio, starting from 0.
the number of consumed samples, not including the current frame (only audio)
the number of samples in the current frame (only audio)
audio sample rate
the PTS of the first frame
the time in seconds of the first frame
tell if the current frame is interlaced
the time in seconds of the current frame
original position in the file of the frame, or undefined if undefined for the current frame
previous input PTS
previous input time in seconds
previous output PTS
previous output time in seconds
wallclock (RTC) time in microseconds. This is deprecated, use time(0) instead.
wallclock (RTC) time at the start of the movie in microseconds
timebase of the input timestamps
setpts=PTS-STARTPTS |
setpts=0.5*PTS |
setpts=2.0*PTS |
setpts=N/(25*TB) |
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' |
setpts=PTS+10/TB |
setpts='(RTCTIME - RTCSTART) / (TB * 1000000)' |
asetpts=N/SR/TB |
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration.
This filter accepts the following options:
The expression which is evaluated into the output timebase.
The value for ‘tb’ is an arithmetic expression representing a rational. The expression can contain the constants "AVTB" (the default timebase), "intb" (the input timebase) and "sr" (the sample rate, audio only). Default value is "intb".
settb=expr=1/25 |
settb=expr=0.1 |
settb=1+0.001 |
settb=2*intb |
settb=AVTB |
Convert input audio to a video output, representing the audio frequency spectrum.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check
the "Video size" section in the ffmpeg-utils manual. Default value is
640x512.
Specify if the spectrum should slide along the window. Default value is
0.
Specify display mode.
It accepts the following values:
all channels are displayed in the same row
all channels are displayed in separate rows
Default value is ‘combined’.
Specify display color mode.
It accepts the following values:
each channel is displayed in a separate color
each channel is is displayed using the same color scheme
Default value is ‘channel’.
Specify scale used for calculating intensity color values.
It accepts the following values:
linear
square root, default
cubic root
logarithmic
Default value is ‘sqrt’.
Set saturation modifier for displayed colors. Negative values provide
alternative color scheme. 0 is no saturation at all.
Saturation must be in [-10.0, 10.0] range.
Default value is 1.
Set window function.
It accepts the following values:
No samples pre-processing (do not expect this to be faster)
Hann window
Hamming window
Blackman window
Default value is hann.
The usage is very similar to the showwaves filter; see the examples in that section.
showspectrum=s=1280x480:scale=log |
ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'
|
Convert input audio to a video output, representing the samples waves.
The filter accepts the following options:
Specify the video size for the output. For the syntax of this option, check the "Video size" section in the ffmpeg-utils manual. Default value is "600x240".
Set display mode.
Available values are:
Draw a point for each sample.
Draw a vertical line for each sample.
Default value is point.
Set the number of samples which are printed on the same column. A larger value will decrease the frame rate. Must be a positive integer. This option can be set only if the value for rate is not explicitly specified.
Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".
amovie=a.mp3,asplit[out0],showwaves[out1] |
aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1] |
Split input into several identical outputs.
asplit works with audio input, split with video.
The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2.
[in] split [out0][out1] |
[in] asplit=3 [out0][out1][out2] |
[in] split [splitout1][splitout2]; [splitout1] crop=100:100:0:0 [cropout]; [splitout2] pad=200:200:100:100 [padout]; |
ffmpeg:
ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT |
Receive commands sent through a libzmq client, and forward them to filters in the filtergraph.
zmq and azmq work as a pass-through filters. zmq
must be inserted between two video filters, azmq between two
audio filters.
To enable these filters you need to install the libzmq library and
headers and configure FFmpeg with --enable-libzmq.
For more information about libzmq see: http://www.zeromq.org/
The zmq and azmq filters work as a libzmq server, which
receives messages sent through a network interface defined by the
‘bind_address’ option.
The received message must be in the form:
TARGET COMMAND [ARG] |
TARGET specifies the target of the command, usually the name of the filter class or a specific filter instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional argument list for the given COMMAND.
Upon reception, the message is processed and the corresponding command is injected into the filtergraph. Depending on the result, the filter will send a reply to the client, adopting the format:
ERROR_CODE ERROR_REASON MESSAGE |
MESSAGE is optional.
Look at ‘tools/zmqsend’ for an example of a zmq client which can be used to send commands processed by these filters.
Consider the following filtergraph generated by ffplay
ffplay -dumpgraph 1 -f lavfi " color=s=100x100:c=red [l]; color=s=100x100:c=blue [r]; nullsrc=s=200x100, zmq [bg]; [bg][l] overlay [bg+l]; [bg+l][r] overlay=x=100 " |
To change the color of the left side of the video, the following command can be used:
echo Parsed_color_0 c yellow | tools/zmqsend |
To change the right side:
echo Parsed_color_1 c pink | tools/zmqsend |
Below is a description of the currently available multimedia sources.
This is the same as movie source, except it selects an audio stream by default.
Read audio and/or video stream(s) from a movie container.
This filter accepts the following options:
The name of the resource to read (not necessarily a file but also a device or a stream accessed through some protocol).
Specifies the format assumed for the movie to read, and can be either the name of a container or an input device. If not specified the format is guessed from movie_name or by probing.
Specifies the seek point in seconds, the frames will be output
starting from this seek point, the parameter is evaluated with
av_strtod so the numerical value may be suffixed by an IS
postfix. Default value is "0".
Specifies the streams to read. Several streams can be specified, separated by "+". The source will then have as many outputs, in the same order. The syntax is explained in the “Stream specifiers” section in the ffmpeg manual. Two special names, "dv" and "da" specify respectively the default (best suited) video and audio stream. Default is "dv", or "da" if the filter is called as "amovie".
Specifies the index of the video stream to read. If the value is -1, the best suited video stream will be automatically selected. Default value is "-1". Deprecated. If the filter is called "amovie", it will select audio instead of video.
Specifies how many times to read the stream in sequence. If the value is less than 1, the stream will be read again and again. Default value is "1".
Note that when the movie is looped the source timestamps are not changed, so it will generate non monotonically increasing timestamps.
This filter allows to overlay a second video on top of main input of a filtergraph as shown in this graph:
input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+
|
movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over]; [in] setpts=PTS-STARTPTS [main]; [main][over] overlay=16:16 [out] |
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over]; [in] setpts=PTS-STARTPTS [main]; [main][over] overlay=16:16 [out] |
movie=dvd.vob:s=v:0+#0x81 [video] [audio] |
ffplay, ffmpeg, ffprobe, ffserver, ffmpeg-utils, ffmpeg-scaler, ffmpeg-resampler, ffmpeg-codecs, ffmpeg-bitstream-filters, ffmpeg-formats, ffmpeg-devices, ffmpeg-protocols, ffmpeg-filters
The FFmpeg developers.
For details about the authorship, see the Git history of the project
(git://source.ffmpeg.org/ffmpeg), e.g. by typing the command
git log in the FFmpeg source directory, or browsing the
online repository at http://source.ffmpeg.org.
Maintainers for the specific components are listed in the file ‘MAINTAINERS’ in the source code tree.