Provided by: ffmpeg_3.4.11-0ubuntu0.1_amd64 
NAME
ffserver - ffserver video server
SYNOPSIS
ffserver [options]
DESCRIPTION
ffserver is a streaming server for both audio and video. It supports several live feeds, streaming from
files and time shifting on live feeds. You can seek to positions in the past on each live feed, provided
you specify a big enough feed storage.
ffserver is configured through a configuration file, which is read at startup. If not explicitly
specified, it will read from /etc/ffserver.conf.
ffserver receives prerecorded files or FFM streams from some ffmpeg instance as input, then streams them
over RTP/RTSP/HTTP.
An ffserver instance will listen on some port as specified in the configuration file. You can launch one
or more instances of ffmpeg and send one or more FFM streams to the port where ffserver is expecting to
receive them. Alternately, you can make ffserver launch such ffmpeg instances at startup.
Input streams are called feeds, and each one is specified by a "<Feed>" section in the configuration
file.
For each feed you can have different output streams in various formats, each one specified by a
"<Stream>" section in the configuration file.
DETAILED DESCRIPTION
ffserver works by forwarding streams encoded by ffmpeg, or pre-recorded streams which are read from disk.
Precisely, ffserver acts as an HTTP server, accepting POST requests from ffmpeg to acquire the stream to
publish, and serving RTSP clients or HTTP clients GET requests with the stream media content.
A feed is an FFM stream created by ffmpeg, and sent to a port where ffserver is listening.
Each feed is identified by a unique name, corresponding to the name of the resource published on
ffserver, and is configured by a dedicated "Feed" section in the configuration file.
The feed publish URL is given by:
http://<ffserver_ip_address>:<http_port>/<feed_name>
where ffserver_ip_address is the IP address of the machine where ffserver is installed, http_port is the
port number of the HTTP server (configured through the HTTPPort option), and feed_name is the name of the
corresponding feed defined in the configuration file.
Each feed is associated to a file which is stored on disk. This stored file is used to send pre-recorded
data to a player as fast as possible when new content is added in real-time to the stream.
A "live-stream" or "stream" is a resource published by ffserver, and made accessible through the HTTP
protocol to clients.
A stream can be connected to a feed, or to a file. In the first case, the published stream is forwarded
from the corresponding feed generated by a running instance of ffmpeg, in the second case the stream is
read from a pre-recorded file.
Each stream is identified by a unique name, corresponding to the name of the resource served by ffserver,
and is configured by a dedicated "Stream" section in the configuration file.
The stream access HTTP URL is given by:
http://<ffserver_ip_address>:<http_port>/<stream_name>[<options>]
The stream access RTSP URL is given by:
http://<ffserver_ip_address>:<rtsp_port>/<stream_name>[<options>]
stream_name is the name of the corresponding stream defined in the configuration file. options is a list
of options specified after the URL which affects how the stream is served by ffserver. http_port and
rtsp_port are the HTTP and RTSP ports configured with the options HTTPPort and RTSPPort respectively.
In case the stream is associated to a feed, the encoding parameters must be configured in the stream
configuration. They are sent to ffmpeg when setting up the encoding. This allows ffserver to define the
encoding parameters used by the ffmpeg encoders.
The ffmpeg override_ffserver commandline option allows one to override the encoding parameters set by the
server.
Multiple streams can be connected to the same feed.
For example, you can have a situation described by the following graph:
_________ __________
| | | |
ffmpeg 1 -----| feed 1 |-----| stream 1 |
\ |_________|\ |__________|
\ \
\ \ __________
\ \ | |
\ \| stream 2 |
\ |__________|
\
\ _________ __________
\ | | | |
\| feed 2 |-----| stream 3 |
|_________| |__________|
_________ __________
| | | |
ffmpeg 2 -----| feed 3 |-----| stream 4 |
|_________| |__________|
_________ __________
| | | |
| file 1 |-----| stream 5 |
|_________| |__________|
FFM, FFM2 formats
FFM and FFM2 are formats used by ffserver. They allow storing a wide variety of video and audio streams
and encoding options, and can store a moving time segment of an infinite movie or a whole movie.
FFM is version specific, and there is limited compatibility of FFM files generated by one version of
ffmpeg/ffserver and another version of ffmpeg/ffserver. It may work but it is not guaranteed to work.
FFM2 is extensible while maintaining compatibility and should work between differing versions of tools.
FFM2 is the default.
Status stream
ffserver supports an HTTP interface which exposes the current status of the server.
Simply point your browser to the address of the special status stream specified in the configuration
file.
For example if you have:
<Stream status.html>
Format status
# Only allow local people to get the status
ACL allow localhost
ACL allow 192.168.0.0 192.168.255.255
</Stream>
then the server will post a page with the status information when the special stream status.html is
requested.
How do I make it work?
As a simple test, just run the following two command lines where INPUTFILE is some file which you can
decode with ffmpeg:
ffserver -f doc/ffserver.conf &
ffmpeg -i INPUTFILE http://localhost:8090/feed1.ffm
At this point you should be able to go to your Windows machine and fire up Windows Media Player (WMP). Go
to Open URL and enter
http://<linuxbox>:8090/test.asf
You should (after a short delay) see video and hear audio.
WARNING: trying to stream test1.mpg doesn't work with WMP as it tries to transfer the entire file before
starting to play. The same is true of AVI files.
You should edit the ffserver.conf file to suit your needs (in terms of frame rates etc). Then install
ffserver and ffmpeg, write a script to start them up, and off you go.
What else can it do?
You can replay video from .ffm files that was recorded earlier. However, there are a number of caveats,
including the fact that the ffserver parameters must match the original parameters used to record the
file. If they do not, then ffserver deletes the file before recording into it. (Now that I write this,
it seems broken).
You can fiddle with many of the codec choices and encoding parameters, and there are a bunch more
parameters that you cannot control. Post a message to the mailing list if there are some 'must have'
parameters. Look in ffserver.conf for a list of the currently available controls.
It will automatically generate the ASX or RAM files that are often used in browsers. These files are
actually redirections to the underlying ASF or RM file. The reason for this is that the browser often
fetches the entire file before starting up the external viewer. The redirection files are very small and
can be transferred quickly. [The stream itself is often 'infinite' and thus the browser tries to download
it and never finishes.]
Tips
* When you connect to a live stream, most players (WMP, RA, etc) want to buffer a certain number of
seconds of material so that they can display the signal continuously. However, ffserver (by default)
starts sending data in realtime. This means that there is a pause of a few seconds while the buffering is
being done by the player. The good news is that this can be cured by adding a '?buffer=5' to the end of
the URL. This means that the stream should start 5 seconds in the past -- and so the first 5 seconds of
the stream are sent as fast as the network will allow. It will then slow down to real time. This
noticeably improves the startup experience.
You can also add a 'Preroll 15' statement into the ffserver.conf that will add the 15 second prebuffering
on all requests that do not otherwise specify a time. In addition, ffserver will skip frames until a
key_frame is found. This further reduces the startup delay by not transferring data that will be
discarded.
Why does the ?buffer / Preroll stop working after a time?
It turns out that (on my machine at least) the number of frames successfully grabbed is marginally less
than the number that ought to be grabbed. This means that the timestamp in the encoded data stream gets
behind realtime. This means that if you say 'Preroll 10', then when the stream gets 10 or more seconds
behind, there is no Preroll left.
Fixing this requires a change in the internals of how timestamps are handled.
Does the "?date=" stuff work.
Yes (subject to the limitation outlined above). Also note that whenever you start ffserver, it deletes
the ffm file (if any parameters have changed), thus wiping out what you had recorded before.
The format of the "?date=xxxxxx" is fairly flexible. You should use one of the following formats (the 'T'
is literal):
* YYYY-MM-DDTHH:MM:SS (localtime)
* YYYY-MM-DDTHH:MM:SSZ (UTC)
You can omit the YYYY-MM-DD, and then it refers to the current day. However note that ?date=16:00:00
refers to 16:00 on the current day -- this may be in the future and so is unlikely to be useful.
You use this by adding the ?date= to the end of the URL for the stream. For example:
http://localhost:8080/test.asf?date=2002-07-26T23:05:00.
OPTIONS
All the numerical options, if not specified otherwise, accept a string representing a number as input,
which may be followed by one of the SI unit prefixes, for example: 'K', 'M', or 'G'.
If 'i' is appended to the SI unit prefix, the complete prefix will be interpreted as a unit prefix for
binary multiples, which are based on powers of 1024 instead of powers of 1000. Appending 'B' to the SI
unit prefix multiplies the value by 8. This allows using, for example: 'KB', 'MiB', 'G' and 'B' as number
suffixes.
Options which do not take arguments are boolean options, and set the corresponding value to true. They
can be set to false by prefixing the option name with "no". For example using "-nofoo" will set the
boolean option with name "foo" to false.
Stream specifiers
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely
specify which stream(s) a given option belongs to.
A stream specifier is a string generally appended to the option name and separated from it by a colon.
E.g. "-codec:a:1 ac3" contains the "a:1" stream specifier, which matches the second audio stream.
Therefore, it would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, so that the option is applied to all of them. E.g. the
stream specifier in "-b:a 128k" matches all audio streams.
An empty stream specifier matches all streams. For example, "-codec copy" or "-codec: copy" would copy
all the streams without reencoding.
Possible forms of stream specifiers are:
stream_index
Matches the stream with this index. E.g. "-threads:1 4" would set the thread count for the second
stream to 4.
stream_type[:stream_index]
stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for subtitle, 'd' for data,
and 't' for attachments. 'v' matches all video streams, 'V' only matches video streams which are not
attached pictures, video thumbnails or cover arts. If stream_index is given, then it matches stream
number stream_index of this type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index]
If stream_index is given, then it matches the stream with number stream_index in the program with the
id program_id. Otherwise, it matches all streams in the program.
#stream_id or i:stream_id
Match the stream by stream id (e.g. PID in MPEG-TS container).
m:key[:value]
Matches streams with the metadata tag key having the specified value. If value is not given, matches
streams that contain the given tag with any value.
u Matches streams with usable configuration, the codec must be defined and the essential information
such as video dimension or audio sample rate must be present.
Note that in ffmpeg, matching by metadata will only work properly for input files.
Generic options
These options are shared amongst the ff* tools.
-L Show license.
-h, -?, -help, --help [arg]
Show help. An optional parameter may be specified to print help about a specific item. If no argument
is specified, only basic (non advanced) tool options are shown.
Possible values of arg are:
long
Print advanced tool options in addition to the basic tool options.
full
Print complete list of options, including shared and private options for encoders, decoders,
demuxers, muxers, filters, etc.
decoder=decoder_name
Print detailed information about the decoder named decoder_name. Use the -decoders option to get
a list of all decoders.
encoder=encoder_name
Print detailed information about the encoder named encoder_name. Use the -encoders option to get
a list of all encoders.
demuxer=demuxer_name
Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a
list of all demuxers and muxers.
muxer=muxer_name
Print detailed information about the muxer named muxer_name. Use the -formats option to get a
list of all muxers and demuxers.
filter=filter_name
Print detailed information about the filter name filter_name. Use the -filters option to get a
list of all filters.
-version
Show version.
-formats
Show available formats (including devices).
-demuxers
Show available demuxers.
-muxers
Show available muxers.
-devices
Show available devices.
-codecs
Show all codecs known to libavcodec.
Note that the term 'codec' is used throughout this documentation as a shortcut for what is more
correctly called a media bitstream format.
-decoders
Show available decoders.
-encoders
Show all available encoders.
-bsfs
Show available bitstream filters.
-protocols
Show available protocols.
-filters
Show available libavfilter filters.
-pix_fmts
Show available pixel formats.
-sample_fmts
Show available sample formats.
-layouts
Show channel names and standard channel layouts.
-colors
Show recognized color names.
-sources device[,opt1=val1[,opt2=val2]...]
Show autodetected sources of the input device. Some devices may provide system-dependent source
names that cannot be autodetected. The returned list cannot be assumed to be always complete.
ffmpeg -sources pulse,server=192.168.0.4
-sinks device[,opt1=val1[,opt2=val2]...]
Show autodetected sinks of the output device. Some devices may provide system-dependent sink names
that cannot be autodetected. The returned list cannot be assumed to be always complete.
ffmpeg -sinks pulse,server=192.168.0.4
-loglevel [repeat+]loglevel | -v [repeat+]loglevel
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 string or a number containing one of the following values:
quiet, -8
Show nothing at all; be silent.
panic, 0
Only show fatal errors which could lead the process to crash, such as an assertion failure. This
is not currently used for anything.
fatal, 8
Only show fatal errors. These are errors after which the process absolutely cannot continue.
error, 16
Show all errors, including ones which can be recovered from.
warning, 24
Show all warnings and errors. Any message related to possibly incorrect or unexpected events will
be shown.
info, 32
Show informative messages during processing. This is in addition to warnings and errors. This is
the default value.
verbose, 40
Same as "info", except more verbose.
debug, 48
Show everything, including debugging information.
trace, 56
By default the program logs to stderr. If coloring is supported by the terminal, colors are used to
mark errors and warnings. Log coloring can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting the environment variable
AV_LOG_FORCE_COLOR. The use of the environment variable NO_COLOR is deprecated and will be dropped
in a future FFmpeg version.
-report
Dump full command line and console output to a file named "program-YYYYMMDD-HHMMSS.log" in the
current directory. This file can be useful for bug reports. It also implies "-loglevel verbose".
Setting the environment variable FFREPORT to any value has the same effect. If the value is a
':'-separated key=value sequence, these options will affect the report; option values must be escaped
if they contain special characters or the options delimiter ':' (see the ``Quoting and escaping''
section in the ffmpeg-utils manual).
The following options are recognized:
file
set the file name to use for the report; %p is expanded to the name of the program, %t is
expanded to a timestamp, "%%" is expanded to a plain "%"
level
set the log verbosity level using a numerical value (see "-loglevel").
For example, to output a report to a file named ffreport.log using a log level of 32 (alias for log
level "info"):
FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output
Errors in parsing the environment variable are not fatal, and will not appear in the report.
-hide_banner
Suppress printing banner.
All FFmpeg tools will normally show a copyright notice, build options and library versions. This
option can be used to suppress printing this information.
-cpuflags flags (global)
Allows setting and clearing cpu flags. This option is intended for testing. Do not use it unless you
know what you're doing.
ffmpeg -cpuflags -sse+mmx ...
ffmpeg -cpuflags mmx ...
ffmpeg -cpuflags 0 ...
Possible flags for this option are:
x86
mmx
mmxext
sse
sse2
sse2slow
sse3
sse3slow
ssse3
atom
sse4.1
sse4.2
avx
avx2
xop
fma3
fma4
3dnow
3dnowext
bmi1
bmi2
cmov
ARM
armv5te
armv6
armv6t2
vfp
vfpv3
neon
setend
AArch64
armv8
vfp
neon
PowerPC
altivec
Specific Processors
pentium2
pentium3
pentium4
k6
k62
athlon
athlonxp
k8
-opencl_bench
This option is used to benchmark all available OpenCL devices and print the results. This option is
only available when FFmpeg has been compiled with "--enable-opencl".
When FFmpeg is configured with "--enable-opencl", the options for the global OpenCL context are set
via -opencl_options. See the "OpenCL Options" section in the ffmpeg-utils manual for the complete
list of supported options. Amongst others, these options include the ability to select a specific
platform and device to run the OpenCL code on. By default, FFmpeg will run on the first device of the
first platform. While the options for the global OpenCL context provide flexibility to the user in
selecting the OpenCL device of their choice, most users would probably want to select the fastest
OpenCL device for their system.
This option assists the selection of the most efficient configuration by identifying the appropriate
device for the user's system. The built-in benchmark is run on all the OpenCL devices and the
performance is measured for each device. The devices in the results list are sorted based on their
performance with the fastest device listed first. The user can subsequently invoke ffmpeg using the
device deemed most appropriate via -opencl_options to obtain the best performance for the OpenCL
accelerated code.
Typical usage to use the fastest OpenCL device involve the following steps.
Run the command:
ffmpeg -opencl_bench
Note down the platform ID (pidx) and device ID (didx) of the first i.e. fastest device in the list.
Select the platform and device using the command:
ffmpeg -opencl_options platform_idx=<pidx>:device_idx=<didx> ...
-opencl_options options (global)
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.
AVOptions
These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the
list of available AVOptions, use the -help option. They are separated into two categories:
generic
These options can be set for any container, codec or device. Generic options are listed under
AVFormatContext options for containers/devices and under AVCodecContext options for codecs.
private
These options are specific to the given container, device or codec. Private options are listed under
their corresponding containers/devices/codecs.
For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the id3v2_version
private option of the MP3 muxer:
ffmpeg -i input.flac -id3v2_version 3 out.mp3
All codec AVOptions are per-stream, and thus a stream specifier should be attached to them.
Note: the -nooption syntax cannot be used for boolean AVOptions, use -option 0/-option 1.
Note: the old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name
is now obsolete and will be removed soon.
Main options
-f configfile
Read configuration file configfile. If not specified it will read by default from /etc/ffserver.conf.
-n Enable no-launch mode. This option disables all the "Launch" directives within the various "<Feed>"
sections. Since ffserver will not launch any ffmpeg instances, you will have to launch them manually.
-d Enable debug mode. This option increases log verbosity, and directs log messages to stdout. When
specified, the CustomLog option is ignored.
CONFIGURATION FILE SYNTAX
ffserver reads a configuration file containing global options and settings for each stream and feed.
The configuration file consists of global options and dedicated sections, which must be introduced by
"<SECTION_NAME ARGS>" on a separate line and must be terminated by a line in the form "</SECTION_NAME>".
ARGS is optional.
Currently the following sections are recognized: Feed, Stream, Redirect.
A line starting with "#" is ignored and treated as a comment.
Name of options and sections are case-insensitive.
ACL syntax
An ACL (Access Control List) specifies the address which are allowed to access a given stream, or to
write a given feed.
It accepts the following forms
• Allow/deny access to address.
ACL ALLOW <address>
ACL DENY <address>
• Allow/deny access to ranges of addresses from first_address to last_address.
ACL ALLOW <first_address> <last_address>
ACL DENY <first_address> <last_address>
You can repeat the ACL allow/deny as often as you like. It is on a per stream basis. The first match
defines the action. If there are no matches, then the default is the inverse of the last ACL statement.
Thus 'ACL allow localhost' only allows access from localhost. 'ACL deny 1.0.0.0 1.255.255.255' would
deny the whole of network 1 and allow everybody else.
Global options
HTTPPort port_number
Port port_number
RTSPPort port_number
HTTPPort sets the HTTP server listening TCP port number, RTSPPort sets the RTSP server listening TCP
port number.
Port is the equivalent of HTTPPort and is deprecated.
You must select a different port from your standard HTTP web server if it is running on the same
computer.
If not specified, no corresponding server will be created.
HTTPBindAddress ip_address
BindAddress ip_address
RTSPBindAddress ip_address
Set address on which the HTTP/RTSP server is bound. Only useful if you have several network
interfaces.
BindAddress is the equivalent of HTTPBindAddress and is deprecated.
MaxHTTPConnections n
Set number of simultaneous HTTP connections that can be handled. It has to be defined before the
MaxClients parameter, since it defines the MaxClients maximum limit.
Default value is 2000.
MaxClients n
Set number of simultaneous requests that can be handled. Since ffserver is very fast, it is more
likely that you will want to leave this high and use MaxBandwidth.
Default value is 5.
MaxBandwidth kbps
Set the maximum amount of kbit/sec that you are prepared to consume when streaming to clients.
Default value is 1000.
CustomLog filename
Set access log file (uses standard Apache log file format). '-' is the standard output.
If not specified ffserver will produce no log.
In case the commandline option -d is specified this option is ignored, and the log is written to
standard output.
NoDaemon
Set no-daemon mode. This option is currently ignored since now ffserver will always work in no-daemon
mode, and is deprecated.
UseDefaults
NoDefaults
Control whether default codec options are used for the all streams or not. Each stream may overwrite
this setting for its own. Default is UseDefaults. The last occurrence overrides the previous if
multiple definitions exist.
Feed section
A Feed section defines a feed provided to ffserver.
Each live feed contains one video and/or audio sequence coming from an ffmpeg encoder or another
ffserver. This sequence may be encoded simultaneously with several codecs at several resolutions.
A feed instance specification is introduced by a line in the form:
<Feed FEED_FILENAME>
where FEED_FILENAME specifies the unique name of the FFM stream.
The following options are recognized within a Feed section.
File filename
ReadOnlyFile filename
Set the path where the feed file is stored on disk.
If not specified, the /tmp/FEED.ffm is assumed, where FEED is the feed name.
If ReadOnlyFile is used the file is marked as read-only and it will not be deleted or updated.
Truncate
Truncate the feed file, rather than appending to it. By default ffserver will append data to the
file, until the maximum file size value is reached (see FileMaxSize option).
FileMaxSize size
Set maximum size of the feed file in bytes. 0 means unlimited. The postfixes "K" (2^10), "M" (2^20),
and "G" (2^30) are recognized.
Default value is 5M.
Launch args
Launch an ffmpeg command when creating ffserver.
args must be a sequence of arguments to be provided to an ffmpeg instance. The first provided
argument is ignored, and it is replaced by a path with the same dirname of the ffserver instance,
followed by the remaining argument and terminated with a path corresponding to the feed.
When the launched process exits, ffserver will launch another program instance.
In case you need a more complex ffmpeg configuration, e.g. if you need to generate multiple FFM feeds
with a single ffmpeg instance, you should launch ffmpeg by hand.
This option is ignored in case the commandline option -n is specified.
ACL spec
Specify the list of IP address which are allowed or denied to write the feed. Multiple ACL options
can be specified.
Stream section
A Stream section defines a stream provided by ffserver, and identified by a single name.
The stream is sent when answering a request containing the stream name.
A stream section must be introduced by the line:
<Stream STREAM_NAME>
where STREAM_NAME specifies the unique name of the stream.
The following options are recognized within a Stream section.
Encoding options are marked with the encoding tag, and they are used to set the encoding parameters, and
are mapped to libavcodec encoding options. Not all encoding options are supported, in particular it is
not possible to set encoder private options. In order to override the encoding options specified by
ffserver, you can use the ffmpeg override_ffserver commandline option.
Only one of the Feed and File options should be set.
Feed feed_name
Set the input feed. feed_name must correspond to an existing feed defined in a "Feed" section.
When this option is set, encoding options are used to setup the encoding operated by the remote
ffmpeg process.
File filename
Set the filename of the pre-recorded input file to stream.
When this option is set, encoding options are ignored and the input file content is re-streamed as
is.
Format format_name
Set the format of the output stream.
Must be the name of a format recognized by FFmpeg. If set to status, it is treated as a status
stream.
InputFormat format_name
Set input format. If not specified, it is automatically guessed.
Preroll n
Set this to the number of seconds backwards in time to start. Note that most players will buffer 5-10
seconds of video, and also you need to allow for a keyframe to appear in the data stream.
Default value is 0.
StartSendOnKey
Do not send stream until it gets the first key frame. By default ffserver will send data immediately.
MaxTime n
Set the number of seconds to run. This value set the maximum duration of the stream a client will be
able to receive.
A value of 0 means that no limit is set on the stream duration.
ACL spec
Set ACL for the stream.
DynamicACL spec
RTSPOption option
MulticastAddress address
MulticastPort port
MulticastTTL integer
NoLoop
FaviconURL url
Set favicon (favourite icon) for the server status page. It is ignored for regular streams.
Author value
Comment value
Copyright value
Title value
Set metadata corresponding to the option. All these options are deprecated in favor of Metadata.
Metadata key value
Set metadata value on the output stream.
UseDefaults
NoDefaults
Control whether default codec options are used for the stream or not. Default is UseDefaults unless
disabled globally.
NoAudio
NoVideo
Suppress audio/video.
AudioCodec codec_name (encoding,audio)
Set audio codec.
AudioBitRate rate (encoding,audio)
Set bitrate for the audio stream in kbits per second.
AudioChannels n (encoding,audio)
Set number of audio channels.
AudioSampleRate n (encoding,audio)
Set sampling frequency for audio. When using low bitrates, you should lower this frequency to 22050
or 11025. The supported frequencies depend on the selected audio codec.
AVOptionAudio [codec:]option value (encoding,audio)
Set generic or private option for audio stream. Private option must be prefixed with codec name or
codec must be defined before.
AVPresetAudio preset (encoding,audio)
Set preset for audio stream.
VideoCodec codec_name (encoding,video)
Set video codec.
VideoBitRate n (encoding,video)
Set bitrate for the video stream in kbits per second.
VideoBitRateRange range (encoding,video)
Set video bitrate range.
A range must be specified in the form minrate-maxrate, and specifies the minrate and maxrate encoding
options expressed in kbits per second.
VideoBitRateRangeTolerance n (encoding,video)
Set video bitrate tolerance in kbits per second.
PixelFormat pixel_format (encoding,video)
Set video pixel format.
Debug integer (encoding,video)
Set video debug encoding option.
Strict integer (encoding,video)
Set video strict encoding option.
VideoBufferSize n (encoding,video)
Set ratecontrol buffer size, expressed in KB.
VideoFrameRate n (encoding,video)
Set number of video frames per second.
VideoSize (encoding,video)
Set size of the video frame, must be an abbreviation or in the form WxH. See the Video size section
in the ffmpeg-utils(1) manual.
Default value is "160x128".
VideoIntraOnly (encoding,video)
Transmit only intra frames (useful for low bitrates, but kills frame rate).
VideoGopSize n (encoding,video)
If non-intra only, an intra frame is transmitted every VideoGopSize frames. Video synchronization can
only begin at an intra frame.
VideoTag tag (encoding,video)
Set video tag.
VideoHighQuality (encoding,video)
Video4MotionVector (encoding,video)
BitExact (encoding,video)
Set bitexact encoding flag.
IdctSimple (encoding,video)
Set simple IDCT algorithm.
Qscale n (encoding,video)
Enable constant quality encoding, and set video qscale (quantization scale) value, expressed in n QP
units.
VideoQMin n (encoding,video)
VideoQMax n (encoding,video)
Set video qmin/qmax.
VideoQDiff integer (encoding,video)
Set video qdiff encoding option.
LumiMask float (encoding,video)
DarkMask float (encoding,video)
Set lumi_mask/dark_mask encoding options.
AVOptionVideo [codec:]option value (encoding,video)
Set generic or private option for video stream. Private option must be prefixed with codec name or
codec must be defined before.
AVPresetVideo preset (encoding,video)
Set preset for video stream.
preset must be the path of a preset file.
Server status stream
A server status stream is a special stream which is used to show statistics about the ffserver
operations.
It must be specified setting the option Format to status.
Redirect section
A redirect section specifies where to redirect the requested URL to another page.
A redirect section must be introduced by the line:
<Redirect NAME>
where NAME is the name of the page which should be redirected.
It only accepts the option URL, which specify the redirection URL.
STREAM EXAMPLES
• Multipart JPEG
<Stream test.mjpg>
Feed feed1.ffm
Format mpjpeg
VideoFrameRate 2
VideoIntraOnly
NoAudio
Strict -1
</Stream>
• Single JPEG
<Stream test.jpg>
Feed feed1.ffm
Format jpeg
VideoFrameRate 2
VideoIntraOnly
VideoSize 352x240
NoAudio
Strict -1
</Stream>
• Flash
<Stream test.swf>
Feed feed1.ffm
Format swf
VideoFrameRate 2
VideoIntraOnly
NoAudio
</Stream>
• ASF compatible
<Stream test.asf>
Feed feed1.ffm
Format asf
VideoFrameRate 15
VideoSize 352x240
VideoBitRate 256
VideoBufferSize 40
VideoGopSize 30
AudioBitRate 64
StartSendOnKey
</Stream>
• MP3 audio
<Stream test.mp3>
Feed feed1.ffm
Format mp2
AudioCodec mp3
AudioBitRate 64
AudioChannels 1
AudioSampleRate 44100
NoVideo
</Stream>
• Ogg Vorbis audio
<Stream test.ogg>
Feed feed1.ffm
Metadata title "Stream title"
AudioBitRate 64
AudioChannels 2
AudioSampleRate 44100
NoVideo
</Stream>
• Real with audio only at 32 kbits
<Stream test.ra>
Feed feed1.ffm
Format rm
AudioBitRate 32
NoVideo
</Stream>
• Real with audio and video at 64 kbits
<Stream test.rm>
Feed feed1.ffm
Format rm
AudioBitRate 32
VideoBitRate 128
VideoFrameRate 25
VideoGopSize 25
</Stream>
• For stream coming from a file: you only need to set the input filename and optionally a new format.
<Stream file.rm>
File "/usr/local/httpd/htdocs/tlive.rm"
NoAudio
</Stream>
<Stream file.asf>
File "/usr/local/httpd/htdocs/test.asf"
NoAudio
Metadata author "Me"
Metadata copyright "Super MegaCorp"
Metadata title "Test stream from disk"
Metadata comment "Test comment"
</Stream>
SYNTAX
This section documents the syntax and formats employed by the FFmpeg libraries and tools.
Quoting and escaping
FFmpeg adopts the following quoting and escaping mechanism, unless explicitly specified. The following
rules are applied:
• ' and \ are special characters (respectively used for quoting and escaping). In addition to them,
there might be other special characters depending on the specific syntax where the escaping and
quoting are employed.
• A special character is escaped by prefixing it with a \.
• All characters enclosed between '' are included literally in the parsed string. The quote character '
itself cannot be quoted, so you may need to close the quote and escape it.
• Leading and trailing whitespaces, unless escaped or quoted, are removed from the parsed string.
Note that you may need to add a second level of escaping when using the command line or a script, which
depends on the syntax of the adopted shell language.
The function "av_get_token" defined in libavutil/avstring.h can be used to parse a token quoted or
escaped according to the rules defined above.
The tool tools/ffescape in the FFmpeg source tree can be used to automatically quote or escape a string
in a script.
Examples
• Escape the string "Crime d'Amour" containing the "'" special character:
Crime d\'Amour
• The string above contains a quote, so the "'" needs to be escaped when quoting it:
'Crime d'\''Amour'
• Include leading or trailing whitespaces using quoting:
' this string starts and ends with whitespaces '
• Escaping and quoting can be mixed together:
' The string '\'string\'' is a string '
• To include a literal \ you can use either escaping or quoting:
'c:\foo' can be written as c:\\foo
Date
The accepted syntax is:
[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
now
If the value is "now" it takes the current time.
Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day
part is not specified it takes the current year-month-day.
Time duration
There are two accepted syntaxes for expressing time duration.
[-][<HH>:]<MM>:<SS>[.<m>...]
HH expresses the number of hours, MM the number of minutes for a maximum of 2 digits, and SS the number
of seconds for a maximum of 2 digits. The m at the end expresses decimal value for SS.
or
[-]<S>+[.<m>...]
S expresses the number of seconds, with the optional decimal part m.
In both expressions, the optional - indicates negative duration.
Examples
The following examples are all valid time duration:
55 55 seconds
12:03:45
12 hours, 03 minutes and 45 seconds
23.189
23.189 seconds
Video size
Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size
abbreviation.
The following abbreviations are recognized:
ntsc
720x480
pal 720x576
qntsc
352x240
qpal
352x288
sntsc
640x480
spal
768x576
film
352x240
ntsc-film
352x240
sqcif
128x96
qcif
176x144
cif 352x288
4cif
704x576
16cif
1408x1152
qqvga
160x120
qvga
320x240
vga 640x480
svga
800x600
xga 1024x768
uxga
1600x1200
qxga
2048x1536
sxga
1280x1024
qsxga
2560x2048
hsxga
5120x4096
wvga
852x480
wxga
1366x768
wsxga
1600x1024
wuxga
1920x1200
woxga
2560x1600
wqsxga
3200x2048
wquxga
3840x2400
whsxga
6400x4096
whuxga
7680x4800
cga 320x200
ega 640x350
hd480
852x480
hd720
1280x720
hd1080
1920x1080
2k 2048x1080
2kflat
1998x1080
2kscope
2048x858
4k 4096x2160
4kflat
3996x2160
4kscope
4096x1716
nhd 640x360
hqvga
240x160
wqvga
400x240
fwqvga
432x240
hvga
480x320
qhd 960x540
2kdci
2048x1080
4kdci
4096x2160
uhd2160
3840x2160
uhd4320
7680x4320
Video rate
Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a
string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video
frame rate abbreviation.
The following abbreviations are recognized:
ntsc
30000/1001
pal 25/1
qntsc
30000/1001
qpal
25/1
sntsc
30000/1001
spal
25/1
film
24/1
ntsc-film
24000/1001
Ratio
A ratio can be expressed as an expression, or in the form numerator:denominator.
Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the
returned value if you want to exclude those values.
The undefined value can be expressed using the "0:0" string.
Color
It can be the name of a color as defined below (case insensitive match) or a "[0x|#]RRGGBB[AA]" sequence,
possibly followed by @ and a string representing the alpha component.
The alpha component may be a string composed by "0x" followed by an hexadecimal number or a decimal
number between 0.0 and 1.0, which represents the opacity value (0x00 or 0.0 means completely transparent,
0xff or 1.0 completely opaque). If the alpha component is not specified then 0xff is assumed.
The string random will result in a random color.
The following names of colors are recognized:
AliceBlue
0xF0F8FF
AntiqueWhite
0xFAEBD7
Aqua
0x00FFFF
Aquamarine
0x7FFFD4
Azure
0xF0FFFF
Beige
0xF5F5DC
Bisque
0xFFE4C4
Black
0x000000
BlanchedAlmond
0xFFEBCD
Blue
0x0000FF
BlueViolet
0x8A2BE2
Brown
0xA52A2A
BurlyWood
0xDEB887
CadetBlue
0x5F9EA0
Chartreuse
0x7FFF00
Chocolate
0xD2691E
Coral
0xFF7F50
CornflowerBlue
0x6495ED
Cornsilk
0xFFF8DC
Crimson
0xDC143C
Cyan
0x00FFFF
DarkBlue
0x00008B
DarkCyan
0x008B8B
DarkGoldenRod
0xB8860B
DarkGray
0xA9A9A9
DarkGreen
0x006400
DarkKhaki
0xBDB76B
DarkMagenta
0x8B008B
DarkOliveGreen
0x556B2F
Darkorange
0xFF8C00
DarkOrchid
0x9932CC
DarkRed
0x8B0000
DarkSalmon
0xE9967A
DarkSeaGreen
0x8FBC8F
DarkSlateBlue
0x483D8B
DarkSlateGray
0x2F4F4F
DarkTurquoise
0x00CED1
DarkViolet
0x9400D3
DeepPink
0xFF1493
DeepSkyBlue
0x00BFFF
DimGray
0x696969
DodgerBlue
0x1E90FF
FireBrick
0xB22222
FloralWhite
0xFFFAF0
ForestGreen
0x228B22
Fuchsia
0xFF00FF
Gainsboro
0xDCDCDC
GhostWhite
0xF8F8FF
Gold
0xFFD700
GoldenRod
0xDAA520
Gray
0x808080
Green
0x008000
GreenYellow
0xADFF2F
HoneyDew
0xF0FFF0
HotPink
0xFF69B4
IndianRed
0xCD5C5C
Indigo
0x4B0082
Ivory
0xFFFFF0
Khaki
0xF0E68C
Lavender
0xE6E6FA
LavenderBlush
0xFFF0F5
LawnGreen
0x7CFC00
LemonChiffon
0xFFFACD
LightBlue
0xADD8E6
LightCoral
0xF08080
LightCyan
0xE0FFFF
LightGoldenRodYellow
0xFAFAD2
LightGreen
0x90EE90
LightGrey
0xD3D3D3
LightPink
0xFFB6C1
LightSalmon
0xFFA07A
LightSeaGreen
0x20B2AA
LightSkyBlue
0x87CEFA
LightSlateGray
0x778899
LightSteelBlue
0xB0C4DE
LightYellow
0xFFFFE0
Lime
0x00FF00
LimeGreen
0x32CD32
Linen
0xFAF0E6
Magenta
0xFF00FF
Maroon
0x800000
MediumAquaMarine
0x66CDAA
MediumBlue
0x0000CD
MediumOrchid
0xBA55D3
MediumPurple
0x9370D8
MediumSeaGreen
0x3CB371
MediumSlateBlue
0x7B68EE
MediumSpringGreen
0x00FA9A
MediumTurquoise
0x48D1CC
MediumVioletRed
0xC71585
MidnightBlue
0x191970
MintCream
0xF5FFFA
MistyRose
0xFFE4E1
Moccasin
0xFFE4B5
NavajoWhite
0xFFDEAD
Navy
0x000080
OldLace
0xFDF5E6
Olive
0x808000
OliveDrab
0x6B8E23
Orange
0xFFA500
OrangeRed
0xFF4500
Orchid
0xDA70D6
PaleGoldenRod
0xEEE8AA
PaleGreen
0x98FB98
PaleTurquoise
0xAFEEEE
PaleVioletRed
0xD87093
PapayaWhip
0xFFEFD5
PeachPuff
0xFFDAB9
Peru
0xCD853F
Pink
0xFFC0CB
Plum
0xDDA0DD
PowderBlue
0xB0E0E6
Purple
0x800080
Red 0xFF0000
RosyBrown
0xBC8F8F
RoyalBlue
0x4169E1
SaddleBrown
0x8B4513
Salmon
0xFA8072
SandyBrown
0xF4A460
SeaGreen
0x2E8B57
SeaShell
0xFFF5EE
Sienna
0xA0522D
Silver
0xC0C0C0
SkyBlue
0x87CEEB
SlateBlue
0x6A5ACD
SlateGray
0x708090
Snow
0xFFFAFA
SpringGreen
0x00FF7F
SteelBlue
0x4682B4
Tan 0xD2B48C
Teal
0x008080
Thistle
0xD8BFD8
Tomato
0xFF6347
Turquoise
0x40E0D0
Violet
0xEE82EE
Wheat
0xF5DEB3
White
0xFFFFFF
WhiteSmoke
0xF5F5F5
Yellow
0xFFFF00
YellowGreen
0x9ACD32
Channel Layout
A channel layout specifies the spatial disposition of the channels in a multi-channel audio stream. To
specify a channel layout, FFmpeg makes use of a special syntax.
Individual channels are identified by an id, as given by the table below:
FL front left
FR front right
FC front center
LFE low frequency
BL back left
BR back right
FLC front left-of-center
FRC front right-of-center
BC back center
SL side left
SR side right
TC top center
TFL top front left
TFC top front center
TFR top front right
TBL top back left
TBC top back center
TBR top back right
DL downmix left
DR downmix right
WL wide left
WR wide right
SDL surround direct left
SDR surround direct right
LFE2
low frequency 2
Standard channel layout compositions can be specified by using the following identifiers:
mono
FC
stereo
FL+FR
2.1 FL+FR+LFE
3.0 FL+FR+FC
3.0(back)
FL+FR+BC
4.0 FL+FR+FC+BC
quad
FL+FR+BL+BR
quad(side)
FL+FR+SL+SR
3.1 FL+FR+FC+LFE
5.0 FL+FR+FC+BL+BR
5.0(side)
FL+FR+FC+SL+SR
4.1 FL+FR+FC+LFE+BC
5.1 FL+FR+FC+LFE+BL+BR
5.1(side)
FL+FR+FC+LFE+SL+SR
6.0 FL+FR+FC+BC+SL+SR
6.0(front)
FL+FR+FLC+FRC+SL+SR
hexagonal
FL+FR+FC+BL+BR+BC
6.1 FL+FR+FC+LFE+BC+SL+SR
6.1 FL+FR+FC+LFE+BL+BR+BC
6.1(front)
FL+FR+LFE+FLC+FRC+SL+SR
7.0 FL+FR+FC+BL+BR+SL+SR
7.0(front)
FL+FR+FC+FLC+FRC+SL+SR
7.1 FL+FR+FC+LFE+BL+BR+SL+SR
7.1(wide)
FL+FR+FC+LFE+BL+BR+FLC+FRC
7.1(wide-side)
FL+FR+FC+LFE+FLC+FRC+SL+SR
octagonal
FL+FR+FC+BL+BR+BC+SL+SR
downmix
DL+DR
A custom channel layout can be specified as a sequence of terms, separated by '+' or '|'. Each term can
be:
• the name of a standard channel layout (e.g. mono, stereo, 4.0, quad, 5.0, etc.)
• the name of a single channel (e.g. FL, FR, FC, LFE, etc.)
• a number of channels, in decimal, followed by 'c', yielding the default channel layout for that
number of channels (see the function "av_get_default_channel_layout"). Note that not all channel
counts have a default layout.
• a number of channels, in decimal, followed by 'C', yielding an unknown channel layout with the
specified number of channels. Note that not all channel layout specification strings support unknown
channel layouts.
• a channel layout mask, in hexadecimal starting with "0x" (see the "AV_CH_*" macros in
libavutil/channel_layout.h.
Before libavutil version 53 the trailing character "c" to specify a number of channels was optional, but
now it is required, while a channel layout mask can also be specified as a decimal number (if and only if
not followed by "c" or "C").
See also the function "av_get_channel_layout" defined in libavutil/channel_layout.h.
EXPRESSION EVALUATION
When evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through
the libavutil/eval.h interface.
An expression may contain unary, binary operators, constants, and functions.
Two expressions expr1 and expr2 can be combined to form another expression "expr1;expr2". expr1 and
expr2 are evaluated in turn, and the new expression evaluates to the value of expr2.
The following binary operators are available: "+", "-", "*", "/", "^".
The following unary operators are available: "+", "-".
The following functions are available:
abs(x)
Compute absolute value of x.
acos(x)
Compute arccosine of x.
asin(x)
Compute arcsine of x.
atan(x)
Compute arctangent of x.
atan2(x, y)
Compute principal value of the arc tangent of y/x.
between(x, min, max)
Return 1 if x is greater than or equal to min and lesser than or equal to max, 0 otherwise.
bitand(x, y)
bitor(x, y)
Compute bitwise and/or operation on x and y.
The results of the evaluation of x and y are converted to integers before executing the bitwise
operation.
Note that both the conversion to integer and the conversion back to floating point can lose
precision. Beware of unexpected results for large numbers (usually 2^53 and larger).
ceil(expr)
Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".
clip(x, min, max)
Return the value of x clipped between min and max.
cos(x)
Compute cosine of x.
cosh(x)
Compute hyperbolic cosine of x.
eq(x, y)
Return 1 if x and y are equivalent, 0 otherwise.
exp(x)
Compute exponential of x (with base "e", the Euler's number).
floor(expr)
Round the value of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is
"-2.0".
gauss(x)
Compute Gauss function of x, corresponding to "exp(-x*x/2) / sqrt(2*PI)".
gcd(x, y)
Return the greatest common divisor of x and y. If both x and y are 0 or either or both are less than
zero then behavior is undefined.
gt(x, y)
Return 1 if x is greater than y, 0 otherwise.
gte(x, y)
Return 1 if x is greater than or equal to y, 0 otherwise.
hypot(x, y)
This function is similar to the C function with the same name; it returns "sqrt(x*x + y*y)", the
length of the hypotenuse of a right triangle with sides of length x and y, or the distance of the
point (x, y) from the origin.
if(x, y)
Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0
otherwise.
if(x, y, z)
Evaluate x, and if the result is non-zero return the evaluation result of y, otherwise the evaluation
result of z.
ifnot(x, y)
Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.
ifnot(x, y, z)
Evaluate x, and if the result is zero return the evaluation result of y, otherwise the evaluation
result of z.
isinf(x)
Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
isnan(x)
Return 1.0 if x is NAN, 0.0 otherwise.
ld(var)
Load the value of the internal variable with number var, which was previously stored with st(var,
expr). The function returns the loaded value.
lerp(x, y, z)
Return linear interpolation between x and y by amount of z.
log(x)
Compute natural logarithm of x.
lt(x, y)
Return 1 if x is lesser than y, 0 otherwise.
lte(x, y)
Return 1 if x is lesser than or equal to y, 0 otherwise.
max(x, y)
Return the maximum between x and y.
min(x, y)
Return the minimum between x and y.
mod(x, y)
Compute the remainder of division of x by y.
not(expr)
Return 1.0 if expr is zero, 0.0 otherwise.
pow(x, y)
Compute the power of x elevated y, it is equivalent to "(x)^(y)".
print(t)
print(t, l)
Print the value of expression t with loglevel l. If l is not specified then a default log level is
used. Returns the value of the expression printed.
Prints t with loglevel l
random(x)
Return a pseudo random value between 0.0 and 1.0. x is the index of the internal variable which will
be used to save the seed/state.
root(expr, max)
Find an input value for which the function represented by expr with argument ld(0) is 0 in the
interval 0..max.
The expression in expr must denote a continuous function or the result is undefined.
ld(0) is used to represent the function input value, which means that the given expression will be
evaluated multiple times with various input values that the expression can access through ld(0). When
the expression evaluates to 0 then the corresponding input value will be returned.
round(expr)
Round the value of expression expr to the nearest integer. For example, "round(1.5)" is "2.0".
sin(x)
Compute sine of x.
sinh(x)
Compute hyperbolic sine of x.
sqrt(expr)
Compute the square root of expr. This is equivalent to "(expr)^.5".
squish(x)
Compute expression "1/(1 + exp(4*x))".
st(var, expr)
Store the value of the expression expr in an internal variable. var specifies the number of the
variable where to store the value, and it is a value ranging from 0 to 9. The function returns the
value stored in the internal variable. Note, Variables are currently not shared between expressions.
tan(x)
Compute tangent of x.
tanh(x)
Compute hyperbolic tangent of x.
taylor(expr, x)
taylor(expr, x, id)
Evaluate a Taylor series at x, given an expression representing the "ld(id)"-th derivative of a
function at 0.
When the series does not converge the result is undefined.
ld(id) is used to represent the derivative order in expr, which means that the given expression will
be evaluated multiple times with various input values that the expression can access through
"ld(id)". If id is not specified then 0 is assumed.
Note, when you have the derivatives at y instead of 0, "taylor(expr, x-y)" can be used.
time(0)
Return the current (wallclock) time in seconds.
trunc(expr)
Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is
"-1.0".
while(cond, expr)
Evaluate expression expr while the expression cond is non-zero, and returns the value of the last
expr evaluation, or NAN if cond was always false.
The following constants are available:
PI area of the unit disc, approximately 3.14
E exp(1) (Euler's number), approximately 2.718
PHI golden ratio (1+sqrt(5))/2, approximately 1.618
Assuming that an expression is considered "true" if it has a non-zero value, note that:
"*" works like AND
"+" works like OR
For example the construct:
if (A AND B) then C
is equivalent to:
if(A*B, C)
In your C code, you can extend the list of unary and binary functions, and define recognized constants,
so that they are available for your expressions.
The evaluator also recognizes the International System unit prefixes. If 'i' is appended after the
prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The 'B'
postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows
using for example 'KB', 'MiB', 'G' and 'B' as number postfix.
The list of available International System prefixes follows, with indication of the corresponding powers
of 10 and of 2.
y 10^-24 / 2^-80
z 10^-21 / 2^-70
a 10^-18 / 2^-60
f 10^-15 / 2^-50
p 10^-12 / 2^-40
n 10^-9 / 2^-30
u 10^-6 / 2^-20
m 10^-3 / 2^-10
c 10^-2
d 10^-1
h 10^2
k 10^3 / 2^10
K 10^3 / 2^10
M 10^6 / 2^20
G 10^9 / 2^30
T 10^12 / 2^40
P 10^15 / 2^40
E 10^18 / 2^50
Z 10^21 / 2^60
Y 10^24 / 2^70
OPENCL OPTIONS
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:
build_options
Set build options used to compile the registered kernels.
See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".
platform_idx
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()".
device_idx
Select the index of the device used 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()".
CODEC OPTIONS
libavcodec provides some generic global options, which can be set on all the encoders and decoders. In
addition each codec may support so-called private options, which are specific for a given codec.
Sometimes, a global option may only affect a specific kind of codec, and may be nonsensical or ignored by
another, so you need to be aware of the meaning of the specified options. Also some options are meant
only for decoding or encoding.
Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in
the "AVCodecContext" options or using the libavutil/opt.h API for programmatic use.
The list of supported options follow:
b integer (encoding,audio,video)
Set bitrate in bits/s. Default value is 200K.
ab integer (encoding,audio)
Set audio bitrate (in bits/s). Default value is 128K.
bt integer (encoding,video)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far
ratecontrol is willing to deviate from the target average bitrate value. This is not related to
min/max bitrate. Lowering tolerance too much has an adverse effect on quality.
flags flags (decoding/encoding,audio,video,subtitles)
Set generic flags.
Possible values:
mv4 Use four motion vector by macroblock (mpeg4).
qpel
Use 1/4 pel motion compensation.
loop
Use loop filter.
qscale
Use fixed qscale.
gmc Use gmc.
mv0 Always try a mb with mv=<0,0>.
input_preserved
pass1
Use internal 2pass ratecontrol in first pass mode.
pass2
Use internal 2pass ratecontrol in second pass mode.
gray
Only decode/encode grayscale.
emu_edge
Do not draw edges.
psnr
Set error[?] variables during encoding.
truncated
naq Normalize adaptive quantization.
ildct
Use interlaced DCT.
low_delay
Force low delay.
global_header
Place global headers in extradata instead of every keyframe.
bitexact
Only write platform-, build- and time-independent data. (except (I)DCT). This ensures that file
and data checksums are reproducible and match between platforms. Its primary use is for
regression testing.
aic Apply H263 advanced intra coding / mpeg4 ac prediction.
cbp Deprecated, use mpegvideo private options instead.
qprd
Deprecated, use mpegvideo private options instead.
ilme
Apply interlaced motion estimation.
cgop
Use closed gop.
me_method integer (encoding,video)
Set motion estimation method.
Possible values:
zero
zero motion estimation (fastest)
full
full motion estimation (slowest)
epzs
EPZS motion estimation (default)
esa esa motion estimation (alias for full)
tesa
tesa motion estimation
dia dia motion estimation (alias for epzs)
log log motion estimation
phods
phods motion estimation
x1 X1 motion estimation
hex hex motion estimation
umh umh motion estimation
iter
iter motion estimation
extradata_size integer
Set extradata size.
time_base rational number
Set codec time base.
It is the fundamental unit of time (in seconds) in terms of which frame timestamps are represented.
For fixed-fps content, timebase should be "1 / frame_rate" and timestamp increments should be
identically 1.
g integer (encoding,video)
Set the group of picture (GOP) size. Default value is 12.
ar integer (decoding/encoding,audio)
Set audio sampling rate (in Hz).
ac integer (decoding/encoding,audio)
Set number of audio channels.
cutoff integer (encoding,audio)
Set cutoff bandwidth. (Supported only by selected encoders, see their respective documentation
sections.)
frame_size integer (encoding,audio)
Set audio frame size.
Each submitted frame except the last must contain exactly frame_size samples per channel. May be 0
when the codec has CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is not restricted.
It is set by some decoders to indicate constant frame size.
frame_number integer
Set the frame number.
delay integer
qcomp float (encoding,video)
Set video quantizer scale compression (VBR). It is used as a constant in the ratecontrol equation.
Recommended range for default rc_eq: 0.0-1.0.
qblur float (encoding,video)
Set video quantizer scale blur (VBR).
qmin integer (encoding,video)
Set min video quantizer scale (VBR). Must be included between -1 and 69, default value is 2.
qmax integer (encoding,video)
Set max video quantizer scale (VBR). Must be included between -1 and 1024, default value is 31.
qdiff integer (encoding,video)
Set max difference between the quantizer scale (VBR).
bf integer (encoding,video)
Set max number of B frames between non-B-frames.
Must be an integer between -1 and 16. 0 means that B-frames are disabled. If a value of -1 is used,
it will choose an automatic value depending on the encoder.
Default value is 0.
b_qfactor float (encoding,video)
Set qp factor between P and B frames.
rc_strategy integer (encoding,video)
Set ratecontrol method.
b_strategy integer (encoding,video)
Set strategy to choose between I/P/B-frames.
ps integer (encoding,video)
Set RTP payload size in bytes.
mv_bits integer
header_bits integer
i_tex_bits integer
p_tex_bits integer
i_count integer
p_count integer
skip_count integer
misc_bits integer
frame_bits integer
codec_tag integer
bug flags (decoding,video)
Workaround not auto detected encoder bugs.
Possible values:
autodetect
old_msmpeg4
some old lavc generated msmpeg4v3 files (no autodetection)
xvid_ilace
Xvid interlacing bug (autodetected if fourcc==XVIX)
ump4
(autodetected if fourcc==UMP4)
no_padding
padding bug (autodetected)
amv
ac_vlc
illegal vlc bug (autodetected per fourcc)
qpel_chroma
std_qpel
old standard qpel (autodetected per fourcc/version)
qpel_chroma2
direct_blocksize
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge
edge padding bug (autodetected per fourcc/version)
hpel_chroma
dc_clip
ms Workaround various bugs in microsoft broken decoders.
trunc
trancated frames
lelim integer (encoding,video)
Set single coefficient elimination threshold for luminance (negative values also consider DC
coefficient).
celim integer (encoding,video)
Set single coefficient elimination threshold for chrominance (negative values also consider dc
coefficient)
strict integer (decoding/encoding,audio,video)
Specify how strictly to follow the standards.
Possible values:
very
strictly conform to an older more strict version of the spec or reference software
strict
strictly conform to all the things in the spec no matter what consequences
normal
unofficial
allow unofficial extensions
experimental
allow non standardized experimental things, experimental (unfinished/work in progress/not well
tested) decoders and encoders. Note: experimental decoders can pose a security risk, do not use
this for decoding untrusted input.
b_qoffset float (encoding,video)
Set QP offset between P and B frames.
err_detect flags (decoding,audio,video)
Set error detection flags.
Possible values:
crccheck
verify embedded CRCs
bitstream
detect bitstream specification deviations
buffer
detect improper bitstream length
explode
abort decoding on minor error detection
ignore_err
ignore decoding errors, and continue decoding. This is useful if you want to analyze the content
of a video and thus want everything to be decoded no matter what. This option will not result in
a video that is pleasing to watch in case of errors.
careful
consider things that violate the spec and have not been seen in the wild as errors
compliant
consider all spec non compliancies as errors
aggressive
consider things that a sane encoder should not do as an error
has_b_frames integer
block_align integer
mpeg_quant integer (encoding,video)
Use MPEG quantizers instead of H.263.
qsquish float (encoding,video)
How to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function).
rc_qmod_amp float (encoding,video)
Set experimental quantizer modulation.
rc_qmod_freq integer (encoding,video)
Set experimental quantizer modulation.
rc_override_count integer
rc_eq string (encoding,video)
Set rate control equation. When computing the expression, besides the standard functions defined in
the section 'Expression Evaluation', the following functions are available: bits2qp(bits),
qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI
isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
maxrate integer (encoding,audio,video)
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
minrate integer (encoding,audio,video)
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use
elsewise.
bufsize integer (encoding,audio,video)
Set ratecontrol buffer size (in bits).
rc_buf_aggressivity float (encoding,video)
Currently useless.
i_qfactor float (encoding,video)
Set QP factor between P and I frames.
i_qoffset float (encoding,video)
Set QP offset between P and I frames.
rc_init_cplx float (encoding,video)
Set initial complexity for 1-pass encoding.
dct integer (encoding,video)
Set DCT algorithm.
Possible values:
auto
autoselect a good one (default)
fastint
fast integer
int accurate integer
mmx
altivec
faan
floating point AAN DCT
lumi_mask float (encoding,video)
Compress bright areas stronger than medium ones.
tcplx_mask float (encoding,video)
Set temporal complexity masking.
scplx_mask float (encoding,video)
Set spatial complexity masking.
p_mask float (encoding,video)
Set inter masking.
dark_mask float (encoding,video)
Compress dark areas stronger than medium ones.
idct integer (decoding/encoding,video)
Select IDCT implementation.
Possible values:
auto
int
simple
simplemmx
simpleauto
Automatically pick a IDCT compatible with the simple one
arm
altivec
sh4
simplearm
simplearmv5te
simplearmv6
simpleneon
simplealpha
ipp
xvidmmx
faani
floating point AAN IDCT
slice_count integer
ec flags (decoding,video)
Set error concealment strategy.
Possible values:
guess_mvs
iterative motion vector (MV) search (slow)
deblock
use strong deblock filter for damaged MBs
favor_inter
favor predicting from the previous frame instead of the current
bits_per_coded_sample integer
pred integer (encoding,video)
Set prediction method.
Possible values:
left
plane
median
aspect rational number (encoding,video)
Set sample aspect ratio.
sar rational number (encoding,video)
Set sample aspect ratio. Alias to aspect.
debug flags (decoding/encoding,audio,video,subtitles)
Print specific debug info.
Possible values:
pict
picture info
rc rate control
bitstream
mb_type
macroblock (MB) type
qp per-block quantization parameter (QP)
mv motion vector
dct_coeff
green_metadata
display complexity metadata for the upcoming frame, GoP or for a given duration.
skip
startcode
pts
er error recognition
mmco
memory management control operations (H.264)
bugs
vis_qp
visualize quantization parameter (QP), lower QP are tinted greener
vis_mb_type
visualize block types
buffers
picture buffer allocations
thread_ops
threading operations
nomc
skip motion compensation
vismv integer (decoding,video)
Visualize motion vectors (MVs).
This option is deprecated, see the codecview filter instead.
Possible values:
pf forward predicted MVs of P-frames
bf forward predicted MVs of B-frames
bb backward predicted MVs of B-frames
cmp integer (encoding,video)
Set full pel me compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
subcmp integer (encoding,video)
Set sub pel me compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
mbcmp integer (encoding,video)
Set macroblock compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
ildctcmp integer (encoding,video)
Set interlaced dct compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
dia_size integer (encoding,video)
Set diamond type & size for motion estimation.
last_pred integer (encoding,video)
Set amount of motion predictors from the previous frame.
preme integer (encoding,video)
Set pre motion estimation.
precmp integer (encoding,video)
Set pre motion estimation compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
pre_dia_size integer (encoding,video)
Set diamond type & size for motion estimation pre-pass.
subq integer (encoding,video)
Set sub pel motion estimation quality.
dtg_active_format integer
me_range integer (encoding,video)
Set limit motion vectors range (1023 for DivX player).
ibias integer (encoding,video)
Set intra quant bias.
pbias integer (encoding,video)
Set inter quant bias.
color_table_id integer
global_quality integer (encoding,audio,video)
coder integer (encoding,video)
Possible values:
vlc variable length coder / huffman coder
ac arithmetic coder
raw raw (no encoding)
rle run-length coder
deflate
deflate-based coder
context integer (encoding,video)
Set context model.
slice_flags integer
xvmc_acceleration integer
mbd integer (encoding,video)
Set macroblock decision algorithm (high quality mode).
Possible values:
simple
use mbcmp (default)
bits
use fewest bits
rd use best rate distortion
stream_codec_tag integer
sc_threshold integer (encoding,video)
Set scene change threshold.
lmin integer (encoding,video)
Set min lagrange factor (VBR).
lmax integer (encoding,video)
Set max lagrange factor (VBR).
nr integer (encoding,video)
Set noise reduction.
rc_init_occupancy integer (encoding,video)
Set number of bits which should be loaded into the rc buffer before decoding starts.
flags2 flags (decoding/encoding,audio,video)
Possible values:
fast
Allow non spec compliant speedup tricks.
sgop
Deprecated, use mpegvideo private options instead.
noout
Skip bitstream encoding.
ignorecrop
Ignore cropping information from sps.
local_header
Place global headers at every keyframe instead of in extradata.
chunks
Frame data might be split into multiple chunks.
showall
Show all frames before the first keyframe.
skiprd
Deprecated, use mpegvideo private options instead.
export_mvs
Export motion vectors into frame side-data (see "AV_FRAME_DATA_MOTION_VECTORS") for codecs that
support it. See also doc/examples/export_mvs.c.
error integer (encoding,video)
qns integer (encoding,video)
Deprecated, use mpegvideo private options instead.
threads integer (decoding/encoding,video)
Set the number of threads to be used, in case the selected codec implementation supports multi-
threading.
Possible values:
auto, 0
automatically select the number of threads to set
Default value is auto.
me_threshold integer (encoding,video)
Set motion estimation threshold.
mb_threshold integer (encoding,video)
Set macroblock threshold.
dc integer (encoding,video)
Set intra_dc_precision.
nssew integer (encoding,video)
Set nsse weight.
skip_top integer (decoding,video)
Set number of macroblock rows at the top which are skipped.
skip_bottom integer (decoding,video)
Set number of macroblock rows at the bottom which are skipped.
profile integer (encoding,audio,video)
Possible values:
unknown
aac_main
aac_low
aac_ssr
aac_ltp
aac_he
aac_he_v2
aac_ld
aac_eld
mpeg2_aac_low
mpeg2_aac_he
mpeg4_sp
mpeg4_core
mpeg4_main
mpeg4_asp
dts
dts_es
dts_96_24
dts_hd_hra
dts_hd_ma
level integer (encoding,audio,video)
Possible values:
unknown
lowres integer (decoding,audio,video)
Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.
skip_threshold integer (encoding,video)
Set frame skip threshold.
skip_factor integer (encoding,video)
Set frame skip factor.
skip_exp integer (encoding,video)
Set frame skip exponent. Negative values behave identical to the corresponding positive ones, except
that the score is normalized. Positive values exist primarily for compatibility reasons and are not
so useful.
skipcmp integer (encoding,video)
Set frame skip compare function.
Possible values:
sad sum of absolute differences, fast (default)
sse sum of squared errors
satd
sum of absolute Hadamard transformed differences
dct sum of absolute DCT transformed differences
psnr
sum of squared quantization errors (avoid, low quality)
bit number of bits needed for the block
rd rate distortion optimal, slow
zero
0
vsad
sum of absolute vertical differences
vsse
sum of squared vertical differences
nsse
noise preserving sum of squared differences
w53 5/3 wavelet, only used in snow
w97 9/7 wavelet, only used in snow
dctmax
chroma
border_mask float (encoding,video)
Increase the quantizer for macroblocks close to borders.
mblmin integer (encoding,video)
Set min macroblock lagrange factor (VBR).
mblmax integer (encoding,video)
Set max macroblock lagrange factor (VBR).
mepc integer (encoding,video)
Set motion estimation bitrate penalty compensation (1.0 = 256).
skip_loop_filter integer (decoding,video)
skip_idct integer (decoding,video)
skip_frame integer (decoding,video)
Make decoder discard processing depending on the frame type selected by the option value.
skip_loop_filter skips frame loop filtering, skip_idct skips frame IDCT/dequantization, skip_frame
skips decoding.
Possible values:
none
Discard no frame.
default
Discard useless frames like 0-sized frames.
noref
Discard all non-reference frames.
bidir
Discard all bidirectional frames.
nokey
Discard all frames excepts keyframes.
all Discard all frames.
Default value is default.
bidir_refine integer (encoding,video)
Refine the two motion vectors used in bidirectional macroblocks.
brd_scale integer (encoding,video)
Downscale frames for dynamic B-frame decision.
keyint_min integer (encoding,video)
Set minimum interval between IDR-frames.
refs integer (encoding,video)
Set reference frames to consider for motion compensation.
chromaoffset integer (encoding,video)
Set chroma qp offset from luma.
trellis integer (encoding,audio,video)
Set rate-distortion optimal quantization.
sc_factor integer (encoding,video)
Set value multiplied by qscale for each frame and added to scene_change_score.
mv0_threshold integer (encoding,video)
b_sensitivity integer (encoding,video)
Adjust sensitivity of b_frame_strategy 1.
compression_level integer (encoding,audio,video)
min_prediction_order integer (encoding,audio)
max_prediction_order integer (encoding,audio)
timecode_frame_start integer (encoding,video)
Set GOP timecode frame start number, in non drop frame format.
request_channels integer (decoding,audio)
Set desired number of audio channels.
bits_per_raw_sample integer
channel_layout integer (decoding/encoding,audio)
Possible values:
request_channel_layout integer (decoding,audio)
Possible values:
rc_max_vbv_use float (encoding,video)
rc_min_vbv_use float (encoding,video)
ticks_per_frame integer (decoding/encoding,audio,video)
color_primaries integer (decoding/encoding,video)
Possible values:
bt709
BT.709
bt470m
BT.470 M
bt470bg
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
film
Film
bt2020
BT.2020
smpte428
smpte428_1
SMPTE ST 428-1
smpte431
SMPTE 431-2
smpte432
SMPTE 432-1
jedec-p22
JEDEC P22
color_trc integer (decoding/encoding,video)
Possible values:
bt709
BT.709
gamma22
BT.470 M
gamma28
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
linear
Linear
log
log100
Log
log_sqrt
log316
Log square root
iec61966_2_4
iec61966-2-4
IEC 61966-2-4
bt1361
bt1361e
BT.1361
iec61966_2_1
iec61966-2-1
IEC 61966-2-1
bt2020_10
bt2020_10bit
BT.2020 - 10 bit
bt2020_12
bt2020_12bit
BT.2020 - 12 bit
smpte2084
SMPTE ST 2084
smpte428
smpte428_1
SMPTE ST 428-1
arib-std-b67
ARIB STD-B67
colorspace integer (decoding/encoding,video)
Possible values:
rgb RGB
bt709
BT.709
fcc FCC
bt470bg
BT.470 BG
smpte170m
SMPTE 170 M
smpte240m
SMPTE 240 M
ycocg
YCOCG
bt2020nc
bt2020_ncl
BT.2020 NCL
bt2020c
bt2020_cl
BT.2020 CL
smpte2085
SMPTE 2085
color_range integer (decoding/encoding,video)
If used as input parameter, it serves as a hint to the decoder, which color_range the input has.
Possible values:
tv
mpeg
MPEG (219*2^(n-8))
pc
jpeg
JPEG (2^n-1)
chroma_sample_location integer (decoding/encoding,video)
Possible values:
left
center
topleft
top
bottomleft
bottom
log_level_offset integer
Set the log level offset.
slices integer (encoding,video)
Number of slices, used in parallelized encoding.
thread_type flags (decoding/encoding,video)
Select which multithreading methods to use.
Use of frame will increase decoding delay by one frame per thread, so clients which cannot provide
future frames should not use it.
Possible values:
slice
Decode more than one part of a single frame at once.
Multithreading using slices works only when the video was encoded with slices.
frame
Decode more than one frame at once.
Default value is slice+frame.
audio_service_type integer (encoding,audio)
Set audio service type.
Possible values:
ma Main Audio Service
ef Effects
vi Visually Impaired
hi Hearing Impaired
di Dialogue
co Commentary
em Emergency
vo Voice Over
ka Karaoke
request_sample_fmt sample_fmt (decoding,audio)
Set sample format audio decoders should prefer. Default value is "none".
pkt_timebase rational number
sub_charenc encoding (decoding,subtitles)
Set the input subtitles character encoding.
field_order field_order (video)
Set/override the field order of the video. Possible values:
progressive
Progressive video
tt Interlaced video, top field coded and displayed first
bb Interlaced video, bottom field coded and displayed first
tb Interlaced video, top coded first, bottom displayed first
bt Interlaced video, bottom coded first, top displayed first
skip_alpha bool (decoding,video)
Set to 1 to disable processing alpha (transparency). This works like the gray flag in the flags
option which skips chroma information instead of alpha. Default is 0.
codec_whitelist list (input)
"," separated list of allowed decoders. By default all are allowed.
dump_separator string (input)
Separator used to separate the fields printed on the command line about the Stream parameters. For
example to separate the fields with newlines and indention:
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
max_pixels integer (decoding/encoding,video)
Maximum number of pixels per image. This value can be used to avoid out of memory failures due to
large images.
apply_cropping bool (decoding,video)
Enable cropping if cropping parameters are multiples of the required alignment for the left and top
parameters. If the alignment is not met the cropping will be partially applied to maintain alignment.
Default is 1 (enabled). Note: The required alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set
and the CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command line. Also hardware
decoders will not apply left/top Cropping.
DECODERS
Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.
When you configure your FFmpeg build, all the supported native decoders are enabled by default. Decoders
requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You
can list all available decoders using the configure option "--list-decoders".
You can disable all the decoders with the configure option "--disable-decoders" and selectively enable /
disable single decoders with the options "--enable-decoder=DECODER" / "--disable-decoder=DECODER".
The option "-decoders" of the ff* tools will display the list of enabled decoders.
VIDEO DECODERS
A description of some of the currently available video decoders follows.
hevc
HEVC / H.265 decoder.
Note: the skip_loop_filter option has effect only at level "all".
rawvideo
Raw video decoder.
This decoder decodes rawvideo streams.
Options
top top_field_first
Specify the assumed field type of the input video.
-1 the video is assumed to be progressive (default)
0 bottom-field-first is assumed
1 top-field-first is assumed
AUDIO DECODERS
A description of some of the currently available audio decoders follows.
ac3
AC-3 audio decoder.
This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio
3 (a.k.a. dnet).
AC-3 Decoder Options
-drc_scale value
Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This
factor is applied exponentially. There are 3 notable scale factor ranges:
drc_scale == 0
DRC disabled. Produces full range audio.
0 < drc_scale <= 1
DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full
range and full compression.
drc_scale > 1
DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds
are enhanced.
flac
FLAC audio decoder.
This decoder aims to implement the complete FLAC specification from Xiph.
FLAC Decoder options
-use_buggy_lpc
The lavc FLAC encoder used to produce buggy streams with high lpc values (like the default value).
This option makes it possible to decode such streams correctly by using lavc's old buggy lpc logic
for decoding.
ffwavesynth
Internal wave synthesizer.
This decoder generates wave patterns according to predefined sequences. Its use is purely internal and
the format of the data it accepts is not publicly documented.
libcelt
libcelt decoder wrapper.
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of
the libcelt headers and library during configuration. You need to explicitly configure the build with
"--enable-libcelt".
libgsm
libgsm decoder wrapper.
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm
headers and library during configuration. You need to explicitly configure the build with
"--enable-libgsm".
This decoder supports both the ordinary GSM and the Microsoft variant.
libilbc
libilbc decoder wrapper.
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the
presence of the libilbc headers and library during configuration. You need to explicitly configure the
build with "--enable-libilbc".
Options
The following option is supported by the libilbc wrapper.
enhance
Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).
libopencore-amrnb
libopencore-amrnb decoder wrapper.
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it
requires the presence of the libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrnb".
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.
libopencore-amrwb
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it
requires the presence of the libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrwb".
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.
libopus
libopus decoder wrapper.
libopus allows libavcodec to decode the Opus Interactive Audio Codec. Requires the presence of the
libopus headers and library during configuration. You need to explicitly configure the build with
"--enable-libopus".
An FFmpeg native decoder for Opus exists, so users can decode Opus without this library.
SUBTITLES DECODERS
dvbsub
Options
compute_clut
-1 Compute clut if no matching CLUT is in the stream.
0 Never compute CLUT
1 Always compute CLUT and override the one provided in the stream.
dvb_substream
Selects the dvb substream, or all substreams if -1 which is default.
dvdsub
This codec decodes the bitmap subtitles used in DVDs; the same subtitles can also be found in VobSub file
pairs and in some Matroska files.
Options
palette
Specify the global palette used by the bitmaps. When stored in VobSub, the palette is normally
specified in the index file; in Matroska, the palette is stored in the codec extra-data in the same
format as in VobSub. In DVDs, the palette is stored in the IFO file, and therefore not available when
reading from dumped VOB files.
The format for this option is a string containing 16 24-bits hexadecimal numbers (without 0x prefix)
separated by comas, for example "0d00ee, ee450d, 101010, eaeaea, 0ce60b, ec14ed, ebff0b, 0d617a,
7b7b7b, d1d1d1, 7b2a0e, 0d950c, 0f007b, cf0dec, cfa80c, 7c127b".
ifo_palette
Specify the IFO file from which the global palette is obtained. (experimental)
forced_subs_only
Only decode subtitle entries marked as forced. Some titles have forced and non-forced subtitles in
the same track. Setting this flag to 1 will only keep the forced subtitles. Default value is 0.
libzvbi-teletext
Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext subtitles. Requires the presence
of the libzvbi headers and library during configuration. You need to explicitly configure the build with
"--enable-libzvbi".
Options
txt_page
List of teletext page numbers to decode. You may use the special * string to match all pages. Pages
that do not match the specified list are dropped. Default value is *.
txt_chop_top
Discards the top teletext line. Default value is 1.
txt_format
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.
txt_left
X offset of generated bitmaps, default is 0.
txt_top
Y offset of generated bitmaps, default is 0.
txt_chop_spaces
Chops leading and trailing spaces and removes empty lines from the generated text. This option is
useful for teletext based subtitles where empty spaces may be present at the start or at the end of
the lines or empty lines may be present between the subtitle lines because of double-sized teletext
characters. Default value is 1.
txt_duration
Sets the display duration of the decoded teletext pages or subtitles in milliseconds. Default value
is 30000 which is 30 seconds.
txt_transparent
Force transparent background of the generated teletext bitmaps. Default value is 0 which means an
opaque background.
txt_opacity
Sets the opacity (0-255) of the teletext background. If txt_transparent is not set, it only affects
characters between a start box and an end box, typically subtitles. Default value is 0 if
txt_transparent is set, 255 otherwise.
ENCODERS
Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.
When you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders
requiring an external library must be enabled manually via the corresponding "--enable-lib" option. You
can list all available encoders using the configure option "--list-encoders".
You can disable all the encoders with the configure option "--disable-encoders" and selectively enable /
disable single encoders with the options "--enable-encoder=ENCODER" / "--disable-encoder=ENCODER".
The option "-encoders" of the ff* tools will display the list of enabled encoders.
AUDIO ENCODERS
A description of some of the currently available audio encoders follows.
aac
Advanced Audio Coding (AAC) encoder.
This encoder is the default AAC encoder, natively implemented into FFmpeg. Its quality is on par or
better than libfdk_aac at the default bitrate of 128kbps. This encoder also implements more options,
profiles and samplerates than other encoders (with only the AAC-HE profile pending to be implemented) so
this encoder has become the default and is the recommended choice.
Options
b Set bit rate in bits/s. Setting this automatically activates constant bit rate (CBR) mode. If this
option is unspecified it is set to 128kbps.
q Set quality for variable bit rate (VBR) mode. This option is valid only using the ffmpeg command-line
tool. For library interface users, use global_quality.
cutoff
Set cutoff frequency. If unspecified will allow the encoder to dynamically adjust the cutoff to
improve clarity on low bitrates.
aac_coder
Set AAC encoder coding method. Possible values:
twoloop
Two loop searching (TLS) method.
This method first sets quantizers depending on band thresholds and then tries to find an optimal
combination by adding or subtracting a specific value from all quantizers and adjusting some
individual quantizer a little. Will tune itself based on whether aac_is, aac_ms and aac_pns are
enabled. This is the default choice for a coder.
anmr
Average noise to mask ratio (ANMR) trellis-based solution.
This is an experimental coder which currently produces a lower quality, is more unstable and is
slower than the default twoloop coder but has potential. Currently has no support for the aac_is
or aac_pns options. Not currently recommended.
fast
Constant quantizer method.
This method sets a constant quantizer for all bands. This is the fastest of all the methods and
has no rate control or support for aac_is or aac_pns. Not recommended.
aac_ms
Sets mid/side coding mode. The default value of "auto" will automatically use M/S with bands which
will benefit from such coding. Can be forced for all bands using the value "enable", which is mainly
useful for debugging or disabled using "disable".
aac_is
Sets intensity stereo coding tool usage. By default, it's enabled and will automatically toggle IS
for similar pairs of stereo bands if it's beneficial. Can be disabled for debugging by setting the
value to "disable".
aac_pns
Uses perceptual noise substitution to replace low entropy high frequency bands with imperceptible
white noise during the decoding process. By default, it's enabled, but can be disabled for debugging
purposes by using "disable".
aac_tns
Enables the use of a multitap FIR filter which spans through the high frequency bands to hide
quantization noise during the encoding process and is reverted by the decoder. As well as decreasing
unpleasant artifacts in the high range this also reduces the entropy in the high bands and allows for
more bits to be used by the mid-low bands. By default it's enabled but can be disabled for debugging
by setting the option to "disable".
aac_ltp
Enables the use of the long term prediction extension which increases coding efficiency in very low
bandwidth situations such as encoding of voice or solo piano music by extending constant harmonic
peaks in bands throughout frames. This option is implied by profile:a aac_low and is incompatible
with aac_pred. Use in conjunction with -ar to decrease the samplerate.
aac_pred
Enables the use of a more traditional style of prediction where the spectral coefficients transmitted
are replaced by the difference of the current coefficients minus the previous "predicted"
coefficients. In theory and sometimes in practice this can improve quality for low to mid bitrate
audio. This option implies the aac_main profile and is incompatible with aac_ltp.
profile
Sets the encoding profile, possible values:
aac_low
The default, AAC "Low-complexity" profile. Is the most compatible and produces decent quality.
mpeg2_aac_low
Equivalent to "-profile:a aac_low -aac_pns 0". PNS was introduced with the MPEG4 specifications.
aac_ltp
Long term prediction profile, is enabled by and will enable the aac_ltp option. Introduced in
MPEG4.
aac_main
Main-type prediction profile, is enabled by and will enable the aac_pred option. Introduced in
MPEG2.
If this option is unspecified it is set to aac_low.
ac3 and ac3_fixed
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented
RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed encoder only uses fixed-point integer math.
This does not mean that one is always faster, just that one or the other may be better suited to a
particular system. The 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.
AC-3 Metadata
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not
affect the audio encoding itself. Some of the options do directly affect or influence the decoding and
playback of the resulting bitstream, while others are just for informational purposes. A few of the
options will add bits to the output stream that could otherwise be used for audio data, and will thus
affect the quality of the output. Those will be indicated accordingly with a note in the option list
below.
These parameters are described in detail in several publicly-available documents.
*<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>
*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>
Metadata Control Options
-per_frame_metadata boolean
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
0 The metadata values set at initialization will be used for every frame in the stream. (default)
1 Metadata values can be changed before encoding each frame.
Downmix Levels
-center_mixlev level
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing
to stereo. This field will only be written to the bitstream if a center channel is present. The value
is specified as a scale factor. There are 3 valid values:
0.707
Apply -3dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6dB gain
-surround_mixlev level
Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo. This field will only be written to the bitstream if one or more surround
channels are present. The value is specified as a scale factor. There are 3 valid values:
0.707
Apply -3dB gain
0.500
Apply -6dB gain (default)
0.000
Silence Surround Channel(s)
Audio Production Information
Audio Production Information is optional information describing the mixing environment. Either none or
both of the fields are written to the bitstream.
-mixing_level number
Mixing Level. Specifies peak sound pressure level (SPL) in the production environment when the mix
was mastered. Valid values are 80 to 111, or -1 for unknown or not indicated. The default value is
-1, but that value cannot be used if the Audio Production Information is written to the bitstream.
Therefore, if the "room_type" option is not the default value, the "mixing_level" option must not be
-1.
-room_type type
Room Type. Describes the equalization used during the final mixing session at the studio or on the
dubbing stage. A large room is a dubbing stage with the industry standard X-curve equalization; a
small room has flat equalization. This field will not be written to the bitstream if both the
"mixing_level" option and the "room_type" option have the default values.
0
notindicated
Not Indicated (default)
1
large
Large Room
2
small
Small Room
Other Metadata Options
-copyright boolean
Copyright Indicator. Specifies whether a copyright exists for this audio.
0
off No Copyright Exists (default)
1
on Copyright Exists
-dialnorm value
Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital
100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets
the average volume of the dialogue to a preset level. The goal is to match volume level between
program sources. A value of -31dB will result in no volume level change, relative to the source
volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31
being the default.
-dsur_mode mode
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field
will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean
the encoder will actually apply Dolby Surround processing.
0
notindicated
Not Indicated (default)
1
off Not Dolby Surround Encoded
2
on Dolby Surround Encoded
-original boolean
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a
copy.
0
off Not Original Source
1
on Original Source (default)
Extended Bitstream Information
The extended bitstream options are part of the Alternate Bit Stream Syntax as specified in Annex D of the
A/52:2010 standard. It is grouped into 2 parts. If any one parameter in a group is specified, all values
in that group will be written to the bitstream. Default values are used for those that are written but
have not been specified. If the mixing levels are written, the decoder will use these values instead of
the ones specified in the "center_mixlev" and "surround_mixlev" options if it supports the Alternate Bit
Stream Syntax.
Extended Bitstream Information - Part 1
-dmix_mode mode
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro
(normal stereo) as the preferred stereo downmix mode.
0
notindicated
Not Indicated (default)
1
ltrt
Lt/Rt Downmix Preferred
2
loro
Lo/Ro Downmix Preferred
-ltrt_cmixlev level
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when
downmixing to stereo in Lt/Rt mode.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
-ltrt_surmixlev level
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo in Lt/Rt mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)
-loro_cmixlev level
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when
downmixing to stereo in Lo/Ro mode.
1.414
Apply +3dB gain
1.189
Apply +1.5dB gain
1.000
Apply 0dB gain
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain (default)
0.500
Apply -6.0dB gain
0.000
Silence Center Channel
-loro_surmixlev level
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when
downmixing to stereo in Lo/Ro mode.
0.841
Apply -1.5dB gain
0.707
Apply -3.0dB gain
0.595
Apply -4.5dB gain
0.500
Apply -6.0dB gain (default)
0.000
Silence Surround Channel(s)
Extended Bitstream Information - Part 2
-dsurex_mode mode
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1).
Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.
0
notindicated
Not Indicated (default)
1
on Dolby Surround EX Off
2
off Dolby Surround EX On
-dheadphone_mode mode
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel
matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually
apply Dolby Headphone processing.
0
notindicated
Not Indicated (default)
1
on Dolby Headphone Off
2
off Dolby Headphone On
-ad_conv_type type
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
0
standard
Standard A/D Converter (default)
1
hdcd
HDCD A/D Converter
Other AC-3 Encoding Options
-stereo_rematrixing boolean
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3
feature that increases quality by selectively encoding the left/right channels as mid/side. This
option is enabled by default, and it is highly recommended that it be left as enabled except for
testing purposes.
cutoff frequency
Set lowpass cutoff frequency. If unspecified, the encoder selects a default determined by various
other encoding parameters.
Floating-Point-Only AC-3 Encoding Options
These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder
due to the corresponding features not being implemented in fixed-point.
-channel_coupling boolean
Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by
combining high frequency information from multiple channels into a single channel. The per-channel
high frequency information is sent with less accuracy in both the frequency and time domains. This
allows more bits to be used for lower frequencies while preserving enough information to reconstruct
the high frequencies. This option is enabled by default for the floating-point encoder and should
generally be left as enabled except for testing purposes or to increase encoding speed.
-1
auto
Selected by Encoder (default)
0
off Disable Channel Coupling
1
on Enable Channel Coupling
-cpl_start_band number
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the
bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the
start band will be determined by the encoder based on the bit rate, sample rate, and channel layout.
This option has no effect if channel coupling is disabled.
-1
auto
Selected by Encoder (default)
flac
FLAC (Free Lossless Audio Codec) Encoder
Options
The following options are supported by FFmpeg's flac encoder.
compression_level
Sets the compression level, which chooses defaults for many other options if they are not set
explicitly. Valid values are from 0 to 12, 5 is the default.
frame_size
Sets the size of the frames in samples per channel.
lpc_coeff_precision
Sets the LPC coefficient precision, valid values are from 1 to 15, 15 is the default.
lpc_type
Sets the first stage LPC algorithm
none
LPC is not used
fixed
fixed LPC coefficients
levinson
cholesky
lpc_passes
Number of passes to use for Cholesky factorization during LPC analysis
min_partition_order
The minimum partition order
max_partition_order
The maximum partition order
prediction_order_method
estimation
2level
4level
8level
search
Bruteforce search
log
ch_mode
Channel mode
auto
The mode is chosen automatically for each frame
indep
Channels are independently coded
left_side
right_side
mid_side
exact_rice_parameters
Chooses if rice parameters are calculated exactly or approximately. if set to 1 then they are chosen
exactly, which slows the code down slightly and improves compression slightly.
multi_dim_quant
Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied after the first
stage to finetune the coefficients. This is quite slow and slightly improves compression.
opus
Opus encoder.
This is a native FFmpeg encoder for the Opus format. Currently its in development and only implements the
CELT part of the codec. Its quality is usually worse and at best is equal to the libopus encoder.
Options
b Set bit rate in bits/s. If unspecified it uses the number of channels and the layout to make a good
guess.
opus_delay
Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly decrease quality.
libfdk_aac
libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.
The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project.
Requires the presence of the libfdk-aac headers and library during configuration. You need to explicitly
configure the build with "--enable-libfdk-aac". The library is also incompatible with GPL, so if you
allow the use of GPL, you should configure with "--enable-gpl --enable-nonfree --enable-libfdk-aac".
This encoder is considered to produce output on par or worse at 128kbps to the the native FFmpeg AAC
encoder but can often produce better sounding audio at identical or lower bitrates and has support for
the AAC-HE profiles.
VBR encoding, enabled through the vbr or flags +qscale options, is experimental and only works with some
combinations of parameters.
Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or higher.
For more information see the fdk-aac project at <http://sourceforge.net/p/opencore-amr/fdk-aac/>.
Options
The following options are mapped on the shared FFmpeg codec options.
b Set bit rate in bits/s. If the bitrate is not explicitly specified, it is automatically set to a
suitable value depending on the selected profile.
In case VBR mode is enabled the option is ignored.
ar Set audio sampling rate (in Hz).
channels
Set the number of audio channels.
flags +qscale
Enable fixed quality, VBR (Variable Bit Rate) mode. Note that VBR is implicitly enabled when the vbr
value is positive.
cutoff
Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value automatically
computed by the library. Default value is 0.
profile
Set audio profile.
The following profiles are recognized:
aac_low
Low Complexity AAC (LC)
aac_he
High Efficiency AAC (HE-AAC)
aac_he_v2
High Efficiency AAC version 2 (HE-AACv2)
aac_ld
Low Delay AAC (LD)
aac_eld
Enhanced Low Delay AAC (ELD)
If not specified it is set to aac_low.
The following are private options of the libfdk_aac encoder.
afterburner
Enable afterburner feature if set to 1, disabled if set to 0. This improves the quality but also the
required processing power.
Default value is 1.
eld_sbr
Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.
Default value is 0.
signaling
Set SBR/PS signaling style.
It can assume one of the following values:
default
choose signaling implicitly (explicit hierarchical by default, implicit if global header is
disabled)
implicit
implicit backwards compatible signaling
explicit_sbr
explicit SBR, implicit PS signaling
explicit_hierarchical
explicit hierarchical signaling
Default value is default.
latm
Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.
Default value is 0.
header_period
Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band configuration buffers
within LATM/LOAS transport layer.
Must be a 16-bits non-negative integer.
Default value is 0.
vbr Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is highest quality. A
value of 0 will disable VBR, and CBR (Constant Bit Rate) is enabled.
Currently only the aac_low profile supports VBR encoding.
VBR modes 1-5 correspond to roughly the following average bit rates:
1 32 kbps/channel
2 40 kbps/channel
3 48-56 kbps/channel
4 64 kbps/channel
5 about 80-96 kbps/channel
Default value is 0.
Examples
• Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4) container:
ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a
• Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC profile:
ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a
libmp3lame
LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.
Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly
configure the build with "--enable-libmp3lame".
See libshine for a fixed-point MP3 encoder, although with a lower quality.
Options
The following options are supported by the libmp3lame wrapper. The lame-equivalent of the options are
listed in parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in kilobits/s.
q (-V)
Set constant quality setting for VBR. This option is valid only using the ffmpeg command-line tool.
For library interface users, use global_quality.
compression_level (-q)
Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning highest quality
but slowest, and 9 meaning fastest while producing the worst quality.
cutoff (--lowpass)
Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the cutoff.
reservoir
Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled by default, but
can be overridden by use --nores option.
joint_stereo (-m j)
Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. Default
value is 1.
abr (--abr)
Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate, while this
options only tells FFmpeg to use ABR still relies on b to set bitrate.
libopencore-amrnb
OpenCORE Adaptive Multi-Rate Narrowband encoder.
Requires the presence of the libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with "--enable-libopencore-amrnb --enable-version3".
This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by
setting strict to unofficial or lower.
Options
b Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will
round to the nearest valid bitrate.
4750
5150
5900
6700
7400
7950
10200
12200
dtx Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0
(disabled).
libopus
libopus Opus Interactive Audio Codec encoder wrapper.
Requires the presence of the libopus headers and library during configuration. You need to explicitly
configure the build with "--enable-libopus".
Option Mapping
Most libopus options are modelled after the opusenc utility from opus-tools. The following is an option
mapping chart describing options supported by the libopus wrapper, and their opusenc-equivalent in
parentheses.
b (bitrate)
Set the bit rate in bits/s. FFmpeg's b option is expressed in bits/s, while opusenc's bitrate in
kilobits/s.
vbr (vbr, hard-cbr, and cvbr)
Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the opusenc equivalent
options in parentheses:
off (hard-cbr)
Use constant bit rate encoding.
on (vbr)
Use variable bit rate encoding (the default).
constrained (cvbr)
Use constrained variable bit rate encoding.
compression_level (comp)
Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0 gives the fastest
encodes but lower quality, while 10 gives the highest quality but slowest encoding. The default is
10.
frame_duration (framesize)
Set maximum frame size, or duration of a frame in milliseconds. The argument must be exactly the
following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower latency but less quality at a
given bitrate. Sizes greater than 20ms are only interesting at fairly low bitrates. The default is
20ms.
packet_loss (expect-loss)
Set expected packet loss percentage. The default is 0.
application (N.A.)
Set intended application type. Valid options are listed below:
voip
Favor improved speech intelligibility.
audio
Favor faithfulness to the input (the default).
lowdelay
Restrict to only the lowest delay modes.
cutoff (N.A.)
Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000, 6000, 8000,
12000, or 20000, corresponding to narrowband, mediumband, wideband, super wideband, and fullband
respectively. The default is 0 (cutoff disabled).
mapping_family (mapping_family)
Set channel mapping family to be used by the encoder. The default value of -1 uses mapping family 0
for mono and stereo inputs, and mapping family 1 otherwise. The default also disables the surround
masking and LFE bandwidth optimzations in libopus, and requires that the input contains 8 channels or
fewer.
Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE bandwidth
optimizations, and 255 for independent streams with an unspecified channel layout.
libshine
Shine Fixed-Point MP3 encoder wrapper.
Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without an FPU, e.g.
armel CPUs, and some phones and tablets. However, as it is more targeted on performance than quality, it
is not on par with LAME and other production-grade encoders quality-wise. Also, according to the
project's homepage, this encoder may not be free of bugs as the code was written a long time ago and the
project was dead for at least 5 years.
This encoder only supports stereo and mono input. This is also CBR-only.
The original project (last updated in early 2007) is at <http://sourceforge.net/projects/libshine-fxp/>.
We only support the updated fork by the Savonet/Liquidsoap project at <https://github.com/savonet/shine>.
Requires the presence of the libshine headers and library during configuration. You need to explicitly
configure the build with "--enable-libshine".
See also libmp3lame.
Options
The following options are supported by the libshine wrapper. The shineenc-equivalent of the options are
listed in parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR. shineenc -b option is expressed in kilobits/s.
libtwolame
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".
Options
The following options are supported by the libtwolame wrapper. The twolame-equivalent options follow the
FFmpeg ones and are in parentheses.
b (-b)
Set bitrate expressed in bits/s for CBR. twolame b option is expressed in kilobits/s. Default value
is 128k.
q (-V)
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.
mode (--mode)
Set the mode of the resulting audio. Possible values:
auto
Choose mode automatically based on the input. This is the default.
stereo
Stereo
joint_stereo
Joint stereo
dual_channel
Dual channel
mono
Mono
psymodel (--psyc-mode)
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.
energy_levels (--energy)
Enable energy levels extensions when set to 1. The default value is 0 (disabled).
error_protection (--protect)
Enable CRC error protection when set to 1. The default value is 0 (disabled).
copyright (--copyright)
Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).
original (--original)
Set MPEG audio original flag when set to 1. The default value is 0 (disabled).
libvo-amrwbenc
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.
Options
b Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to
the nearest valid bitrate.
6600
8850
12650
14250
15850
18250
19850
23050
23850
dtx Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0
(disabled).
libvorbis
libvorbis encoder wrapper.
Requires the presence of the libvorbisenc headers and library during configuration. You need to
explicitly configure the build with "--enable-libvorbis".
Options
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).
b (-b)
Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.
q (-q)
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.
cutoff (--advanced-encode-option lowpass_frequency=N)
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).
minrate (-m)
Set minimum bitrate expressed in bits/s. oggenc -m is expressed in kilobits/s.
maxrate (-M)
Set maximum bitrate expressed in bits/s. oggenc -M is expressed in kilobits/s. This only has effect
on ABR mode.
iblock (--advanced-encode-option impulse_noisetune=N)
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.
libwavpack
A wrapper providing WavPack encoding through libwavpack.
Only lossless mode using 32-bit integer samples is supported currently.
Requires the presence of the libwavpack headers and library during configuration. You need to explicitly
configure the build with "--enable-libwavpack".
Note that a libavcodec-native encoder for the WavPack codec exists so users can encode audios with this
codec without using this encoder. See wavpackenc.
Options
wavpack command line utility's corresponding options are listed in parentheses, if any.
frame_size (--blocksize)
Default is 32768.
compression_level
Set speed vs. compression tradeoff. Acceptable arguments are listed below:
0 (-f)
Fast mode.
1 Normal (default) settings.
2 (-h)
High quality.
3 (-hh)
Very high quality.
4-8 (-hh -xEXTRAPROC)
Same as 3, but with extra processing enabled.
4 is the same as -x2 and 8 is the same as -x6.
mjpeg
Motion JPEG encoder.
Options
huffman
Set the huffman encoding strategy. Possible values:
default
Use the default huffman tables. This is the default strategy.
optimal
Compute and use optimal huffman tables.
wavpack
WavPack lossless audio encoder.
This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack, but there is
virtually no reason to use that encoder.
See also libwavpack.
Options
The equivalent options for wavpack command line utility are listed in parentheses.
Shared options
The following shared options are effective for this encoder. Only special notes about this particular
encoder will be documented here. For the general meaning of the options, see the Codec Options chapter.
frame_size (--blocksize)
For this encoder, the range for this option is between 128 and 131072. Default is automatically
decided based on sample rate and number of channel.
For the complete formula of calculating default, see libavcodec/wavpackenc.c.
compression_level (-f, -h, -hh, and -x)
This option's syntax is consistent with libwavpack's.
Private options
joint_stereo (-j)
Set whether to enable joint stereo. Valid values are:
on (1)
Force mid/side audio encoding.
off (0)
Force left/right audio encoding.
auto
Let the encoder decide automatically.
optimize_mono
Set whether to enable optimization for mono. This option is only effective for non-mono streams.
Available values:
on enabled
off disabled
VIDEO ENCODERS
A description of some of the currently available video encoders follows.
Hap
Vidvox Hap video encoder.
Options
format integer
Specifies the Hap format to encode.
hap
hap_alpha
hap_q
Default value is hap.
chunks integer
Specifies the number of chunks to split frames into, between 1 and 64. This permits multithreaded
decoding of large frames, potentially at the cost of data-rate. The encoder may modify this value to
divide frames evenly.
Default value is 1.
compressor integer
Specifies the second-stage compressor to use. If set to none, chunks will be limited to 1, as chunked
uncompressed frames offer no benefit.
none
snappy
Default value is snappy.
jpeg2000
The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the encoding
quality. Lossless encoding can be selected with "-pred 1".
Options
format
Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-rgb pix_fmts.
libkvazaar
Kvazaar H.265/HEVC encoder.
Requires the presence of the libkvazaar headers and library during configuration. You need to explicitly
configure the build with --enable-libkvazaar.
Options
b Set target video bitrate in bit/s and enable rate control.
kvazaar-params
Set kvazaar parameters as a list of name=value pairs separated by commas (,). See kvazaar
documentation for a list of options.
libopenh264
Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libopenh264 headers and library during configuration. You need
to explicitly configure the build with "--enable-libopenh264". The library is detected using pkg-config.
For more information about the library see <http://www.openh264.org>.
Options
The following FFmpeg global options affect the configurations of the libopenh264 encoder.
b Set the bitrate (as a number of bits per second).
g Set the GOP size.
maxrate
Set the max bitrate (as a number of bits per second).
flags +global_header
Set global header in the bitstream.
slices
Set the number of slices, used in parallelized encoding. Default value is 0. This is only used when
slice_mode is set to fixed.
slice_mode
Set slice mode. Can assume one of the following possible values:
fixed
a fixed number of slices
rowmb
one slice per row of macroblocks
auto
automatic number of slices according to number of threads
dyn dynamic slicing
Default value is auto.
loopfilter
Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.
profile
Set profile restrictions. If set to the value of main enable CABAC (set the
"SEncParamExt.iEntropyCodingModeFlag" flag to 1).
max_nal_size
Set maximum NAL size in bytes.
allow_skip_frames
Allow skipping frames to hit the target bitrate if set to 1.
libtheora
libtheora Theora encoder wrapper.
Requires the presence of the libtheora headers and library during configuration. You need to explicitly
configure the build with "--enable-libtheora".
For more information about the libtheora project see <http://www.theora.org/>.
Options
The following global options are mapped to internal libtheora options which affect the quality and the
bitrate of the encoded stream.
b Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode. In case VBR (Variable Bit Rate)
mode is enabled this option is ignored.
flags
Used to enable constant quality mode (VBR) encoding through the qscale flag, and to enable the
"pass1" and "pass2" modes.
g Set the GOP size.
global_quality
Set the global quality as an integer in lambda units.
Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to QP units by
dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
value in the native libtheora range [0-63]. A higher value corresponds to a higher quality.
q Enable VBR mode when set to a non-negative value, and set constant quality value as a double floating
point value in QP units.
The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in the native
libtheora range [0-63].
This option is valid only using the ffmpeg command-line tool. For library interface users, use
global_quality.
Examples
• Set maximum constant quality (VBR) encoding with ffmpeg:
ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg
• Use ffmpeg to convert a CBR 1000 kbps Theora video stream:
ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg
libvpx
VP8/VP9 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration. You need to explicitly
configure the build with "--enable-libvpx".
Options
The following options are supported by the libvpx wrapper. The vpxenc-equivalent options or values are
listed in parentheses for easy migration.
To reduce the duplication of documentation, only the private options and some others requiring special
attention are documented here. For the documentation of the undocumented generic options, see the Codec
Options chapter.
To get more documentation of the libvpx options, invoke the command ffmpeg -h encoder=libvpx, ffmpeg -h
encoder=libvpx-vp9 or vpxenc --help. Further information is available in the libvpx API documentation.
b (target-bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while vpxenc's target-
bitrate is in kilobits/s.
g (kf-max-dist)
keyint_min (kf-min-dist)
qmin (min-q)
qmax (max-q)
bufsize (buf-sz, buf-optimal-sz)
Set ratecontrol buffer size (in bits). Note vpxenc's options are specified in milliseconds, the
libvpx wrapper converts this value as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz =
bufsize * 1000 / bitrate * 5 / 6".
rc_init_occupancy (buf-initial-sz)
Set number of bits which should be loaded into the rc buffer before decoding starts. Note vpxenc's
option is specified in milliseconds, the libvpx wrapper converts this value as follows:
"rc_init_occupancy * 1000 / bitrate".
undershoot-pct
Set datarate undershoot (min) percentage of the target bitrate.
overshoot-pct
Set datarate overshoot (max) percentage of the target bitrate.
skip_threshold (drop-frame)
qcomp (bias-pct)
maxrate (maxsection-pct)
Set GOP max bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target
bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100 / bitrate)".
minrate (minsection-pct)
Set GOP min bitrate in bits/s. Note vpxenc's option is specified as a percentage of the target
bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100 / bitrate)".
minrate, maxrate, b end-usage=cbr
"(minrate == maxrate == bitrate)".
crf (end-usage=cq, cq-level)
tune (tune)
psnr (psnr)
ssim (ssim)
quality, deadline (deadline)
best
Use best quality deadline. Poorly named and quite slow, this option should be avoided as it may
give worse quality output than good.
good
Use good quality deadline. This is a good trade-off between speed and quality when used with the
cpu-used option.
realtime
Use realtime quality deadline.
speed, cpu-used (cpu-used)
Set quality/speed ratio modifier. Higher values speed up the encode at the cost of quality.
nr (noise-sensitivity)
static-thresh
Set a change threshold on blocks below which they will be skipped by the encoder.
slices (token-parts)
Note that FFmpeg's slices option gives the total number of partitions, while vpxenc's token-parts is
given as "log2(partitions)".
max-intra-rate
Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means unlimited.
force_key_frames
"VPX_EFLAG_FORCE_KF"
Alternate reference frame related
auto-alt-ref
Enable use of alternate reference frames (2-pass only).
arnr-max-frames
Set altref noise reduction max frame count.
arnr-type
Set altref noise reduction filter type: backward, forward, centered.
arnr-strength
Set altref noise reduction filter strength.
rc-lookahead, lag-in-frames (lag-in-frames)
Set number of frames to look ahead for frametype and ratecontrol.
error-resilient
Enable error resiliency features.
VP9-specific options
lossless
Enable lossless mode.
tile-columns
Set number of tile columns to use. Note this is given as "log2(tile_columns)". For example, 8
tile columns would be requested by setting the tile-columns option to 3.
tile-rows
Set number of tile rows to use. Note this is given as "log2(tile_rows)". For example, 4 tile
rows would be requested by setting the tile-rows option to 2.
frame-parallel
Enable frame parallel decodability features.
aq-mode
Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3: cyclic refresh,
4: equator360).
colorspace color-space
Set input color space. The VP9 bitstream supports signaling the following colorspaces:
rgb sRGB
bt709 bt709
unspecified unknown
bt470bg bt601
smpte170m smpte170
smpte240m smpte240
bt2020_ncl bt2020
row-mt boolean
Enable row based multi-threading.
For more information about libvpx see: <http://www.webmproject.org/>
libwebp
libwebp WebP Image encoder wrapper
libwebp is Google's official encoder for WebP images. It can encode in either lossy or lossless mode.
Lossy images are essentially a wrapper around a VP8 frame. Lossless images are a separate codec developed
by Google.
Pixel Format
Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations of the format
and libwebp. Alpha is supported for either mode. Because of API limitations, if RGB is passed in when
encoding lossy or YUV is passed in for encoding lossless, the pixel format will automatically be
converted using functions from libwebp. This is not ideal and is done only for convenience.
Options
-lossless boolean
Enables/Disables use of lossless mode. Default is 0.
-compression_level integer
For lossy, this is a quality/speed tradeoff. Higher values give better quality for a given size at
the cost of increased encoding time. For lossless, this is a size/speed tradeoff. Higher values give
smaller size at the cost of increased encoding time. More specifically, it controls the number of
extra algorithms and compression tools used, and varies the combination of these tools. This maps to
the method option in libwebp. The valid range is 0 to 6. Default is 4.
-qscale float
For lossy encoding, this controls image quality, 0 to 100. For lossless encoding, this controls the
effort and time spent at compressing more. The default value is 75. Note that for usage via
libavcodec, this option is called global_quality and must be multiplied by FF_QP2LAMBDA.
-preset type
Configuration preset. This does some automatic settings based on the general type of the image.
none
Do not use a preset.
default
Use the encoder default.
picture
Digital picture, like portrait, inner shot
photo
Outdoor photograph, with natural lighting
drawing
Hand or line drawing, with high-contrast details
icon
Small-sized colorful images
text
Text-like
libx264, libx264rgb
x264 H.264/MPEG-4 AVC encoder wrapper.
This encoder requires the presence of the libx264 headers and library during configuration. You need to
explicitly configure the build with "--enable-libx264".
libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial transform,
adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF), lossless mode, psy
optimizations for detail retention (adaptive quantization, psy-RD, psy-trellis).
Many libx264 encoder options are mapped to FFmpeg global codec options, while unique encoder options are
provided through private options. Additionally the x264opts and x264-params private options allows one to
pass a list of key=value tuples as accepted by the libx264 "x264_param_parse" function.
The x264 project website is at <http://www.videolan.org/developers/x264.html>.
The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats as input
instead of YUV.
Supported Pixel Formats
x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's configure time.
FFmpeg only supports one bit depth in one particular build. In other words, it is not possible to build
one FFmpeg with multiple versions of x264 with different bit depths.
Options
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
--fullhelp or consult the libx264 documentation.
b (bitrate)
Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while x264's bitrate is in
kilobits/s.
bf (bframes)
g (keyint)
qmin (qpmin)
Minimum quantizer scale.
qmax (qpmax)
Maximum quantizer scale.
qdiff (qpstep)
Maximum difference between quantizer scales.
qblur (qblur)
Quantizer curve blur
qcomp (qcomp)
Quantizer curve compression factor
refs (ref)
Number of reference frames each P-frame can use. The range is from 0-16.
sc_threshold (scenecut)
Sets the threshold for the scene change detection.
trellis (trellis)
Performs Trellis quantization to increase efficiency. Enabled by default.
nr (nr)
me_range (merange)
Maximum range of the motion search in pixels.
me_method (me)
Set motion estimation method. Possible values in the decreasing order of speed:
dia (dia)
epzs (dia)
Diamond search with radius 1 (fastest). epzs is an alias for dia.
hex (hex)
Hexagonal search with radius 2.
umh (umh)
Uneven multi-hexagon search.
esa (esa)
Exhaustive search.
tesa (tesa)
Hadamard exhaustive search (slowest).
forced-idr
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces
it to choose an IDR-frame.
subq (subme)
Sub-pixel motion estimation method.
b_strategy (b-adapt)
Adaptive B-frame placement decision algorithm. Use only on first-pass.
keyint_min (min-keyint)
Minimum GOP size.
coder
Set entropy encoder. Possible values:
ac Enable CABAC.
vlc Enable CAVLC and disable CABAC. It generates the same effect as x264's --no-cabac option.
cmp Set full pixel motion estimation comparison algorithm. Possible values:
chroma
Enable chroma in motion estimation.
sad Ignore chroma in motion estimation. It generates the same effect as x264's --no-chroma-me option.
threads (threads)
Number of encoding threads.
thread_type
Set multithreading technique. Possible values:
slice
Slice-based multithreading. It generates the same effect as x264's --sliced-threads option.
frame
Frame-based multithreading.
flags
Set encoding flags. It can be used to disable closed GOP and enable open GOP by setting it to
"-cgop". The result is similar to the behavior of x264's --open-gop option.
rc_init_occupancy (vbv-init)
preset (preset)
Set the encoding preset.
tune (tune)
Set tuning of the encoding params.
profile (profile)
Set profile restrictions.
fastfirstpass
Enable fast settings when encoding first pass, when set to 1. When set to 0, it has the same effect
of x264's --slow-firstpass option.
crf (crf)
Set the quality for constant quality mode.
crf_max (crf-max)
In CRF mode, prevents VBV from lowering quality beyond this point.
qp (qp)
Set constant quantization rate control method parameter.
aq-mode (aq-mode)
Set AQ method. Possible values:
none (0)
Disabled.
variance (1)
Variance AQ (complexity mask).
autovariance (2)
Auto-variance AQ (experimental).
aq-strength (aq-strength)
Set AQ strength, reduce blocking and blurring in flat and textured areas.
psy Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as x264's
--no-psy option.
psy-rd (psy-rd)
Set strength of psychovisual optimization, in psy-rd:psy-trellis format.
rc-lookahead (rc-lookahead)
Set number of frames to look ahead for frametype and ratecontrol.
weightb
Enable weighted prediction for B-frames when set to 1. When set to 0, it has the same effect as
x264's --no-weightb option.
weightp (weightp)
Set weighted prediction method for P-frames. Possible values:
none (0)
Disabled
simple (1)
Enable only weighted refs
smart (2)
Enable both weighted refs and duplicates
ssim (ssim)
Enable calculation and printing SSIM stats after the encoding.
intra-refresh (intra-refresh)
Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.
avcintra-class (class)
Configure the encoder to generate AVC-Intra. Valid values are 50,100 and 200
bluray-compat (bluray-compat)
Configure the encoder to be compatible with the bluray standard. It is a shorthand for setting
"bluray-compat=1 force-cfr=1".
b-bias (b-bias)
Set the influence on how often B-frames are used.
b-pyramid (b-pyramid)
Set method for keeping of some B-frames as references. Possible values:
none (none)
Disabled.
strict (strict)
Strictly hierarchical pyramid.
normal (normal)
Non-strict (not Blu-ray compatible).
mixed-refs
Enable the use of one reference per partition, as opposed to one reference per macroblock when set to
1. When set to 0, it has the same effect as x264's --no-mixed-refs option.
8x8dct
Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set to 0, it has
the same effect as x264's --no-8x8dct option.
fast-pskip
Enable early SKIP detection on P-frames when set to 1. When set to 0, it has the same effect as
x264's --no-fast-pskip option.
aud (aud)
Enable use of access unit delimiters when set to 1.
mbtree
Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same effect as x264's
--no-mbtree option.
deblock (deblock)
Set loop filter parameters, in alpha:beta form.
cplxblur (cplxblur)
Set fluctuations reduction in QP (before curve compression).
partitions (partitions)
Set partitions to consider as a comma-separated list of. Possible values in the list:
p8x8
8x8 P-frame partition.
p4x4
4x4 P-frame partition.
b8x8
4x4 B-frame partition.
i8x8
8x8 I-frame partition.
i4x4
4x4 I-frame partition. (Enabling p4x4 requires p8x8 to be enabled. Enabling i8x8 requires
adaptive spatial transform (8x8dct option) to be enabled.)
none (none)
Do not consider any partitions.
all (all)
Consider every partition.
direct-pred (direct)
Set direct MV prediction mode. Possible values:
none (none)
Disable MV prediction.
spatial (spatial)
Enable spatial predicting.
temporal (temporal)
Enable temporal predicting.
auto (auto)
Automatically decided.
slice-max-size (slice-max-size)
Set the limit of the size of each slice in bytes. If not specified but RTP payload size (ps) is
specified, that is used.
stats (stats)
Set the file name for multi-pass stats.
nal-hrd (nal-hrd)
Set signal HRD information (requires vbv-bufsize to be set). Possible values:
none (none)
Disable HRD information signaling.
vbr (vbr)
Variable bit rate.
cbr (cbr)
Constant bit rate (not allowed in MP4 container).
x264opts (N.A.)
Set any x264 option, see x264 --fullhelp for a list.
Argument is a list of key=value couples separated by ":". In filter and psy-rd options that use ":"
as a separator themselves, use "," instead. They accept it as well since long ago but this is kept
undocumented for some reason.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv
a53cc boolean
Import closed captions (which must be ATSC compatible format) into output. Only the mpeg2 and h264
decoders provide these. Default is 1 (on).
x264-params (N.A.)
Override the x264 configuration using a :-separated list of key=value parameters.
This option is functionally the same as the x264opts, but is duplicated for compatibility with the
Libav fork.
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT
Encoding ffpresets for common usages are provided so they can be used with the general presets system
(e.g. passing the pre option).
libx265
x265 H.265/HEVC encoder wrapper.
This encoder requires the presence of the libx265 headers and library during configuration. You need to
explicitly configure the build with --enable-libx265.
Options
preset
Set the x265 preset.
tune
Set the x265 tune parameter.
forced-idr
Normally, when forcing a I-frame type, the encoder can select any type of I-frame. This option forces
it to choose an IDR-frame.
x265-params
Set x265 options using a list of key=value couples separated by ":". See x265 --help for a list of
options.
For example to specify libx265 encoding options with -x265-params:
ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4
libxvid
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and library during configuration. You need
to explicitly configure the build with "--enable-libxvid --enable-gpl".
The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users can encode to this format without
this library.
Options
The following options are supported by the libxvid wrapper. Some of the following options are listed but
are not documented, and correspond to shared codec options. See the Codec Options chapter for their
documentation. The other shared options which are not listed have no effect for the libxvid encoder.
b
g
qmin
qmax
mpeg_quant
threads
bf
b_qfactor
b_qoffset
flags
Set specific encoding flags. Possible values:
mv4 Use four motion vector by macroblock.
aic Enable high quality AC prediction.
gray
Only encode grayscale.
gmc Enable the use of global motion compensation (GMC).
qpel
Enable quarter-pixel motion compensation.
cgop
Enable closed GOP.
global_header
Place global headers in extradata instead of every keyframe.
trellis
me_method
Set motion estimation method. Possible values in decreasing order of speed and increasing order of
quality:
zero
Use no motion estimation (default).
phods
x1
log Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks.
x1 and log are aliases for phods.
epzs
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.
full
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.
mbd Set macroblock decision algorithm. Possible values in the increasing order of quality:
simple
Use macroblock comparing function algorithm (default).
bits
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
rd Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel
refinement for 8x8 blocks, and rate distortion-based search using square pattern.
lumi_aq
Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).
variance_aq
Enable variance adaptive quantization when set to 1. Default is 0 (disabled).
When combined with lumi_aq, the resulting quality will not be better than any of the two specified
individually. In other words, the resulting quality will be the worse one of the two effects.
ssim
Set structural similarity (SSIM) displaying method. Possible values:
off Disable displaying of SSIM information.
avg Output average SSIM at the end of encoding to stdout. The format of showing the average SSIM is:
Average SSIM: %f
For users who are not familiar with C, %f means a float number, or a decimal (e.g. 0.939232).
frame
Output both per-frame SSIM data during encoding and average SSIM at the end of encoding to
stdout. The format of per-frame information is:
SSIM: avg: %1.3f min: %1.3f max: %1.3f
For users who are not familiar with C, %1.3f means a float number rounded to 3 digits after the
dot (e.g. 0.932).
ssim_acc
Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives the most
accurate result and 4 computes the fastest.
mpeg2
MPEG-2 video encoder.
Options
seq_disp_ext integer
Specifies if the encoder should write a sequence_display_extension to the output.
-1
auto
Decide automatically to write it or not (this is the default) by checking if the data to be
written is different from the default or unspecified values.
0
never
Never write it.
1
always
Always write it.
png
PNG image encoder.
Private options
dpi integer
Set physical density of pixels, in dots per inch, unset by default
dpm integer
Set physical density of pixels, in dots per meter, unset by default
ProRes
Apple ProRes encoder.
FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder. The used encoder can be chosen
with the "-vcodec" option.
Private Options for prores-ks
profile integer
Select the ProRes profile to encode
proxy
lt
standard
hq
4444
4444xq
quant_mat integer
Select quantization matrix.
auto
default
proxy
lt
standard
hq
If set to auto, the matrix matching the profile will be picked. If not set, the matrix providing the
highest quality, default, will be picked.
bits_per_mb integer
How many bits to allot for coding one macroblock. Different profiles use between 200 and 2400 bits
per macroblock, the maximum is 8000.
mbs_per_slice integer
Number of macroblocks in each slice (1-8); the default value (8) should be good in almost all
situations.
vendor string
Override the 4-byte vendor ID. A custom vendor ID like apl0 would claim the stream was produced by
the Apple encoder.
alpha_bits integer
Specify number of bits for alpha component. Possible values are 0, 8 and 16. Use 0 to disable alpha
plane coding.
Speed considerations
In the default mode of operation the encoder has to honor frame constraints (i.e. not produce frames with
size bigger than requested) while still making output picture as good as possible. A frame containing a
lot of small details is harder to compress and the encoder would spend more time searching for
appropriate quantizers for each slice.
Setting a higher bits_per_mb limit will improve the speed.
For the fastest encoding speed set the qscale parameter (4 is the recommended value) and do not set a
size constraint.
QSV encoders
The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)
The ratecontrol method is selected as follows:
• When global_quality is specified, a quality-based mode is used. Specifically this means either
- CQP - constant quantizer scale, when the qscale codec flag is also set (the -qscale ffmpeg
option).
- LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option is also set.
- ICQ -- intelligent constant quality otherwise.
• Otherwise, a bitrate-based mode is used. For all of those, you should specify at least the desired
average bitrate with the b option.
- LA - VBR with lookahead, when the look_ahead option is specified.
- VCM - video conferencing mode, when the vcm option is set.
- CBR - constant bitrate, when maxrate is specified and equal to the average bitrate.
- VBR - variable bitrate, when maxrate is specified, but is higher than the average bitrate.
- AVBR - average VBR mode, when maxrate is not specified. This mode is further configured by the
avbr_accuracy and avbr_convergence options.
Note that depending on your system, a different mode than the one you specified may be selected by the
encoder. Set the verbosity level to verbose or higher to see the actual settings used by the QSV runtime.
Additional libavcodec global options are mapped to MSDK options as follows:
• g/gop_size -> GopPicSize
• bf/max_b_frames+1 -> GopRefDist
• rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB
• slices -> NumSlice
• refs -> NumRefFrame
• b_strategy/b_frame_strategy -> BRefType
• cgop/CLOSED_GOP codec flag -> GopOptFlag
• For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset set the difference between QPP and
QPI, and QPP and QPB respectively.
• Setting the coder option to the value vlc will make the H.264 encoder use CAVLC instead of CABAC.
snow
Options
iterative_dia_size
dia size for the iterative motion estimation
VAAPI encoders
Wrappers for hardware encoders accessible via VAAPI.
These encoders only accept input in VAAPI hardware surfaces. If you have input in software frames, use
the hwupload filter to upload them to the GPU.
The following standard libavcodec options are used:
• g / gop_size
• bf / max_b_frames
• profile
• level
• b / bit_rate
• maxrate / rc_max_rate
• bufsize / rc_buffer_size
• rc_init_occupancy / rc_initial_buffer_occupancy
• compression_level
Speed / quality tradeoff: higher values are faster / worse quality.
• q / global_quality
Size / quality tradeoff: higher values are smaller / worse quality.
• qmin (only: qmax is not supported)
• i_qfactor / i_quant_factor
• i_qoffset / i_quant_offset
• b_qfactor / b_quant_factor
• b_qoffset / b_quant_offset
h264_vaapi
profile sets the value of profile_idc and the constraint_set*_flags. level sets the value of
level_idc.
low_power
Use low-power encoding mode.
coder
Set entropy encoder (default is cabac). Possible values:
ac
cabac
Use CABAC.
vlc
cavlc
Use CAVLC.
hevc_vaapi
profile and level set the values of general_profile_idc and general_level_idc respectively.
mjpeg_vaapi
Always encodes using the standard quantisation and huffman tables - global_quality scales the
standard quantisation table (range 1-100).
mpeg2_vaapi
profile and level set the value of profile_and_level_indication.
No rate control is supported.
vp8_vaapi
B-frames are not supported.
global_quality sets the q_idx used for non-key frames (range 0-127).
loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.
vp9_vaapi
global_quality sets the q_idx used for P-frames (range 0-255).
loop_filter_level
loop_filter_sharpness
Manually set the loop filter parameters.
B-frames are supported, but the output stream is always in encode order rather than display order.
If B-frames are enabled, it may be necessary to use the vp9_raw_reorder bitstream filter to modify
the output stream to display frames in the correct order.
Only normal frames are produced - the vp9_superframe bitstream filter may be required to produce a
stream usable with all decoders.
vc2
SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional broadcasting but
since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes
it suitable for other tasks which require low overhead and low compression (like screen recording).
Options
b Sets target video bitrate. Usually that's around 1:6 of the uncompressed video bitrate (e.g. for
1920x1080 50fps yuv422p10 that's around 400Mbps). Higher values (close to the uncompressed bitrate)
turn on lossless compression mode.
field_order
Enables field coding when set (e.g. to tt - top field first) for interlaced inputs. Should increase
compression with interlaced content as it splits the fields and encodes each separately.
wavelet_depth
Sets the total amount of wavelet transforms to apply, between 1 and 5 (default). Lower values reduce
compression and quality. Less capable decoders may not be able to handle values of wavelet_depth over
3.
wavelet_type
Sets the transform type. Currently only 5_3 (LeGall) and 9_7 (Deslauriers-Dubuc) are implemented,
with 9_7 being the one with better compression and thus is the default.
slice_width
slice_height
Sets the slice size for each slice. Larger values result in better compression. For compatibility
with other more limited decoders use slice_width of 32 and slice_height of 8.
tolerance
Sets the undershoot tolerance of the rate control system in percent. This is to prevent an expensive
search from being run.
qm Sets the quantization matrix preset to use by default or when wavelet_depth is set to 5
- default Uses the default quantization matrix from the specifications, extended with values for
the fifth level. This provides a good balance between keeping detail and omitting artifacts.
- flat Use a completely zeroed out quantization matrix. This increases PSNR but might reduce
perception. Use in bogus benchmarks.
- color Reduces detail but attempts to preserve color at extremely low bitrates.
SUBTITLES ENCODERS
dvdsub
This codec encodes the bitmap subtitle format that is used in DVDs. Typically they are stored in VOBSUB
file pairs (*.idx + *.sub), and they can also be used in Matroska files.
Options
even_rows_fix
When set to 1, enable a work-around that makes the number of pixel rows even in all subtitles. This
fixes a problem with some players that cut off the bottom row if the number is odd. The work-around
just adds a fully transparent row if needed. The overhead is low, typically one byte per subtitle on
average.
By default, this work-around is disabled.
BITSTREAM FILTERS
When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can
list all available ones using the configure option "--list-bsfs".
You can disable all the bitstream filters using the configure option "--disable-bsfs", and selectively
enable any bitstream filter using the option "--enable-bsf=BSF", or you can disable a particular
bitstream filter using the option "--disable-bsf=BSF".
The option "-bsfs" of the ff* tools will display the list of all the supported bitstream filters included
in your build.
The ff* tools have a -bsf option applied per stream, taking a comma-separated list of filters, whose
parameters follow the filter name after a '='.
ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT
Below is a description of the currently available bitstream filters, with their parameters, if any.
aac_adtstoasc
Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.
This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes the ADTS
header.
This filter is required for example when copying an AAC stream from a raw ADTS AAC or an MPEG-TS
container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats such as 3GP or M4A.
Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.
chomp
Remove zero padding at the end of a packet.
dca_core
Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.
dump_extra
Add extradata to the beginning of the filtered packets.
The additional argument specifies which packets should be filtered. It accepts the values:
a add extradata to all key packets, but only if local_header is set in the flags2 codec context field
k add extradata to all key packets
e 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
extract_extradata
Extract the in-band extradata.
Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC (VPS/)SPS/PPS) to
be transmitted either "in-band" (i.e. as a part of the bitstream containing the coded frames) or "out of
band" (e.g. on the container level). This latter form is called "extradata" in FFmpeg terminology.
This bitstream filter detects the in-band headers and makes them available as extradata.
remove
When this option is enabled, the long-term headers are removed from the bitstream after extraction.
h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex
B of the ITU-T H.264 specification).
This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer
"mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the
command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw H.264 (muxer "h264")
output formats.
hevc_mp4toannexb
Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as defined in the
Annex B of the ITU-T H.265 specification).
This is required by some streaming formats, typically the MPEG-2 transport stream format (muxer
"mpegts").
For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg, you can use the
command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts
Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw HEVC/H.265 (muxer
"h265" or "hevc") output formats.
imxdump
Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This filter only
applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7 and newer with the
appropriate -tag:v.
For example, to remux 30 MB/sec NTSC IMX to MOV:
ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov
mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be
extracted without loss, e.g. by
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg
Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for
decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the
MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table.
The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not
arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular
JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea
how to decompress the data. The exact table necessary is given in the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1
header ID and lacking a DHT segment) to produce fully qualified JPEG images.
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
exiftran -i -9 frame*.jpg
ffmpeg -i frame_%d.jpg -c:v copy rotated.avi
mjpegadump
Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.
mov2textsub
Extract a representable text file from MOV subtitles, stripping the metadata header from each subtitle
packet.
See also the text2movsub filter.
mp3decomp
Decompress non-standard compressed MP3 audio headers.
mpeg4_unpack_bframes
Unpack DivX-style packed B-frames.
DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken Video for
Windows subsystem. They use more space, can cause minor AV sync issues, require more CPU power to decode
(unless the player has some decoded picture queue to compensate the 2,0,2,0 frame per packet style) and
cause trouble if copied into a standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not
be able to decode them, since they are not valid MPEG-4.
For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames using ffmpeg,
you can use the command:
ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi
noise
Damages the contents of packets or simply drops them without damaging the container. Can be used for
fuzzing or testing error resilience/concealment.
Parameters:
amount
A numeral string, whose value is related to how often output bytes will be modified. Therefore,
values below or equal to 0 are forbidden, and the lower the more frequent bytes will be modified,
with 1 meaning every byte is modified.
dropamount
A numeral string, whose value is related to how often packets will be dropped. Therefore, values
below or equal to 0 are forbidden, and the lower the more frequent packets will be dropped, with 1
meaning every packet is dropped.
The following example applies the modification to every byte but does not drop any packets.
ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv
null
This bitstream filter passes the packets through unchanged.
remove_extra
Remove extradata from packets.
It accepts the following parameter:
freq
Set which frame types to remove extradata from.
k Remove extradata from non-keyframes only.
keyframe
Remove extradata from keyframes only.
e, all
Remove extradata from all frames.
text2movsub
Convert text subtitles to MOV subtitles (as used by the "mov_text" codec) with metadata headers.
See also the mov2textsub filter.
vp9_superframe
Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of split/segmented VP9
streams where the alt-ref frame was split from its visible counterpart.
vp9_superframe_split
Split VP9 superframes into single frames.
vp9_raw_reorder
Given a VP9 stream with correct timestamps but possibly out of order, insert additional show-existing-
frame packets to correct the ordering.
FORMAT OPTIONS
The libavformat library provides some generic global options, which can be set on all the muxers and
demuxers. In addition each muxer or demuxer may support so-called private options, which are specific for
that component.
Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in
the "AVFormatContext" options or using the libavutil/opt.h API for programmatic use.
The list of supported options follows:
avioflags flags (input/output)
Possible values:
direct
Reduce buffering.
probesize integer (input)
Set probing size in bytes, i.e. the size of the data to analyze to get stream information. A higher
value will enable detecting more information in case it is dispersed into the stream, but will
increase latency. Must be an integer not lesser than 32. It is 5000000 by default.
packetsize integer (output)
Set packet size.
fflags flags (input/output)
Set format flags.
Possible values:
ignidx
Ignore index.
fastseek
Enable fast, but inaccurate seeks for some formats.
genpts
Generate PTS.
nofillin
Do not fill in missing values that can be exactly calculated.
noparse
Disable AVParsers, this needs "+nofillin" too.
igndts
Ignore DTS.
discardcorrupt
Discard corrupted frames.
sortdts
Try to interleave output packets by DTS.
keepside
Do not merge side data.
latm
Enable RTP MP4A-LATM payload.
nobuffer
Reduce the latency introduced by optional buffering
bitexact
Only write platform-, build- and time-independent data. This ensures that file and data
checksums are reproducible and match between platforms. Its primary use is for regression
testing.
shortest
Stop muxing at the end of the shortest stream. It may be needed to increase max_interleave_delta
to avoid flushing the longer streams before EOF.
seek2any integer (input)
Allow seeking to non-keyframes on demuxer level when supported if set to 1. Default is 0.
analyzeduration integer (input)
Specify how many microseconds are analyzed to probe the input. A higher value will enable detecting
more accurate information, but will increase latency. It defaults to 5,000,000 microseconds = 5
seconds.
cryptokey hexadecimal string (input)
Set decryption key.
indexmem integer (input)
Set max memory used for timestamp index (per stream).
rtbufsize integer (input)
Set max memory used for buffering real-time frames.
fdebug flags (input/output)
Print specific debug info.
Possible values:
ts
max_delay integer (input/output)
Set maximum muxing or demuxing delay in microseconds.
fpsprobesize integer (input)
Set number of frames used to probe fps.
audio_preload integer (output)
Set microseconds by which audio packets should be interleaved earlier.
chunk_duration integer (output)
Set microseconds for each chunk.
chunk_size integer (output)
Set size in bytes for each chunk.
err_detect, f_err_detect flags (input)
Set error detection flags. "f_err_detect" is deprecated and should be used only via the ffmpeg tool.
Possible values:
crccheck
Verify embedded CRCs.
bitstream
Detect bitstream specification deviations.
buffer
Detect improper bitstream length.
explode
Abort decoding on minor error detection.
careful
Consider things that violate the spec and have not been seen in the wild as errors.
compliant
Consider all spec non compliancies as errors.
aggressive
Consider things that a sane encoder should not do as an error.
max_interleave_delta integer (output)
Set maximum buffering duration for interleaving. The duration is expressed in microseconds, and
defaults to 1000000 (1 second).
To ensure all the streams are interleaved correctly, libavformat will wait until it has at least one
packet for each stream before actually writing any packets to the output file. When some streams are
"sparse" (i.e. there are large gaps between successive packets), this can result in excessive
buffering.
This field specifies the maximum difference between the timestamps of the first and the last packet
in the muxing queue, above which libavformat will output a packet regardless of whether it has queued
a packet for all the streams.
If set to 0, libavformat will continue buffering packets until it has a packet for each stream,
regardless of the maximum timestamp difference between the buffered packets.
use_wallclock_as_timestamps integer (input)
Use wallclock as timestamps if set to 1. Default is 0.
avoid_negative_ts integer (output)
Possible values:
make_non_negative
Shift timestamps to make them non-negative. Also note that this affects only leading negative
timestamps, and not non-monotonic negative timestamps.
make_zero
Shift timestamps so that the first timestamp is 0.
auto (default)
Enables shifting when required by the target format.
disabled
Disables shifting of timestamp.
When shifting is enabled, all output timestamps are shifted by the same amount. Audio, video, and
subtitles desynching and relative timestamp differences are preserved compared to how they would have
been without shifting.
skip_initial_bytes integer (input)
Set number of bytes to skip before reading header and frames if set to 1. Default is 0.
correct_ts_overflow integer (input)
Correct single timestamp overflows if set to 1. Default is 1.
flush_packets integer (output)
Flush the underlying I/O stream after each packet. Default is -1 (auto), which means that the
underlying protocol will decide, 1 enables it, and has the effect of reducing the latency, 0 disables
it and may increase IO throughput in some cases.
output_ts_offset offset (output)
Set the output time offset.
offset must be a time duration specification, see the Time duration section in the ffmpeg-utils(1)
manual.
The offset is added by the muxer to the output timestamps.
Specifying a positive offset means that the corresponding streams are delayed bt the time duration
specified in offset. Default value is 0 (meaning that no offset is applied).
format_whitelist list (input)
"," separated list of allowed demuxers. By default all are allowed.
dump_separator string (input)
Separator used to separate the fields printed on the command line about the Stream parameters. For
example to separate the fields with newlines and indention:
ffprobe -dump_separator "
" -i ~/videos/matrixbench_mpeg2.mpg
max_streams integer (input)
Specifies the maximum number of streams. This can be used to reject files that would require too many
resources due to a large number of streams.
Format stream specifiers
Format stream specifiers allow selection of one or more streams that match specific properties.
Possible forms of stream specifiers are:
stream_index
Matches the stream with this index.
stream_type[:stream_index]
stream_type is one of following: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data, and
't' for attachments. If stream_index is given, then it matches the stream number stream_index of this
type. Otherwise, it matches all streams of this type.
p:program_id[:stream_index]
If stream_index is given, then it matches the stream with number stream_index in the program with the
id program_id. Otherwise, it matches all streams in the program.
#stream_id
Matches the stream by a format-specific ID.
The exact semantics of stream specifiers is defined by the "avformat_match_stream_specifier()" function
declared in the libavformat/avformat.h header.
DEMUXERS
Demuxers are configured elements in FFmpeg that can read the multimedia streams from a particular type of
file.
When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all
available ones using the configure option "--list-demuxers".
You can disable all the demuxers using the configure option "--disable-demuxers", and selectively enable
a single demuxer with the option "--enable-demuxer=DEMUXER", or disable it with the option
"--disable-demuxer=DEMUXER".
The option "-demuxers" of the ff* tools will display the list of enabled demuxers. Use "-formats" to view
a combined list of enabled demuxers and muxers.
The description of some of the currently available demuxers follows.
aa
Audible Format 2, 3, and 4 demuxer.
This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.
applehttp
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant
index number. By setting the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the caller
can decide which variant streams to actually receive. The total bitrate of the variant that the stream
belongs to is available in a metadata key named "variant_bitrate".
apng
Animated Portable Network Graphics demuxer.
This demuxer is used to demux APNG files. All headers, but the PNG signature, up to (but not including)
the first fcTL chunk are transmitted as extradata. Frames are then split as being all the chunks between
two fcTL ones, or between the last fcTL and IEND chunks.
-ignore_loop bool
Ignore the loop variable in the file if set.
-max_fps int
Maximum framerate in frames per second (0 for no limit).
-default_fps int
Default framerate in frames per second when none is specified in the file (0 meaning as fast as
possible).
asf
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
-no_resync_search bool
Do not try to resynchronize by looking for a certain optional start code.
concat
Virtual concatenation script demuxer.
This demuxer reads a list of files and other directives from a text file and demuxes them one after the
other, as if all their packets had been muxed together.
The timestamps in the files are adjusted so that the first file starts at 0 and each next file starts
where the previous one finishes. Note that it is done globally and may cause gaps if all streams do not
have exactly the same length.
All files must have the same streams (same codecs, same time base, etc.).
The duration of each file is used to adjust the timestamps of the next file: if the duration is incorrect
(because it was computed using the bit-rate or because the file is truncated, for example), it can cause
artifacts. The "duration" directive can be used to override the duration stored in each file.
Syntax
The script is a text file in extended-ASCII, with one directive per line. Empty lines, leading spaces
and lines starting with '#' are ignored. The following directive is recognized:
"file path"
Path to a file to read; special characters and spaces must be escaped with backslash or single
quotes.
All subsequent file-related directives apply to that file.
"ffconcat version 1.0"
Identify the script type and version. It also sets the safe option to 1 if it was -1.
To make FFmpeg recognize the format automatically, this directive must appear exactly as is (no extra
space or byte-order-mark) on the very first line of the script.
"duration dur"
Duration of the file. This information can be specified from the file; specifying it here may be more
efficient or help if the information from the file is not available or accurate.
If the duration is set for all files, then it is possible to seek in the whole concatenated video.
"inpoint timestamp"
In point of the file. When the demuxer opens the file it instantly seeks to the specified timestamp.
Seeking is done so that all streams can be presented successfully at In point.
This directive works best with intra frame codecs, because for non-intra frame ones you will usually
get extra packets before the actual In point and the decoded content will most likely contain frames
before In point too.
For each file, packets before the file In point will have timestamps less than the calculated start
timestamp of the file (negative in case of the first file), and the duration of the files (if not
specified by the "duration" directive) will be reduced based on their specified In point.
Because of potential packets before the specified In point, packet timestamps may overlap between two
concatenated files.
"outpoint timestamp"
Out point of the file. When the demuxer reaches the specified decoding timestamp in any of the
streams, it handles it as an end of file condition and skips the current and all the remaining
packets from all streams.
Out point is exclusive, which means that the demuxer will not output packets with a decoding
timestamp greater or equal to Out point.
This directive works best with intra frame codecs and formats where all streams are tightly
interleaved. For non-intra frame codecs you will usually get additional packets with presentation
timestamp after Out point therefore the decoded content will most likely contain frames after Out
point too. If your streams are not tightly interleaved you may not get all the packets from all
streams before Out point and you may only will be able to decode the earliest stream until Out point.
The duration of the files (if not specified by the "duration" directive) will be reduced based on
their specified Out point.
"file_packet_metadata key=value"
Metadata of the packets of the file. The specified metadata will be set for each file packet. You can
specify this directive multiple times to add multiple metadata entries.
"stream"
Introduce a stream in the virtual file. All subsequent stream-related directives apply to the last
introduced stream. Some streams properties must be set in order to allow identifying the matching
streams in the subfiles. If no streams are defined in the script, the streams from the first file
are copied.
"exact_stream_id id"
Set the id of the stream. If this directive is given, the string with the corresponding id in the
subfiles will be used. This is especially useful for MPEG-PS (VOB) files, where the order of the
streams is not reliable.
Options
This demuxer accepts the following option:
safe
If set to 1, reject unsafe file paths. A file path is considered safe if it does not contain a
protocol specification and is relative and all components only contain characters from the portable
character set (letters, digits, period, underscore and hyphen) and have no period at the beginning of
a component.
If set to 0, any file name is accepted.
The default is 1.
-1 is equivalent to 1 if the format was automatically probed and 0 otherwise.
auto_convert
If set to 1, try to perform automatic conversions on packet data to make the streams concatenable.
The default is 1.
Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to H.264 streams in
MP4 format. This is necessary in particular if there are resolution changes.
segment_time_metadata
If set to 1, every packet will contain the lavf.concat.start_time and the lavf.concat.duration packet
metadata values which are the start_time and the duration of the respective file segments in the
concatenated output expressed in microseconds. The duration metadata is only set if it is known based
on the concat file. The default is 0.
Examples
• Use absolute filenames and include some comments:
# my first filename
file /mnt/share/file-1.wav
# my second filename including whitespace
file '/mnt/share/file 2.wav'
# my third filename including whitespace plus single quote
file '/mnt/share/file 3'\''.wav'
• Allow for input format auto-probing, use safe filenames and set the duration of the first file:
ffconcat version 1.0
file file-1.wav
duration 20.0
file subdir/file-2.wav
flv, live_flv
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams. In case of live network streams, if you
force format, you may use live_flv option instead of flv to survive timestamp discontinuities.
ffmpeg -f flv -i myfile.flv ...
ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....
-flv_metadata bool
Allocate the streams according to the onMetaData array content.
gif
Animated GIF demuxer.
It accepts the following options:
min_delay
Set the minimum valid delay between frames in hundredths of seconds. Range is 0 to 6000. Default
value is 2.
max_gif_delay
Set the maximum valid delay between frames in hundredth of seconds. Range is 0 to 65535. Default
value is 65535 (nearly eleven minutes), the maximum value allowed by the specification.
default_delay
Set the default delay between frames in hundredths of seconds. Range is 0 to 6000. Default value is
10.
ignore_loop
GIF files can contain information to loop a certain number of times (or infinitely). If ignore_loop
is set to 1, then the loop setting from the input will be ignored and looping will not occur. If set
to 0, then looping will occur and will cycle the number of times according to the GIF. Default value
is 1.
For example, with the overlay filter, place an infinitely looping GIF over another video:
ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv
Note that in the above example the shortest option for overlay filter is used to end the output video at
the length of the shortest input file, which in this case is input.mp4 as the GIF in this example loops
infinitely.
hls
HLS demuxer
It accepts the following options:
live_start_index
segment index to start live streams at (negative values are from the end).
allowed_extensions
',' separated list of file extensions that hls is allowed to access.
max_reload
Maximum number of times a insufficient list is attempted to be reloaded. Default value is 1000.
image2
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern. The syntax and meaning of the
pattern is specified by the option pattern_type.
The pattern may contain a suffix which is used to automatically determine the format of the images
contained in the files.
The size, the pixel format, and the format of each image must be the same for all the files in the
sequence.
This demuxer accepts the following options:
framerate
Set the frame rate for the video stream. It defaults to 25.
loop
If set to 1, loop over the input. Default value is 0.
pattern_type
Select the pattern type used to interpret the provided filename.
pattern_type accepts one of the following values.
none
Disable pattern matching, therefore the video will only contain the specified image. You should
use this option if you do not want to create sequences from multiple images and your filenames
may contain special pattern characters.
sequence
Select a sequence pattern type, used to specify a sequence of files indexed by sequential
numbers.
A sequence pattern may contain the string "%d" or "%0Nd", which specifies the position of the
characters representing a sequential number in each filename matched by the pattern. If the form
"%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the
total number of 0-padded digits representing the number. The literal character '%' can be
specified in the pattern with the string "%%".
If the sequence pattern contains "%d" or "%0Nd", the first filename of the file list specified by
the pattern must contain a number inclusively contained between start_number and
start_number+start_number_range-1, and all the following numbers must be sequential.
For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form
img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a
sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, ..., i%m%g-10.jpg, etc.
Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a
single image file img.jpeg you can employ the command:
ffmpeg -i img.jpeg img.png
glob
Select a glob wildcard pattern type.
The pattern is interpreted like a "glob()" pattern. This is only selectable if libavformat was
compiled with globbing support.
glob_sequence (deprecated, will be removed)
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and the provided pattern
contains at least one glob meta character among "%*?[]{}" that is preceded by an unescaped "%",
the pattern is interpreted like a "glob()" pattern, otherwise it is interpreted like a sequence
pattern.
All glob special characters "%*?[]{}" must be prefixed with "%". To escape a literal "%" you
shall use "%%".
For example the pattern "foo-%*.jpeg" will match all the filenames prefixed by "foo-" and
terminating with ".jpeg", and "foo-%?%?%?.jpeg" will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating with ".jpeg".
This pattern type is deprecated in favor of glob and sequence.
Default value is glob_sequence.
pixel_format
Set the pixel format of the images to read. If not specified the pixel format is guessed from the
first image file in the sequence.
start_number
Set the index of the file matched by the image file pattern to start to read from. Default value is
0.
start_number_range
Set the index interval range to check when looking for the first image file in the sequence, starting
from start_number. Default value is 5.
ts_from_file
If set to 1, will set frame timestamp to modification time of image file. Note that monotonity of
timestamps is not provided: images go in the same order as without this option. Default value is 0.
If set to 2, will set frame timestamp to the modification time of the image file in nanosecond
precision.
video_size
Set the video size of the images to read. If not specified the video size is guessed from the first
image file in the sequence.
Examples
• Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ...,
assuming an input frame rate of 10 frames per second:
ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv
• As above, but start by reading from a file with index 100 in the sequence:
ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv
• Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png"
suffix:
ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv
libgme
The Game Music Emu library is a collection of video game music file emulators.
See <http://code.google.com/p/game-music-emu/> for more information.
Some files have multiple tracks. The demuxer will pick the first track by default. The track_index option
can be used to select a different track. Track indexes start at 0. The demuxer exports the number of
tracks as tracks meta data entry.
For very large files, the max_size option may have to be adjusted.
libopenmpt
libopenmpt based module demuxer
See <https://lib.openmpt.org/libopenmpt/> for more information.
Some files have multiple subsongs (tracks) this can be set with the subsong option.
It accepts the following options:
subsong
Set the subsong index. This can be either 'all', 'auto', or the index of the subsong. Subsong
indexes start at 0. The default is 'auto'.
The default value is to let libopenmpt choose.
layout
Set the channel layout. Valid values are 1, 2, and 4 channel layouts. The default value is STEREO.
sample_rate
Set the sample rate for libopenmpt to output. Range is from 1000 to INT_MAX. The value default is
48000.
mov/mp4/3gp/QuickTime
QuickTime / MP4 demuxer.
This demuxer accepts the following options:
enable_drefs
Enable loading of external tracks, disabled by default. Enabling this can theoretically leak
information in some use cases.
use_absolute_path
Allows loading of external tracks via absolute paths, disabled by default. Enabling this poses a
security risk. It should only be enabled if the source is known to be non malicious.
mpegts
MPEG-2 transport stream demuxer.
This demuxer accepts the following options:
resync_size
Set size limit for looking up a new synchronization. Default value is 65536.
fix_teletext_pts
Override teletext packet PTS and DTS values with the timestamps calculated from the PCR of the first
program which the teletext stream is part of and is not discarded. Default value is 1, set this
option to 0 if you want your teletext packet PTS and DTS values untouched.
ts_packetsize
Output option carrying the raw packet size in bytes. Show the detected raw packet size, cannot be
set by the user.
scan_all_pmts
Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means automatic
setting, 1 means enabled, 0 means disabled). Default value is -1.
mpjpeg
MJPEG encapsulated in multi-part MIME demuxer.
This demuxer allows reading of MJPEG, where each frame is represented as a part of
multipart/x-mixed-replace stream.
strict_mime_boundary
Default implementation applies a relaxed standard to multi-part MIME boundary detection, to prevent
regression with numerous existing endpoints not generating a proper MIME MJPEG stream. Turning this
option on by setting it to 1 will result in a stricter check of the boundary value.
rawvideo
Raw video demuxer.
This demuxer allows one to read raw video data. Since there is no header specifying the assumed video
parameters, the user must specify them in order to be able to decode the data correctly.
This demuxer accepts the following options:
framerate
Set input video frame rate. Default value is 25.
pixel_format
Set the input video pixel format. Default value is "yuv420p".
video_size
Set the input video size. This value must be specified explicitly.
For example to read a rawvideo file input.raw with ffplay, assuming a pixel format of "rgb24", a video
size of "320x240", and a frame rate of 10 images per second, use the command:
ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw
sbg
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen <http://uazu.net/sbagen/> to generate binaural
beats sessions. A SBG script looks like that:
-SE
a: 300-2.5/3 440+4.5/0
b: 300-2.5/0 440+4.5/3
off: -
NOW == a
+0:07:00 == b
+0:14:00 == a
+0:21:00 == b
+0:30:00 off
A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps
(including the script start time) or only relative ones, then its layout is fixed, and the conversion is
straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference
for relative timestamps will be taken from the current time of day at the time the script is read, and
the script layout will be frozen according to that reference. That means that if the script is directly
played, the actual times will match the absolute timestamps up to the sound controller's clock accuracy,
but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.
tedcaptions
JSON captions used for <http://www.ted.com/>.
TED does not provide links to the captions, but they can be guessed from the page. The file
tools/bookmarklets.html from the FFmpeg source tree contains a bookmarklet to expose them.
This demuxer accepts the following option:
start_time
Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is used to sync
the captions with the downloadable videos, because they include a 15s intro.
Example: convert the captions to a format most players understand:
ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt
MUXERS
Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of
file.
When you configure your FFmpeg build, all the supported muxers are enabled by default. You can list all
available muxers using the configure option "--list-muxers".
You can disable all the muxers with the configure option "--disable-muxers" and selectively enable /
disable single muxers with the options "--enable-muxer=MUXER" / "--disable-muxer=MUXER".
The option "-muxers" of the ff* tools will display the list of enabled muxers. Use "-formats" to view a
combined list of enabled demuxers and muxers.
A description of some of the currently available muxers follows.
aiff
Audio Interchange File Format muxer.
Options
It accepts the following options:
write_id3v2
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
id3v2_version
Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are
supported. The default is version 4.
asf
Advanced Systems Format muxer.
Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.
Options
It accepts the following options:
packet_size
Set the muxer packet size. By tuning this setting you may reduce data fragmentation or muxer overhead
depending on your source. Default value is 3200, minimum is 100, maximum is 64k.
avi
Audio Video Interleaved muxer.
Options
It accepts the following options:
reserve_index_space
Reserve the specified amount of bytes for the OpenDML master index of each stream within the file
header. By default additional master indexes are embedded within the data packets if there is no
space left in the first master index and are linked together as a chain of indexes. This index
structure can cause problems for some use cases, e.g. third-party software strictly relying on the
OpenDML index specification or when file seeking is slow. Reserving enough index space in the file
header avoids these problems.
The required index space depends on the output file size and should be about 16 bytes per gigabyte.
When this option is omitted or set to zero the necessary index space is guessed.
write_channel_mask
Write the channel layout mask into the audio stream header.
This option is enabled by default. Disabling the channel mask can be useful in specific scenarios,
e.g. when merging multiple audio streams into one for compatibility with software that only supports
a single audio stream in AVI (see the "amerge" section in the ffmpeg-filters manual).
chromaprint
Chromaprint fingerprinter
This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for the provided
audio data. It takes a single signed native-endian 16-bit raw audio stream.
Options
silence_threshold
Threshold for detecting silence, ranges from 0 to 32767. -1 for default (required for use with the
AcoustID service).
algorithm
Algorithm index to fingerprint with.
fp_format
Format to output the fingerprint as. Accepts the following options:
raw Binary raw fingerprint
compressed
Binary compressed fingerprint
base64
Base64 compressed fingerprint
crc
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio
frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal
number 0-padded to 8 digits containing the CRC for all the decoded input frames.
See also the framecrc muxer.
Examples
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 -
flv
Adobe Flash Video Format muxer.
This muxer accepts the following options:
flvflags flags
Possible values:
aac_seq_header_detect
Place AAC sequence header based on audio stream data.
no_sequence_end
Disable sequence end tag.
no_metadata
Disable metadata tag.
no_duration_filesize
Disable duration and filesize in metadata when they are equal to zero at the end of stream. (Be
used to non-seekable living stream).
add_keyframe_index
Used to facilitate seeking; particularly for HTTP pseudo streaming.
dash
Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments and manifest files according to
the MPEG-DASH standard ISO/IEC 23009-1:2014.
For more information see:
• ISO DASH Specification:
<http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>
• WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>
It creates a MPD manifest file and segment files for each stream.
The segment filename might contain pre-defined identifiers used with SegmentTemplate as defined in
section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$", "$Number$",
"$Bandwidth$" and "$Time$".
ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
-b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
-profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
-b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
-window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
-f dash /path/to/out.mpd
-min_seg_duration microseconds
Set the segment length in microseconds.
-window_size size
Set the maximum number of segments kept in the manifest.
-extra_window_size size
Set the maximum number of segments kept outside of the manifest before removing from disk.
-remove_at_exit remove
Enable (1) or disable (0) removal of all segments when finished.
-use_template template
Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
-use_timeline timeline
Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
-single_file single_file
Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
-single_file_name file_name
DASH-templated name to be used for baseURL. Implies single_file set to "1".
-init_seg_name init_name
DASH-templated name to used for the initialization segment. Default is
"init-stream$RepresentationID$.m4s"
-media_seg_name segment_name
DASH-templated name to used for the media segments. Default is
"chunk-stream$RepresentationID$-$Number%05d$.m4s"
-utc_timing_url utc_url
URL of the page that will return the UTC timestamp in ISO format. Example:
"https://time.akamai.com/?iso"
-adaptation_sets adaptation_sets
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being
the IDs of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream
identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet for each stream.
framecrc
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>, 0x<CRC>
CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.
Examples
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.
framehash
Per-packet hash testing format.
This muxer computes and prints a cryptographic hash for each audio and video packet. This can be used for
packet-by-packet equality checks without having to individually do a binary comparison on each.
By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before
computing the hash, but the output of explicit conversions to other codecs can also be used. It uses the
SHA-256 cryptographic hash function by default, but supports several other algorithms.
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>, <hash>
hash is a hexadecimal number representing the computed hash for the packet.
hash algorithm
Use the cryptographic hash function specified by the string algorithm. Supported values include
"MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
(default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".
Examples
To compute the SHA-256 hash of the audio and video frames in INPUT, converted to raw audio and video
packets, and store it in the file out.sha256:
ffmpeg -i INPUT -f framehash out.sha256
To print the information to stdout, using the MD5 hash function, use the command:
ffmpeg -i INPUT -f framehash -hash md5 -
See also the hash muxer.
framemd5
Per-packet MD5 testing format.
This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5 hash function.
Examples
To compute the MD5 hash 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 framehash and md5 muxers.
gif
Animated GIF muxer.
It accepts the following options:
loop
Set the number of times to loop the output. Use "-1" for no loop, 0 for looping indefinitely
(default).
final_delay
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 into 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 large time base: the delay between two frames can therefore not be
smaller than one centi second.
hash
Hash testing format.
This muxer computes and prints a cryptographic hash of all the input audio and video frames. This can be
used for equality checks without having to do a complete binary comparison.
By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before
computing the hash, but the output of explicit conversions to other codecs can also be used. Timestamps
are ignored. It uses the SHA-256 cryptographic hash function by default, but supports several other
algorithms.
The output of the muxer consists of a single line of the form: algo=hash, where algo is a short string
representing the hash function used, and hash is a hexadecimal number representing the computed hash.
hash algorithm
Use the cryptographic hash function specified by the string algorithm. Supported values include
"MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
(default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32" and "adler32".
Examples
To compute the SHA-256 hash of the input converted to raw audio and video, and store it in the file
out.sha256:
ffmpeg -i INPUT -f hash out.sha256
To print an MD5 hash to stdout use the command:
ffmpeg -i INPUT -f hash -hash md5 -
See also the framehash muxer.
hls
Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming (HLS)
specification.
It creates a playlist file, and one or more segment files. The output filename specifies the playlist
filename.
By default, the muxer creates a file for each segment produced. These files have the same name as the
playlist, followed by a sequential number and a .ts extension.
For example, to convert an input file with ffmpeg:
ffmpeg -i in.nut out.m3u8
This example will produce the playlist, out.m3u8, and segment files: out0.ts, out1.ts, out2.ts, etc.
See also the segment muxer, which provides a more generic and flexible implementation of a segmenter, and
can be used to perform HLS segmentation.
Options
This muxer supports the following options:
hls_init_time seconds
Set the initial target segment length in seconds. Default value is 0. Segment will be cut on the
next key frame after this time has passed on the first m3u8 list. After the initial playlist is
filled ffmpeg will cut segments at duration equal to "hls_time"
hls_time seconds
Set the target segment length in seconds. Default value is 2. Segment will be cut on the next key
frame after this time has passed.
hls_list_size size
Set the maximum number of playlist entries. If set to 0 the list file will contain all the segments.
Default value is 5.
hls_ts_options options_list
Set output format options using a :-separated list of key=value parameters. Values containing ":"
special characters must be escaped.
hls_wrap wrap
This is a deprecated option, you can use "hls_list_size" and "hls_flags delete_segments" instead it
This option is useful to avoid to fill the disk with many segment files, and limits the maximum
number of segment files written to disk to wrap.
hls_start_number_source
Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") according to the specified source.
Unless "hls_flags single_file" is set, it also specifies source of starting sequence numbers of
segment and subtitle filenames. In any case, if "hls_flags append_list" is set and read playlist
sequence number is greater than the specified start sequence number, then that value will be used as
start value.
It accepts the following values:
generic (default)
Set the starting sequence numbers according to start_number option value.
epoch
The start number will be the seconds since epoch (1970-01-01 00:00:00)
datetime
The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g. 20161231235759.
start_number number
Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") from the specified number when
hls_start_number_source value is generic. (This is the default case.) Unless "hls_flags single_file"
is set, it also specifies starting sequence numbers of segment and subtitle filenames. Default value
is 0.
hls_allow_cache allowcache
Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.
hls_base_url baseurl
Append baseurl to every entry in the playlist. Useful to generate playlists with absolute paths.
Note that the playlist sequence number must be unique for each segment and it is not to be confused
with the segment filename sequence number which can be cyclic, for example if the wrap option is
specified.
hls_segment_filename filename
Set the segment filename. Unless "hls_flags single_file" is set, filename is used as a string format
with the segment number:
ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment files: file000.ts, file001.ts,
file002.ts, etc.
filename may contain full path or relative path specification, but only the file name part without
any path info will be contained in the m3u8 segment list. Should a relative path be specified, the
path of the created segment files will be relative to the current working directory. When
use_localtime_mkdir is set, the whole expanded value of filename will be written into the m3u8
segment list.
use_localtime
Use strftime() on filename to expand the segment filename with localtime. The segment number is also
available in this mode, but to use it, you need to specify second_level_segment_index hls_flag and
%%d will be the specifier.
ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment files: file-20160215-1455569023.ts,
file-20160215-1455569024.ts, etc. Note: On some systems/environments, the %s specifier is not
available. See
"strftime()" documentation.
ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8
This example will produce the playlist, out.m3u8, and segment files: file-20160215-0001.ts,
file-20160215-0002.ts, etc.
use_localtime_mkdir
Used together with -use_localtime, it will create all subdirectories which is expanded in filename.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory 201560215 (if it does not exist), and then produce the playlist,
out.m3u8, and segment files: 20160215/file-20160215-1455569023.ts,
20160215/file-20160215-1455569024.ts, etc.
ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8
This example will create a directory hierarchy 2016/02/15 (if any of them do not exist), and then
produce the playlist, out.m3u8, and segment files: 2016/02/15/file-20160215-1455569023.ts,
2016/02/15/file-20160215-1455569024.ts, etc.
hls_key_info_file key_info_file
Use the information in key_info_file for segment encryption. The first line of key_info_file
specifies the key URI written to the playlist. The key URL is used to access the encryption key
during playback. The second line specifies the path to the key file used to obtain the key during the
encryption process. The key file is read as a single packed array of 16 octets in binary format. The
optional third line specifies the initialization vector (IV) as a hexadecimal string to be used
instead of the segment sequence number (default) for encryption. Changes to key_info_file will result
in segment encryption with the new key/IV and an entry in the playlist for the new key URI/IV if
"hls_flags periodic_rekey" is enabled.
Key info file format:
<key URI>
<key file path>
<IV> (optional)
Example key URIs:
http://server/file.key
/path/to/file.key
file.key
Example key file paths:
file.key
/path/to/file.key
Example IV:
0123456789ABCDEF0123456789ABCDEF
Key info file example:
http://server/file.key
/path/to/file.key
0123456789ABCDEF0123456789ABCDEF
Example shell script:
#!/bin/sh
BASE_URL=${1:-'.'}
openssl rand 16 > file.key
echo $BASE_URL/file.key > file.keyinfo
echo file.key >> file.keyinfo
echo $(openssl rand -hex 16) >> file.keyinfo
ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
-hls_key_info_file file.keyinfo out.m3u8
-hls_enc enc
Enable (1) or disable (0) the AES128 encryption. When enabled every segment generated is encrypted
and the encryption key is saved as playlist name.key.
-hls_enc_key key
Hex-coded 16byte key to encrypt the segments, by default it is randomly generated.
-hls_enc_key_url keyurl
If set, keyurl is prepended instead of baseurl to the key filename in the playlist.
-hls_enc_iv iv
Hex-coded 16byte initialization vector for every segment instead of the autogenerated ones.
hls_segment_type flags
Possible values:
mpegts
If this flag is set, the hls segment files will format to mpegts. the mpegts files is used in
all hls versions.
fmp4
If this flag is set, the hls segment files will format to fragment mp4 looks like dash. the fmp4
files is used in hls after version 7.
hls_fmp4_init_filename filename
set filename to the fragment files header file, default filename is init.mp4.
hls_flags flags
Possible values:
single_file
If this flag is set, the muxer will store all segments in a single MPEG-TS file, and will use
byte ranges in the playlist. HLS playlists generated with this way will have the version number
4. For example:
ffmpeg -i in.nut -hls_flags single_file out.m3u8
Will produce the playlist, out.m3u8, and a single segment file, out.ts.
delete_segments
Segment files removed from the playlist are deleted after a period of time equal to the duration
of the segment plus the duration of the playlist.
append_list
Append new segments into the end of old segment list, and remove the "#EXT-X-ENDLIST" from the
old segment list.
round_durations
Round the duration info in the playlist file segment info to integer values, instead of using
floating point.
discont_start
Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the first segment's information.
omit_endlist
Do not append the "EXT-X-ENDLIST" tag at the end of the playlist.
periodic_rekey
The file specified by "hls_key_info_file" will be checked periodically and detect updates to the
encryption info. Be sure to replace this file atomically, including the file containing the AES
encryption key.
split_by_time
Allow segments to start on frames other than keyframes. This improves behavior on some players
when the time between keyframes is inconsistent, but may make things worse on others, and can
cause some oddities during seeking. This flag should be used with the "hls_time" option.
program_date_time
Generate "EXT-X-PROGRAM-DATE-TIME" tags.
second_level_segment_index
Makes it possible to use segment indexes as %%d in hls_segment_filename expression besides
date/time values when use_localtime is on. To get fixed width numbers with trailing zeroes,
%%0xd format is available where x is the required width.
second_level_segment_size
Makes it possible to use segment sizes (counted in bytes) as %%s in hls_segment_filename
expression besides date/time values when use_localtime is on. To get fixed width numbers with
trailing zeroes, %%0xs format is available where x is the required width.
second_level_segment_duration
Makes it possible to use segment duration (calculated in microseconds) as %%t in
hls_segment_filename expression besides date/time values when use_localtime is on. To get fixed
width numbers with trailing zeroes, %%0xt format is available where x is the required width.
ffmpeg -i sample.mpeg \
-f hls -hls_time 3 -hls_list_size 5 \
-hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
-use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8
This will produce segments like this: segment_20170102194334_0003_00122200_0000003000000.ts,
segment_20170102194334_0004_00120072_0000003000000.ts etc.
temp_file
Write segment data to filename.tmp and rename to filename only once the segment is complete. A
webserver serving up segments can be configured to reject requests to *.tmp to prevent access to
in-progress segments before they have been added to the m3u8 playlist.
hls_playlist_type event
Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces hls_list_size to 0; the playlist can
only be appended to.
hls_playlist_type vod
Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces hls_list_size to 0; the playlist must not
change.
method
Use the given HTTP method to create the hls files.
ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8
This example will upload all the mpegts segment files to the HTTP server using the HTTP PUT method,
and update the m3u8 files every "refresh" times using the same method. Note that the HTTP server
must support the given method for uploading files.
http_user_agent
Override User-Agent field in HTTP header. Applicable only for HTTP output.
ico
ICO file muxer.
Microsoft's icon file format (ICO) has some strict limitations that should be noted:
• Size cannot exceed 256 pixels in any dimension
• Only BMP and PNG images can be stored
• If a BMP image is used, it must be one of the following pixel formats:
BMP Bit Depth FFmpeg Pixel Format
1bit pal8
4bit pal8
8bit pal8
16bit rgb555le
24bit bgr24
32bit bgra
• If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
• If a PNG image is used, it must use the rgba pixel format
image2
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.
Examples
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 cfr -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 cfr -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 start of the input video you can employ the command:
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg
The strftime option allows you to expand the filename with date and time information. Check the
documentation of the "strftime()" function for the syntax.
For example to generate image files from the "strftime()" "%Y-%m-%d_%H-%M-%S" pattern, the following
ffmpeg command can be used:
ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"
Options
start_number
Start the sequence from the specified number. Default value is 1.
update
If set to 1, the filename will always be interpreted as just a filename, not a pattern, and the
corresponding file will be continuously overwritten with new images. Default value is 0.
strftime
If set to 1, expand the filename with date and time information from "strftime()". Default value is
0.
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
Matroska container muxer.
This muxer implements the matroska and webm container specs.
Metadata
The recognized metadata settings in this muxer are:
title
Set title name provided to a single track.
language
Specify the language of the track in the Matroska languages form.
The language can be either the 3 letters bibliographic ISO-639-2 (ISO 639-2/B) form (like "fre" for
French), or a language code mixed with a country code for specialities in languages (like "fre-ca"
for Canadian French).
stereo_mode
Set stereo 3D video layout of two views in a single video track.
The following values are recognized:
mono
video is not stereo
left_right
Both views are arranged side by side, Left-eye view is on the left
bottom_top
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
top_bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
checkerboard_rl
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
checkerboard_lr
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
row_interleaved_rl
Each view is constituted by a row based interleaving, Right-eye view is first row
row_interleaved_lr
Each view is constituted by a row based interleaving, Left-eye view is first row
col_interleaved_rl
Both views are arranged in a column based interleaving manner, Right-eye view is first column
col_interleaved_lr
Both views are arranged in a column based interleaving manner, Left-eye view is first column
anaglyph_cyan_red
All frames are in anaglyph format viewable through red-cyan filters
right_left
Both views are arranged side by side, Right-eye view is on the left
anaglyph_green_magenta
All frames are in anaglyph format viewable through green-magenta filters
block_lr
Both eyes laced in one Block, Left-eye view is first
block_rl
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
Options
This muxer supports the following options:
reserve_index_space
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
MD5 testing format.
This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash function.
Examples
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 hash and framemd5 muxers.
mov, mp4, ismv
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.
Options
Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments:
-moov_size bytes
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.
-movflags frag_keyframe
Start a new fragment at each video keyframe.
-frag_duration duration
Create fragments that are duration microseconds long.
-frag_size size
Create fragments that contain up to size bytes of payload data.
-movflags frag_custom
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.)
-min_frag_duration duration
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:
-movflags empty_moov
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.
This option is implicitly set when writing ismv (Smooth Streaming) files.
-movflags separate_moof
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.
-movflags faststart
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.
-movflags rtphint
Add RTP hinting tracks to the output file.
-movflags disable_chpl
Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track
are written to the file. With this option set, only the QuickTime chapter track will be written. Nero
chapters can cause failures when the file is reprocessed with certain tagging programs, like mp3Tag
2.61a and iTunes 11.3, most likely other versions are affected as well.
-movflags omit_tfhd_offset
Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute
byte positions in the file/streams.
-movflags default_base_moof
Similarly to the omit_tfhd_offset, this flag avoids writing the absolute base_data_offset field in
tfhd atoms, but does so by using the new default-base-is-moof flag instead. This flag is new from
14496-12:2012. This may make the fragments easier to parse in certain circumstances (avoiding basing
track fragment location calculations on the implicit end of the previous track fragment).
-write_tmcd
Specify "on" to force writing a timecode track, "off" to disable it and "auto" to write a timecode
track only for mov and mp4 output (default).
-movflags negative_cts_offsets
Enables utilization of version 1 of the CTTS box, in which the CTS offsets can be negative. This
enables the initial sample to have DTS/CTS of zero, and reduces the need for edit lists for some
cases such as video tracks with B-frames. Additionally, eases conformance with the DASH-IF
interoperability guidelines.
Example
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)
Audible AAX
Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4 byte activation
secret.
ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4
mp3
The MP3 muxer writes a raw MP3 stream with the following optional features:
• An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and 2.4 are supported,
the "id3v2_version" private option controls which one is used (3 or 4). Setting "id3v2_version" to 0
disables the ID3v2 header completely.
The muxer supports writing attached pictures (APIC frames) to the ID3v2 header. 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.
• A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by default, but will be
written only if the output is seekable. The "write_xing" private option can be used to disable it.
The frame contains various information that may be useful to the decoder, like the audio duration or
encoder delay.
• A legacy ID3v1 tag at the end of the file (disabled by default). It may be enabled with the
"write_id3v1" private option, but as its capabilities are very limited, its usage is not recommended.
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
Write a "clean" MP3 without any extra features:
ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3
mpegts
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
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.
Options
The muxer options are:
mpegts_transport_stream_id integer
Set the transport_stream_id. This identifies a transponder in DVB. Default is 0x0001.
mpegts_original_network_id integer
Set the original_network_id. 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. Default
is 0x0001.
mpegts_service_id integer
Set the service_id, also known as program in DVB. Default is 0x0001.
mpegts_service_type integer
Set the program service_type. Default is "digital_tv". Accepts the following options:
hex_value
Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.
digital_tv
Digital TV service.
digital_radio
Digital Radio service.
teletext
Teletext service.
advanced_codec_digital_radio
Advanced Codec Digital Radio service.
mpeg2_digital_hdtv
MPEG2 Digital HDTV service.
advanced_codec_digital_sdtv
Advanced Codec Digital SDTV service.
advanced_codec_digital_hdtv
Advanced Codec Digital HDTV service.
mpegts_pmt_start_pid integer
Set the first PID for PMT. Default is 0x1000. Max is 0x1f00.
mpegts_start_pid integer
Set the first PID for data packets. Default is 0x0100. Max is 0x0f00.
mpegts_m2ts_mode boolean
Enable m2ts mode if set to 1. Default value is "-1" which disables m2ts mode.
muxrate integer
Set a constant muxrate. Default is VBR.
pes_payload_size integer
Set minimum PES packet payload in bytes. Default is 2930.
mpegts_flags flags
Set mpegts flags. Accepts the following options:
resend_headers
Reemit PAT/PMT before writing the next packet.
latm
Use LATM packetization for AAC.
pat_pmt_at_frames
Reemit PAT and PMT at each video frame.
system_b
Conform to System B (DVB) instead of System A (ATSC).
initial_discontinuity
Mark the initial packet of each stream as discontinuity.
resend_headers integer
Reemit PAT/PMT before writing the next packet. This option is deprecated: use mpegts_flags instead.
mpegts_copyts boolean
Preserve original timestamps, if value is set to 1. Default value is "-1", which results in shifting
timestamps so that they start from 0.
omit_video_pes_length boolean
Omit the PES packet length for video packets. Default is 1 (true).
pcr_period integer
Override the default PCR retransmission time in milliseconds. Ignored if variable muxrate is
selected. Default is 20.
pat_period double
Maximum time in seconds between PAT/PMT tables.
sdt_period double
Maximum time in seconds between SDT tables.
tables_version integer
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
...
Example
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" \
out.ts
mxf, mxf_d10
MXF muxer.
Options
The muxer options are:
store_user_comments bool
Set if user comments should be stored if available or never. IRT D-10 does not allow user comments.
The default is thus to write them for mxf but not for mxf_d10
null
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 -
nut
-syncpoints flags
Change the syncpoint usage in nut:
default use the normal low-overhead seeking aids.
none do not use the syncpoints at all, reducing the overhead but making the stream non-seekable;
Use of this option is not recommended, as the resulting files are very damage
sensitive and seeking is not possible. Also in general the overhead from
syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
all growing data tables, allowing to mux endless streams with limited memory
and without these disadvantages.
timestamped extend the syncpoint with a wallclock field.
The none and timestamped flags are experimental.
-write_index bool
Write index at the end, the default is to write an index.
ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor
ogg
Ogg container muxer.
-page_duration duration
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.
-serial_offset value
Serial value from which to set the streams serial number. Setting it to different and sufficiently
large values ensures that the produced ogg files can be safely chained.
segment, stream_segment, ssegment
Basic stream segmenter.
This 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, or by using a "strftime" template if the strftime
option is enabled.
"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.
See also the hls muxer, which provides a more specific implementation for HLS segmentation.
Options
The segment muxer supports the following options:
increment_tc 1|0
if set to 1, increment timecode between each segment If this is selected, the input need to have a
timecode in the first video stream. Default value is 0.
reference_stream specifier
Set the reference stream, as specified by the string specifier. If specifier is set to "auto", the
reference is chosen 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".
segment_format format
Override the inner container format, by default it is guessed by the filename extension.
segment_format_options options_list
Set output format options using a :-separated list of key=value parameters. Values containing the ":"
special character must be escaped.
segment_list name
Generate also a listfile named name. If not specified no listfile is generated.
segment_list_flags flags
Set flags affecting the segment list generation.
It currently supports the following flags:
cache
Allow caching (only affects M3U8 list files).
live
Allow live-friendly file generation.
segment_list_size size
Update the list file so that it contains at most size segments. If 0 the list file will contain all
the segments. Default value is 0.
segment_list_entry_prefix prefix
Prepend prefix to each entry. Useful to generate absolute paths. By default no prefix is applied.
segment_list_type type
Select the listing format.
The following values are recognized:
flat
Generate a flat list for the created segments, one segment per line.
csv, ext
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.
ffconcat
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.
m3u8
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.
segment_time time
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.
segment_atclocktime 1|0
If set to "1" split at regular clock time intervals starting from 00:00 o'clock. The time value
specified in segment_time is used for setting the length of the splitting interval.
For example with segment_time set to "900" this makes it possible to create files at 12:00 o'clock,
12:15, 12:30, etc.
Default value is "0".
segment_clocktime_offset duration
Delay the segment splitting times with the specified duration when using segment_atclocktime.
For example with segment_time set to "900" and segment_clocktime_offset set to "300" this makes it
possible to create files at 12:05, 12:20, 12:35, etc.
Default value is "0".
segment_clocktime_wrap_duration duration
Force the segmenter to only start a new segment if a packet reaches the muxer within the specified
duration after the segmenting clock time. This way you can make the segmenter more resilient to
backward local time jumps, such as leap seconds or transition to standard time from daylight savings
time.
Default is the maximum possible duration which means starting a new segment regardless of the elapsed
time since the last clock time.
segment_time_delta delta
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.
segment_times times
Specify a list of split points. times contains a list of comma separated duration specifications, in
increasing order. See also the segment_time option.
segment_frames frames
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.
segment_wrap limit
Wrap around segment index once it reaches limit.
segment_start_number number
Set the sequence number of the first segment. Defaults to 0.
strftime 1|0
Use the "strftime" function to define the name of the new segments to write. If this is selected, the
output segment name must contain a "strftime" function template. Default value is 0.
break_non_keyframes 1|0
If enabled, allow segments to start on frames other than keyframes. This improves behavior on some
players when the time between keyframes is inconsistent, but may make things worse on others, and can
cause some oddities during seeking. Defaults to 0.
reset_timestamps 1|0
Reset timestamps at the beginning 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.
initial_offset offset
Specify timestamp offset to apply to the output packet timestamps. The argument must be a time
duration specification, and defaults to 0.
write_empty_segments 1|0
If enabled, write an empty segment if there are no packets during the period a segment would usually
span. Otherwise, the segment will be filled with the next packet written. Defaults to 0.
Examples
• Remux the content of file in.mkv to a list of segments out-000.nut, out-001.nut, etc., and write the
list of generated segments to out.list:
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut
• Segment input and set output format options for the output segments:
ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4
• Segment the input file according to the split points specified by the segment_times option:
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
• Use the 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.
• Segment the input file by splitting the input file according to the frame numbers sequence specified
with the segment_frames option:
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut
• Convert the in.mkv to TS segments using the "libx264" and "aac" encoders:
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts
• Segment the input file, and create an M3U8 live playlist (can be used as live HLS source):
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
smoothstreaming
Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving with conventional
web server.
window_size
Specify the number of fragments kept in the manifest. Default 0 (keep all).
extra_window_size
Specify the number of fragments kept outside of the manifest before removing from disk. Default 5.
lookahead_count
Specify the number of lookahead fragments. Default 2.
min_frag_duration
Specify the minimum fragment duration (in microseconds). Default 5000000.
remove_at_exit
Specify whether to remove all fragments when finished. Default 0 (do not remove).
fifo
The fifo pseudo-muxer allows the separation of encoding and muxing by using first-in-first-out queue and
running the actual muxer in a separate thread. This is especially useful in combination with the tee
muxer and can be used to send data to several destinations with different reliability/writing
speed/latency.
API users should be aware that callback functions (interrupt_callback, io_open and io_close) used within
its AVFormatContext must be thread-safe.
The behavior of the fifo muxer if the queue fills up or if the output fails is selectable,
• output can be transparently restarted with configurable delay between retries based on real time or
time of the processed stream.
• encoding can be blocked during temporary failure, or continue transparently dropping packets in case
fifo queue fills up.
fifo_format
Specify the format name. Useful if it cannot be guessed from the output name suffix.
queue_size
Specify size of the queue (number of packets). Default value is 60.
format_opts
Specify format options for the underlying muxer. Muxer options can be specified as a list of
key=value pairs separated by ':'.
drop_pkts_on_overflow bool
If set to 1 (true), in case the fifo queue fills up, packets will be dropped rather than blocking the
encoder. This makes it possible to continue streaming without delaying the input, at the cost of
omitting part of the stream. By default this option is set to 0 (false), so in such cases the encoder
will be blocked until the muxer processes some of the packets and none of them is lost.
attempt_recovery bool
If failure occurs, attempt to recover the output. This is especially useful when used with network
output, since it makes it possible to restart streaming transparently. By default this option is set
to 0 (false).
max_recovery_attempts
Sets maximum number of successive unsuccessful recovery attempts after which the output fails
permanently. By default this option is set to 0 (unlimited).
recovery_wait_time duration
Waiting time before the next recovery attempt after previous unsuccessful recovery attempt. Default
value is 5 seconds.
recovery_wait_streamtime bool
If set to 0 (false), the real time is used when waiting for the recovery attempt (i.e. the recovery
will be attempted after at least recovery_wait_time seconds). If set to 1 (true), the time of the
processed stream is taken into account instead (i.e. the recovery will be attempted after at least
recovery_wait_time seconds of the stream is omitted). By default, this option is set to 0 (false).
recover_any_error bool
If set to 1 (true), recovery will be attempted regardless of type of the error causing the failure.
By default this option is set to 0 (false) and in case of certain (usually permanent) errors the
recovery is not attempted even when attempt_recovery is set to 1.
restart_with_keyframe bool
Specify whether to wait for the keyframe after recovering from queue overflow or failure. This option
is set to 0 (false) by default.
Examples
• Stream something to rtmp server, continue processing the stream at real-time rate even in case of
temporary failure (network outage) and attempt to recover streaming every second indefinitely.
ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
-drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name
tee
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.
use_fifo bool
If set to 1, slave outputs will be processed in separate thread using fifo muxer. This allows to
compensate for different speed/latency/reliability of outputs and setup transparent recovery. By
default this feature is turned off.
fifo_options
Options to pass to fifo pseudo-muxer instances. See fifo.
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(1) 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:
f Specify the format name. Useful if it cannot be guessed from the output name suffix.
bsfs[/spec]
Specify a list of bitstream filters to apply to the specified output.
use_fifo bool
This allows to override tee muxer use_fifo option for individual slave muxer.
fifo_options
This allows to override tee muxer fifo_options for individual slave muxer. See fifo.
It is possible to specify to which streams a given bitstream filter applies, by appending a stream
specifier to the option separated by "/". spec must be a stream specifier (see Format stream
specifiers). If the stream specifier is not specified, the bitstream filters will be applied to all
streams in the output.
Several bitstream filters can be specified, separated by ",".
select
Select the streams that should be mapped to the slave output, specified by a stream specifier. If not
specified, this defaults to all the input streams. You may use multiple stream specifiers separated
by commas (",") e.g.: "a:0,v"
onfail
Specify behaviour on output failure. This can be set to either "abort" (which is default) or
"ignore". "abort" will cause whole process to fail in case of failure on this slave output. "ignore"
will ignore failure on this output, so other outputs will continue without being affected.
Examples
• Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP (the streams
need to be explicitly mapped):
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
• As above, but continue streaming even if output to local file fails (for example local drive fills
up):
ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
"[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"
• Use ffmpeg to encode the input, and send the output to three different destinations. The "dump_extra"
bitstream filter is used to add extradata information to all the output video keyframes packets, as
requested by the MPEG-TS format. The select option is applied to out.aac in order to make it contain
only audio packets.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"
• As below, but select only stream "a:1" for the audio output. Note that a second level escaping must
be performed, as ":" is a special character used to separate options.
ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
-f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']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.
webm_dash_manifest
WebM DASH Manifest muxer.
This muxer implements the WebM DASH Manifest specification to generate the DASH manifest XML. It also
supports manifest generation for DASH live streams.
For more information see:
• WebM DASH Specification:
<https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>
• ISO DASH Specification:
<http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>
Options
This muxer supports the following options:
adaptation_sets
This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x and y are the
unique identifiers of the adaptation sets and a,b,c,d and e are the indices of the corresponding
audio and video streams. Any number of adaptation sets can be added using this option.
live
Set this to 1 to create a live stream DASH Manifest. Default: 0.
chunk_start_index
Start index of the first chunk. This will go in the startNumber attribute of the SegmentTemplate
element in the manifest. Default: 0.
chunk_duration_ms
Duration of each chunk in milliseconds. This will go in the duration attribute of the SegmentTemplate
element in the manifest. Default: 1000.
utc_timing_url
URL of the page that will return the UTC timestamp in ISO format. This will go in the value attribute
of the UTCTiming element in the manifest. Default: None.
time_shift_buffer_depth
Smallest time (in seconds) shifting buffer for which any Representation is guaranteed to be
available. This will go in the timeShiftBufferDepth attribute of the MPD element. Default: 60.
minimum_update_period
Minimum update period (in seconds) of the manifest. This will go in the minimumUpdatePeriod attribute
of the MPD element. Default: 0.
Example
ffmpeg -f webm_dash_manifest -i video1.webm \
-f webm_dash_manifest -i video2.webm \
-f webm_dash_manifest -i audio1.webm \
-f webm_dash_manifest -i audio2.webm \
-map 0 -map 1 -map 2 -map 3 \
-c copy \
-f webm_dash_manifest \
-adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
manifest.xml
webm_chunk
WebM Live Chunk Muxer.
This muxer writes out WebM headers and chunks as separate files which can be consumed by clients that
support WebM Live streams via DASH.
Options
This muxer supports the following options:
chunk_start_index
Index of the first chunk (defaults to 0).
header
Filename of the header where the initialization data will be written.
audio_chunk_duration
Duration of each audio chunk in milliseconds (defaults to 5000).
Example
ffmpeg -f v4l2 -i /dev/video0 \
-f alsa -i hw:0 \
-map 0:0 \
-c:v libvpx-vp9 \
-s 640x360 -keyint_min 30 -g 30 \
-f webm_chunk \
-header webm_live_video_360.hdr \
-chunk_start_index 1 \
webm_live_video_360_%d.chk \
-map 1:0 \
-c:a libvorbis \
-b:a 128k \
-f webm_chunk \
-header webm_live_audio_128.hdr \
-chunk_start_index 1 \
-audio_chunk_duration 1000 \
webm_live_audio_128_%d.chk
METADATA
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then
load it back using the metadata muxer/demuxer.
The file format is as follows:
1. A file consists of a header and a number of metadata tags divided into sections, each on its own
line.
2. The header is a ;FFMETADATA string, followed by a version number (now 1).
3. Metadata tags are of the form key=value
4. Immediately after header follows global metadata
5. After global metadata there may be sections with per-stream/per-chapter metadata.
6. A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in brackets ([, ]) and
ends with next section or end of file.
7. At the beginning of a chapter section there may be an optional timebase to be used for start/end
values. It must be in form TIMEBASE=num/den, where num and den are integers. If the timebase is
missing then start/end times are assumed to be in milliseconds.
Next a chapter section must contain chapter start and end times in form START=num, END=num, where num
is a positive integer.
8. Empty lines and lines starting with ; or # are ignored.
9. Metadata keys or values containing special characters (=, ;, #, \ and a newline) must be escaped with
a backslash \.
10. Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag (in the
example above key is foo , value is
bar).
A ffmetadata file might look like this:
;FFMETADATA1
title=bike\\shed
;this is a comment
artist=FFmpeg troll team
[CHAPTER]
TIMEBASE=1/1000
START=0
#chapter ends at 0:01:00
END=60000
title=chapter \#1
[STREAM]
title=multi\
line
By using the ffmetadata muxer and demuxer it is possible to extract metadata from an input file to an
ffmetadata file, and then transcode the file into an output file with the edited ffmetadata file.
Extracting an ffmetadata file with ffmpeg goes as follows:
ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE
Reinserting edited metadata information from the FFMETADATAFILE file can be done as:
ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT
PROTOCOL OPTIONS
The libavformat library provides some generic global options, which can be set on all the protocols. In
addition each protocol may support so-called private options, which are specific for that component.
Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in
the "AVFormatContext" options or using the libavutil/opt.h API for programmatic use.
The list of supported options follows:
protocol_whitelist list (input)
Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols prefixed by "-"
are disabled. All protocols are allowed by default but protocols used by an another protocol (nested
protocols) are restricted to a per protocol subset.
PROTOCOLS
Protocols are configured elements in FFmpeg that enable access to resources that require specific
protocols.
When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list
all available ones using the configure option "--list-protocols".
You can disable all the protocols using the configure option "--disable-protocols", and selectively
enable a protocol using the option "--enable-protocol=PROTOCOL", or you can disable a particular protocol
using the option "--disable-protocol=PROTOCOL".
The option "-protocols" of the ff* tools will display the list of supported protocols.
All protocols accept the following options:
rw_timeout
Maximum time to wait for (network) read/write operations to complete, in microseconds.
A description of the currently available protocols follows.
async
Asynchronous data filling wrapper for input stream.
Fill data in a background thread, to decouple I/O operation from demux thread.
async:<URL>
async:http://host/resource
async:cache:http://host/resource
bluray
Read BluRay playlist.
The accepted options are:
angle
BluRay angle
chapter
Start chapter (1...N)
playlist
Playlist to read (BDMV/PLAYLIST/?????.mpls)
Examples:
Read longest playlist from BluRay mounted to /mnt/bluray:
bluray:/mnt/bluray
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray
cache
Caching wrapper for input stream.
Cache the input stream to temporary file. It brings seeking capability to live streams.
cache:<URL>
concat
Physical concatenation protocol.
Read and seek from many resources in sequence as if they were a unique resource.
A URL accepted by this protocol has the syntax:
concat:<URL1>|<URL2>|...|<URLN>
where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying
a distinct protocol.
For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay use the
command:
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg
Note that you may need to escape the character "|" which is special for many shells.
crypto
AES-encrypted stream reading protocol.
The accepted options are:
key Set the AES decryption key binary block from given hexadecimal representation.
iv Set the AES decryption initialization vector binary block from given hexadecimal representation.
Accepted URL formats:
crypto:<URL>
crypto+<URL>
data
Data in-line in the URI. See <http://en.wikipedia.org/wiki/Data_URI_scheme>.
For example, to convert a GIF file given inline with ffmpeg:
ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png
file
File access protocol.
Read from or write to a file.
A file URL can have the form:
file:<filename>
where filename is the path of the file to read.
An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on the build, an
URL that looks like a Windows path with the drive letter at the beginning will also be assumed to be a
file URL (usually not the case in builds for unix-like systems).
For example to read from a file input.mpeg with ffmpeg use the command:
ffmpeg -i file:input.mpeg output.mpeg
This protocol accepts the following options:
truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
blocksize
Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not
limiting the requested block size. Setting this value reasonably low improves user termination
request reaction time, which is valuable for files on slow medium.
ftp
FTP (File Transfer Protocol).
Read from or write to remote resources using FTP protocol.
Following syntax is required.
ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
This protocol accepts the following options.
timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By
default it is set to -1, which means that the timeout is not specified.
ftp-anonymous-password
Password used when login as anonymous user. Typically an e-mail address should be used.
ftp-write-seekable
Control seekability of connection during encoding. If set to 1 the resource is supposed to be
seekable, if set to 0 it is assumed not to be seekable. Default value is 0.
NOTE: Protocol can be used as output, but it is recommended to not do it, unless special care is taken
(tests, customized server configuration etc.). Different FTP servers behave in different way during seek
operation. ff* tools may produce incomplete content due to server limitations.
This protocol accepts the following options:
follow
If set to 1, the protocol will retry reading at the end of the file, allowing reading files that
still are being written. In order for this to terminate, you either need to use the rw_timeout
option, or use the interrupt callback (for API users).
gopher
Gopher protocol.
hls
Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing
the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The
nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either
"file" or "http".
hls+http://host/path/to/remote/resource.m3u8
hls+file://path/to/local/resource.m3u8
Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the
issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8
files.
http
HTTP (Hyper Text Transfer Protocol).
This protocol accepts the following options:
seekable
Control seekability of connection. If set to 1 the resource is supposed to be seekable, if set to 0
it is assumed not to be seekable, if set to -1 it will try to autodetect if it is seekable. Default
value is -1.
chunked_post
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
content_type
Set a specific content type for the POST messages or for listen mode.
http_proxy
set HTTP proxy to tunnel through e.g. http://example.com:1234
headers
Set custom HTTP headers, can override built in default headers. The value must be a string encoding
the headers.
multiple_requests
Use persistent connections if set to 1, default is 0.
post_data
Set custom HTTP post data.
user_agent
Override the User-Agent header. If not specified the protocol will use a string describing the
libavformat build. ("Lavf/<version>")
user-agent
This is a deprecated option, you can use user_agent instead it.
timeout
Set timeout in microseconds of socket I/O operations used by the underlying low level operation. By
default it is set to -1, which means that the timeout is not specified.
reconnect_at_eof
If set then eof is treated like an error and causes reconnection, this is useful for live / endless
streams.
reconnect_streamed
If set then even streamed/non seekable streams will be reconnected on errors.
reconnect_delay_max
Sets the maximum delay in seconds after which to give up reconnecting
mime_type
Export the MIME type.
icy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the
metadata has to be retrieved by the application by reading the icy_metadata_headers and
icy_metadata_packet options. The default is 1.
icy_metadata_headers
If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by
newline characters.
icy_metadata_packet
If the server supports ICY metadata, and icy was set to 1, this contains the last non-empty metadata
packet sent by the server. It should be polled in regular intervals by applications interested in
mid-stream metadata updates.
cookies
Set the cookies to be sent in future requests. The format of each cookie is the same as the value of
a Set-Cookie HTTP response field. Multiple cookies can be delimited by a newline character.
offset
Set initial byte offset.
end_offset
Try to limit the request to bytes preceding this offset.
method
When used as a client option it sets the HTTP method for the request.
When used as a server option it sets the HTTP method that is going to be expected from the client(s).
If the expected and the received HTTP method do not match the client will be given a Bad Request
response. When unset the HTTP method is not checked for now. This will be replaced by autodetection
in the future.
listen
If set to 1 enables experimental HTTP server. This can be used to send data when used as an output
option, or read data from a client with HTTP POST when used as an input option. If set to 2 enables
experimental multi-client HTTP server. This is not yet implemented in ffmpeg.c or ffserver.c and thus
must not be used as a command line option.
# Server side (sending):
ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>
# Client side (receiving):
ffmpeg -i http://<server>:<port> -c copy somefile.ogg
# Client can also be done with wget:
wget http://<server>:<port> -O somefile.ogg
# Server side (receiving):
ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg
# Client side (sending):
ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>
# Client can also be done with wget:
wget --post-file=somefile.ogg http://<server>:<port>
HTTP Cookies
Some HTTP requests will be denied unless cookie values are passed in with the request. The cookies option
allows these cookies to be specified. At the very least, each cookie must specify a value along with a
path and domain. HTTP requests that match both the domain and path will automatically include the cookie
value in the HTTP Cookie header field. Multiple cookies can be delimited by a newline.
The required syntax to play a stream specifying a cookie is:
ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8
Icecast
Icecast protocol (stream to Icecast servers)
This protocol accepts the following options:
ice_genre
Set the stream genre.
ice_name
Set the stream name.
ice_description
Set the stream description.
ice_url
Set the stream website URL.
ice_public
Set if the stream should be public. The default is 0 (not public).
user_agent
Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.
password
Set the Icecast mountpoint password.
content_type
Set the stream content type. This must be set if it is different from audio/mpeg.
legacy_icecast
This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the
SOURCE method.
icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>
mmst
MMS (Microsoft Media Server) protocol over TCP.
mmsh
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
mmsh://<server>[:<port>][/<app>][/<playpath>]
md5
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes this to the designated output or
stdout if none is specified. It can be used to test muxers without writing an actual file.
Some examples follow.
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
ffmpeg -i input.flv -f avi -y md5:output.avi.md5
# Write the MD5 hash of the encoded AVI file to stdout.
ffmpeg -i input.flv -f avi -y md5:
Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with
the MD5 output protocol.
pipe
UNIX pipe access protocol.
Read and write from UNIX pipes.
The accepted syntax is:
pipe:[<number>]
number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2
for stderr). If number is not specified, by default the stdout file descriptor will be used for writing,
stdin for reading.
For example to read from stdin with ffmpeg:
cat test.wav | ffmpeg -i pipe:0
# ...this is the same as...
cat test.wav | ffmpeg -i pipe:
For writing to stdout with ffmpeg:
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
# ...this is the same as...
ffmpeg -i test.wav -f avi pipe: | cat > test.avi
This protocol accepts the following options:
blocksize
Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which results in not
limiting the requested block size. Setting this value reasonably low improves user termination
request reaction time, which is valuable if data transmission is slow.
Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail
with the pipe output protocol.
prompeg
Pro-MPEG Code of Practice #3 Release 2 FEC protocol.
The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2 Transport
Streams sent over RTP.
This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp" protocol.
The required syntax is:
-f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>
The destination UDP ports are "port + 2" for the column FEC stream and "port + 4" for the row FEC stream.
This protocol accepts the following options:
l=n The number of columns (4-20, LxD <= 100)
d=n The number of rows (4-20, LxD <= 100)
Example usage:
-f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>
rtmp
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.
The required syntax is:
rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]
The accepted parameters are:
username
An optional username (mostly for publishing).
password
An optional password (mostly for publishing).
server
The address of the RTMP server.
port
The number of the TCP port to use (by default is 1935).
app It is the name of the application to access. It usually corresponds to the path where the application
is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.). You can override the value
parsed from the URI through the "rtmp_app" option, too.
playpath
It is the path or name of the resource to play with reference to the application specified in app,
may be prefixed by "mp4:". You can override the value parsed from the URI through the "rtmp_playpath"
option, too.
listen
Act as a server, listening for an incoming connection.
timeout
Maximum time to wait for the incoming connection. Implies listen.
Additionally, the following parameters can be set via command line options (or in code via "AVOption"s):
rtmp_app
Name of application to connect on the RTMP server. This option overrides the parameter specified in
the URI.
rtmp_buffer
Set the client buffer time in milliseconds. The default is 3000.
rtmp_conn
Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1 S:authMe O:1
NN:code:1.23 NS:flag:ok O:0". Each value is prefixed by a single character denoting the type, B for
Boolean, N for number, S for string, O for object, or Z for null, followed by a colon. For Booleans
the data must be either 0 or 1 for FALSE or TRUE, respectively. Likewise for Objects the data must
be 0 or 1 to end or begin an object, respectively. Data items in subobjects may be named, by
prefixing the type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This
option may be used multiple times to construct arbitrary AMF sequences.
rtmp_flashver
Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When
publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)
rtmp_flush_interval
Number of packets flushed in the same request (RTMPT only). The default is 10.
rtmp_live
Specify that the media is a live stream. No resuming or seeking in live streams is possible. The
default value is "any", which means the subscriber first tries to play the live stream specified in
the playpath. If a live stream of that name is not found, it plays the recorded stream. The other
possible values are "live" and "recorded".
rtmp_pageurl
URL of the web page in which the media was embedded. By default no value will be sent.
rtmp_playpath
Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
rtmp_subscribe
Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option
is specified or if rtmp_live is set to live.
rtmp_swfhash
SHA256 hash of the decompressed SWF file (32 bytes).
rtmp_swfsize
Size of the decompressed SWF file, required for SWFVerification.
rtmp_swfurl
URL of the SWF player for the media. By default no value will be sent.
rtmp_swfverify
URL to player swf file, compute hash/size automatically.
rtmp_tcurl
URL of the target stream. Defaults to proto://host[:port]/app.
For example to read with ffplay a multimedia resource named "sample" from the application "vod" from an
RTMP server "myserver":
ffplay rtmp://myserver/vod/sample
To publish to a password protected server, passing the playpath and app names separately:
ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/
rtmpe
Encrypted Real-Time Messaging Protocol.
The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within
standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
pair of RC4 keys.
rtmps
Real-Time Messaging Protocol over a secure SSL connection.
The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted
connection.
rtmpt
Real-Time Messaging Protocol tunneled through HTTP.
The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content
within HTTP requests to traverse firewalls.
rtmpte
Encrypted Real-Time Messaging Protocol tunneled through HTTP.
The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming
multimedia content within HTTP requests to traverse firewalls.
rtmpts
Real-Time Messaging Protocol tunneled through HTTPS.
The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content
within HTTPS requests to traverse firewalls.
libsmbclient
libsmbclient permits one to manipulate CIFS/SMB network resources.
Following syntax is required.
smb://[[domain:]user[:password@]]server[/share[/path[/file]]]
This protocol accepts the following options.
timeout
Set timeout in milliseconds of socket I/O operations used by the underlying low level operation. By
default it is set to -1, which means that the timeout is not specified.
truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
workgroup
Set the workgroup used for making connections. By default workgroup is not specified.
For more information see: <http://www.samba.org/>.
libssh
Secure File Transfer Protocol via libssh
Read from or write to remote resources using SFTP protocol.
Following syntax is required.
sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg
This protocol accepts the following options.
timeout
Set timeout of socket I/O operations used by the underlying low level operation. By default it is set
to -1, which means that the timeout is not specified.
truncate
Truncate existing files on write, if set to 1. A value of 0 prevents truncating. Default value is 1.
private_key
Specify the path of the file containing private key to use during authorization. By default libssh
searches for keys in the ~/.ssh/ directory.
Example: Play a file stored on remote server.
ffplay sftp://user:password@server_address:22/home/user/resource.mpeg
librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
Real-Time Messaging Protocol and its variants supported through librtmp.
Requires the presence of the librtmp headers and library during configuration. You need to explicitly
configure the build with "--enable-librtmp". If enabled this will replace the native RTMP protocol.
This protocol provides most client functions and a few server functions needed to support RTMP, RTMP
tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of
these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
<rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts"
corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified
for the RTMP native protocol. options contains a list of space-separated options of the form key=val.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using ffmpeg:
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream
To play the same stream using ffplay:
ffplay "rtmp://myserver/live/mystream live=1"
rtp
Real-time Transport Protocol.
The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]
port specifies the RTP port to use.
The following URL options are supported:
ttl=n
Set the TTL (Time-To-Live) value (for multicast only).
rtcpport=n
Set the remote RTCP port to n.
localrtpport=n
Set the local RTP port to n.
localrtcpport=n'
Set the local RTCP port to n.
pkt_size=n
Set max packet size (in bytes) to n.
connect=0|1
Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0).
sources=ip[,ip]
List allowed source IP addresses.
block=ip[,ip]
List disallowed (blocked) source IP addresses.
write_to_source=0|1
Send packets to the source address of the latest received packet (if set to 1) or to a default remote
address (if set to 0).
localport=n
Set the local RTP port to n.
This is a deprecated option. Instead, localrtpport should be used.
Important notes:
1. If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.
2. If localrtpport (the local RTP port) is not set any available port will be used for the local RTP and
RTCP ports.
3. If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port value plus 1.
rtsp
Real-Time Streaming Protocol.
RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer
supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and
Real-RTSP (with data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin
Streaming Server and Mischa Spiegelmock's <https://github.com/revmischa/rtsp-server>).
The required syntax for a RTSP url is:
rtsp://<hostname>[:<port>]/<path>
Options can be set on the ffmpeg/ffplay command line, or set in code via "AVOption"s or in
"avformat_open_input".
The following options are supported.
initial_pause
Do not start playing the stream immediately if set to 1. Default value is 0.
rtsp_transport
Set RTSP transport protocols.
It accepts the following values:
udp Use UDP as lower transport protocol.
tcp Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
udp_multicast
Use UDP multicast as lower transport protocol.
http
Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.
Multiple lower transport protocols may be specified, in that case they are tried one at a time (if
the setup of one fails, the next one is tried). For the muxer, only the tcp and udp options are
supported.
rtsp_flags
Set RTSP flags.
The following values are accepted:
filter_src
Accept packets only from negotiated peer address and port.
listen
Act as a server, listening for an incoming connection.
prefer_tcp
Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.
Default value is none.
allowed_media_types
Set media types to accept from the server.
The following flags are accepted:
video
audio
data
By default it accepts all media types.
min_port
Set minimum local UDP port. Default value is 5000.
max_port
Set maximum local UDP port. Default value is 65000.
timeout
Set maximum timeout (in seconds) to wait for incoming connections.
A value of -1 means infinite (default). This option implies the rtsp_flags set to listen.
reorder_queue_size
Set number of packets to buffer for handling of reordered packets.
stimeout
Set socket TCP I/O timeout in microseconds.
user-agent
Override User-Agent header. If not specified, it defaults to the libavformat identifier string.
When receiving data over UDP, the demuxer tries to reorder received packets (since they may arrive out of
order, or packets may get lost totally). This can be disabled by setting the maximum demuxing delay to
zero (via the "max_delay" field of AVFormatContext).
When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be chosen with
"-vst" n and "-ast" n for video and audio respectively, and can be switched on the fly by pressing "v"
and "a".
Examples
The following examples all make use of the ffplay and ffmpeg tools.
• Watch a stream over UDP, with a max reordering delay of 0.5 seconds:
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4
• Watch a stream tunneled over HTTP:
ffplay -rtsp_transport http rtsp://server/video.mp4
• Send a stream in realtime to a RTSP server, for others to watch:
ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp
• Receive a stream in realtime:
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>
sap
Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it
is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams
regularly on a separate port.
Muxer
The syntax for a SAP url given to the muxer is:
sap://<destination>[:<port>][?<options>]
The RTP packets are sent to destination on port port, or to port 5004 if no port is specified. options
is a "&"-separated list. The following options are supported:
announce_addr=address
Specify the destination IP address for sending the announcements to. If omitted, the announcements
are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or
ff0e::2:7ffe if destination is an IPv6 address.
announce_port=port
Specify the port to send the announcements on, defaults to 9875 if not specified.
ttl=ttl
Specify the time to live value for the announcements and RTP packets, defaults to 255.
same_port=0|1
If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent
on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires
this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving
requires all streams to be sent on unique ports.
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1
Similarly, for watching in ffplay:
ffmpeg -re -i <input> -f sap sap://224.0.0.255
And for watching in ffplay, over IPv6:
ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]
Demuxer
The syntax for a SAP url given to the demuxer is:
sap://[<address>][:<port>]
address is the multicast address to listen for announcements on, if omitted, the default 224.2.127.254
(sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port. Once an announcement is received,
it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
ffplay sap://
To play back the first stream announced on one the default IPv6 SAP multicast address:
ffplay sap://[ff0e::2:7ffe]
sctp
Stream Control Transmission Protocol.
The accepted URL syntax is:
sctp://<host>:<port>[?<options>]
The protocol accepts the following options:
listen
If set to any value, listen for an incoming connection. Outgoing connection is done by default.
max_streams
Set the maximum number of streams. By default no limit is set.
srtp
Secure Real-time Transport Protocol.
The accepted options are:
srtp_in_suite
srtp_out_suite
Select input and output encoding suites.
Supported values:
AES_CM_128_HMAC_SHA1_80
SRTP_AES128_CM_HMAC_SHA1_80
AES_CM_128_HMAC_SHA1_32
SRTP_AES128_CM_HMAC_SHA1_32
srtp_in_params
srtp_out_params
Set input and output encoding parameters, which are expressed by a base64-encoded representation of a
binary block. The first 16 bytes of this binary block are used as master key, the following 14 bytes
are used as master salt.
subfile
Virtually extract a segment of a file or another stream. The underlying stream must be seekable.
Accepted options:
start
Start offset of the extracted segment, in bytes.
end End offset of the extracted segment, in bytes.
Examples:
Extract a chapter from a DVD VOB file (start and end sectors obtained externally and multiplied by 2048):
subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB
Play an AVI file directly from a TAR archive:
subfile,,start,183241728,end,366490624,,:archive.tar
tee
Writes the output to multiple protocols. The individual outputs are separated by |
tee:file://path/to/local/this.avi|file://path/to/local/that.avi
tcp
Transmission Control Protocol.
The required syntax for a TCP url is:
tcp://<hostname>:<port>[?<options>]
options contains a list of &-separated options of the form key=val.
The list of supported options follows.
listen=1|0
Listen for an incoming connection. Default value is 0.
timeout=microseconds
Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise
error.
listen_timeout=milliseconds
Set listen timeout, expressed in milliseconds.
recv_buffer_size=bytes
Set receive buffer size, expressed bytes.
send_buffer_size=bytes
Set send buffer size, expressed bytes.
The following example shows how to setup a listening TCP connection with ffmpeg, which is then accessed
with ffplay:
ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
ffplay tcp://<hostname>:<port>
tls
Transport Layer Security (TLS) / Secure Sockets Layer (SSL)
The required syntax for a TLS/SSL url is:
tls://<hostname>:<port>[?<options>]
The following parameters can be set via command line options (or in code via "AVOption"s):
ca_file, cafile=filename
A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS
library contains a default this might not need to be specified for verification to work, but not all
libraries and setups have defaults built in. The file must be in OpenSSL PEM format.
tls_verify=1|0
If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this
currently only makes sure that the peer certificate is signed by one of the root certificates in the
CA database, but it does not validate that the certificate actually matches the host name we are
trying to connect to. (With 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.
cert_file, cert=filename
A file containing a certificate to use in the handshake with the peer. (When operating as server, in
listen mode, this is more often required by the peer, while client certificates only are mandated in
certain setups.)
key_file, key=filename
A file containing the private key for the certificate.
listen=1|0
If enabled, listen for connections on the provided port, and assume the server role in the handshake
instead of the client role.
Example command lines:
To create a TLS/SSL server that serves an input stream.
ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>
To play back a stream from the TLS/SSL server using ffplay:
ffplay tls://<hostname>:<port>
udp
User Datagram Protocol.
The required syntax for an UDP URL is:
udp://<hostname>:<port>[?<options>]
options contains a list of &-separated options of the form key=val.
In case threading is enabled on the system, a circular buffer is used to store the incoming data, which
allows one to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal
options are related to this buffer.
The list of supported options follows.
buffer_size=size
Set the UDP maximum socket buffer size in bytes. This is used to set either the receive or send
buffer size, depending on what the socket is used for. Default is 64KB. See also fifo_size.
bitrate=bitrate
If set to nonzero, the output will have the specified constant bitrate if the input has enough
packets to sustain it.
burst_bits=bits
When using bitrate this specifies the maximum number of bits in packet bursts.
localport=port
Override the local UDP port to bind with.
localaddr=addr
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.
pkt_size=size
Set the size in bytes of UDP packets.
reuse=1|0
Explicitly allow or disallow reusing UDP sockets.
ttl=ttl
Set the time to live value (for multicast only).
connect=1|0
Initialize the UDP socket with "connect()". In this case, the destination address can't be changed
with ff_udp_set_remote_url later. If the destination address isn't known at the start, this option
can be specified in ff_udp_set_remote_url, too. This allows finding out the source address for the
packets with getsockname, and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received. For receiving, this gives the benefit of only receiving packets from the
specified peer address/port.
sources=address[,address]
Only receive packets sent to the multicast group from one of the specified sender IP addresses.
block=address[,address]
Ignore packets sent to the multicast group from the specified sender IP addresses.
fifo_size=units
Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes.
If not specified defaults to 7*4096.
overrun_nonfatal=1|0
Survive in case of UDP receiving circular buffer overrun. Default value is 0.
timeout=microseconds
Set raise error timeout, expressed in microseconds.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise
error.
broadcast=1|0
Explicitly allow or disallow UDP broadcasting.
Note that broadcasting may not work properly on networks having a broadcast storm protection.
Examples
• Use ffmpeg to stream over UDP to a remote endpoint:
ffmpeg -i <input> -f <format> udp://<hostname>:<port>
• Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a large input
buffer:
ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535
• Use ffmpeg to receive over UDP from a remote endpoint:
ffmpeg -i udp://[<multicast-address>]:<port> ...
unix
Unix local socket
The required syntax for a Unix socket URL is:
unix://<filepath>
The following parameters can be set via command line options (or in code via "AVOption"s):
timeout
Timeout in ms.
listen
Create the Unix socket in listening mode.
DEVICE OPTIONS
The libavdevice library provides the same interface as libavformat. Namely, an input device is considered
like a demuxer, and an output device like a muxer, and the interface and generic device options are the
same provided by libavformat (see the ffmpeg-formats manual).
In addition each input or output device may support so-called private options, which are specific for
that component.
Options may be set by specifying -option value in the FFmpeg tools, or by setting the value explicitly in
the device "AVFormatContext" options or using the libavutil/opt.h API for programmatic use.
INPUT DEVICES
Input devices are configured elements in FFmpeg which enable accessing the data coming from a multimedia
device attached to your system.
When you configure your FFmpeg build, all the supported input devices are enabled by default. You can
list all available ones using the configure option "--list-indevs".
You can disable all the input devices using the configure option "--disable-indevs", and selectively
enable an input device using the option "--enable-indev=INDEV", or you can disable a particular input
device using the option "--disable-indev=INDEV".
The option "-devices" of the ff* tools will display the list of supported input devices.
A description of the currently available input devices follows.
alsa
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound installed on your system.
This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA
card identifier.
An ALSA identifier has the syntax:
hw:<CARD>[,<DEV>[,<SUBDEV>]]
where the DEV and SUBDEV components are optional.
The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and
subdevice number (-1 means any).
To see the list of cards currently recognized by your system check the files /proc/asound/cards and
/proc/asound/devices.
For example to capture with ffmpeg from an ALSA device with card id 0, you may run the command:
ffmpeg -f alsa -i hw:0 alsaout.wav
For more information see: <http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>
Options
sample_rate
Set the sample rate in Hz. Default is 48000.
channels
Set the number of channels. Default is 2.
avfoundation
AVFoundation input device.
AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >= 10.7 as well as
on iOS.
The input filename has to be given in the following syntax:
-i "[[VIDEO]:[AUDIO]]"
The first entry selects the video input while the latter selects the audio input. The stream has to be
specified by the device name or the device index as shown by the device list. Alternatively, the video
and/or audio input device can be chosen by index using the
B<-video_device_index E<lt>INDEXE<gt>>
and/or
B<-audio_device_index E<lt>INDEXE<gt>>
, overriding any device name or index given in the input filename.
All available devices can be enumerated by using -list_devices true, listing all device names and
corresponding indices.
There are two device name aliases:
"default"
Select the AVFoundation default device of the corresponding type.
"none"
Do not record the corresponding media type. This is equivalent to specifying an empty device name or
index.
Options
AVFoundation supports the following options:
-list_devices <TRUE|FALSE>
If set to true, a list of all available input devices is given showing all device names and indices.
-video_device_index <INDEX>
Specify the video device by its index. Overrides anything given in the input filename.
-audio_device_index <INDEX>
Specify the audio device by its index. Overrides anything given in the input filename.
-pixel_format <FORMAT>
Request the video device to use a specific pixel format. If the specified format is not supported, a
list of available formats is given and the first one in this list is used instead. Available pixel
formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
yuv420p, nv12, yuyv422, gray"
-framerate
Set the grabbing frame rate. Default is "ntsc", corresponding to a frame rate of "30000/1001".
-video_size
Set the video frame size.
-capture_cursor
Capture the mouse pointer. Default is 0.
-capture_mouse_clicks
Capture the screen mouse clicks. Default is 0.
Examples
• Print the list of AVFoundation supported devices and exit:
$ ffmpeg -f avfoundation -list_devices true -i ""
• Record video from video device 0 and audio from audio device 0 into out.avi:
$ ffmpeg -f avfoundation -i "0:0" out.avi
• Record video from video device 2 and audio from audio device 1 into out.avi:
$ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi
• Record video from the system default video device using the pixel format bgr0 and do not record any
audio into out.avi:
$ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi
bktr
BSD video input device.
Options
framerate
Set the frame rate.
video_size
Set the video frame size. Default is "vga".
standard
Available values are:
pal
ntsc
secam
paln
palm
ntscj
decklink
The decklink input device provides capture capabilities for Blackmagic DeckLink devices.
To enable this input device, you need the Blackmagic DeckLink SDK and you need to configure with the
appropriate "--extra-cflags" and "--extra-ldflags". On Windows, you need to run the IDL files through
widl.
DeckLink is very picky about the formats it supports. Pixel format of the input can be set with
raw_format. Framerate and video size must be determined for your device with -list_formats 1. Audio
sample rate is always 48 kHz and the number of channels can be 2, 8 or 16. Note that all audio channels
are bundled in one single audio track.
Options
list_devices
If set to true, print a list of devices and exit. Defaults to false.
list_formats
If set to true, print a list of supported formats and exit. Defaults to false.
format_code <FourCC>
This sets the input video format to the format given by the FourCC. To see the supported values of
your device(s) use list_formats. Note that there is a FourCC 'pal ' that can also be used as pal (3
letters).
bm_v210
This is a deprecated option, you can use raw_format instead. If set to 1, video is captured in 10
bit v210 instead of uyvy422. Not all Blackmagic devices support this option.
raw_format
Set the pixel format of the captured video. Available values are:
uyvy422
yuv422p10
argb
bgra
rgb10
teletext_lines
If set to nonzero, an additional teletext stream will be captured from the vertical ancillary data.
Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In case of HD sources, OP47 packets
are decoded.
This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22, and lines 318
to 335. Line 6 is the LSB in the mask. Selected lines which do not contain teletext information will
be ignored. You can use the special all constant to select all possible lines, or standard to skip
lines 6, 318 and 319, which are not compatible with all receivers.
For SD sources, ffmpeg needs to be compiled with "--enable-libzvbi". For HD sources, on older
(pre-4K) DeckLink card models you have to capture in 10 bit mode.
channels
Defines number of audio channels to capture. Must be 2, 8 or 16. Defaults to 2.
duplex_mode
Sets the decklink device duplex mode. Must be unset, half or full. Defaults to unset.
video_input
Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component, composite or s_video.
Defaults to unset.
audio_input
Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr, analog_rca or
microphone. Defaults to unset.
video_pts
Sets the video packet timestamp source. Must be video, audio, reference or wallclock. Defaults to
video.
audio_pts
Sets the audio packet timestamp source. Must be video, audio, reference or wallclock. Defaults to
audio.
draw_bars
If set to true, color bars are drawn in the event of a signal loss. Defaults to true.
queue_size
Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming frames will be
dropped. Defaults to 1073741824.
Examples
• List input devices:
ffmpeg -f decklink -list_devices 1 -i dummy
• List supported formats:
ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'
• Capture video clip at 1080i50:
ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi
• Capture video clip at 1080i50 10 bit:
ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
• Capture video clip at 1080i50 with 16 audio channels:
ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi
kmsgrab
KMS video input device.
Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM object that can
be passed to other hardware functions.
Requires either DRM master or CAP_SYS_ADMIN to run.
If you don't understand what all of that means, you probably don't want this. Look at x11grab instead.
Options
device
DRM device to capture on. Defaults to /dev/dri/card0.
format
Pixel format of the framebuffer. Defaults to bgr0.
format_modifier
Format modifier to signal on output frames. This is necessary to import correctly into some APIs,
but can't be autodetected. See the libdrm documentation for possible values.
crtc_id
KMS CRTC ID to define the capture source. The first active plane on the given CRTC will be used.
plane_id
KMS plane ID to define the capture source. Defaults to the first active plane found if neither
crtc_id nor plane_id are specified.
framerate
Framerate to capture at. This is not synchronised to any page flipping or framebuffer changes - it
just defines the interval at which the framebuffer is sampled. Sampling faster than the framebuffer
update rate will generate independent frames with the same content. Defaults to 30.
Examples
• Capture from the first active plane, download the result to normal frames and encode. This will only
work if the framebuffer is both linear and mappable - if not, the result may be scrambled or fail to
download.
ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4
• Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode as H.264.
ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4
libndi_newtek
The libndi_newtek input device provides capture capabilities for using NDI (Network Device Interface,
standard created by NewTek).
Input filename is a NDI source name that could be found by sending -find_sources 1 to command line - it
has no specific syntax but human-readable formatted.
To enable this input device, you need the NDI SDK and you need to configure with the appropriate
"--extra-cflags" and "--extra-ldflags".
Options
find_sources
If set to true, print a list of found/available NDI sources and exit. Defaults to false.
wait_sources
Override time to wait until the number of online sources have changed. Defaults to 0.5.
allow_video_fields
When this flag is false, all video that you receive will be progressive. Defaults to true.
Examples
• List input devices:
ffmpeg -f libndi_newtek -find_sources 1 -i dummy
• Restream to NDI:
ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2
dshow
Windows DirectShow input device.
DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. Currently only audio and
video devices are supported.
Multiple devices may be opened as separate inputs, but they may also be opened on the same input, which
should improve synchronism between them.
The input name should be in the format:
<TYPE>=<NAME>[:<TYPE>=<NAME>]
where TYPE can be either audio or video, and NAME is the device's name or alternative name..
Options
If no options are specified, the device's defaults are used. If the device does not support the
requested options, it will fail to open.
video_size
Set the video size in the captured video.
framerate
Set the frame rate in the captured video.
sample_rate
Set the sample rate (in Hz) of the captured audio.
sample_size
Set the sample size (in bits) of the captured audio.
channels
Set the number of channels in the captured audio.
list_devices
If set to true, print a list of devices and exit.
list_options
If set to true, print a list of selected device's options and exit.
video_device_number
Set video device number for devices with the same name (starts at 0, defaults to 0).
audio_device_number
Set audio device number for devices with the same name (starts at 0, defaults to 0).
pixel_format
Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or
set to rawvideo.
audio_buffer_size
Set audio device buffer size in milliseconds (which can directly impact latency, depending on the
device). Defaults to using the audio device's default buffer size (typically some multiple of
500ms). Setting this value too low can degrade performance. See also
<http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>
video_pin_name
Select video capture pin to use by name or alternative name.
audio_pin_name
Select audio capture pin to use by name or alternative name.
crossbar_video_input_pin_number
Select video input pin number for crossbar device. This will be routed to the crossbar device's Video
Decoder output pin. Note that changing this value can affect future invocations (sets a new default)
until system reboot occurs.
crossbar_audio_input_pin_number
Select audio input pin number for crossbar device. This will be routed to the crossbar device's Audio
Decoder output pin. Note that changing this value can affect future invocations (sets a new default)
until system reboot occurs.
show_video_device_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
change video filter properties and configurations manually. Note that for crossbar devices,
adjusting values in this dialog may be needed at times to toggle between PAL (25 fps) and NTSC
(29.97) input frame rates, sizes, interlacing, etc. Changing these values can enable different scan
rates/frame rates and avoiding green bars at the bottom, flickering scan lines, etc. Note that with
some devices, changing these properties can also affect future invocations (sets new defaults) until
system reboot occurs.
show_audio_device_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
change audio filter properties and configurations manually.
show_video_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
manually modify crossbar pin routings, when it opens a video device.
show_audio_crossbar_connection_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
manually modify crossbar pin routings, when it opens an audio device.
show_analog_tv_tuner_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
manually modify TV channels and frequencies.
show_analog_tv_tuner_audio_dialog
If set to true, before capture starts, popup a display dialog to the end user, allowing them to
manually modify TV audio (like mono vs. stereo, Language A,B or C).
audio_device_load
Load an audio capture filter device from file instead of searching it by name. It may load additional
parameters too, if the filter supports the serialization of its properties to. To use this an audio
capture source has to be specified, but it can be anything even fake one.
audio_device_save
Save the currently used audio capture filter device and its parameters (if the filter supports it) to
a file. If a file with the same name exists it will be overwritten.
video_device_load
Load a video capture filter device from file instead of searching it by name. It may load additional
parameters too, if the filter supports the serialization of its properties to. To use this a video
capture source has to be specified, but it can be anything even fake one.
video_device_save
Save the currently used video capture filter device and its parameters (if the filter supports it) to
a file. If a file with the same name exists it will be overwritten.
Examples
• Print the list of DirectShow supported devices and exit:
$ ffmpeg -list_devices true -f dshow -i dummy
• Open video device Camera:
$ ffmpeg -f dshow -i video="Camera"
• Open second video device with name Camera:
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera"
• Open video device Camera and audio device Microphone:
$ ffmpeg -f dshow -i video="Camera":audio="Microphone"
• Print the list of supported options in selected device and exit:
$ ffmpeg -list_options true -f dshow -i video="Camera"
• Specify pin names to capture by name or alternative name, specify alternative device name:
$ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"
• Configure a crossbar device, specifying crossbar pins, allow user to adjust video capture properties
at startup:
$ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"
fbdev
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer
monitor, typically on the console. It is accessed through a file device node, usually /dev/fb0.
For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source
tree.
See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).
To record from the framebuffer device /dev/fb0 with ffmpeg:
ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi
You can take a single screenshot image with the command:
ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg
Options
framerate
Set the frame rate. Default is 25.
gdigrab
Win32 GDI-based screen capture device.
This device allows you to capture a region of the display on Windows.
There are two options for the input filename:
desktop
or
title=<window_title>
The first option will capture the entire desktop, or a fixed region of the desktop. The second option
will instead capture the contents of a single window, regardless of its position on the screen.
For example, to grab the entire desktop using ffmpeg:
ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg
Grab a 640x480 region at position "10,20":
ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg
Grab the contents of the window named "Calculator"
ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg
Options
draw_mouse
Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer. Default value is
1.
framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of "30000/1001".
show_region
Show grabbed region on screen.
If show_region is specified with 1, then the grabbing region will be indicated on screen. With this
option, it is easy to know what is being grabbed if only a portion of the screen is grabbed.
Note that show_region is incompatible with grabbing the contents of a single window.
For example:
ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg
video_size
Set the video frame size. The default is to capture the full screen if desktop is selected, or the
full window size if title=window_title is selected.
offset_x
When capturing a region with video_size, set the distance from the left edge of the screen or
desktop.
Note that the offset calculation is from the top left corner of the primary monitor on Windows. If
you have a monitor positioned to the left of your primary monitor, you will need to use a negative
offset_x value to move the region to that monitor.
offset_y
When capturing a region with video_size, set the distance from the top edge of the screen or desktop.
Note that the offset calculation is from the top left corner of the primary monitor on Windows. If
you have a monitor positioned above your primary monitor, you will need to use a negative offset_y
value to move the region to that monitor.
iec61883
FireWire DV/HDV input device using libiec61883.
To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on your system.
Use the configure option "--enable-libiec61883" to compile with the device enabled.
The iec61883 capture device supports capturing from a video device connected via IEEE1394 (FireWire),
using libiec61883 and the new Linux FireWire stack (juju). This is the default DV/HDV input method in
Linux Kernel 2.6.37 and later, since the old FireWire stack was removed.
Specify the FireWire port to be used as input file, or "auto" to choose the first port connected.
Options
dvtype
Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if
usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa)
will not work and result in undefined behavior. The values auto, dv and hdv are supported.
dvbuffer
Set maximum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it
is not frame exact, since HDV does not have a fixed frame size.
dvguid
Select the capture device by specifying its GUID. Capturing will only be performed from the specified
device and fails if no device with the given GUID is found. This is useful to select the input if
multiple devices are connected at the same time. Look at /sys/bus/firewire/devices to find out the
GUIDs.
Examples
• Grab and show the input of a FireWire DV/HDV device.
ffplay -f iec61883 -i auto
• Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the
source is HDV.
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg
jack
JACK input device.
To enable this input device during configuration you need libjack installed on your system.
A JACK input device creates one or more JACK writable clients, one for each audio channel, with name
client_name:input_N, where client_name is the name provided by the application, and N is a number which
identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.
Once you have created one or more JACK readable clients, you need to connect them to one or more JACK
writable clients.
To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect programs, or do it
through a graphical interface, for example with qjackctl.
To list the JACK clients and their properties you can invoke the command jack_lsp.
Follows an example which shows how to capture a JACK readable client with ffmpeg.
# Create a JACK writable client with name "ffmpeg".
$ ffmpeg -f jack -i ffmpeg -y out.wav
# Start the sample jack_metro readable client.
$ jack_metro -b 120 -d 0.2 -f 4000
# List the current JACK clients.
$ jack_lsp -c
system:capture_1
system:capture_2
system:playback_1
system:playback_2
ffmpeg:input_1
metro:120_bpm
# Connect metro to the ffmpeg writable client.
$ jack_connect metro:120_bpm ffmpeg:input_1
For more information read: <http://jackaudio.org/>
Options
channels
Set the number of channels. Default is 2.
lavfi
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter filtergraph.
For each filtergraph open output, the input device will create a corresponding stream which is mapped to
the generated output. Currently only video data is supported. The filtergraph is specified through the
option graph.
Options
graph
Specify the filtergraph to use as input. Each video open output must be labelled by a unique string
of the form "outN", where N is a number starting from 0 corresponding to the mapped input stream
generated by the device. The first unlabelled output is automatically assigned to the "out0" label,
but all the others need to be specified explicitly.
The suffix "+subcc" can be appended to the output label to create an extra stream with the closed
captions packets attached to that output (experimental; only for EIA-608 / CEA-708 for now). The
subcc streams are created after all the normal streams, in the order of the corresponding stream.
For example, if there is "out19+subcc", "out7+subcc" and up to "out42", the stream #43 is subcc for
stream #7 and stream #44 is subcc for stream #19.
If not specified defaults to the filename specified for the input device.
graph_file
Set the filename of the filtergraph to be read and sent to the other filters. Syntax of the
filtergraph is the same as the one specified by the option graph.
dumpgraph
Dump graph to stderr.
Examples
• Create a color video stream and play it back with ffplay:
ffplay -f lavfi -graph "color=c=pink [out0]" dummy
• As the previous example, but use filename for specifying the graph description, and omit the "out0"
label:
ffplay -f lavfi color=c=pink
• Create three different video test filtered sources and play them:
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3
• Read an audio stream from a file using the amovie source and play it back with ffplay:
ffplay -f lavfi "amovie=test.wav"
• Read an audio stream and a video stream and play it back with ffplay:
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"
• Dump decoded frames to images and closed captions to a file (experimental):
ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin
libcdio
Audio-CD input device based on libcdio.
To enable this input device during configuration you need libcdio installed on your system. It requires
the configure option "--enable-libcdio".
This device allows playing and grabbing from an Audio-CD.
For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you may run the command:
ffmpeg -f libcdio -i /dev/sr0 cd.wav
Options
speed
Set drive reading speed. Default value is 0.
The speed is specified CD-ROM speed units. The speed is set through the libcdio
"cdio_cddap_speed_set" function. On many CD-ROM drives, specifying a value too large will result in
using the fastest speed.
paranoia_mode
Set paranoia recovery mode flags. It accepts one of the following values:
disable
verify
overlap
neverskip
full
Default value is disable.
For more information about the available recovery modes, consult the paranoia project documentation.
libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.
Requires the configure option "--enable-libdc1394".
openal
The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL headers and libraries installed on your
system, and need to configure FFmpeg with "--enable-openal".
OpenAL headers and libraries should be provided as part of your OpenAL implementation, or as an
additional download (an SDK). Depending on your installation you may need to specify additional flags via
the "--extra-cflags" and "--extra-ldflags" for allowing the build system to locate the OpenAL headers and
libraries.
An incomplete list of OpenAL implementations follows:
Creative
The official Windows implementation, providing hardware acceleration with supported devices and
software fallback. See <http://openal.org/>.
OpenAL Soft
Portable, open source (LGPL) software implementation. Includes backends for the most common sound
APIs on the Windows, Linux, Solaris, and BSD operating systems. See
<http://kcat.strangesoft.net/openal.html>.
Apple
OpenAL is part of Core Audio, the official Mac OS X Audio interface. See
<http://developer.apple.com/technologies/mac/audio-and-video.html>
This device allows one to capture from an audio input device handled through OpenAL.
You need to specify the name of the device to capture in the provided filename. If the empty string is
provided, the device will automatically select the default device. You can get the list of the supported
devices by using the option list_devices.
Options
channels
Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are
currently supported. Defaults to 2.
sample_size
Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently
supported. Defaults to 16.
sample_rate
Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.
list_devices
If set to true, print a list of devices and exit. Defaults to false.
Examples
Print the list of OpenAL supported devices and exit:
$ ffmpeg -list_devices true -f openal -i dummy out.ogg
Capture from the OpenAL device DR-BT101 via PulseAudio:
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg
Capture from the default device (note the empty string '' as filename):
$ ffmpeg -f openal -i '' out.ogg
Capture from two devices simultaneously, writing to two different files, within the same ffmpeg command:
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg
Note: not all OpenAL implementations support multiple simultaneous capture - try the latest OpenAL Soft
if the above does not work.
oss
Open Sound System input device.
The filename to provide to the input device is the device node representing the OSS input device, and is
usually set to /dev/dsp.
For example to grab from /dev/dsp using ffmpeg use the command:
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav
For more information about OSS see: <http://manuals.opensound.com/usersguide/dsp.html>
Options
sample_rate
Set the sample rate in Hz. Default is 48000.
channels
Set the number of channels. Default is 2.
pulse
PulseAudio input device.
To enable this output device you need to configure FFmpeg with "--enable-libpulse".
The filename to provide to the input device is a source device or the string "default"
To list the PulseAudio source devices and their properties you can invoke the command pactl list sources.
More information about PulseAudio can be found on <http://www.pulseaudio.org>.
Options
server
Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not
provided.
name
Specify the application name PulseAudio will use when showing active clients, by default it is the
"LIBAVFORMAT_IDENT" string.
stream_name
Specify the stream name PulseAudio will use when showing active streams, by default it is "record".
sample_rate
Specify the samplerate in Hz, by default 48kHz is used.
channels
Specify the channels in use, by default 2 (stereo) is set.
frame_size
Specify the number of bytes per frame, by default it is set to 1024.
fragment_size
Specify the minimal buffering fragment in PulseAudio, it will affect the audio latency. By default it
is unset.
wallclock
Set the initial PTS using the current time. Default is 1.
Examples
Record a stream from default device:
ffmpeg -f pulse -i default /tmp/pulse.wav
sndio
sndio input device.
To enable this input device during configuration you need libsndio installed on your system.
The filename to provide to the input device is the device node representing the sndio input device, and
is usually set to /dev/audio0.
For example to grab from /dev/audio0 using ffmpeg use the command:
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav
Options
sample_rate
Set the sample rate in Hz. Default is 48000.
channels
Set the number of channels. Default is 2.
video4linux2, v4l2
Video4Linux2 input video device.
"v4l2" can be used as alias for "video4linux2".
If FFmpeg is built with v4l-utils support (by using the "--enable-libv4l2" configure option), it is
possible to use it with the "-use_libv4l2" input device option.
The name of the device to grab is a file device node, usually Linux systems tend to automatically create
such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind
/dev/videoN, where N is a number associated to the device.
Video4Linux2 devices usually support a limited set of widthxheight sizes and frame rates. You can check
which are supported using -list_formats all for Video4Linux2 devices. Some devices, like TV cards,
support one or more standards. It is possible to list all the supported standards using -list_standards
all.
The time base for the timestamps is 1 microsecond. Depending on the kernel version and configuration, the
timestamps may be derived from the real time clock (origin at the Unix Epoch) or the monotonic clock
(origin usually at boot time, unaffected by NTP or manual changes to the clock). The -timestamps abs or
-ts abs option can be used to force conversion into the real time clock.
Some usage examples of the video4linux2 device with ffmpeg and ffplay:
• List supported formats for a video4linux2 device:
ffplay -f video4linux2 -list_formats all /dev/video0
• Grab and show the input of a video4linux2 device:
ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0
• Grab and record the input of a video4linux2 device, leave the frame rate and size as previously set:
ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg
For more information about Video4Linux, check <http://linuxtv.org/>.
Options
standard
Set the standard. Must be the name of a supported standard. To get a list of the supported standards,
use the list_standards option.
channel
Set the input channel number. Default to -1, which means using the previously selected channel.
video_size
Set the video frame size. The argument must be a string in the form WIDTHxHEIGHT or a valid size
abbreviation.
pixel_format
Select the pixel format (only valid for raw video input).
input_format
Set the preferred pixel format (for raw video) or a codec name. This option allows one to select the
input format, when several are available.
framerate
Set the preferred video frame rate.
list_formats
List available formats (supported pixel formats, codecs, and frame sizes) and exit.
Available values are:
all Show all available (compressed and non-compressed) formats.
raw Show only raw video (non-compressed) formats.
compressed
Show only compressed formats.
list_standards
List supported standards and exit.
Available values are:
all Show all supported standards.
timestamps, ts
Set type of timestamps for grabbed frames.
Available values are:
default
Use timestamps from the kernel.
abs Use absolute timestamps (wall clock).
mono2abs
Force conversion from monotonic to absolute timestamps.
Default value is "default".
use_libv4l2
Use libv4l2 (v4l-utils) conversion functions. Default is 0.
vfwcap
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from 0 to 9. You may use "list" as
filename to print a list of drivers. Any other filename will be interpreted as device number 0.
Options
video_size
Set the video frame size.
framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of "30000/1001".
x11grab
X11 video input device.
To enable this input device during configuration you need libxcb installed on your system. It will be
automatically detected during configuration.
This device allows one to capture a region of an X11 display.
The filename passed as input has the syntax:
[<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]
hostname:display_number.screen_number specifies the X11 display name of the screen to grab from. hostname
can be omitted, and defaults to "localhost". The environment variable DISPLAY contains the default
display name.
x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the
X11 screen. They default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the xdpyinfo program for getting basic information about the properties of your X11 display (e.g.
grep for "name" or "dimensions").
For example to grab from :0.0 using ffmpeg:
ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg
Grab at position "10,20":
ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
Options
draw_mouse
Specify whether to draw the mouse pointer. A value of 0 specifies not to draw the pointer. Default
value is 1.
follow_mouse
Make the grabbed area follow the mouse. The argument can be "centered" or a number of pixels PIXELS.
When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the
pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches
within PIXELS (greater than zero) to the edge of region.
For example:
ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg
To follow only when the mouse pointer reaches within 100 pixels to edge:
ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg
framerate
Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of "30000/1001".
show_region
Show grabbed region on screen.
If show_region is specified with 1, then the grabbing region will be indicated on screen. With this
option, it is easy to know what is being grabbed if only a portion of the screen is grabbed.
region_border
Set the region border thickness if -show_region 1 is used. Range is 1 to 128 and default is 3 (XCB-
based x11grab only).
For example:
ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg
With follow_mouse:
ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg
video_size
Set the video frame size. Default value is "vga".
grab_x
grab_y
Set the grabbing region coordinates. They are expressed as offset from the top left corner of the X11
window and correspond to the x_offset and y_offset parameters in the device name. The default value
for both options is 0.
OUTPUT DEVICES
Output devices are configured elements in FFmpeg that can write multimedia data to an output device
attached to your system.
When you configure your FFmpeg build, all the supported output devices are enabled by default. You can
list all available ones using the configure option "--list-outdevs".
You can disable all the output devices using the configure option "--disable-outdevs", and selectively
enable an output device using the option "--enable-outdev=OUTDEV", or you can disable a particular input
device using the option "--disable-outdev=OUTDEV".
The option "-devices" of the ff* tools will display the list of enabled output devices.
A description of the currently available output devices follows.
alsa
ALSA (Advanced Linux Sound Architecture) output device.
Examples
• Play a file on default ALSA device:
ffmpeg -i INPUT -f alsa default
• Play a file on soundcard 1, audio device 7:
ffmpeg -i INPUT -f alsa hw:1,7
caca
CACA output device.
This output device allows one to show a video stream in CACA window. Only one CACA window is allowed per
application, so you can have only one instance of this output device in an application.
To enable this output device you need to configure FFmpeg with "--enable-libcaca". libcaca is a graphics
library that outputs text instead of pixels.
For more information about libcaca, check: <http://caca.zoy.org/wiki/libcaca>
Options
window_title
Set the CACA window title, if not specified default to the filename specified for the output device.
window_size
Set the CACA window size, can be a string of the form widthxheight or a video size abbreviation. If
not specified it defaults to the size of the input video.
driver
Set display driver.
algorithm
Set dithering algorithm. Dithering is necessary because the picture being rendered has usually far
more colours than the available palette. The accepted values are listed with "-list_dither
algorithms".
antialias
Set antialias method. Antialiasing smoothens the rendered image and avoids the commonly seen
staircase effect. The accepted values are listed with "-list_dither antialiases".
charset
Set which characters are going to be used when rendering text. The accepted values are listed with
"-list_dither charsets".
color
Set color to be used when rendering text. The accepted values are listed with "-list_dither colors".
list_drivers
If set to true, print a list of available drivers and exit.
list_dither
List available dither options related to the argument. The argument must be one of "algorithms",
"antialiases", "charsets", "colors".
Examples
• The following command shows the ffmpeg output is an CACA window, forcing its size to 80x25:
ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -
• Show the list of available drivers and exit:
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -
• Show the list of available dither colors and exit:
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -
decklink
The decklink output device provides playback capabilities for Blackmagic DeckLink devices.
To enable this output device, you need the Blackmagic DeckLink SDK and you need to configure with the
appropriate "--extra-cflags" and "--extra-ldflags". On Windows, you need to run the IDL files through
widl.
DeckLink is very picky about the formats it supports. Pixel format is always uyvy422, framerate, field
order and video size must be determined for your device with -list_formats 1. Audio sample rate is always
48 kHz.
Options
list_devices
If set to true, print a list of devices and exit. Defaults to false.
list_formats
If set to true, print a list of supported formats and exit. Defaults to false.
preroll
Amount of time to preroll video in seconds. Defaults to 0.5.
Examples
• List output devices:
ffmpeg -i test.avi -f decklink -list_devices 1 dummy
• List supported formats:
ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'
• Play video clip:
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'
• Play video clip with non-standard framerate or video size:
ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'
libndi_newtek
The libndi_newtek output device provides playback capabilities for using NDI (Network Device Interface,
standard created by NewTek).
Output filename is a NDI name.
To enable this output device, you need the NDI SDK and you need to configure with the appropriate
"--extra-cflags" and "--extra-ldflags".
NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and rgb0.
Options
reference_level
The audio reference level in dB. This specifies how many dB above the reference level (+4dBU) is the
full range of 16 bit audio. Defaults to 0.
clock_video
These specify whether video "clock" themselves. Defaults to false.
clock_audio
These specify whether audio "clock" themselves. Defaults to false.
Examples
• Play video clip:
ffmpeg -i "udp://@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1
fbdev
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.
Options
xoffset
yoffset
Set x/y coordinate of top left corner. Default is 0.
Examples
Play a file on framebuffer device /dev/fb0. Required pixel format depends on current framebuffer
settings.
ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0
See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).
opengl
OpenGL output device.
To enable this output device you need to configure FFmpeg with "--enable-opengl".
This output device allows one to render to OpenGL context. Context may be provided by application or
default SDL window is created.
When device renders to external context, application must implement handlers for following messages:
"AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" - create OpenGL context on current thread.
"AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
"AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers. "AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER" - destroy
OpenGL context. Application is also required to inform a device about current resolution by sending
"AV_APP_TO_DEV_WINDOW_SIZE" message.
Options
background
Set background color. Black is a default.
no_window
Disables default SDL window when set to non-zero value. Application must provide OpenGL context and
both "window_size_cb" and "window_swap_buffers_cb" callbacks when set.
window_title
Set the SDL window title, if not specified default to the filename specified for the output device.
Ignored when no_window is set.
window_size
Set preferred window size, can be a string of the form widthxheight or a video size abbreviation. If
not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
Mostly usable when no_window is not set.
Examples
Play a file on SDL window using OpenGL rendering:
ffmpeg -i INPUT -f opengl "window title"
oss
OSS (Open Sound System) output device.
pulse
PulseAudio output device.
To enable this output device you need to configure FFmpeg with "--enable-libpulse".
More information about PulseAudio can be found on <http://www.pulseaudio.org>
Options
server
Connect to a specific PulseAudio server, specified by an IP address. Default server is used when not
provided.
name
Specify the application name PulseAudio will use when showing active clients, by default it is the
"LIBAVFORMAT_IDENT" string.
stream_name
Specify the stream name PulseAudio will use when showing active streams, by default it is set to the
specified output name.
device
Specify the device to use. Default device is used when not provided. List of output devices can be
obtained with command pactl list sinks.
buffer_size
buffer_duration
Control the size and duration of the PulseAudio buffer. A small buffer gives more control, but
requires more frequent updates.
buffer_size specifies size in bytes while buffer_duration specifies duration in milliseconds.
When both options are provided then the highest value is used (duration is recalculated to bytes
using stream parameters). If they are set to 0 (which is default), the device will use the default
PulseAudio duration value. By default PulseAudio set buffer duration to around 2 seconds.
prebuf
Specify pre-buffering size in bytes. The server does not start with playback before at least prebuf
bytes are available in the buffer. By default this option is initialized to the same value as
buffer_size or buffer_duration (whichever is bigger).
minreq
Specify minimum request size in bytes. The server does not request less than minreq bytes from the
client, instead waits until the buffer is free enough to request more bytes at once. It is
recommended to not set this option, which will initialize this to a value that is deemed sensible by
the server.
Examples
Play a file on default device on default server:
ffmpeg -i INPUT -f pulse "stream name"
sdl
SDL (Simple DirectMedia Layer) output device.
This output device allows one to show a video stream in an SDL window. Only one SDL window is allowed per
application, so you can have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system when configuring your build.
For more information about SDL, check: <http://www.libsdl.org/>
Options
window_title
Set the SDL window title, if not specified default to the filename specified for the output device.
icon_title
Set the name of the iconified SDL window, if not specified it is set to the same value of
window_title.
window_size
Set the SDL window size, can be a string of the form widthxheight or a video size abbreviation. If
not specified it defaults to the size of the input video, downscaled according to the aspect ratio.
window_fullscreen
Set fullscreen mode when non-zero value is provided. Default value is zero.
Interactive commands
The window created by the device can be controlled through the following interactive commands.
q, ESC
Quit the device immediately.
Examples
The following command shows the ffmpeg output is an SDL window, forcing its size to the qcif format:
ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"
sndio
sndio audio output device.
xv
XV (XVideo) output device.
This output device allows one to show a video stream in a X Window System window.
Options
display_name
Specify the hardware display name, which determines the display and communications domain to be used.
The display name or DISPLAY environment variable can be a string in the format
hostname[:number[.screen_number]].
hostname specifies the name of the host machine on which the display is physically attached. number
specifies the number of the display server on that host machine. screen_number specifies the screen
to be used on that server.
If unspecified, it defaults to the value of the DISPLAY environment variable.
For example, "dual-headed:0.1" would specify screen 1 of display 0 on the machine named
``dual-headed''.
Check the X11 specification for more detailed information about the display name format.
window_id
When set to non-zero value then device doesn't create new window, but uses existing one with provided
window_id. By default this options is set to zero and device creates its own window.
window_size
Set the created window size, can be a string of the form widthxheight or a video size abbreviation.
If not specified it defaults to the size of the input video. Ignored when window_id is set.
window_x
window_y
Set the X and Y window offsets for the created window. They are both set to 0 by default. The values
may be ignored by the window manager. Ignored when window_id is set.
window_title
Set the window title, if not specified default to the filename specified for the output device.
Ignored when window_id is set.
For more information about XVideo see <http://www.x.org/>.
Examples
• Decode, display and encode video input with ffmpeg at the same time:
ffmpeg -i INPUT OUTPUT -f xv display
• Decode and display the input video to multiple X11 windows:
ffmpeg -i INPUT -f xv normal -vf negate -f xv negated
RESAMPLER OPTIONS
The audio resampler supports the following named options.
Options may be set by specifying -option value in the FFmpeg tools, option=value for the aresample
filter, by setting the value explicitly in the "SwrContext" options or using the libavutil/opt.h API for
programmatic use.
ich, in_channel_count
Set the number of input channels. Default value is 0. Setting this value is not mandatory if the
corresponding channel layout in_channel_layout is set.
och, out_channel_count
Set the number of output channels. Default value is 0. Setting this value is not mandatory if the
corresponding channel layout out_channel_layout is set.
uch, used_channel_count
Set the number of used input channels. Default value is 0. This option is only used for special
remapping.
isr, in_sample_rate
Set the input sample rate. Default value is 0.
osr, out_sample_rate
Set the output sample rate. Default value is 0.
isf, in_sample_fmt
Specify the input sample format. It is set by default to "none".
osf, out_sample_fmt
Specify the output sample format. It is set by default to "none".
tsf, internal_sample_fmt
Set the internal sample format. Default value is "none". This will automatically be chosen when it
is not explicitly set.
icl, in_channel_layout
ocl, out_channel_layout
Set the input/output channel layout.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
clev, center_mix_level
Set the center mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
slev, surround_mix_level
Set the surround mix level. It is a value expressed in deciBel, and must be in the interval [-32,32].
lfe_mix_level
Set LFE mix into non LFE level. It is used when there is a LFE input but no LFE output. It is a value
expressed in deciBel, and must be in the interval [-32,32].
rmvol, rematrix_volume
Set rematrix volume. Default value is 1.0.
rematrix_maxval
Set maximum output value for rematrixing. This can be used to prevent clipping vs. preventing volume
reduction. A value of 1.0 prevents clipping.
flags, swr_flags
Set flags used by the converter. Default value is 0.
It supports the following individual flags:
res force resampling, this flag forces resampling to be used even when the input and output sample
rates match.
dither_scale
Set the dither scale. Default value is 1.
dither_method
Set dither method. Default value is 0.
Supported values:
rectangular
select rectangular dither
triangular
select triangular dither
triangular_hp
select triangular dither with high pass
lipshitz
select Lipshitz noise shaping dither.
shibata
select Shibata noise shaping dither.
low_shibata
select low Shibata noise shaping dither.
high_shibata
select high Shibata noise shaping dither.
f_weighted
select f-weighted noise shaping dither
modified_e_weighted
select modified-e-weighted noise shaping dither
improved_e_weighted
select improved-e-weighted noise shaping dither
resampler
Set resampling engine. Default value is swr.
Supported values:
swr select the native SW Resampler; filter options precision and cheby are not applicable in this
case.
soxr
select the SoX Resampler (where available); compensation, and filter options filter_size,
phase_shift, exact_rational, filter_type & kaiser_beta, are not applicable in this case.
filter_size
For swr only, set resampling filter size, default value is 32.
phase_shift
For swr only, set resampling phase shift, default value is 10, and must be in the interval [0,30].
linear_interp
Use linear interpolation when enabled (the default). Disable it if you want to preserve speed instead
of quality when exact_rational fails.
exact_rational
For swr only, when enabled, try to use exact phase_count based on input and output sample rate.
However, if it is larger than "1 << phase_shift", the phase_count will be "1 << phase_shift" as
fallback. Default is enabled.
cutoff
Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float value between 0 and 1.
Default value is 0.97 with swr, and 0.91 with soxr (which, with a sample-rate of 44100, preserves the
entire audio band to 20kHz).
precision
For soxr only, the precision in bits to which the resampled signal will be calculated. The default
value of 20 (which, with suitable dithering, is appropriate for a destination bit-depth of 16) gives
SoX's 'High Quality'; a value of 28 gives SoX's 'Very High Quality'.
cheby
For soxr only, selects passband rolloff none (Chebyshev) & higher-precision approximation for
'irrational' ratios. Default value is 0.
async
For swr only, simple 1 parameter audio sync to timestamps using stretching, squeezing, filling and
trimming. Setting this to 1 will enable filling and trimming, larger values represent the maximum
amount in samples that the data may be stretched or squeezed for each second. Default value is 0,
thus no compensation is applied to make the samples match the audio timestamps.
first_pts
For swr only, assume the first pts should be this value. The time unit is 1 / sample rate. This
allows for padding/trimming at the start of stream. By default, no assumption is made about the first
frame's expected pts, so no padding or trimming is done. For example, this could be set to 0 to pad
the beginning with silence if an audio stream starts after the video stream or to trim any samples
with a negative pts due to encoder delay.
min_comp
For swr only, set the minimum difference between timestamps and audio data (in seconds) to trigger
stretching/squeezing/filling or trimming of the data to make it match the timestamps. The default is
that stretching/squeezing/filling and trimming is disabled (min_comp = "FLT_MAX").
min_hard_comp
For swr only, set the minimum difference between timestamps and audio data (in seconds) to trigger
adding/dropping samples to make it match the timestamps. This option effectively is a threshold to
select between hard (trim/fill) and soft (squeeze/stretch) compensation. Note that all compensation
is by default disabled through min_comp. The default is 0.1.
comp_duration
For swr only, set duration (in seconds) over which data is stretched/squeezed to make it match the
timestamps. Must be a non-negative double float value, default value is 1.0.
max_soft_comp
For swr only, set maximum factor by which data is stretched/squeezed to make it match the timestamps.
Must be a non-negative double float value, default value is 0.
matrix_encoding
Select matrixed stereo encoding.
It accepts the following values:
none
select none
dolby
select Dolby
dplii
select Dolby Pro Logic II
Default value is "none".
filter_type
For swr only, select resampling filter type. This only affects resampling operations.
It accepts the following values:
cubic
select cubic
blackman_nuttall
select Blackman Nuttall windowed sinc
kaiser
select Kaiser windowed sinc
kaiser_beta
For swr only, set Kaiser window beta value. Must be a double float value in the interval [2,16],
default value is 9.
output_sample_bits
For swr only, set number of used output sample bits for dithering. Must be an integer in the interval
[0,64], default value is 0, which means it's not used.
SCALER OPTIONS
The video scaler supports the following named options.
Options may be set by specifying -option value in the FFmpeg tools. For programmatic use, they can be set
explicitly in the "SwsContext" options or through the libavutil/opt.h API.
sws_flags
Set the scaler flags. This is also used to set the scaling algorithm. Only a single algorithm should
be selected. Default value is bicubic.
It accepts the following values:
fast_bilinear
Select fast bilinear scaling algorithm.
bilinear
Select bilinear scaling algorithm.
bicubic
Select bicubic scaling algorithm.
experimental
Select experimental scaling algorithm.
neighbor
Select nearest neighbor rescaling algorithm.
area
Select averaging area rescaling algorithm.
bicublin
Select bicubic scaling algorithm for the luma component, bilinear for chroma components.
gauss
Select Gaussian rescaling algorithm.
sinc
Select sinc rescaling algorithm.
lanczos
Select Lanczos rescaling algorithm.
spline
Select natural bicubic spline rescaling algorithm.
print_info
Enable printing/debug logging.
accurate_rnd
Enable accurate rounding.
full_chroma_int
Enable full chroma interpolation.
full_chroma_inp
Select full chroma input.
bitexact
Enable bitexact output.
srcw
Set source width.
srch
Set source height.
dstw
Set destination width.
dsth
Set destination height.
src_format
Set source pixel format (must be expressed as an integer).
dst_format
Set destination pixel format (must be expressed as an integer).
src_range
Select source range.
dst_range
Select destination range.
param0, param1
Set scaling algorithm parameters. The specified values are specific of some scaling algorithms and
ignored by others. The specified values are floating point number values.
sws_dither
Set the dithering algorithm. Accepts one of the following values. Default value is auto.
auto
automatic choice
none
no dithering
bayer
bayer dither
ed error diffusion dither
a_dither
arithmetic dither, based using addition
x_dither
arithmetic dither, based using xor (more random/less apparent patterning that a_dither).
alphablend
Set the alpha blending to use when the input has alpha but the output does not. Default value is
none.
uniform_color
Blend onto a uniform background color
checkerboard
Blend onto a checkerboard
none
No blending
FILTERING INTRODUCTION
Filtering in FFmpeg is enabled through the libavfilter library.
In libavfilter, a filter can have multiple inputs and multiple outputs. To illustrate the sorts of
things that are possible, we consider the following filtergraph.
[main]
input --> split ---------------------> overlay --> output
| ^
|[tmp] [flip]|
+-----> crop --> vflip -------+
This filtergraph splits the input stream in two streams, then sends one stream through the crop filter
and the vflip filter, before merging it back with the other stream by overlaying it on top. You can use
the following command to achieve this:
ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT
The result will be that the top half of the video is mirrored onto the bottom half of the output video.
Filters in the same linear chain are separated by commas, and distinct linear chains of filters are
separated by semicolons. In our example, crop,vflip are in one linear chain, split and overlay are
separately in another. The points where the linear chains join are labelled by names enclosed in square
brackets. In the example, the split filter generates two outputs that are associated to the labels [main]
and [tmp].
The stream sent to the second output of split, labelled as [tmp], is processed through the crop filter,
which crops away the lower half part of the video, and then vertically flipped. The overlay filter takes
in input the first unchanged output of the split filter (which was labelled as [main]), and overlay on
its lower half the output generated by the crop,vflip filterchain.
Some filters take in input a list of parameters: they are specified after the filter name and an equal
sign, and are separated from each other by a colon.
There exist so-called source filters that do not have an audio/video input, and sink filters that will
not have audio/video output.
GRAPH
The graph2dot program included in the FFmpeg tools directory can be used to parse a filtergraph
description and issue a corresponding textual representation in the dot language.
Invoke the command:
graph2dot -h
to see how to use graph2dot.
You can then pass the dot description to the dot program (from the graphviz suite of programs) and obtain
a graphical representation of the filtergraph.
For example the sequence of commands:
echo <GRAPH_DESCRIPTION> | \
tools/graph2dot -o graph.tmp && \
dot -Tpng graph.tmp -o graph.png && \
display graph.png
can be used to create and display an image representing the graph described by the GRAPH_DESCRIPTION
string. Note that this string must be a complete self-contained graph, with its inputs and outputs
explicitly defined. For example if your command line is of the form:
ffmpeg -i infile -vf scale=640:360 outfile
your GRAPH_DESCRIPTION string will need to be of the form:
nullsrc,scale=640:360,nullsink
you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific
input file.
FILTERGRAPH DESCRIPTION
A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple
links between a pair of filters. Each link has one input pad on one side connecting it to one filter from
which it takes its input, and one output pad on the other side connecting it to one filter accepting its
output.
Each filter in a filtergraph is an instance of a filter class registered in the application, which
defines the features and the number of input and output pads of the filter.
A filter with no input pads is called a "source", and a filter with no output pads is called a "sink".
Filtergraph syntax
A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and
-filter_complex options in ffmpeg and -vf/-af in ffplay, and by the "avfilter_graph_parse_ptr()" function
defined in libavfilter/avfilter.h.
A filterchain consists of a sequence of connected filters, each one connected to the previous one in the
sequence. A filterchain is represented by a list of ","-separated filter descriptions.
A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list
of ";"-separated filterchain descriptions.
A filter is represented by a string of the form:
[in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]
filter_name is the name of the filter class of which the described filter is an instance of, and has to
be the name of one of the filter classes registered in the program optionally followed by "@id". The
name of the filter class is optionally followed by a string "=arguments".
arguments is a string which contains the parameters used to initialize the filter instance. It may have
one of two forms:
• A ':'-separated list of key=value pairs.
• A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order
they are declared. E.g. the "fade" filter declares three options in this order -- type, start_frame
and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option
type, 0 to start_frame and 30 to nb_frames.
• A ':'-separated list of mixed direct value and long key=value pairs. The direct value must precede
the key=value pairs, and follow the same constraints order of the previous point. The following
key=value pairs can be set in any preferred order.
If the option value itself is a list of items (e.g. the "format" filter takes a list of pixel formats),
the items in the list are usually separated by |.
The list of arguments can be quoted using the character ' as initial and ending mark, and the character \
for escaping the characters within the quoted text; otherwise the argument string is considered
terminated when the next special character (belonging to the set []=;,) is encountered.
The name and arguments of the filter are optionally preceded and followed by a list of link labels. A
link label allows one to name a link and associate it to a filter output or input pad. The preceding
labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1
... out_link_M, are associated to the output pads.
When two link labels with the same name are found in the filtergraph, a link between the corresponding
input and output pad is created.
If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next
filter in the filterchain. For example in the filterchain
nullsrc, split[L1], [L2]overlay, nullsink
the split filter instance has two output pads, and the overlay filter instance two input pads. The first
output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second
output pad of split is linked to the second input pad of overlay, which are both unlabelled.
In a filter description, if the input label of the first filter is not specified, "in" is assumed; if the
output label of the last filter is not specified, "out" is assumed.
In a complete filterchain all the unlabelled filter input and output pads must be connected. A
filtergraph is considered valid if all the filter input and output pads of all the filterchains are
connected.
Libavfilter will automatically insert scale filters where format conversion is required. It is possible
to specify swscale flags for those automatically inserted scalers by prepending "sws_flags=flags;" to the
filtergraph description.
Here is a BNF description of the filtergraph syntax:
<NAME> ::= sequence of alphanumeric characters and '_'
<FILTER_NAME> ::= <NAME>["@"<NAME>]
<LINKLABEL> ::= "[" <NAME> "]"
<LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>]
<FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
<FILTER> ::= [<LINKLABELS>] <FILTER_NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
<FILTERCHAIN> ::= <FILTER> [,<FILTERCHAIN>]
<FILTERGRAPH> ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]
Notes on filtergraph escaping
Filtergraph description composition entails several levels of escaping. See the "Quoting and escaping"
section in the ffmpeg-utils(1) manual for more information about the employed escaping procedure.
A first level escaping affects the content of each filter option value, which may contain the special
character ":" used to separate values, or one of the escaping characters "\'".
A second level escaping affects the whole filter description, which may contain the escaping characters
"\'" or the special characters "[],;" used by the filtergraph description.
Finally, when you specify a filtergraph on a shell commandline, you need to perform a third level
escaping for the shell special characters contained within it.
For example, consider the following string to be embedded in the drawtext filter description text value:
this is a 'string': may contain one, or more, special characters
This string contains the "'" special escaping character, and the ":" special character, so it needs to be
escaped in this way:
text=this is a \'string\'\: may contain one, or more, special characters
A second level of escaping is required when embedding the filter description in a filtergraph
description, in order to escape all the filtergraph special characters. Thus the example above becomes:
drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters
(note that in addition to the "\'" escaping special characters, also "," needs to be escaped).
Finally an additional level of escaping is needed when writing the filtergraph description in a shell
command, which depends on the escaping rules of the adopted shell. For example, assuming that "\" is
special and needs to be escaped with another "\", the previous string will finally result in:
-vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"
TIMELINE EDITING
Some filters support a generic enable option. For the filters supporting timeline editing, this option
can be set to an expression which is evaluated before sending a frame to the filter. If the evaluation is
non-zero, the filter will be enabled, otherwise the frame will be sent unchanged to the next filter in
the filtergraph.
The expression accepts the following values:
t timestamp expressed in seconds, NAN if the input timestamp is unknown
n sequential number of the input frame, starting from 0
pos the position in the file of the input frame, NAN if unknown
w
h width and height of the input frame if video
Additionally, these filters support an enable command that can be used to re-define the expression.
Like any other filtering option, the enable option follows the same rules.
For example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a curves filter
starting at 3 seconds:
smartblur = enable='between(t,10,3*60)',
curves = enable='gte(t,3)' : preset=cross_process
See "ffmpeg -filters" to view which filters have timeline support.
OPTIONS FOR FILTERS WITH SEVERAL INPUTS
Some filters with several inputs support a common set of options. These options can only be set by name,
not with the short notation.
eof_action
The action to take when EOF is encountered on the secondary input; it accepts one of the following
values:
repeat
Repeat the last frame (the default).
endall
End both streams.
pass
Pass the main input through.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
repeatlast
If set to 1, force the filter to extend the last frame of secondary streams until the end of the
primary stream. A value of 0 disables this behavior. Default value is 1.
AUDIO FILTERS
When you configure your FFmpeg build, you can disable any of the existing filters using
"--disable-filters". The configure output will show the audio filters included in your build.
Below is a description of the currently available audio filters.
acompressor
A compressor is mainly used to reduce the dynamic range of a signal. Especially modern music is mostly
compressed at a high ratio to improve the overall loudness. It's done to get the highest attention of a
listener, "fatten" the sound and bring more "power" to the track. If a signal is compressed too much it
may sound dull or "dead" afterwards or it may start to "pump" (which could be a powerful effect but can
also destroy a track completely). The right compression is the key to reach a professional sound and is
the high art of mixing and mastering. Because of its complex settings it may take a long time to get the
right feeling for this kind of effect.
Compression is done by detecting the volume above a chosen level "threshold" and dividing it by the
factor set with "ratio". So if you set the threshold to -12dB and your signal reaches -6dB a ratio of
2:1 will result in a signal at -9dB. Because an exact manipulation of the signal would cause distortion
of the waveform the reduction can be levelled over the time. This is done by setting "Attack" and
"Release". "attack" determines how long the signal has to rise above the threshold before any reduction
will occur and "release" sets the time the signal has to fall below the threshold to reduce the reduction
again. Shorter signals than the chosen attack time will be left untouched. The overall reduction of the
signal can be made up afterwards with the "makeup" setting. So compressing the peaks of a signal about
6dB and raising the makeup to this level results in a signal twice as loud than the source. To gain a
softer entry in the compression the "knee" flattens the hard edge at the threshold in the range of the
chosen decibels.
The filter accepts the following options:
level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold
If a signal of stream rises above this level it will affect the gain reduction. By default it is
0.125. Range is between 0.00097563 and 1.
ratio
Set a ratio by which the signal is reduced. 1:2 means that if the level rose 4dB above the threshold,
it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction starts.
Default is 20. Range is between 0.01 and 2000.
release
Amount of milliseconds the signal has to fall below the threshold before reduction is decreased
again. Default is 250. Range is between 0.01 and 9000.
makeup
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1
to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843.
Range is between 1 and 8.
link
Choose if the "average" level between all channels of input stream or the louder("maximum") channel
of input stream affects the reduction. Default is "average".
detection
Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms"
which is mostly smoother.
mix How much to use compressed signal in output. Default is 1. Range is between 0 and 1.
acopy
Copy the input audio source unchanged to the output. This is mainly useful for testing purposes.
acrossfade
Apply cross fade from one input audio stream to another input audio stream. The cross fade is applied
for specified duration near the end of first stream.
The filter accepts the following options:
nb_samples, ns
Specify the number of samples for which the cross fade effect has to last. At the end of the cross
fade effect the first input audio will be completely silent. Default is 44100.
duration, d
Specify the duration of the cross fade effect. See the Time duration section in the ffmpeg-utils(1)
manual for the accepted syntax. By default the duration is determined by nb_samples. If set this
option is used instead of nb_samples.
overlap, o
Should first stream end overlap with second stream start. Default is enabled.
curve1
Set curve for cross fade transition for first stream.
curve2
Set curve for cross fade transition for second stream.
For description of available curve types see afade filter description.
Examples
• Cross fade from one input to another:
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac
• Cross fade from one input to another but without overlapping:
ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac
acrusher
Reduce audio bit resolution.
This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly reduce number of
bits an audio signal is sampled with. This doesn't change the bit depth at all, it just produces the
effect. Material reduced in bit depth sounds more harsh and "digital". This filter is able to even round
to continuous values instead of discrete bit depths. Additionally it has a D/C offset which results in
different crushing of the lower and the upper half of the signal. An Anti-Aliasing setting is able to
produce "softer" crushing sounds.
Another feature of this filter is the logarithmic mode. This setting switches from linear distances
between bits to logarithmic ones. The result is a much more "natural" sounding crusher which doesn't
gate low signals for example. The human ear has a logarithmic perception, too so this kind of crushing is
much more pleasant. Logarithmic crushing is also able to get anti-aliased.
The filter accepts the following options:
level_in
Set level in.
level_out
Set level out.
bits
Set bit reduction.
mix Set mixing amount.
mode
Can be linear: "lin" or logarithmic: "log".
dc Set DC.
aa Set anti-aliasing.
samples
Set sample reduction.
lfo Enable LFO. By default disabled.
lforange
Set LFO range.
lforate
Set LFO rate.
adelay
Delay one or more audio channels.
Samples in delayed channel are filled with silence.
The filter accepts the following option:
delays
Set list of delays in milliseconds for each channel separated by '|'. Unused delays will be silently
ignored. If number of given delays is smaller than number of channels all remaining channels will not
be delayed. If you want to delay exact number of samples, append 'S' to number.
Examples
• Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the second channel
(and any other channels that may be present) unchanged.
adelay=1500|0|500
• Delay second channel by 500 samples, the third channel by 700 samples and leave the first channel
(and any other channels that may be present) unchanged.
adelay=0|500S|700S
aecho
Apply echoing to the input audio.
Echoes are reflected sound and can occur naturally amongst mountains (and sometimes large buildings) when
talking or shouting; digital echo effects emulate this behaviour and are often used to help fill out the
sound of a single instrument or vocal. The time difference between the original signal and the reflection
is the "delay", and the loudness of the reflected signal is the "decay". Multiple echoes can have
different delays and decays.
A description of the accepted parameters follows.
in_gain
Set input gain of reflected signal. Default is 0.6.
out_gain
Set output gain of reflected signal. Default is 0.3.
delays
Set list of time intervals in milliseconds between original signal and reflections separated by '|'.
Allowed range for each "delay" is "(0 - 90000.0]". Default is 1000.
decays
Set list of loudness of reflected signals separated by '|'. Allowed range for each "decay" is "(0 -
1.0]". Default is 0.5.
Examples
• Make it sound as if there are twice as many instruments as are actually playing:
aecho=0.8:0.88:60:0.4
• If delay is very short, then it sound like a (metallic) robot playing music:
aecho=0.8:0.88:6:0.4
• A longer delay will sound like an open air concert in the mountains:
aecho=0.8:0.9:1000:0.3
• Same as above but with one more mountain:
aecho=0.8:0.9:1000|1800:0.3|0.25
aemphasis
Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs with different
filter curves. E.g. to store music on vinyl the signal has to be altered by a filter first to even out
the disadvantages of this recording medium. Once the material is played back the inverse filter has to
be applied to restore the distortion of the frequency response.
The filter accepts the following options:
level_in
Set input gain.
level_out
Set output gain.
mode
Set filter mode. For restoring material use "reproduction" mode, otherwise use "production" mode.
Default is "reproduction" mode.
type
Set filter type. Selects medium. Can be one of the following:
col select Columbia.
emi select EMI.
bsi select BSI (78RPM).
riaa
select RIAA.
cd select Compact Disc (CD).
50fm
select 50Xs (FM).
75fm
select 75Xs (FM).
50kf
select 50Xs (FM-KF).
75kf
select 75Xs (FM-KF).
aeval
Modify an audio signal according to the specified expressions.
This filter accepts one or more expressions (one for each channel), which are evaluated and used to
modify a corresponding audio signal.
It accepts the following parameters:
exprs
Set the '|'-separated expressions list for each separate channel. If the number of input channels is
greater than the number of expressions, the last specified expression is used for the remaining
output channels.
channel_layout, c
Set output channel layout. If not specified, the channel layout is specified by the number of
expressions. If set to same, it will use by default the same input channel layout.
Each expression in exprs can contain the following constants and functions:
ch channel number of the current expression
n number of the evaluated sample, starting from 0
s sample rate
t time of the evaluated sample expressed in seconds
nb_in_channels
nb_out_channels
input and output number of channels
val(CH)
the value of input channel with number CH
Note: this filter is slow. For faster processing you should use a dedicated filter.
Examples
• Half volume:
aeval=val(ch)/2:c=same
• Invert phase of the second channel:
aeval=val(0)|-val(1)
afade
Apply fade-in/out effect to input audio.
A description of the accepted parameters follows.
type, t
Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out effect. Default is
"in".
start_sample, ss
Specify the number of the start sample for starting to apply the fade effect. Default is 0.
nb_samples, ns
Specify the number of samples for which the fade effect has to last. At the end of the fade-in effect
the output audio will have the same volume as the input audio, at the end of the fade-out transition
the output audio will be silence. Default is 44100.
start_time, st
Specify the start time of the fade effect. Default is 0. The value must be specified as a time
duration; see the Time duration section in the ffmpeg-utils(1) manual for the accepted syntax. If
set this option is used instead of start_sample.
duration, d
Specify the duration of the fade effect. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax. At the end of the fade-in effect the output audio will have the same volume
as the input audio, at the end of the fade-out transition the output audio will be silence. By
default the duration is determined by nb_samples. If set this option is used instead of nb_samples.
curve
Set curve for fade transition.
It accepts the following values:
tri select triangular, linear slope (default)
qsin
select quarter of sine wave
hsin
select half of sine wave
esin
select exponential sine wave
log select logarithmic
ipar
select inverted parabola
qua select quadratic
cub select cubic
squ select square root
cbr select cubic root
par select parabola
exp select exponential
iqsin
select inverted quarter of sine wave
ihsin
select inverted half of sine wave
dese
select double-exponential seat
desi
select double-exponential sigmoid
Examples
• Fade in first 15 seconds of audio:
afade=t=in:ss=0:d=15
• Fade out last 25 seconds of a 900 seconds audio:
afade=t=out:st=875:d=25
afftfilt
Apply arbitrary expressions to samples in frequency domain.
real
Set frequency domain real expression for each separate channel separated by '|'. Default is "1". If
the number of input channels is greater than the number of expressions, the last specified expression
is used for the remaining output channels.
imag
Set frequency domain imaginary expression for each separate channel separated by '|'. If not set,
real option is used.
Each expression in real and imag can contain the following constants:
sr sample rate
b current frequency bin number
nb number of available bins
ch channel number of the current expression
chs number of channels
pts current frame pts
win_size
Set window size.
It accepts the following values:
w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536
Default is "w4096"
win_func
Set window function. Default is "hann".
overlap
Set window overlap. If set to 1, the recommended overlap for selected window function will be picked.
Default is 0.75.
Examples
• Leave almost only low frequencies in audio:
afftfilt="1-clip((b/nb)*b,0,1)"
afir
Apply an arbitrary Frequency Impulse Response filter.
This filter is designed for applying long FIR filters, up to 30 seconds long.
It can be used as component for digital crossover filters, room equalization, cross talk cancellation,
wavefield synthesis, auralization, ambiophonics and ambisonics.
This filter uses second stream as FIR coefficients. If second stream holds single channel, it will be
used for all input channels in first stream, otherwise number of channels in second stream must be same
as number of channels in first stream.
It accepts the following parameters:
dry Set dry gain. This sets input gain.
wet Set wet gain. This sets final output gain.
length
Set Impulse Response filter length. Default is 1, which means whole IR is processed.
again
Enable applying gain measured from power of IR.
Examples
• Apply reverb to stream using mono IR file as second input, complete command using ffmpeg:
ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav
aformat
Set output format constraints for the input audio. The framework will negotiate the most appropriate
format to minimize conversions.
It accepts the following parameters:
sample_fmts
A '|'-separated list of requested sample formats.
sample_rates
A '|'-separated list of requested sample rates.
channel_layouts
A '|'-separated list of requested channel layouts.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
If a parameter is omitted, all values are allowed.
Force the output to either unsigned 8-bit or signed 16-bit stereo
aformat=sample_fmts=u8|s16:channel_layouts=stereo
agate
A gate is mainly used to reduce lower parts of a signal. This kind of signal processing reduces
disturbing noise between useful signals.
Gating is done by detecting the volume below a chosen level threshold and dividing it by the factor set
with ratio. The bottom of the noise floor is set via range. Because an exact manipulation of the signal
would cause distortion of the waveform the reduction can be levelled over time. This is done by setting
attack and release.
attack determines how long the signal has to fall below the threshold before any reduction will occur and
release sets the time the signal has to rise above the threshold to reduce the reduction again. Shorter
signals than the chosen attack time will be left untouched.
level_in
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed
range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range
is from 0 to 1.
ratio
Set a ratio by which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction stops.
Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold before the reduction is increased
again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is
2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like one. Default is "rms". Can be
"peak" or "rms".
link
Choose if the average level between all channels or the louder channel affects the reduction.
Default is "average". Can be "average" or "maximum".
alimiter
The limiter prevents an input signal from rising over a desired threshold. This limiter uses lookahead
technology to prevent your signal from distorting. It means that there is a small delay after the signal
is processed. Keep in mind that the delay it produces is the attack time you set.
The filter accepts the following options:
level_in
Set input gain. Default is 1.
level_out
Set output gain. Default is 1.
limit
Don't let signals above this level pass the limiter. Default is 1.
attack
The limiter will reach its attenuation level in this amount of time in milliseconds. Default is 5
milliseconds.
release
Come back from limiting to attenuation 1.0 in this amount of milliseconds. Default is 50
milliseconds.
asc When gain reduction is always needed ASC takes care of releasing to an average reduction level rather
than reaching a reduction of 0 in the release time.
asc_level
Select how much the release time is affected by ASC, 0 means nearly no changes in release time while
1 produces higher release times.
level
Auto level output signal. Default is enabled. This normalizes audio back to 0dB if enabled.
Depending on picked setting it is recommended to upsample input 2x or 4x times with aresample before
applying this filter.
allpass
Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-width width. An
all-pass filter changes the audio's frequency to phase relationship without changing its frequency to
amplitude relationship.
The filter accepts the following options:
frequency, f
Set frequency in Hz.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units.
channels, c
Specify which channels to filter, by default all available are filtered.
aloop
Loop audio samples.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal number of samples.
start
Set first sample of loop.
amerge
Merge two or more audio streams into a single multi-channel stream.
The filter accepts the following options:
inputs
Set the number of inputs. Default is 2.
If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the
output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of
the inputs are not disjoint, the output will have all the channels of the first input then all the
channels of the second input, in that order, and the channel layout of the output will be the default
value corresponding to the total number of channels.
For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output
will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel
of the first input, b1 is the first channel of the second input).
On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2,
b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected
value.
All inputs must have the same sample rate, and format.
If inputs do not have the same duration, the output will stop with the shortest.
Examples
• Merge two mono files into a stereo stream:
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge
• Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:
ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv
amix
Mixes multiple audio inputs into a single output.
Note that this filter only supports float samples (the amerge and pan audio filters support many
formats). If the amix input has integer samples then aresample will be automatically inserted to perform
the conversion to float samples.
For example
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT
will mix 3 input audio streams to a single output with the same duration as the first input and a dropout
transition time of 3 seconds.
It accepts the following parameters:
inputs
The number of inputs. If unspecified, it defaults to 2.
duration
How to determine the end-of-stream.
longest
The duration of the longest input. (default)
shortest
The duration of the shortest input.
first
The duration of the first input.
dropout_transition
The transition time, in seconds, for volume renormalization when an input stream ends. The default
value is 2 seconds.
anequalizer
High-order parametric multiband equalizer for each channel.
It accepts the following parameters:
params
This option string is in format: "cchn f=cf w=w g=g t=f | ..." Each equalizer band is separated by
'|'.
chn Set channel number to which equalization will be applied. If input doesn't have that channel the
entry is ignored.
f Set central frequency for band. If input doesn't have that frequency the entry is ignored.
w Set band width in hertz.
g Set band gain in dB.
t Set filter type for band, optional, can be:
0 Butterworth, this is default.
1 Chebyshev type 1.
2 Chebyshev type 2.
curves
With this option activated frequency response of anequalizer is displayed in video stream.
size
Set video stream size. Only useful if curves option is activated.
mgain
Set max gain that will be displayed. Only useful if curves option is activated. Setting this to a
reasonable value makes it possible to display gain which is derived from neighbour bands which are
too close to each other and thus produce higher gain when both are activated.
fscale
Set frequency scale used to draw frequency response in video output. Can be linear or logarithmic.
Default is logarithmic.
colors
Set color for each channel curve which is going to be displayed in video stream. This is list of
color names separated by space or by '|'. Unrecognised or missing colors will be replaced by white
color.
Examples
• Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels using Chebyshev
type 1 filter:
anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1
Commands
This filter supports the following commands:
change
Alter existing filter parameters. Syntax for the commands is : "fN|f=freq|w=width|g=gain"
fN is existing filter number, starting from 0, if no such filter is available error is returned.
freq set new frequency parameter. width set new width parameter in herz. gain set new gain
parameter in dB.
Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer change
0|f=200|w=50|g=1',anequalizer=...
anull
Pass the audio source unchanged to the output.
apad
Pad the end of an audio stream with silence.
This can be used together with ffmpeg -shortest to extend audio streams to the same length as the video
stream.
A description of the accepted options follows.
packet_size
Set silence packet size. Default value is 4096.
pad_len
Set the number of samples of silence to add to the end. After the value is reached, the stream is
terminated. This option is mutually exclusive with whole_len.
whole_len
Set the minimum total number of samples in the output audio stream. If the value is longer than the
input audio length, silence is added to the end, until the value is reached. This option is mutually
exclusive with pad_len.
If neither the pad_len nor the whole_len option is set, the filter will add silence to the end of the
input stream indefinitely.
Examples
• Add 1024 samples of silence to the end of the input:
apad=pad_len=1024
• Make sure the audio output will contain at least 10000 samples, pad the input with silence if
required:
apad=whole_len=10000
• Use ffmpeg to pad the audio input with silence, so that the video stream will always result the
shortest and will be converted until the end in the output file when using the shortest option:
ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT
aphaser
Add a phasing effect to the input audio.
A phaser filter creates series of peaks and troughs in the frequency spectrum. The position of the peaks
and troughs are modulated so that they vary over time, creating a sweeping effect.
A description of the accepted parameters follows.
in_gain
Set input gain. Default is 0.4.
out_gain
Set output gain. Default is 0.74
delay
Set delay in milliseconds. Default is 3.0.
decay
Set decay. Default is 0.4.
speed
Set modulation speed in Hz. Default is 0.5.
type
Set modulation type. Default is triangular.
It accepts the following values:
triangular, t
sinusoidal, s
apulsator
Audio pulsator is something between an autopanner and a tremolo. But it can produce funny stereo effects
as well. Pulsator changes the volume of the left and right channel based on a LFO (low frequency
oscillator) with different waveforms and shifted phases. This filter have the ability to define an
offset between left and right channel. An offset of 0 means that both LFO shapes match each other. The
left and right channel are altered equally - a conventional tremolo. An offset of 50% means that the
shape of the right channel is exactly shifted in phase (or moved backwards about half of the frequency) -
pulsator acts as an autopanner. At 1 both curves match again. Every setting in between moves the phase
shift gapless between all stages and produces some "bypassing" sounds with sine and triangle waveforms.
The more you set the offset near 1 (starting from the 0.5) the faster the signal passes from the left to
the right speaker.
The filter accepts the following options:
level_in
Set input gain. By default it is 1. Range is [0.015625 - 64].
level_out
Set output gain. By default it is 1. Range is [0.015625 - 64].
mode
Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or sawdown. Default
is sine.
amount
Set modulation. Define how much of original signal is affected by the LFO.
offset_l
Set left channel offset. Default is 0. Allowed range is [0 - 1].
offset_r
Set right channel offset. Default is 0.5. Allowed range is [0 - 1].
width
Set pulse width. Default is 1. Allowed range is [0 - 2].
timing
Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.
bpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to bpm.
ms Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to ms.
hz Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing is set to hz.
aresample
Resample the input audio to the specified parameters, using the libswresample library. If none are
specified then the filter will automatically convert between its input and output.
This filter is also able to stretch/squeeze the audio data to make it match the timestamps or to inject
silence / cut out audio to make it match the timestamps, do a combination of both or do neither.
The filter accepts the syntax [sample_rate:]resampler_options, where sample_rate expresses a sample rate
and resampler_options is a list of key=value pairs, separated by ":". See the the "Resampler Options"
section in the ffmpeg-resampler(1) manual for the complete list of supported options.
Examples
• Resample the input audio to 44100Hz:
aresample=44100
• Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per second
compensation:
aresample=async=1000
areverse
Reverse an audio clip.
Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.
Examples
• Take the first 5 seconds of a clip, and reverse it.
atrim=end=5,areverse
asetnsamples
Set the number of samples per each output audio frame.
The last output packet may contain a different number of samples, as the filter will flush all the
remaining samples when the input audio signals its end.
The filter accepts the following options:
nb_out_samples, n
Set the number of frames per each output audio frame. The number is intended as the number of samples
per each channel. Default value is 1024.
pad, p
If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will
contain the same number of samples as the previous ones. Default value is 1.
For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use:
asetnsamples=n=1234:p=0
asetrate
Set the sample rate without altering the PCM data. This will result in a change of speed and pitch.
The filter accepts the following options:
sample_rate, r
Set the output sample rate. Default is 44100 Hz.
ashowinfo
Show a line containing various information for each input audio frame. The input audio is not modified.
The shown line contains a sequence of key/value pairs of the form key:value.
The following values are shown in the output:
n The (sequential) number of the input frame, starting from 0.
pts The presentation timestamp of the input frame, in time base units; the time base depends on the
filter input pad, and is usually 1/sample_rate.
pts_time
The presentation timestamp of the input frame in seconds.
pos position of the frame in the input stream, -1 if this information in unavailable and/or meaningless
(for example in case of synthetic audio)
fmt The sample format.
chlayout
The channel layout.
rate
The sample rate for the audio frame.
nb_samples
The number of samples (per channel) in the frame.
checksum
The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the data is
treated as if all the planes were concatenated.
plane_checksums
A list of Adler-32 checksums for each data plane.
astats
Display time domain statistical information about the audio channels. Statistics are calculated and
displayed for each audio channel and, where applicable, an overall figure is also given.
It accepts the following option:
length
Short window length in seconds, used for peak and trough RMS measurement. Default is 0.05 (50
milliseconds). Allowed range is "[0.1 - 10]".
metadata
Set metadata injection. All the metadata keys are prefixed with "lavfi.astats.X", where "X" is
channel number starting from 1 or string "Overall". Default is disabled.
Available keys for each channel are: DC_offset Min_level Max_level Min_difference Max_difference
Mean_difference RMS_difference Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count
Bit_depth Dynamic_range
and for Overall: DC_offset Min_level Max_level Min_difference Max_difference Mean_difference
RMS_difference Peak_level RMS_level RMS_peak RMS_trough Flat_factor Peak_count Bit_depth
Number_of_samples
For example full key look like this "lavfi.astats.1.DC_offset" or this
"lavfi.astats.Overall.Peak_count".
For description what each key means read below.
reset
Set number of frame after which stats are going to be recalculated. Default is disabled.
A description of each shown parameter follows:
DC offset
Mean amplitude displacement from zero.
Min level
Minimal sample level.
Max level
Maximal sample level.
Min difference
Minimal difference between two consecutive samples.
Max difference
Maximal difference between two consecutive samples.
Mean difference
Mean difference between two consecutive samples. The average of each difference between two
consecutive samples.
RMS difference
Root Mean Square difference between two consecutive samples.
Peak level dB
RMS level dB
Standard peak and RMS level measured in dBFS.
RMS peak dB
RMS trough dB
Peak and trough values for RMS level measured over a short window.
Crest factor
Standard ratio of peak to RMS level (note: not in dB).
Flat factor
Flatness (i.e. consecutive samples with the same value) of the signal at its peak levels (i.e. either
Min level or Max level).
Peak count
Number of occasions (not the number of samples) that the signal attained either Min level or Max
level.
Bit depth
Overall bit depth of audio. Number of bits used for each sample.
Dynamic range
Measured dynamic range of audio in dB.
atempo
Adjust audio tempo.
The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume
nominal 1.0 tempo. Tempo must be in the [0.5, 2.0] range.
Examples
• Slow down audio to 80% tempo:
atempo=0.8
• To speed up audio to 125% tempo:
atempo=1.25
atrim
Trim the input so that the output contains one continuous subpart of the input.
It accepts the following parameters:
start
Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp
start will be the first sample in the output.
end Specify time of the first audio sample that will be dropped, i.e. the audio sample immediately
preceding the one with the timestamp end will be the last sample in the output.
start_pts
Same as start, except this option sets the start timestamp in samples instead of seconds.
end_pts
Same as end, except this option sets the end timestamp in samples instead of seconds.
duration
The maximum duration of the output in seconds.
start_sample
The number of the first sample that should be output.
end_sample
The number of the first sample that should be dropped.
start, end, and duration are expressed as time duration specifications; see the Time duration section in
the ffmpeg-utils(1) manual.
Note that the first two sets of the start/end options and the duration option look at the frame
timestamp, while the _sample options simply count the samples that pass through the filter. So
start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or
do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the
output timestamps start at zero, insert the asetpts filter after the atrim filter.
If multiple start or end options are set, this filter tries to be greedy and keep all samples that match
at least one of the specified constraints. To keep only the part that matches all the constraints at
once, chain multiple atrim filters.
The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to
keep everything before the specified time.
Examples:
• Drop everything except the second minute of input:
ffmpeg -i INPUT -af atrim=60:120
• Keep only the first 1000 samples:
ffmpeg -i INPUT -af atrim=end_sample=1000
bandpass
Apply a two-pole Butterworth band-pass filter with central frequency frequency, and (3dB-point) band-
width width. The csg option selects a constant skirt gain (peak gain = Q) instead of the default:
constant 0dB peak gain. The filter roll off at 6dB per octave (20dB per decade).
The filter accepts the following options:
frequency, f
Set the filter's central frequency. Default is 3000.
csg Constant skirt gain if set to 1. Defaults to 0.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units.
channels, c
Specify which channels to filter, by default all available are filtered.
bandreject
Apply a two-pole Butterworth band-reject filter with central frequency frequency, and (3dB-point) band-
width width. The filter roll off at 6dB per octave (20dB per decade).
The filter accepts the following options:
frequency, f
Set the filter's central frequency. Default is 3000.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units.
channels, c
Specify which channels to filter, by default all available are filtered.
bass
Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter with a response
similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
The filter accepts the following options:
gain, g
Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a large boost).
Beware of clipping when using a positive gain.
frequency, f
Set the filter's central frequency and so can be used to extend or reduce the frequency range to be
boosted or cut. The default value is 100 Hz.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Determine how steep is the filter's shelf transition.
channels, c
Specify which channels to filter, by default all available are filtered.
biquad
Apply a biquad IIR filter with the given coefficients. Where b0, b1, b2 and a0, a1, a2 are the numerator
and denominator coefficients respectively. and channels, c specify which channels to filter, by default
all available are filtered.
bs2b
Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libbs2b".
It accepts the following parameters:
profile
Pre-defined crossfeed level.
default
Default level (fcut=700, feed=50).
cmoy
Chu Moy circuit (fcut=700, feed=60).
jmeier
Jan Meier circuit (fcut=650, feed=95).
fcut
Cut frequency (in Hz).
feed
Feed level (in Hz).
channelmap
Remap input channels to new locations.
It accepts the following parameters:
map Map channels from input to output. The argument is a '|'-separated list of mappings, each in the
"in_channel-out_channel" or in_channel form. in_channel can be either the name of the input channel
(e.g. FL for front left) or its index in the input channel layout. out_channel is the name of the
output channel or its index in the output channel layout. If out_channel is not given then it is
implicitly an index, starting with zero and increasing by one for each mapping.
channel_layout
The channel layout of the output stream.
If no mapping is present, the filter will implicitly map input channels to output channels, preserving
indices.
For example, assuming a 5.1+downmix input MOV file,
ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav
will create an output WAV file tagged as stereo from the downmix channels of the input.
To fix a 5.1 WAV improperly encoded in AAC's native channel order
ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav
channelsplit
Split each channel from an input audio stream into a separate output stream.
It accepts the following parameters:
channel_layout
The channel layout of the input stream. The default is "stereo".
For example, assuming a stereo input MP3 file,
ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv
will create an output Matroska file with two audio streams, one containing only the left channel and the
other the right channel.
Split a 5.1 WAV file into per-channel files:
ffmpeg -i in.wav -filter_complex
'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
-map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
side_right.wav
chorus
Add a chorus effect to the audio.
Can make a single vocal sound like a chorus, but can also be applied to instrumentation.
Chorus resembles an echo effect with a short delay, but whereas with echo the delay is constant, with
chorus, it is varied using using sinusoidal or triangular modulation. The modulation depth defines the
range the modulated delay is played before or after the delay. Hence the delayed sound will sound slower
or faster, that is the delayed sound tuned around the original one, like in a chorus where some vocals
are slightly off key.
It accepts the following parameters:
in_gain
Set input gain. Default is 0.4.
out_gain
Set output gain. Default is 0.4.
delays
Set delays. A typical delay is around 40ms to 60ms.
decays
Set decays.
speeds
Set speeds.
depths
Set depths.
Examples
• A single delay:
chorus=0.7:0.9:55:0.4:0.25:2
• Two delays:
chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3
• Fuller sounding chorus with three delays:
chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3
compand
Compress or expand the audio's dynamic range.
It accepts the following parameters:
attacks
decays
A list of times in seconds for each channel over which the instantaneous level of the input signal is
averaged to determine its volume. attacks refers to increase of volume and decays refers to decrease
of volume. For most situations, the attack time (response to the audio getting louder) should be
shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden
soft audio. A typical value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds.
If specified number of attacks & decays is lower than number of channels, the last set attack/decay
will be used for all remaining channels.
points
A list of points for the transfer function, specified in dB relative to the maximum possible signal
amplitude. Each key points list must be defined using the following syntax: "x0/y0|x1/y1|x2/y2|...."
or "x0/y0 x1/y1 x2/y2 ...."
The input values must be in strictly increasing order but the transfer function does not have to be
monotonically rising. The point "0/0" is assumed but may be overridden (by "0/out-dBn"). Typical
values for the transfer function are "-70/-70|-60/-20|1/0".
soft-knee
Set the curve radius in dB for all joints. It defaults to 0.01.
gain
Set the additional gain in dB to be applied at all points on the transfer function. This allows for
easy adjustment of the overall gain. It defaults to 0.
volume
Set an initial volume, in dB, to be assumed for each channel when filtering starts. This permits the
user to supply a nominal level initially, so that, for example, a very large gain is not applied to
initial signal levels before the companding has begun to operate. A typical value for audio which is
initially quiet is -90 dB. It defaults to 0.
delay
Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed before being
fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows
the filter to effectively operate in predictive rather than reactive mode. It defaults to 0.
Examples
• Make music with both quiet and loud passages suitable for listening to in a noisy environment:
compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
Another example for audio with whisper and explosion parts:
compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0
• A noise gate for when the noise is at a lower level than the signal:
compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
• Here is another noise gate, this time for when the noise is at a higher level than the signal (making
it, in some ways, similar to squelch):
compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
• 2:1 compression starting at -6dB:
compand=points=-80/-80|-6/-6|0/-3.8|20/3.5
• 2:1 compression starting at -9dB:
compand=points=-80/-80|-9/-9|0/-5.3|20/2.9
• 2:1 compression starting at -12dB:
compand=points=-80/-80|-12/-12|0/-6.8|20/1.9
• 2:1 compression starting at -18dB:
compand=points=-80/-80|-18/-18|0/-9.8|20/0.7
• 3:1 compression starting at -15dB:
compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2
• Compressor/Gate:
compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6
• Expander:
compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3
• Hard limiter at -6dB:
compand=attacks=0:points=-80/-80|-6/-6|20/-6
• Hard limiter at -12dB:
compand=attacks=0:points=-80/-80|-12/-12|20/-12
• Hard noise gate at -35 dB:
compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20
• Soft limiter:
compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8
compensationdelay
Compensation Delay Line is a metric based delay to compensate differing positions of microphones or
speakers.
For example, you have recorded guitar with two microphones placed in different location. Because the
front of sound wave has fixed speed in normal conditions, the phasing of microphones can vary and depends
on their location and interposition. The best sound mix can be achieved when these microphones are in
phase (synchronized). Note that distance of ~30 cm between microphones makes one microphone to capture
signal in antiphase to another microphone. That makes the final mix sounding moody. This filter helps to
solve phasing problems by adding different delays to each microphone track and make them synchronized.
The best result can be reached when you take one track as base and synchronize other tracks one by one
with it. Remember that synchronization/delay tolerance depends on sample rate, too. Higher sample rates
will give more tolerance.
It accepts the following parameters:
mm Set millimeters distance. This is compensation distance for fine tuning. Default is 0.
cm Set cm distance. This is compensation distance for tightening distance setup. Default is 0.
m Set meters distance. This is compensation distance for hard distance setup. Default is 0.
dry Set dry amount. Amount of unprocessed (dry) signal. Default is 0.
wet Set wet amount. Amount of processed (wet) signal. Default is 1.
temp
Set temperature degree in Celsius. This is the temperature of the environment. Default is 20.
crossfeed
Apply headphone crossfeed filter.
Crossfeed is the process of blending the left and right channels of stereo audio recording. It is mainly
used to reduce extreme stereo separation of low frequencies.
The intent is to produce more speaker like sound to the listener.
The filter accepts the following options:
strength
Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1. This sets gain of low shelf
filter for side part of stereo image. Default is -6dB. Max allowed is -30db when strength is set to
1.
range
Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1. This sets cut off frequency
of low shelf filter. Default is cut off near 1550 Hz. With range set to 1 cut off frequency is set to
2100 Hz.
level_in
Set input gain. Default is 0.9.
level_out
Set output gain. Default is 1.
crystalizer
Simple algorithm to expand audio dynamic range.
The filter accepts the following options:
i Sets the intensity of effect (default: 2.0). Must be in range between 0.0 (unchanged sound) to 10.0
(maximum effect).
c Enable clipping. By default is enabled.
dcshift
Apply a DC shift to the audio.
This can be useful to remove a DC offset (caused perhaps by a hardware problem in the recording chain)
from the audio. The effect of a DC offset is reduced headroom and hence volume. The astats filter can be
used to determine if a signal has a DC offset.
shift
Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the audio.
limitergain
Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to prevent
clipping.
dynaudnorm
Dynamic Audio Normalizer.
This filter applies a certain amount of gain to the input audio in order to bring its peak magnitude to a
target level (e.g. 0 dBFS). However, in contrast to more "simple" normalization algorithms, the Dynamic
Audio Normalizer *dynamically* re-adjusts the gain factor to the input audio. This allows for applying
extra gain to the "quiet" sections of the audio while avoiding distortions or clipping the "loud"
sections. In other words: The Dynamic Audio Normalizer will "even out" the volume of quiet and loud
sections, in the sense that the volume of each section is brought to the same target level. Note,
however, that the Dynamic Audio Normalizer achieves this goal *without* applying "dynamic range
compressing". It will retain 100% of the dynamic range *within* each section of the audio file.
f Set the frame length in milliseconds. In range from 10 to 8000 milliseconds. Default is 500
milliseconds. The Dynamic Audio Normalizer processes the input audio in small chunks, referred to as
frames. This is required, because a peak magnitude has no meaning for just a single sample value.
Instead, we need to determine the peak magnitude for a contiguous sequence of sample values. While a
"standard" normalizer would simply use the peak magnitude of the complete file, the Dynamic Audio
Normalizer determines the peak magnitude individually for each frame. The length of a frame is
specified in milliseconds. By default, the Dynamic Audio Normalizer uses a frame length of 500
milliseconds, which has been found to give good results with most files. Note that the exact frame
length, in number of samples, will be determined automatically, based on the sampling rate of the
individual input audio file.
g Set the Gaussian filter window size. In range from 3 to 301, must be odd number. Default is 31.
Probably the most important parameter of the Dynamic Audio Normalizer is the "window size" of the
Gaussian smoothing filter. The filter's window size is specified in frames, centered around the
current frame. For the sake of simplicity, this must be an odd number. Consequently, the default
value of 31 takes into account the current frame, as well as the 15 preceding frames and the 15
subsequent frames. Using a larger window results in a stronger smoothing effect and thus in less gain
variation, i.e. slower gain adaptation. Conversely, using a smaller window results in a weaker
smoothing effect and thus in more gain variation, i.e. faster gain adaptation. In other words, the
more you increase this value, the more the Dynamic Audio Normalizer will behave like a "traditional"
normalization filter. On the contrary, the more you decrease this value, the more the Dynamic Audio
Normalizer will behave like a dynamic range compressor.
p Set the target peak value. This specifies the highest permissible magnitude level for the normalized
audio input. This filter will try to approach the target peak magnitude as closely as possible, but
at the same time it also makes sure that the normalized signal will never exceed the peak magnitude.
A frame's maximum local gain factor is imposed directly by the target peak magnitude. The default
value is 0.95 and thus leaves a headroom of 5%*. It is not recommended to go above this value.
m Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0. The Dynamic Audio
Normalizer determines the maximum possible (local) gain factor for each input frame, i.e. the maximum
gain factor that does not result in clipping or distortion. The maximum gain factor is determined by
the frame's highest magnitude sample. However, the Dynamic Audio Normalizer additionally bounds the
frame's maximum gain factor by a predetermined (global) maximum gain factor. This is done in order to
avoid excessive gain factors in "silent" or almost silent frames. By default, the maximum gain factor
is 10.0, For most inputs the default value should be sufficient and it usually is not recommended to
increase this value. Though, for input with an extremely low overall volume level, it may be
necessary to allow even higher gain factors. Note, however, that the Dynamic Audio Normalizer does
not simply apply a "hard" threshold (i.e. cut off values above the threshold). Instead, a "sigmoid"
threshold function will be applied. This way, the gain factors will smoothly approach the threshold
value, but never exceed that value.
r Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled. By default, the Dynamic
Audio Normalizer performs "peak" normalization. This means that the maximum local gain factor for
each frame is defined (only) by the frame's highest magnitude sample. This way, the samples can be
amplified as much as possible without exceeding the maximum signal level, i.e. without clipping.
Optionally, however, the Dynamic Audio Normalizer can also take into account the frame's root mean
square, abbreviated RMS. In electrical engineering, the RMS is commonly used to determine the power
of a time-varying signal. It is therefore considered that the RMS is a better approximation of the
"perceived loudness" than just looking at the signal's peak magnitude. Consequently, by adjusting all
frames to a constant RMS value, a uniform "perceived loudness" can be established. If a target RMS
value has been specified, a frame's local gain factor is defined as the factor that would result in
exactly that RMS value. Note, however, that the maximum local gain factor is still restricted by the
frame's highest magnitude sample, in order to prevent clipping.
n Enable channels coupling. By default is enabled. By default, the Dynamic Audio Normalizer will
amplify all channels by the same amount. This means the same gain factor will be applied to all
channels, i.e. the maximum possible gain factor is determined by the "loudest" channel. However, in
some recordings, it may happen that the volume of the different channels is uneven, e.g. one channel
may be "quieter" than the other one(s). In this case, this option can be used to disable the channel
coupling. This way, the gain factor will be determined independently for each channel, depending only
on the individual channel's highest magnitude sample. This allows for harmonizing the volume of the
different channels.
c Enable DC bias correction. By default is disabled. An audio signal (in the time domain) is a
sequence of sample values. In the Dynamic Audio Normalizer these sample values are represented in
the -1.0 to 1.0 range, regardless of the original input format. Normally, the audio signal, or
"waveform", should be centered around the zero point. That means if we calculate the mean value of
all samples in a file, or in a single frame, then the result should be 0.0 or at least very close to
that value. If, however, there is a significant deviation of the mean value from 0.0, in either
positive or negative direction, this is referred to as a DC bias or DC offset. Since a DC bias is
clearly undesirable, the Dynamic Audio Normalizer provides optional DC bias correction. With DC bias
correction enabled, the Dynamic Audio Normalizer will determine the mean value, or "DC correction"
offset, of each input frame and subtract that value from all of the frame's sample values which
ensures those samples are centered around 0.0 again. Also, in order to avoid "gaps" at the frame
boundaries, the DC correction offset values will be interpolated smoothly between neighbouring
frames.
b Enable alternative boundary mode. By default is disabled. The Dynamic Audio Normalizer takes into
account a certain neighbourhood around each frame. This includes the preceding frames as well as the
subsequent frames. However, for the "boundary" frames, located at the very beginning and at the very
end of the audio file, not all neighbouring frames are available. In particular, for the first few
frames in the audio file, the preceding frames are not known. And, similarly, for the last few frames
in the audio file, the subsequent frames are not known. Thus, the question arises which gain factors
should be assumed for the missing frames in the "boundary" region. The Dynamic Audio Normalizer
implements two modes to deal with this situation. The default boundary mode assumes a gain factor of
exactly 1.0 for the missing frames, resulting in a smooth "fade in" and "fade out" at the beginning
and at the end of the input, respectively.
s Set the compress factor. In range from 0.0 to 30.0. Default is 0.0. By default, the Dynamic Audio
Normalizer does not apply "traditional" compression. This means that signal peaks will not be pruned
and thus the full dynamic range will be retained within each local neighbourhood. However, in some
cases it may be desirable to combine the Dynamic Audio Normalizer's normalization algorithm with a
more "traditional" compression. For this purpose, the Dynamic Audio Normalizer provides an optional
compression (thresholding) function. If (and only if) the compression feature is enabled, all input
frames will be processed by a soft knee thresholding function prior to the actual normalization
process. Put simply, the thresholding function is going to prune all samples whose magnitude exceeds
a certain threshold value. However, the Dynamic Audio Normalizer does not simply apply a fixed
threshold value. Instead, the threshold value will be adjusted for each individual frame. In
general, smaller parameters result in stronger compression, and vice versa. Values below 3.0 are not
recommended, because audible distortion may appear.
earwax
Make audio easier to listen to on headphones.
This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on
headphones the stereo image is moved from inside your head (standard for headphones) to outside and in
front of the listener (standard for speakers).
Ported from SoX.
equalizer
Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at and around a
selected frequency can be increased or decreased, whilst (unlike bandpass and bandreject filters) that at
all other frequencies is unchanged.
In order to produce complex equalisation curves, this filter can be given several times, each with a
different central frequency.
The filter accepts the following options:
frequency, f
Set the filter's central frequency in Hz.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units.
gain, g
Set the required gain or attenuation in dB. Beware of clipping when using a positive gain.
channels, c
Specify which channels to filter, by default all available are filtered.
Examples
• Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:
equalizer=f=1000:t=h:width=200:g=-10
• Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:
equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5
extrastereo
Linearly increases the difference between left and right channels which adds some sort of "live" effect
to playback.
The filter accepts the following options:
m Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both channels), with
1.0 sound will be unchanged, with -1.0 left and right channels will be swapped.
c Enable clipping. By default is enabled.
firequalizer
Apply FIR Equalization using arbitrary frequency response.
The filter accepts the following option:
gain
Set gain curve equation (in dB). The expression can contain variables:
f the evaluated frequency
sr sample rate
ch channel number, set to 0 when multichannels evaluation is disabled
chid
channel id, see libavutil/channel_layout.h, set to the first channel id when multichannels
evaluation is disabled
chs number of channels
chlayout
channel_layout, see libavutil/channel_layout.h
and functions:
gain_interpolate(f)
interpolate gain on frequency f based on gain_entry
cubic_interpolate(f)
same as gain_interpolate, but smoother
This option is also available as command. Default is gain_interpolate(f).
gain_entry
Set gain entry for gain_interpolate function. The expression can contain functions:
entry(f, g)
store gain entry at frequency f with value g
This option is also available as command.
delay
Set filter delay in seconds. Higher value means more accurate. Default is 0.01.
accuracy
Set filter accuracy in Hz. Lower value means more accurate. Default is 5.
wfunc
Set window function. Acceptable values are:
rectangular
rectangular window, useful when gain curve is already smooth
hann
hann window (default)
hamming
hamming window
blackman
blackman window
nuttall3
3-terms continuous 1st derivative nuttall window
mnuttall3
minimum 3-terms discontinuous nuttall window
nuttall
4-terms continuous 1st derivative nuttall window
bnuttall
minimum 4-terms discontinuous nuttall (blackman-nuttall) window
bharris
blackman-harris window
tukey
tukey window
fixed
If enabled, use fixed number of audio samples. This improves speed when filtering with large delay.
Default is disabled.
multi
Enable multichannels evaluation on gain. Default is disabled.
zero_phase
Enable zero phase mode by subtracting timestamp to compensate delay. Default is disabled.
scale
Set scale used by gain. Acceptable values are:
linlin
linear frequency, linear gain
linlog
linear frequency, logarithmic (in dB) gain (default)
loglin
logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain
loglog
logarithmic frequency, logarithmic gain
dumpfile
Set file for dumping, suitable for gnuplot.
dumpscale
Set scale for dumpfile. Acceptable values are same with scale option. Default is linlog.
fft2
Enable 2-channel convolution using complex FFT. This improves speed significantly. Default is
disabled.
min_phase
Enable minimum phase impulse response. Default is disabled.
Examples
• lowpass at 1000 Hz:
firequalizer=gain='if(lt(f,1000), 0, -INF)'
• lowpass at 1000 Hz with gain_entry:
firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'
• custom equalization:
firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'
• higher delay with zero phase to compensate delay:
firequalizer=delay=0.1:fixed=on:zero_phase=on
• lowpass on left channel, highpass on right channel:
firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
:gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on
flanger
Apply a flanging effect to the audio.
The filter accepts the following options:
delay
Set base delay in milliseconds. Range from 0 to 30. Default value is 0.
depth
Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.
regen
Set percentage regeneration (delayed signal feedback). Range from -95 to 95. Default value is 0.
width
Set percentage of delayed signal mixed with original. Range from 0 to 100. Default value is 71.
speed
Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.
shape
Set swept wave shape, can be triangular or sinusoidal. Default value is sinusoidal.
phase
Set swept wave percentage-shift for multi channel. Range from 0 to 100. Default value is 25.
interp
Set delay-line interpolation, linear or quadratic. Default is linear.
haas
Apply Haas effect to audio.
Note that this makes most sense to apply on mono signals. With this filter applied to mono signals it
give some directionality and stretches its stereo image.
The filter accepts the following options:
level_in
Set input level. By default is 1, or 0dB
level_out
Set output level. By default is 1, or 0dB.
side_gain
Set gain applied to side part of signal. By default is 1.
middle_source
Set kind of middle source. Can be one of the following:
left
Pick left channel.
right
Pick right channel.
mid Pick middle part signal of stereo image.
side
Pick side part signal of stereo image.
middle_phase
Change middle phase. By default is disabled.
left_delay
Set left channel delay. By default is 2.05 milliseconds.
left_balance
Set left channel balance. By default is -1.
left_gain
Set left channel gain. By default is 1.
left_phase
Change left phase. By default is disabled.
right_delay
Set right channel delay. By defaults is 2.12 milliseconds.
right_balance
Set right channel balance. By default is 1.
right_gain
Set right channel gain. By default is 1.
right_phase
Change right phase. By default is enabled.
hdcd
Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with embedded HDCD codes is
expanded into a 20-bit PCM stream.
The filter supports the Peak Extend and Low-level Gain Adjustment features of HDCD, and detects the
Transient Filter flag.
ffmpeg -i HDCD16.flac -af hdcd OUT24.flac
When using the filter with wav, note the default encoding for wav is 16-bit, so the resulting 20-bit
stream will be truncated back to 16-bit. Use something like -acodec pcm_s24le after the filter to get
24-bit PCM output.
ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav
The filter accepts the following options:
disable_autoconvert
Disable any automatic format conversion or resampling in the filter graph.
process_stereo
Process the stereo channels together. If target_gain does not match between channels, consider it
invalid and use the last valid target_gain.
cdt_ms
Set the code detect timer period in ms.
force_pe
Always extend peaks above -3dBFS even if PE isn't signaled.
analyze_mode
Replace audio with a solid tone and adjust the amplitude to signal some specific aspect of the
decoding process. The output file can be loaded in an audio editor alongside the original to aid
analysis.
"analyze_mode=pe:force_pe=true" can be used to see all samples above the PE level.
Modes are:
0, off
Disabled
1, lle
Gain adjustment level at each sample
2, pe
Samples where peak extend occurs
3, cdt
Samples where the code detect timer is active
4, tgm
Samples where the target gain does not match between channels
headphone
Apply head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for binaural
listening via headphones. The HRIRs are provided via additional streams, for each channel one stereo
input stream is needed.
The filter accepts the following options:
map Set mapping of input streams for convolution. The argument is a '|'-separated list of channel names
in order as they are given as additional stream inputs for filter. This also specify number of input
streams. Number of input streams must be not less than number of channels in first stream plus one.
gain
Set gain applied to audio. Value is in dB. Default is 0.
type
Set processing type. Can be time or freq. time is processing audio in time domain which is slow.
freq is processing audio in frequency domain which is fast. Default is freq.
lfe Set custom gain for LFE channels. Value is in dB. Default is 0.
Examples
• Full example using wav files as coefficients with amovie filters for 7.1 downmix, each amovie filter
use stereo file with IR coefficients as input. The files give coefficients for each position of
virtual loudspeaker:
ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
output.wav
highpass
Apply a high-pass filter with 3dB point frequency. The filter can be either single-pole, or double-pole
(the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).
The filter accepts the following options:
frequency, f
Set frequency in Hz. Default is 3000.
poles, p
Set number of poles. Default is 2.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The
default is 0.707q and gives a Butterworth response.
channels, c
Specify which channels to filter, by default all available are filtered.
join
Join multiple input streams into one multi-channel stream.
It accepts the following parameters:
inputs
The number of input streams. It defaults to 2.
channel_layout
The desired output channel layout. It defaults to stereo.
map Map channels from inputs to output. The argument is a '|'-separated list of mappings, each in the
"input_idx.in_channel-out_channel" form. input_idx is the 0-based index of the input stream.
in_channel can be either the name of the input channel (e.g. FL for front left) or its index in the
specified input stream. out_channel is the name of the output channel.
The filter will attempt to guess the mappings when they are not specified explicitly. It does so by first
trying to find an unused matching input channel and if that fails it picks the first unused input
channel.
Join 3 inputs (with properly set channel layouts):
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT
Build a 5.1 output from 6 single-channel streams:
ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
out
ladspa
Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.
To enable compilation of this filter you need to configure FFmpeg with "--enable-ladspa".
file, f
Specifies the name of LADSPA plugin library to load. If the environment variable LADSPA_PATH is
defined, the LADSPA plugin is searched in each one of the directories specified by the colon
separated list in LADSPA_PATH, otherwise in the standard LADSPA paths, which are in this order:
HOME/.ladspa/lib/, /usr/local/lib/ladspa/, /usr/lib/ladspa/.
plugin, p
Specifies the plugin within the library. Some libraries contain only one plugin, but others contain
many of them. If this is not set filter will list all available plugins within the specified library.
controls, c
Set the '|' separated list of controls which are zero or more floating point values that determine
the behavior of the loaded plugin (for example delay, threshold or gain). Controls need to be
defined using the following syntax: c0=value0|c1=value1|c2=value2|..., where valuei is the value set
on the i-th control. Alternatively they can be also defined using the following syntax:
value0|value1|value2|..., where valuei is the value set on the i-th control. If controls is set to
"help", all available controls and their valid ranges are printed.
sample_rate, s
Specify the sample rate, default to 44100. Only used if plugin have zero inputs.
nb_samples, n
Set the number of samples per channel per each output frame, default is 1024. Only used if plugin
have zero inputs.
duration, d
Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1)
manual for the accepted syntax. Note that the resulting duration may be greater than the specified
duration, as the generated audio is always cut at the end of a complete frame. If not specified, or
the expressed duration is negative, the audio is supposed to be generated forever. Only used if
plugin have zero inputs.
Examples
• List all available plugins within amp (LADSPA example plugin) library:
ladspa=file=amp
• List all available controls and their valid ranges for "vcf_notch" plugin from "VCF" library:
ladspa=f=vcf:p=vcf_notch:c=help
• Simulate low quality audio equipment using "Computer Music Toolkit" (CMT) plugin library:
ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12
• Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):
ladspa=file=tap_reverb:tap_reverb
• Generate white noise, with 0.2 amplitude:
ladspa=file=cmt:noise_source_white:c=c0=.2
• Generate 20 bpm clicks using plugin "C* Click - Metronome" from the "C* Audio Plugin Suite" (CAPS)
library:
ladspa=file=caps:Click:c=c1=20'
• Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:
ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2
• Increase volume by 20dB using fast lookahead limiter from Steve Harris "SWH Plugins" collection:
ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2
• Attenuate low frequencies using Multiband EQ from Steve Harris "SWH Plugins" collection:
ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0
• Reduce stereo image using "Narrower" from the "C* Audio Plugin Suite" (CAPS) library:
ladspa=caps:Narrower
• Another white noise, now using "C* Audio Plugin Suite" (CAPS) library:
ladspa=caps:White:.2
• Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:
ladspa=caps:Fractal:c=c1=1
• Dynamic volume normalization using "VLevel" plugin:
ladspa=vlevel-ladspa:vlevel_mono
Commands
This filter supports the following commands:
cN Modify the N-th control value.
If the specified value is not valid, it is ignored and prior one is kept.
loudnorm
EBU R128 loudness normalization. Includes both dynamic and linear normalization modes. Support for both
single pass (livestreams, files) and double pass (files) modes. This algorithm can target IL, LRA, and
maximum true peak. To accurately detect true peaks, the audio stream will be upsampled to 192 kHz unless
the normalization mode is linear. Use the "-ar" option or "aresample" filter to explicitly set an output
sample rate.
The filter accepts the following options:
I, i
Set integrated loudness target. Range is -70.0 - -5.0. Default value is -24.0.
LRA, lra
Set loudness range target. Range is 1.0 - 20.0. Default value is 7.0.
TP, tp
Set maximum true peak. Range is -9.0 - +0.0. Default value is -2.0.
measured_I, measured_i
Measured IL of input file. Range is -99.0 - +0.0.
measured_LRA, measured_lra
Measured LRA of input file. Range is 0.0 - 99.0.
measured_TP, measured_tp
Measured true peak of input file. Range is -99.0 - +99.0.
measured_thresh
Measured threshold of input file. Range is -99.0 - +0.0.
offset
Set offset gain. Gain is applied before the true-peak limiter. Range is -99.0 - +99.0. Default is
+0.0.
linear
Normalize linearly if possible. measured_I, measured_LRA, measured_TP, and measured_thresh must also
to be specified in order to use this mode. Options are true or false. Default is true.
dual_mono
Treat mono input files as "dual-mono". If a mono file is intended for playback on a stereo system,
its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not affected by this option. Options are
true or false. Default is false.
print_format
Set print format for stats. Options are summary, json, or none. Default value is none.
lowpass
Apply a low-pass filter with 3dB point frequency. The filter can be either single-pole or double-pole
(the default). The filter roll off at 6dB per pole per octave (20dB per pole per decade).
The filter accepts the following options:
frequency, f
Set frequency in Hz. Default is 500.
poles, p
Set number of poles. Default is 2.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Specify the band-width of a filter in width_type units. Applies only to double-pole filter. The
default is 0.707q and gives a Butterworth response.
channels, c
Specify which channels to filter, by default all available are filtered.
Examples
• Lowpass only LFE channel, it LFE is not present it does nothing:
lowpass=c=LFE
pan
Mix channels with specific gain levels. The filter accepts the output channel layout followed by a set of
channels definitions.
This filter is also designed to efficiently remap the channels of an audio stream.
The filter accepts parameters of the form: "l|outdef|outdef|..."
l output channel layout or number of channels
outdef
output channel specification, of the form: "out_name=[gain*]in_name[(+-)[gain*]in_name...]"
out_name
output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)
gain
multiplicative coefficient for the channel, 1 leaving the volume unchanged
in_name
input channel to use, see out_name for details; it is not possible to mix named and numbered input
channels
If the `=' in a channel specification is replaced by `<', then the gains for that specification will be
renormalized so that the total is 1, thus avoiding clipping noise.
Mixing examples
For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:
pan=1c|c0=0.9*c0+0.1*c1
A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround:
pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR
Note that ffmpeg integrates a default down-mix (and up-mix) system that should be preferred (see "-ac"
option) unless you have very specific needs.
Remapping examples
The channel remapping will be effective if, and only if:
*<gain coefficients are zeroes or ones,>
*<only one input per channel output,>
If all these conditions are satisfied, the filter will notify the user ("Pure channel mapping detected"),
and use an optimized and lossless method to do the remapping.
For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels:
pan="stereo| c0=FL | c1=FR"
Given the same source, you can also switch front left and front right channels and keep the input channel
layout:
pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"
If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo
channel layout) with:
pan="stereo|c1=c1"
Still with a stereo audio stream input, you can copy the right channel in both front left and right:
pan="stereo| c0=FR | c1=FR"
replaygain
ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it unchanged. At
end of filtering it displays "track_gain" and "track_peak".
resample
Convert the audio sample format, sample rate and channel layout. It is not meant to be used directly.
rubberband
Apply time-stretching and pitch-shifting with librubberband.
The filter accepts the following options:
tempo
Set tempo scale factor.
pitch
Set pitch scale factor.
transients
Set transients detector. Possible values are:
crisp
mixed
smooth
detector
Set detector. Possible values are:
compound
percussive
soft
phase
Set phase. Possible values are:
laminar
independent
window
Set processing window size. Possible values are:
standard
short
long
smoothing
Set smoothing. Possible values are:
off
on
formant
Enable formant preservation when shift pitching. Possible values are:
shifted
preserved
pitchq
Set pitch quality. Possible values are:
quality
speed
consistency
channels
Set channels. Possible values are:
apart
together
sidechaincompress
This filter acts like normal compressor but has the ability to compress detected signal using second
input signal. It needs two input streams and returns one output stream. First input stream will be
processed depending on second stream signal. The filtered signal then can be filtered with other filters
in later stages of processing. See pan and amerge filter.
The filter accepts the following options:
level_in
Set input gain. Default is 1. Range is between 0.015625 and 64.
threshold
If a signal of second stream raises above this level it will affect the gain reduction of first
stream. By default is 0.125. Range is between 0.00097563 and 1.
ratio
Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB above the
threshold, it will be only 2dB above after the reduction. Default is 2. Range is between 1 and 20.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction starts.
Default is 20. Range is between 0.01 and 2000.
release
Amount of milliseconds the signal has to fall below the threshold before reduction is decreased
again. Default is 250. Range is between 0.01 and 9000.
makeup
Set the amount by how much signal will be amplified after processing. Default is 1. Range is from 1
to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is 2.82843.
Range is between 1 and 8.
link
Choose if the "average" level between all channels of side-chain stream or the louder("maximum")
channel of side-chain stream affects the reduction. Default is "average".
detection
Should the exact signal be taken in case of "peak" or an RMS one in case of "rms". Default is "rms"
which is mainly smoother.
level_sc
Set sidechain gain. Default is 1. Range is between 0.015625 and 64.
mix How much to use compressed signal in output. Default is 1. Range is between 0 and 1.
Examples
• Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the signal of 2nd
input and later compressed signal to be merged with 2nd input:
ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"
sidechaingate
A sidechain gate acts like a normal (wideband) gate but has the ability to filter the detected signal
before sending it to the gain reduction stage. Normally a gate uses the full range signal to detect a
level above the threshold. For example: If you cut all lower frequencies from your sidechain signal the
gate will decrease the volume of your track only if not enough highs appear. With this technique you are
able to reduce the resonation of a natural drum or remove "rumbling" of muted strokes from a heavily
distorted guitar. It needs two input streams and returns one output stream. First input stream will be
processed depending on second stream signal.
The filter accepts the following options:
level_in
Set input level before filtering. Default is 1. Allowed range is from 0.015625 to 64.
range
Set the level of gain reduction when the signal is below the threshold. Default is 0.06125. Allowed
range is from 0 to 1.
threshold
If a signal rises above this level the gain reduction is released. Default is 0.125. Allowed range
is from 0 to 1.
ratio
Set a ratio about which the signal is reduced. Default is 2. Allowed range is from 1 to 9000.
attack
Amount of milliseconds the signal has to rise above the threshold before gain reduction stops.
Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
release
Amount of milliseconds the signal has to fall below the threshold before the reduction is increased
again. Default is 250 milliseconds. Allowed range is from 0.01 to 9000.
makeup
Set amount of amplification of signal after processing. Default is 1. Allowed range is from 1 to 64.
knee
Curve the sharp knee around the threshold to enter gain reduction more softly. Default is
2.828427125. Allowed range is from 1 to 8.
detection
Choose if exact signal should be taken for detection or an RMS like one. Default is rms. Can be peak
or rms.
link
Choose if the average level between all channels or the louder channel affects the reduction.
Default is average. Can be average or maximum.
level_sc
Set sidechain gain. Default is 1. Range is from 0.015625 to 64.
silencedetect
Detect silence in an audio stream.
This filter logs a message when it detects that the input audio volume is less or equal to a noise
tolerance value for a duration greater or equal to the minimum detected noise duration.
The printed times and duration are expressed in seconds.
The filter accepts the following options:
duration, d
Set silence duration until notification (default is 2 seconds).
noise, n
Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified value) or
amplitude ratio. Default is -60dB, or 0.001.
Examples
• Detect 5 seconds of silence with -50dB noise tolerance:
silencedetect=n=-50dB:d=5
• Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in silence.mp3:
ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -
silenceremove
Remove silence from the beginning, middle or end of the audio.
The filter accepts the following options:
start_periods
This value is used to indicate if audio should be trimmed at beginning of the audio. A value of zero
indicates no silence should be trimmed from the beginning. When specifying a non-zero value, it trims
audio up until it finds non-silence. Normally, when trimming silence from beginning of audio the
start_periods will be 1 but it can be increased to higher values to trim all audio up to specific
count of non-silence periods. Default value is 0.
start_duration
Specify the amount of time that non-silence must be detected before it stops trimming audio. By
increasing the duration, bursts of noises can be treated as silence and trimmed off. Default value is
0.
start_threshold
This indicates what sample value should be treated as silence. For digital audio, a value of 0 may be
fine but for audio recorded from analog, you may wish to increase the value to account for background
noise. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio.
Default value is 0.
stop_periods
Set the count for trimming silence from the end of audio. To remove silence from the middle of a
file, specify a stop_periods that is negative. This value is then treated as a positive value and is
used to indicate the effect should restart processing as specified by start_periods, making it
suitable for removing periods of silence in the middle of the audio. Default value is 0.
stop_duration
Specify a duration of silence that must exist before audio is not copied any more. By specifying a
higher duration, silence that is wanted can be left in the audio. Default value is 0.
stop_threshold
This is the same as start_threshold but for trimming silence from the end of audio. Can be specified
in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default value is 0.
leave_silence
This indicates that stop_duration length of audio should be left intact at the beginning of each
period of silence. For example, if you want to remove long pauses between words but do not want to
remove the pauses completely. Default value is 0.
detection
Set how is silence detected. Can be "rms" or "peak". Second is faster and works better with digital
silence which is exactly 0. Default value is "rms".
window
Set ratio used to calculate size of window for detecting silence. Default value is 0.02. Allowed
range is from 0 to 10.
Examples
• The following example shows how this filter can be used to start a recording that does not contain
the delay at the start which usually occurs between pressing the record button and the start of the
performance:
silenceremove=1:5:0.02
• Trim all silence encountered from beginning to end where there is more than 1 second of silence in
audio:
silenceremove=0:0:0:-1:1:-90dB
sofalizer
SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers around the user for
binaural listening via headphones (audio formats up to 9 channels supported). The HRTFs are stored in
SOFA files (see <http://www.sofacoustics.org/> for a database). SOFAlizer is developed at the Acoustics
Research Institute (ARI) of the Austrian Academy of Sciences.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libmysofa".
The filter accepts the following options:
sofa
Set the SOFA file used for rendering.
gain
Set gain applied to audio. Value is in dB. Default is 0.
rotation
Set rotation of virtual loudspeakers in deg. Default is 0.
elevation
Set elevation of virtual speakers in deg. Default is 0.
radius
Set distance in meters between loudspeakers and the listener with near-field HRTFs. Default is 1.
type
Set processing type. Can be time or freq. time is processing audio in time domain which is slow.
freq is processing audio in frequency domain which is fast. Default is freq.
speakers
Set custom positions of virtual loudspeakers. Syntax for this option is: <CH> <AZIM> <ELEV>[|<CH>
<AZIM> <ELEV>|...]. Each virtual loudspeaker is described with short channel name following with
azimuth and elevation in degrees. Each virtual loudspeaker description is separated by '|'. For
example to override front left and front right channel positions use: 'speakers=FL 45 15|FR 345 15'.
Descriptions with unrecognised channel names are ignored.
lfegain
Set custom gain for LFE channels. Value is in dB. Default is 0.
Examples
• Using ClubFritz6 sofa file:
sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1
• Using ClubFritz12 sofa file and bigger radius with small rotation:
sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5
• Similar as above but with custom speaker positions for front left, front right, back left and back
right and also with custom gain:
"sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"
stereotools
This filter has some handy utilities to manage stereo signals, for converting M/S stereo recordings to
L/R signal while having control over the parameters or spreading the stereo image of master track.
The filter accepts the following options:
level_in
Set input level before filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to
64.
level_out
Set output level after filtering for both channels. Defaults is 1. Allowed range is from 0.015625 to
64.
balance_in
Set input balance between both channels. Default is 0. Allowed range is from -1 to 1.
balance_out
Set output balance between both channels. Default is 0. Allowed range is from -1 to 1.
softclip
Enable softclipping. Results in analog distortion instead of harsh digital 0dB clipping. Disabled by
default.
mutel
Mute the left channel. Disabled by default.
muter
Mute the right channel. Disabled by default.
phasel
Change the phase of the left channel. Disabled by default.
phaser
Change the phase of the right channel. Disabled by default.
mode
Set stereo mode. Available values are:
lr>lr
Left/Right to Left/Right, this is default.
lr>ms
Left/Right to Mid/Side.
ms>lr
Mid/Side to Left/Right.
lr>ll
Left/Right to Left/Left.
lr>rr
Left/Right to Right/Right.
lr>l+r
Left/Right to Left + Right.
lr>rl
Left/Right to Right/Left.
ms>ll
Mid/Side to Left/Left.
ms>rr
Mid/Side to Right/Right.
slev
Set level of side signal. Default is 1. Allowed range is from 0.015625 to 64.
sbal
Set balance of side signal. Default is 0. Allowed range is from -1 to 1.
mlev
Set level of the middle signal. Default is 1. Allowed range is from 0.015625 to 64.
mpan
Set middle signal pan. Default is 0. Allowed range is from -1 to 1.
base
Set stereo base between mono and inversed channels. Default is 0. Allowed range is from -1 to 1.
delay
Set delay in milliseconds how much to delay left from right channel and vice versa. Default is 0.
Allowed range is from -20 to 20.
sclevel
Set S/C level. Default is 1. Allowed range is from 1 to 100.
phase
Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.
bmode_in, bmode_out
Set balance mode for balance_in/balance_out option.
Can be one of the following:
balance
Classic balance mode. Attenuate one channel at time. Gain is raised up to 1.
amplitude
Similar as classic mode above but gain is raised up to 2.
power
Equal power distribution, from -6dB to +6dB range.
Examples
• Apply karaoke like effect:
stereotools=mlev=0.015625
• Convert M/S signal to L/R:
"stereotools=mode=ms>lr"
stereowiden
This filter enhance the stereo effect by suppressing signal common to both channels and by delaying the
signal of left into right and vice versa, thereby widening the stereo effect.
The filter accepts the following options:
delay
Time in milliseconds of the delay of left signal into right and vice versa. Default is 20
milliseconds.
feedback
Amount of gain in delayed signal into right and vice versa. Gives a delay effect of left signal in
right output and vice versa which gives widening effect. Default is 0.3.
crossfeed
Cross feed of left into right with inverted phase. This helps in suppressing the mono. If the value
is 1 it will cancel all the signal common to both channels. Default is 0.3.
drymix
Set level of input signal of original channel. Default is 0.8.
superequalizer
Apply 18 band equalizer.
The filter accepts the following options:
1b Set 65Hz band gain.
2b Set 92Hz band gain.
3b Set 131Hz band gain.
4b Set 185Hz band gain.
5b Set 262Hz band gain.
6b Set 370Hz band gain.
7b Set 523Hz band gain.
8b Set 740Hz band gain.
9b Set 1047Hz band gain.
10b Set 1480Hz band gain.
11b Set 2093Hz band gain.
12b Set 2960Hz band gain.
13b Set 4186Hz band gain.
14b Set 5920Hz band gain.
15b Set 8372Hz band gain.
16b Set 11840Hz band gain.
17b Set 16744Hz band gain.
18b Set 20000Hz band gain.
surround
Apply audio surround upmix filter.
This filter allows to produce multichannel output from audio stream.
The filter accepts the following options:
chl_out
Set output channel layout. By default, this is 5.1.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
chl_in
Set input channel layout. By default, this is stereo.
See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.
level_in
Set input volume level. By default, this is 1.
level_out
Set output volume level. By default, this is 1.
lfe Enable LFE channel output if output channel layout has it. By default, this is enabled.
lfe_low
Set LFE low cut off frequency. By default, this is 128 Hz.
lfe_high
Set LFE high cut off frequency. By default, this is 256 Hz.
fc_in
Set front center input volume. By default, this is 1.
fc_out
Set front center output volume. By default, this is 1.
lfe_in
Set LFE input volume. By default, this is 1.
lfe_out
Set LFE output volume. By default, this is 1.
treble
Boost or cut treble (upper) frequencies of the audio using a two-pole shelving filter with a response
similar to that of a standard hi-fi's tone-controls. This is also known as shelving equalisation (EQ).
The filter accepts the following options:
gain, g
Give the gain at whichever is the lower of ~22 kHz and the Nyquist frequency. Its useful range is
about -20 (for a large cut) to +20 (for a large boost). Beware of clipping when using a positive
gain.
frequency, f
Set the filter's central frequency and so can be used to extend or reduce the frequency range to be
boosted or cut. The default value is 3000 Hz.
width_type, t
Set method to specify band-width of filter.
h Hz
q Q-Factor
o octave
s slope
width, w
Determine how steep is the filter's shelf transition.
channels, c
Specify which channels to filter, by default all available are filtered.
tremolo
Sinusoidal amplitude modulation.
The filter accepts the following options:
f Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz or lower) will
result in a tremolo effect. This filter may also be used as a ring modulator by specifying a
modulation frequency higher than 20 Hz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.
vibrato
Sinusoidal phase modulation.
The filter accepts the following options:
f Modulation frequency in Hertz. Range is 0.1 - 20000.0. Default value is 5.0 Hz.
d Depth of modulation as a percentage. Range is 0.0 - 1.0. Default value is 0.5.
volume
Adjust the input audio volume.
It accepts the following parameters:
volume
Set audio volume expression.
Output values are clipped to the maximum value.
The output audio volume is given by the relation:
<output_volume> = <volume> * <input_volume>
The default value for volume is "1.0".
precision
This parameter represents the mathematical precision.
It determines which input sample formats will be allowed, which affects the precision of the volume
scaling.
fixed
8-bit fixed-point; this limits input sample format to U8, S16, and S32.
float
32-bit floating-point; this limits input sample format to FLT. (default)
double
64-bit floating-point; this limits input sample format to DBL.
replaygain
Choose the behaviour on encountering ReplayGain side data in input frames.
drop
Remove ReplayGain side data, ignoring its contents (the default).
ignore
Ignore ReplayGain side data, but leave it in the frame.
track
Prefer the track gain, if present.
album
Prefer the album gain, if present.
replaygain_preamp
Pre-amplification gain in dB to apply to the selected replaygain gain.
Default value for replaygain_preamp is 0.0.
eval
Set when the volume expression is evaluated.
It accepts the following values:
once
only evaluate expression once during the filter initialization, or when the volume command is
sent
frame
evaluate expression for each incoming frame
Default value is once.
The volume expression can contain the following parameters.
n frame number (starting at zero)
nb_channels
number of channels
nb_consumed_samples
number of samples consumed by the filter
nb_samples
number of samples in the current frame
pos original frame position in the file
pts frame PTS
sample_rate
sample rate
startpts
PTS at start of stream
startt
time at start of stream
t frame time
tb timestamp timebase
volume
last set volume value
Note that when eval is set to once only the sample_rate and tb variables are available, all other
variables will evaluate to NAN.
Commands
This filter supports the following commands:
volume
Modify the volume expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
replaygain_noclip
Prevent clipping by limiting the gain applied.
Default value for replaygain_noclip is 1.
Examples
• Halve the input audio volume:
volume=volume=0.5
volume=volume=1/2
volume=volume=-6.0206dB
In all the above example the named key for volume can be omitted, for example like in:
volume=0.5
• Increase input audio power by 6 decibels using fixed-point precision:
volume=volume=6dB:precision=fixed
• Fade volume after time 10 with an annihilation period of 5 seconds:
volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame
volumedetect
Detect the volume of the input video.
The filter has no parameters. The input is not modified. Statistics about the volume will be printed in
the log when the input stream end is reached.
In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis),
and the beginning of a histogram of the registered volume values (from the maximum value to a cumulated
1/1000 of the samples).
All volumes are in decibels relative to the maximum PCM value.
Examples
Here is an excerpt of the output:
[Parsed_volumedetect_0 0xa23120] mean_volume: -27 dB
[Parsed_volumedetect_0 0xa23120] max_volume: -4 dB
[Parsed_volumedetect_0 0xa23120] histogram_4db: 6
[Parsed_volumedetect_0 0xa23120] histogram_5db: 62
[Parsed_volumedetect_0 0xa23120] histogram_6db: 286
[Parsed_volumedetect_0 0xa23120] histogram_7db: 1042
[Parsed_volumedetect_0 0xa23120] histogram_8db: 2551
[Parsed_volumedetect_0 0xa23120] histogram_9db: 4609
[Parsed_volumedetect_0 0xa23120] histogram_10db: 8409
It means that:
• The mean square energy is approximately -27 dB, or 10^-2.7.
• The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.
• There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.
In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes
clipping for 6 samples, etc.
AUDIO SOURCES
Below is a description of the currently available audio sources.
abuffer
Buffer audio frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in
libavfilter/asrc_abuffer.h.
It accepts the following parameters:
time_base
The timebase which will be used for timestamps of submitted frames. It must be either a floating-
point number or in numerator/denominator form.
sample_rate
The sample rate of the incoming audio buffers.
sample_fmt
The sample format of the incoming audio buffers. Either a sample format name or its corresponding
integer representation from the enum AVSampleFormat in libavutil/samplefmt.h
channel_layout
The channel layout of the incoming audio buffers. Either a channel layout name from
channel_layout_map in libavutil/channel_layout.c or its corresponding integer representation from the
AV_CH_LAYOUT_* macros in libavutil/channel_layout.h
channels
The number of channels of the incoming audio buffers. If both channels and channel_layout are
specified, then they must be consistent.
Examples
abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo
will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with
name "s16p" corresponds to the number 6 and the "stereo" channel layout corresponds to the value 0x3,
this is equivalent to:
abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3
aevalsrc
Generate an audio signal specified by an expression.
This source accepts in input one or more expressions (one for each channel), which are evaluated and used
to generate a corresponding audio signal.
This source accepts the following options:
exprs
Set the '|'-separated expressions list for each separate channel. In case the channel_layout option
is not specified, the selected channel layout depends on the number of provided expressions.
Otherwise the last specified expression is applied to the remaining output channels.
channel_layout, c
Set the channel layout. The number of channels in the specified layout must be equal to the number of
specified expressions.
duration, d
Set the minimum duration of the sourced audio. See the Time duration section in the ffmpeg-utils(1)
manual for the accepted syntax. Note that the resulting duration may be greater than the specified
duration, as the generated audio is always cut at the end of a complete frame.
If not specified, or the expressed duration is negative, the audio is supposed to be generated
forever.
nb_samples, n
Set the number of samples per channel per each output frame, default to 1024.
sample_rate, s
Specify the sample rate, default to 44100.
Each expression in exprs can contain the following constants:
n number of the evaluated sample, starting from 0
t time of the evaluated sample expressed in seconds, starting from 0
s sample rate
Examples
• Generate silence:
aevalsrc=0
• Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:
aevalsrc="sin(440*2*PI*t):s=8000"
• Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:
aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"
• Generate white noise:
aevalsrc="-2+random(0)"
• Generate an amplitude modulated signal:
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"
• Generate 2.5 Hz binaural beats on a 360 Hz carrier:
aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"
anullsrc
The null audio source, return unprocessed audio frames. It is mainly useful as a template and to be
employed in analysis / debugging tools, or as the source for filters which ignore the input data (for
example the sox synth filter).
This source accepts the following options:
channel_layout, cl
Specifies the channel layout, and can be either an integer or a string representing a channel layout.
The default value of channel_layout is "stereo".
Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping between strings
and channel layout values.
sample_rate, r
Specifies the sample rate, and defaults to 44100.
nb_samples, n
Set the number of samples per requested frames.
Examples
• Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.
anullsrc=r=48000:cl=4
• Do the same operation with a more obvious syntax:
anullsrc=r=48000:cl=mono
All the parameters need to be explicitly defined.
flite
Synthesize a voice utterance using the libflite library.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libflite".
Note that the flite library is not thread-safe.
The filter accepts the following options:
list_voices
If set to 1, list the names of the available voices and exit immediately. Default value is 0.
nb_samples, n
Set the maximum number of samples per frame. Default value is 512.
textfile
Set the filename containing the text to speak.
text
Set the text to speak.
voice, v
Set the voice to use for the speech synthesis. Default value is "kal". See also the list_voices
option.
Examples
• Read from file speech.txt, and synthesize the text using the standard flite voice:
flite=textfile=speech.txt
• Read the specified text selecting the "slt" voice:
flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
• Input text to ffmpeg:
ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt
• Make ffplay speak the specified text, using "flite" and the "lavfi" device:
ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'
For more information about libflite, check: <http://www.speech.cs.cmu.edu/flite/>
anoisesrc
Generate a noise audio signal.
The filter accepts the following options:
sample_rate, r
Specify the sample rate. Default value is 48000 Hz.
amplitude, a
Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.
duration, d
Specify the duration of the generated audio stream. Not specifying this option results in noise with
an infinite length.
color, colour, c
Specify the color of noise. Available noise colors are white, pink, brown, blue and violet. Default
color is white.
seed, s
Specify a value used to seed the PRNG.
nb_samples, n
Set the number of samples per each output frame, default is 1024.
Examples
• Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of 0.5:
anoisesrc=d=60:c=pink:r=44100:a=0.5
sine
Generate an audio signal made of a sine wave with amplitude 1/8.
The audio signal is bit-exact.
The filter accepts the following options:
frequency, f
Set the carrier frequency. Default is 440 Hz.
beep_factor, b
Enable a periodic beep every second with frequency beep_factor times the carrier frequency. Default
is 0, meaning the beep is disabled.
sample_rate, r
Specify the sample rate, default is 44100.
duration, d
Specify the duration of the generated audio stream.
samples_per_frame
Set the number of samples per output frame.
The expression can contain the following constants:
n The (sequential) number of the output audio frame, starting from 0.
pts The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.
t The PTS of the output audio frame, expressed in seconds.
TB The timebase of the output audio frames.
Default is 1024.
Examples
• Generate a simple 440 Hz sine wave:
sine
• Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:
sine=220:4:d=5
sine=f=220:b=4:d=5
sine=frequency=220:beep_factor=4:duration=5
• Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602" NTSC pattern:
sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'
AUDIO SINKS
Below is a description of the currently available audio sinks.
abuffersink
Buffer audio frames, and make them available to the end of filter chain.
This sink is mainly intended for programmatic use, in particular through the interface defined in
libavfilter/buffersink.h or the options system.
It accepts a pointer to an AVABufferSinkContext structure, which defines the incoming buffers' formats,
to be passed as the opaque parameter to "avfilter_init_filter" for initialization.
anullsink
Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for
use in analysis / debugging tools.
VIDEO FILTERS
When you configure your FFmpeg build, you can disable any of the existing filters using
"--disable-filters". The configure output will show the video filters included in your build.
Below is a description of the currently available video filters.
alphaextract
Extract the alpha component from the input as a grayscale video. This is especially useful with the
alphamerge filter.
alphamerge
Add or replace the alpha component of the primary input with the grayscale value of a second input. This
is intended for use with alphaextract to allow the transmission or storage of frame sequences that have
alpha in a format that doesn't support an alpha channel.
For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with
alphaextract, you might use:
movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]
Since this filter is designed for reconstruction, it operates on frame sequences without considering
timestamps, and terminates when either input reaches end of stream. This will cause problems if your
encoding pipeline drops frames. If you're trying to apply an image as an overlay to a video stream,
consider the overlay filter instead.
ass
Same as the subtitles filter, except that it doesn't require libavcodec and libavformat to work. On the
other hand, it is limited to ASS (Advanced Substation Alpha) subtitles files.
This filter accepts the following option in addition to the common options from the subtitles filter:
shaping
Set the shaping engine
Available values are:
auto
The default libass shaping engine, which is the best available.
simple
Fast, font-agnostic shaper that can do only substitutions
complex
Slower shaper using OpenType for substitutions and positioning
The default is "auto".
atadenoise
Apply an Adaptive Temporal Averaging Denoiser to the video input.
The filter accepts the following options:
0a Set threshold A for 1st plane. Default is 0.02. Valid range is 0 to 0.3.
0b Set threshold B for 1st plane. Default is 0.04. Valid range is 0 to 5.
1a Set threshold A for 2nd plane. Default is 0.02. Valid range is 0 to 0.3.
1b Set threshold B for 2nd plane. Default is 0.04. Valid range is 0 to 5.
2a Set threshold A for 3rd plane. Default is 0.02. Valid range is 0 to 0.3.
2b Set threshold B for 3rd plane. Default is 0.04. Valid range is 0 to 5.
Threshold A is designed to react on abrupt changes in the input signal and threshold B is designed to
react on continuous changes in the input signal.
s Set number of frames filter will use for averaging. Default is 33. Must be odd number in range [5,
129].
p Set what planes of frame filter will use for averaging. Default is all.
avgblur
Apply average blur filter.
The filter accepts the following options:
sizeX
Set horizontal kernel size.
planes
Set which planes to filter. By default all planes are filtered.
sizeY
Set vertical kernel size, if zero it will be same as "sizeX". Default is 0.
bbox
Compute the bounding box for the non-black pixels in the input frame luminance plane.
This filter computes the bounding box containing all the pixels with a luminance value greater than the
minimum allowed value. The parameters describing the bounding box are printed on the filter log.
The filter accepts the following option:
min_val
Set the minimal luminance value. Default is 16.
bitplanenoise
Show and measure bit plane noise.
The filter accepts the following options:
bitplane
Set which plane to analyze. Default is 1.
filter
Filter out noisy pixels from "bitplane" set above. Default is disabled.
blackdetect
Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions,
commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the
detected black interval expressed in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
The filter accepts the following options:
black_min_duration, d
Set the minimum detected black duration expressed in seconds. It must be a non-negative floating
point number.
Default value is 2.0.
picture_black_ratio_th, pic_th
Set the threshold for considering a picture "black". Express the minimum value for the ratio:
<nb_black_pixels> / <nb_pixels>
for which a picture is considered black. Default value is 0.98.
pixel_black_th, pix_th
Set the threshold for considering a pixel "black".
The threshold expresses the maximum pixel luminance value for which a pixel is considered "black".
The provided value is scaled according to the following equation:
<absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>
luminance_range_size and luminance_minimum_value depend on the input video format, the range is
[0-255] for YUV full-range formats and [16-235] for YUV non full-range formats.
Default value is 0.10.
The following example sets the maximum pixel threshold to the minimum value, and detects only black
intervals of 2 or more seconds:
blackdetect=d=2:pix_th=0.00
blackframe
Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or
commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness,
the position in the file if known or -1 and the timestamp in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
This filter exports frame metadata "lavfi.blackframe.pblack". The value represents the percentage of
pixels in the picture that are below the threshold value.
It accepts the following parameters:
amount
The percentage of the pixels that have to be below the threshold; it defaults to 98.
threshold, thresh
The threshold below which a pixel value is considered black; it defaults to 32.
blend, tblend
Blend two video frames into each other.
The "blend" filter takes two input streams and outputs one stream, the first input is the "top" layer and
second input is "bottom" layer. By default, the output terminates when the longest input terminates.
The "tblend" (time blend) filter takes two consecutive frames from one single stream, and outputs the
result obtained by blending the new frame on top of the old frame.
A description of the accepted options follows.
c0_mode
c1_mode
c2_mode
c3_mode
all_mode
Set blend mode for specific pixel component or all pixel components in case of all_mode. Default
value is "normal".
Available values for component modes are:
addition
grainmerge
and
average
burn
darken
difference
grainextract
divide
dodge
freeze
exclusion
extremity
glow
hardlight
hardmix
heat
lighten
linearlight
multiply
multiply128
negation
normal
or
overlay
phoenix
pinlight
reflect
screen
softlight
subtract
vividlight
xor
c0_opacity
c1_opacity
c2_opacity
c3_opacity
all_opacity
Set blend opacity for specific pixel component or all pixel components in case of all_opacity. Only
used in combination with pixel component blend modes.
c0_expr
c1_expr
c2_expr
c3_expr
all_expr
Set blend expression for specific pixel component or all pixel components in case of all_expr. Note
that related mode options will be ignored if those are set.
The expressions can use the following variables:
N The sequential number of the filtered frame, starting from 0.
X
Y the coordinates of the current sample
W
H the width and height of currently filtered plane
SW
SH Width and height scale depending on the currently filtered plane. It is the ratio between the
corresponding luma plane number of pixels and the current plane ones. E.g. for YUV4:2:0 the
values are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.
T Time of the current frame, expressed in seconds.
TOP, A
Value of pixel component at current location for first video frame (top layer).
BOTTOM, B
Value of pixel component at current location for second video frame (bottom layer).
The "blend" filter also supports the framesync options.
Examples
• Apply transition from bottom layer to top layer in first 10 seconds:
blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'
• Apply linear horizontal transition from top layer to bottom layer:
blend=all_expr='A*(X/W)+B*(1-X/W)'
• Apply 1x1 checkerboard effect:
blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'
• Apply uncover left effect:
blend=all_expr='if(gte(N*SW+X,W),A,B)'
• Apply uncover down effect:
blend=all_expr='if(gte(Y-N*SH,0),A,B)'
• Apply uncover up-left effect:
blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'
• Split diagonally video and shows top and bottom layer on each side:
blend=all_expr='if(gt(X,Y*(W/H)),A,B)'
• Display differences between the current and the previous frame:
tblend=all_mode=grainextract
boxblur
Apply a boxblur algorithm to the input video.
It accepts the following parameters:
luma_radius, lr
luma_power, lp
chroma_radius, cr
chroma_power, cp
alpha_radius, ar
alpha_power, ap
A description of the accepted options follows.
luma_radius, lr
chroma_radius, cr
alpha_radius, ar
Set an expression for the box radius in pixels used for blurring the corresponding input plane.
The radius value must be a non-negative number, and must not be greater than the value of the
expression "min(w,h)/2" for the luma and alpha planes, and of "min(cw,ch)/2" for the chroma planes.
Default value for luma_radius is "2". If not specified, chroma_radius and alpha_radius default to the
corresponding value set for luma_radius.
The expressions can contain the following constants:
w
h The input width and height in pixels.
cw
ch The input chroma image width and height in pixels.
hsub
vsub
The horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p",
hsub is 2 and vsub is 1.
luma_power, lp
chroma_power, cp
alpha_power, ap
Specify how many times the boxblur filter is applied to the corresponding plane.
Default value for luma_power is 2. If not specified, chroma_power and alpha_power default to the
corresponding value set for luma_power.
A value of 0 will disable the effect.
Examples
• Apply a boxblur filter with the luma, chroma, and alpha radii set to 2:
boxblur=luma_radius=2:luma_power=1
boxblur=2:1
• Set the luma radius to 2, and alpha and chroma radius to 0:
boxblur=2:1:cr=0:ar=0
• Set the luma and chroma radii to a fraction of the video dimension:
boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1
bwdif
Deinterlace the input video ("bwdif" stands for "Bob Weaver Deinterlacing Filter").
Motion adaptive deinterlacing based on yadif with the use of w3fdif and cubic interpolation algorithms.
It accepts the following parameters:
mode
The interlacing mode to adopt. It accepts one of the following values:
0, send_frame
Output one frame for each frame.
1, send_field
Output one frame for each field.
The default value is "send_field".
parity
The picture field parity assumed for the input interlaced video. It accepts one of the following
values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.
The default value is "auto". If the interlacing is unknown or the decoder does not export this
information, top field first will be assumed.
deint
Specify which frames to deinterlace. Accept one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
chromakey
YUV colorspace color/chroma keying.
The filter accepts the following options:
color
The color which will be replaced with transparency.
similarity
Similarity percentage with the key color.
0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage.
0.0 makes pixels either fully transparent, or not transparent at all.
Higher values result in semi-transparent pixels, with a higher transparency the more similar the
pixels color is to the key color.
yuv Signals that the color passed is already in YUV instead of RGB.
Literal colors like "green" or "red" don't make sense with this enabled anymore. This can be used to
pass exact YUV values as hexadecimal numbers.
Examples
• Make every green pixel in the input image transparent:
ffmpeg -i input.png -vf chromakey=green out.png
• Overlay a greenscreen-video on top of a static black background.
ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv
ciescope
Display CIE color diagram with pixels overlaid onto it.
The filter accepts the following options:
system
Set color system.
ntsc, 470m
ebu, 470bg
smpte
240m
apple
widergb
cie1931
rec709, hdtv
uhdtv, rec2020
cie Set CIE system.
xyy
ucs
luv
gamuts
Set what gamuts to draw.
See "system" option for available values.
size, s
Set ciescope size, by default set to 512.
intensity, i
Set intensity used to map input pixel values to CIE diagram.
contrast
Set contrast used to draw tongue colors that are out of active color system gamut.
corrgamma
Correct gamma displayed on scope, by default enabled.
showwhite
Show white point on CIE diagram, by default disabled.
gamma
Set input gamma. Used only with XYZ input color space.
codecview
Visualize information exported by some codecs.
Some codecs can export information through frames using side-data or other means. For example, some MPEG
based codecs export motion vectors through the export_mvs flag in the codec flags2 option.
The filter accepts the following option:
mv Set motion vectors to visualize.
Available flags for mv are:
pf forward predicted MVs of P-frames
bf forward predicted MVs of B-frames
bb backward predicted MVs of B-frames
qp Display quantization parameters using the chroma planes.
mv_type, mvt
Set motion vectors type to visualize. Includes MVs from all frames unless specified by frame_type
option.
Available flags for mv_type are:
fp forward predicted MVs
bp backward predicted MVs
frame_type, ft
Set frame type to visualize motion vectors of.
Available flags for frame_type are:
if intra-coded frames (I-frames)
pf predicted frames (P-frames)
bf bi-directionally predicted frames (B-frames)
Examples
• Visualize forward predicted MVs of all frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp
• Visualize multi-directionals MVs of P and B-Frames using ffplay:
ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb
colorbalance
Modify intensity of primary colors (red, green and blue) of input frames.
The filter allows an input frame to be adjusted in the shadows, midtones or highlights regions for the
red-cyan, green-magenta or blue-yellow balance.
A positive adjustment value shifts the balance towards the primary color, a negative value towards the
complementary color.
The filter accepts the following options:
rs
gs
bs Adjust red, green and blue shadows (darkest pixels).
rm
gm
bm Adjust red, green and blue midtones (medium pixels).
rh
gh
bh Adjust red, green and blue highlights (brightest pixels).
Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.
Examples
• Add red color cast to shadows:
colorbalance=rs=.3
colorkey
RGB colorspace color keying.
The filter accepts the following options:
color
The color which will be replaced with transparency.
similarity
Similarity percentage with the key color.
0.01 matches only the exact key color, while 1.0 matches everything.
blend
Blend percentage.
0.0 makes pixels either fully transparent, or not transparent at all.
Higher values result in semi-transparent pixels, with a higher transparency the more similar the
pixels color is to the key color.
Examples
• Make every green pixel in the input image transparent:
ffmpeg -i input.png -vf colorkey=green out.png
• Overlay a greenscreen-video on top of a static background image.
ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv
colorlevels
Adjust video input frames using levels.
The filter accepts the following options:
rimin
gimin
bimin
aimin
Adjust red, green, blue and alpha input black point. Allowed ranges for options are "[-1.0, 1.0]".
Defaults are 0.
rimax
gimax
bimax
aimax
Adjust red, green, blue and alpha input white point. Allowed ranges for options are "[-1.0, 1.0]".
Defaults are 1.
Input levels are used to lighten highlights (bright tones), darken shadows (dark tones), change the
balance of bright and dark tones.
romin
gomin
bomin
aomin
Adjust red, green, blue and alpha output black point. Allowed ranges for options are "[0, 1.0]".
Defaults are 0.
romax
gomax
bomax
aomax
Adjust red, green, blue and alpha output white point. Allowed ranges for options are "[0, 1.0]".
Defaults are 1.
Output levels allows manual selection of a constrained output level range.
Examples
• Make video output darker:
colorlevels=rimin=0.058:gimin=0.058:bimin=0.058
• Increase contrast:
colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96
• Make video output lighter:
colorlevels=rimax=0.902:gimax=0.902:bimax=0.902
• Increase brightness:
colorlevels=romin=0.5:gomin=0.5:bomin=0.5
colorchannelmixer
Adjust video input frames by re-mixing color channels.
This filter modifies a color channel by adding the values associated to the other channels of the same
pixels. For example if the value to modify is red, the output value will be:
<red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>
The filter accepts the following options:
rr
rg
rb
ra Adjust contribution of input red, green, blue and alpha channels for output red channel. Default is
1 for rr, and 0 for rg, rb and ra.
gr
gg
gb
ga Adjust contribution of input red, green, blue and alpha channels for output green channel. Default
is 1 for gg, and 0 for gr, gb and ga.
br
bg
bb
ba Adjust contribution of input red, green, blue and alpha channels for output blue channel. Default is
1 for bb, and 0 for br, bg and ba.
ar
ag
ab
aa Adjust contribution of input red, green, blue and alpha channels for output alpha channel. Default
is 1 for aa, and 0 for ar, ag and ab.
Allowed ranges for options are "[-2.0, 2.0]".
Examples
• Convert source to grayscale:
colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3
• Simulate sepia tones:
colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131
colormatrix
Convert color matrix.
The filter accepts the following options:
src
dst Specify the source and destination color matrix. Both values must be specified.
The accepted values are:
bt709
BT.709
fcc FCC
bt601
BT.601
bt470
BT.470
bt470bg
BT.470BG
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
bt2020
BT.2020
For example to convert from BT.601 to SMPTE-240M, use the command:
colormatrix=bt601:smpte240m
colorspace
Convert colorspace, transfer characteristics or color primaries. Input video needs to have an even size.
The filter accepts the following options:
all Specify all color properties at once.
The accepted values are:
bt470m
BT.470M
bt470bg
BT.470BG
bt601-6-525
BT.601-6 525
bt601-6-625
BT.601-6 625
bt709
BT.709
smpte170m
SMPTE-170M
smpte240m
SMPTE-240M
bt2020
BT.2020
space
Specify output colorspace.
The accepted values are:
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
ycgco
YCgCo
bt2020ncl
BT.2020 with non-constant luminance
trc Specify output transfer characteristics.
The accepted values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG
gamma22
Constant gamma of 2.2
gamma28
Constant gamma of 2.8
smpte170m
SMPTE-170M, BT.601-6 625 or BT.601-6 525
smpte240m
SMPTE-240M
srgb
SRGB
iec61966-2-1
iec61966-2-1
iec61966-2-4
iec61966-2-4
xvycc
xvycc
bt2020-10
BT.2020 for 10-bits content
bt2020-12
BT.2020 for 12-bits content
primaries
Specify output color primaries.
The accepted values are:
bt709
BT.709
bt470m
BT.470M
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
film
film
smpte431
SMPTE-431
smpte432
SMPTE-432
bt2020
BT.2020
jedec-p22
JEDEC P22 phosphors
range
Specify output color range.
The accepted values are:
tv TV (restricted) range
mpeg
MPEG (restricted) range
pc PC (full) range
jpeg
JPEG (full) range
format
Specify output color format.
The accepted values are:
yuv420p
YUV 4:2:0 planar 8-bits
yuv420p10
YUV 4:2:0 planar 10-bits
yuv420p12
YUV 4:2:0 planar 12-bits
yuv422p
YUV 4:2:2 planar 8-bits
yuv422p10
YUV 4:2:2 planar 10-bits
yuv422p12
YUV 4:2:2 planar 12-bits
yuv444p
YUV 4:4:4 planar 8-bits
yuv444p10
YUV 4:4:4 planar 10-bits
yuv444p12
YUV 4:4:4 planar 12-bits
fast
Do a fast conversion, which skips gamma/primary correction. This will take significantly less CPU,
but will be mathematically incorrect. To get output compatible with that produced by the colormatrix
filter, use fast=1.
dither
Specify dithering mode.
The accepted values are:
none
No dithering
fsb Floyd-Steinberg dithering
wpadapt
Whitepoint adaptation mode.
The accepted values are:
bradford
Bradford whitepoint adaptation
vonkries
von Kries whitepoint adaptation
identity
identity whitepoint adaptation (i.e. no whitepoint adaptation)
iall
Override all input properties at once. Same accepted values as all.
ispace
Override input colorspace. Same accepted values as space.
iprimaries
Override input color primaries. Same accepted values as primaries.
itrc
Override input transfer characteristics. Same accepted values as trc.
irange
Override input color range. Same accepted values as range.
The filter converts the transfer characteristics, color space and color primaries to the specified user
values. The output value, if not specified, is set to a default value based on the "all" property. If
that property is also not specified, the filter will log an error. The output color range and format
default to the same value as the input color range and format. The input transfer characteristics, color
space, color primaries and color range should be set on the input data. If any of these are missing, the
filter will log an error and no conversion will take place.
For example to convert the input to SMPTE-240M, use the command:
colorspace=smpte240m
convolution
Apply convolution 3x3 or 5x5 filter.
The filter accepts the following options:
0m
1m
2m
3m Set matrix for each plane. Matrix is sequence of 9 or 25 signed integers.
0rdiv
1rdiv
2rdiv
3rdiv
Set multiplier for calculated value for each plane.
0bias
1bias
2bias
3bias
Set bias for each plane. This value is added to the result of the multiplication. Useful for making
the overall image brighter or darker. Default is 0.0.
Examples
• Apply sharpen:
convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"
• Apply blur:
convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"
• Apply edge enhance:
convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"
• Apply edge detect:
convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"
• Apply laplacian edge detector which includes diagonals:
convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"
• Apply emboss:
convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"
convolve
Apply 2D convolution of video stream in frequency domain using second stream as impulse.
The filter accepts the following options:
planes
Set which planes to process.
impulse
Set which impulse video frames will be processed, can be first or all. Default is all.
The "convolve" filter also supports the framesync options.
copy
Copy the input video source unchanged to the output. This is mainly useful for testing purposes.
coreimage
Video filtering on GPU using Apple's CoreImage API on OSX.
Hardware acceleration is based on an OpenGL context. Usually, this means it is processed by video
hardware. However, software-based OpenGL implementations exist which means there is no guarantee for
hardware processing. It depends on the respective OSX.
There are many filters and image generators provided by Apple that come with a large variety of options.
The filter has to be referenced by its name along with its options.
The coreimage filter accepts the following options:
list_filters
List all available filters and generators along with all their respective options as well as possible
minimum and maximum values along with the default values.
list_filters=true
filter
Specify all filters by their respective name and options. Use list_filters to determine all valid
filter names and options. Numerical options are specified by a float value and are automatically
clamped to their respective value range. Vector and color options have to be specified by a list of
space separated float values. Character escaping has to be done. A special option name "default" is
available to use default options for a filter.
It is required to specify either "default" or at least one of the filter options. All omitted
options are used with their default values. The syntax of the filter string is as follows:
filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]
output_rect
Specify a rectangle where the output of the filter chain is copied into the input image. It is given
by a list of space separated float values:
output_rect=x\ y\ width\ height
If not given, the output rectangle equals the dimensions of the input image. The output rectangle is
automatically cropped at the borders of the input image. Negative values are valid for each
component.
output_rect=25\ 25\ 100\ 100
Several filters can be chained for successive processing without GPU-HOST transfers allowing for fast
processing of complex filter chains. Currently, only filters with zero (generators) or exactly one
(filters) input image and one output image are supported. Also, transition filters are not yet usable as
intended.
Some filters generate output images with additional padding depending on the respective filter kernel.
The padding is automatically removed to ensure the filter output has the same size as the input image.
For image generators, the size of the output image is determined by the previous output image of the
filter chain or the input image of the whole filterchain, respectively. The generators do not use the
pixel information of this image to generate their output. However, the generated output is blended onto
this image, resulting in partial or complete coverage of the output image.
The coreimagesrc video source can be used for generating input images which are directly fed into the
filter chain. By using it, providing input images by another video source or an input video is not
required.
Examples
• List all filters available:
coreimage=list_filters=true
• Use the CIBoxBlur filter with default options to blur an image:
coreimage=filter=CIBoxBlur@default
• Use a filter chain with CISepiaTone at default values and CIVignetteEffect with its center at 100x100
and a radius of 50 pixels:
coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50
• Use nullsrc and CIQRCodeGenerator to create a QR code for the FFmpeg homepage, given as complete and
escaped command-line for Apple's standard bash shell:
ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
crop
Crop the input video to given dimensions.
It accepts the following parameters:
w, out_w
The width of the output video. It defaults to "iw". This expression is evaluated only once during
the filter configuration, or when the w or out_w command is sent.
h, out_h
The height of the output video. It defaults to "ih". This expression is evaluated only once during
the filter configuration, or when the h or out_h command is sent.
x The horizontal position, in the input video, of the left edge of the output video. It defaults to
"(in_w-out_w)/2". This expression is evaluated per-frame.
y The vertical position, in the input video, of the top edge of the output video. It defaults to
"(in_h-out_h)/2". This expression is evaluated per-frame.
keep_aspect
If set to 1 will force the output display aspect ratio to be the same of the input, by changing the
output sample aspect ratio. It defaults to 0.
exact
Enable exact cropping. If enabled, subsampled videos will be cropped at exact width/height/x/y as
specified and will not be rounded to nearest smaller value. It defaults to 0.
The out_w, out_h, x, y parameters are expressions containing the following constants:
x
y The computed values for x and y. They are evaluated for each new frame.
in_w
in_h
The input width and height.
iw
ih These are the same as in_w and in_h.
out_w
out_h
The output (cropped) width and height.
ow
oh These are the same as out_w and out_h.
a same as iw / ih
sar input sample aspect ratio
dar input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2
and vsub is 1.
n The number of the input frame, starting from 0.
pos the position in the file of the input frame, NAN if unknown
t The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.
The expression for out_w may depend on the value of out_h, and the expression for out_h may depend on
out_w, but they cannot depend on x and y, as x and y are evaluated after out_w and out_h.
The x and y parameters specify the expressions for the position of the top-left corner of the output
(non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is
approximated to the nearest valid value.
The expression for x may depend on y, and the expression for y may depend on x.
Examples
• Crop area with size 100x100 at position (12,34).
crop=100:100:12:34
Using named options, the example above becomes:
crop=w=100:h=100:x=12:y=34
• Crop the central input area with size 100x100:
crop=100:100
• Crop the central input area with size 2/3 of the input video:
crop=2/3*in_w:2/3*in_h
• Crop the input video central square:
crop=out_w=in_h
crop=in_h
• Delimit the rectangle with the top-left corner placed at position 100:100 and the right-bottom corner
corresponding to the right-bottom corner of the input image.
crop=in_w-100:in_h-100:100:100
• Crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom borders
crop=in_w-2*10:in_h-2*20
• Keep only the bottom right quarter of the input image:
crop=in_w/2:in_h/2:in_w/2:in_h/2
• Crop height for getting Greek harmony:
crop=in_w:1/PHI*in_w
• Apply trembling effect:
crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)
• Apply erratic camera effect depending on timestamp:
crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"
• Set x depending on the value of y:
crop=in_w/2:in_h/2:y:10+10*sin(n/10)
Commands
This filter supports the following commands:
w, out_w
h, out_h
x
y Set width/height of the output video and the horizontal/vertical position in the input video. The
command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
cropdetect
Auto-detect the crop size.
It calculates the necessary cropping parameters and prints the recommended parameters via the logging
system. The detected dimensions correspond to the non-black area of the input video.
It accepts the following parameters:
limit
Set higher black value threshold, which can be optionally specified from nothing (0) to everything
(255 for 8-bit based formats). An intensity value greater to the set value is considered non-black.
It defaults to 24. You can also specify a value between 0.0 and 1.0 which will be scaled depending
on the bitdepth of the pixel format.
round
The value which the width/height should be divisible by. It defaults to 16. The offset is
automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2
video). 16 is best when encoding to most video codecs.
reset_count, reset
Set the counter that determines after how many frames cropdetect will reset the previously detected
largest video area and start over to detect the current optimal crop area. Default value is 0.
This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns
the largest area encountered during playback.
curves
Apply color adjustments using curves.
This filter is similar to the Adobe Photoshop and GIMP curves tools. Each component (red, green and blue)
has its values defined by N key points tied from each other using a smooth curve. The x-axis represents
the pixel values from the input frame, and the y-axis the new pixel values to be set for the output
frame.
By default, a component curve is defined by the two points (0;0) and (1;1). This creates a straight line
where each original pixel value is "adjusted" to its own value, which means no change to the image.
The filter allows you to redefine these two points and add some more. A new curve (using a natural cubic
spline interpolation) will be define to pass smoothly through all these new coordinates. The new defined
points needs to be strictly increasing over the x-axis, and their x and y values must be in the [0;1]
interval. If the computed curves happened to go outside the vector spaces, the values will be clipped
accordingly.
The filter accepts the following options:
preset
Select one of the available color presets. This option can be used in addition to the r, g, b
parameters; in this case, the later options takes priority on the preset values. Available presets
are:
none
color_negative
cross_process
darker
increase_contrast
lighter
linear_contrast
medium_contrast
negative
strong_contrast
vintage
Default is "none".
master, m
Set the master key points. These points will define a second pass mapping. It is sometimes called a
"luminance" or "value" mapping. It can be used with r, g, b or all since it acts like a post-
processing LUT.
red, r
Set the key points for the red component.
green, g
Set the key points for the green component.
blue, b
Set the key points for the blue component.
all Set the key points for all components (not including master). Can be used in addition to the other
key points component options. In this case, the unset component(s) will fallback on this all setting.
psfile
Specify a Photoshop curves file (".acv") to import the settings from.
plot
Save Gnuplot script of the curves in specified file.
To avoid some filtergraph syntax conflicts, each key points list need to be defined using the following
syntax: "x0/y0 x1/y1 x2/y2 ...".
Examples
• Increase slightly the middle level of blue:
curves=blue='0/0 0.5/0.58 1/1'
• Vintage effect:
curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'
Here we obtain the following coordinates for each components:
red "(0;0.11) (0.42;0.51) (1;0.95)"
green
"(0;0) (0.50;0.48) (1;1)"
blue
"(0;0.22) (0.49;0.44) (1;0.80)"
• The previous example can also be achieved with the associated built-in preset:
curves=preset=vintage
• Or simply:
curves=vintage
• Use a Photoshop preset and redefine the points of the green component:
curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'
• Check out the curves of the "cross_process" profile using ffmpeg and gnuplot:
ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
gnuplot -p /tmp/curves.plt
datascope
Video data analysis filter.
This filter shows hexadecimal pixel values of part of video.
The filter accepts the following options:
size, s
Set output video size.
x Set x offset from where to pick pixels.
y Set y offset from where to pick pixels.
mode
Set scope mode, can be one of the following:
mono
Draw hexadecimal pixel values with white color on black background.
color
Draw hexadecimal pixel values with input video pixel color on black background.
color2
Draw hexadecimal pixel values on color background picked from input video, the text color is
picked in such way so its always visible.
axis
Draw rows and columns numbers on left and top of video.
opacity
Set background opacity.
dctdnoiz
Denoise frames using 2D DCT (frequency domain filtering).
This filter is not designed for real time.
The filter accepts the following options:
sigma, s
Set the noise sigma constant.
This sigma defines a hard threshold of "3 * sigma"; every DCT coefficient (absolute value) below this
threshold with be dropped.
If you need a more advanced filtering, see expr.
Default is 0.
overlap
Set number overlapping pixels for each block. Since the filter can be slow, you may want to reduce
this value, at the cost of a less effective filter and the risk of various artefacts.
If the overlapping value doesn't permit processing the whole input width or height, a warning will be
displayed and according borders won't be denoised.
Default value is blocksize-1, which is the best possible setting.
expr, e
Set the coefficient factor expression.
For each coefficient of a DCT block, this expression will be evaluated as a multiplier value for the
coefficient.
If this is option is set, the sigma option will be ignored.
The absolute value of the coefficient can be accessed through the c variable.
n Set the blocksize using the number of bits. "1<<n" defines the blocksize, which is the width and
height of the processed blocks.
The default value is 3 (8x8) and can be raised to 4 for a blocksize of 16x16. Note that changing this
setting has huge consequences on the speed processing. Also, a larger block size does not necessarily
means a better de-noising.
Examples
Apply a denoise with a sigma of 4.5:
dctdnoiz=4.5
The same operation can be achieved using the expression system:
dctdnoiz=e='gte(c, 4.5*3)'
Violent denoise using a block size of "16x16":
dctdnoiz=15:n=4
deband
Remove banding artifacts from input video. It works by replacing banded pixels with average value of
referenced pixels.
The filter accepts the following options:
1thr
2thr
3thr
4thr
Set banding detection threshold for each plane. Default is 0.02. Valid range is 0.00003 to 0.5. If
difference between current pixel and reference pixel is less than threshold, it will be considered as
banded.
range, r
Banding detection range in pixels. Default is 16. If positive, random number in range 0 to set value
will be used. If negative, exact absolute value will be used. The range defines square of four
pixels around current pixel.
direction, d
Set direction in radians from which four pixel will be compared. If positive, random direction from 0
to set direction will be picked. If negative, exact of absolute value will be picked. For example
direction 0, -PI or -2*PI radians will pick only pixels on same row and -PI/2 will pick only pixels
on same column.
blur, b
If enabled, current pixel is compared with average value of all four surrounding pixels. The default
is enabled. If disabled current pixel is compared with all four surrounding pixels. The pixel is
considered banded if only all four differences with surrounding pixels are less than threshold.
coupling, c
If enabled, current pixel is changed if and only if all pixel components are banded, e.g. banding
detection threshold is triggered for all color components. The default is disabled.
decimate
Drop duplicated frames at regular intervals.
The filter accepts the following options:
cycle
Set the number of frames from which one will be dropped. Setting this to N means one frame in every
batch of N frames will be dropped. Default is 5.
dupthresh
Set the threshold for duplicate detection. If the difference metric for a frame is less than or equal
to this value, then it is declared as duplicate. Default is 1.1
scthresh
Set scene change threshold. Default is 15.
blockx
blocky
Set the size of the x and y-axis blocks used during metric calculations. Larger blocks give better
noise suppression, but also give worse detection of small movements. Must be a power of two. Default
is 32.
ppsrc
Mark main input as a pre-processed input and activate clean source input stream. This allows the
input to be pre-processed with various filters to help the metrics calculation while keeping the
frame selection lossless. When set to 1, the first stream is for the pre-processed input, and the
second stream is the clean source from where the kept frames are chosen. Default is 0.
chroma
Set whether or not chroma is considered in the metric calculations. Default is 1.
deflate
Apply deflate effect to the video.
This filter replaces the pixel by the local(3x3) average by taking into account only values lower than
the pixel.
It accepts the following options:
threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
deflicker
Remove temporal frame luminance variations.
It accepts the following options:
size, s
Set moving-average filter size in frames. Default is 5. Allowed range is 2 - 129.
mode, m
Set averaging mode to smooth temporal luminance variations.
Available values are:
am Arithmetic mean
gm Geometric mean
hm Harmonic mean
qm Quadratic mean
cm Cubic mean
pm Power mean
median
Median
bypass
Do not actually modify frame. Useful when one only wants metadata.
dejudder
Remove judder produced by partially interlaced telecined content.
Judder can be introduced, for instance, by pullup filter. If the original source was partially telecined
content then the output of "pullup,dejudder" will have a variable frame rate. May change the recorded
frame rate of the container. Aside from that change, this filter will not affect constant frame rate
video.
The option available in this filter is:
cycle
Specify the length of the window over which the judder repeats.
Accepts any integer greater than 1. Useful values are:
4 If the original was telecined from 24 to 30 fps (Film to NTSC).
5 If the original was telecined from 25 to 30 fps (PAL to NTSC).
20 If a mixture of the two.
The default is 4.
delogo
Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a rectangle
covering the logo and watch it disappear (and sometimes something even uglier appear - your mileage may
vary).
It accepts the following parameters:
x
y Specify the top left corner coordinates of the logo. They must be specified.
w
h Specify the width and height of the logo to clear. They must be specified.
band, t
Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 1.
This option is deprecated, setting higher values should no longer be necessary and is not
recommended.
show
When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, and h
parameters. The default value is 0.
The rectangle is drawn on the outermost pixels which will be (partly) replaced with interpolated
values. The values of the next pixels immediately outside this rectangle in each direction will be
used to compute the interpolated pixel values inside the rectangle.
Examples
• Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, and a band of
size 10:
delogo=x=0:y=0:w=100:h=77:band=10
deshake
Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove camera shake
from hand-holding a camera, bumping a tripod, moving on a vehicle, etc.
The filter accepts the following options:
x
y
w
h Specify a rectangular area where to limit the search for motion vectors. If desired the search for
motion vectors can be limited to a rectangular area of the frame defined by its top left corner,
width and height. These parameters have the same meaning as the drawbox filter which can be used to
visualise the position of the bounding box.
This is useful when simultaneous movement of subjects within the frame might be confused for camera
motion by the motion vector search.
If any or all of x, y, w and h are set to -1 then the full frame is used. This allows later options
to be set without specifying the bounding box for the motion vector search.
Default - search the whole frame.
rx
ry Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.
edge
Specify how to generate pixels to fill blanks at the edge of the frame. Available values are:
blank, 0
Fill zeroes at blank locations
original, 1
Original image at blank locations
clamp, 2
Extruded edge value at blank locations
mirror, 3
Mirrored edge at blank locations
Default value is mirror.
blocksize
Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.
contrast
Specify the contrast threshold for blocks. Only blocks with more than the specified contrast
(difference between darkest and lightest pixels) will be considered. Range 1-255, default 125.
search
Specify the search strategy. Available values are:
exhaustive, 0
Set exhaustive search
less, 1
Set less exhaustive search.
Default value is exhaustive.
filename
If set then a detailed log of the motion search is written to the specified file.
opencl
If set to 1, specify using OpenCL capabilities, only available if FFmpeg was configured with
"--enable-opencl". Default value is 0.
despill
Remove unwanted contamination of foreground colors, caused by reflected color of greenscreen or
bluescreen.
This filter accepts the following options:
type
Set what type of despill to use.
mix Set how spillmap will be generated.
expand
Set how much to get rid of still remaining spill.
red Controls amount of red in spill area.
green
Controls amount of green in spill area. Should be -1 for greenscreen.
blue
Controls amount of blue in spill area. Should be -1 for bluescreen.
brightness
Controls brightness of spill area, preserving colors.
alpha
Modify alpha from generated spillmap.
detelecine
Apply an exact inverse of the telecine operation. It requires a predefined pattern specified using the
pattern option which must be the same as that passed to the telecine filter.
This filter accepts the following options:
first_field
top, t
top field first
bottom, b
bottom field first The default value is "top".
pattern
A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.
start_frame
A number representing position of the first frame with respect to the telecine pattern. This is to be
used if the stream is cut. The default value is 0.
dilation
Apply dilation effect to the video.
This filter replaces the pixel by the local(3x3) maximum.
It accepts the following options:
threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.
Flags to local 3x3 coordinates maps like this:
1 2 3
4 5
6 7 8
displace
Displace pixels as indicated by second and third input stream.
It takes three input streams and outputs one stream, the first input is the source, and second and third
input are displacement maps.
The second input specifies how much to displace pixels along the x-axis, while the third input specifies
how much to displace pixels along the y-axis. If one of displacement map streams terminates, last frame
from that displacement map will be used.
Note that once generated, displacements maps can be reused over and over again.
A description of the accepted options follows.
edge
Set displace behavior for pixels that are out of range.
Available values are:
blank
Missing pixels are replaced by black pixels.
smear
Adjacent pixels will spread out to replace missing pixels.
wrap
Out of range pixels are wrapped so they point to pixels of other side.
mirror
Out of range pixels will be replaced with mirrored pixels.
Default is smear.
Examples
• Add ripple effect to rgb input of video size hd720:
ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT
• Add wave effect to rgb input of video size hd720:
ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT
drawbox
Draw a colored box on the input image.
It accepts the following parameters:
x
y The expressions which specify the top left corner coordinates of the box. It defaults to 0.
width, w
height, h
The expressions which specify the width and height of the box; if 0 they are interpreted as the input
width and height. It defaults to 0.
color, c
Specify the color of the box to write. For the general syntax of this option, check the "Color"
section in the ffmpeg-utils manual. If the special value "invert" is used, the box edge color is the
same as the video with inverted luma.
thickness, t
The expression which sets the thickness of the box edge. 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:
dar The input display aspect ratio, it is the same as (w / h) * sar.
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2
and vsub is 1.
in_h, ih
in_w, iw
The input width and height.
sar The input sample aspect ratio.
x
y The x and y offset coordinates where the box is drawn.
w
h The width and height of the drawn box.
t The thickness of the drawn box.
These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example
specify "y=x/dar" or "h=w/dar".
Examples
• Draw a black box around the edge of the input image:
drawbox
• Draw a box with color red and an opacity of 50%:
drawbox=10:20:200:60:red@0.5
The previous example can be specified as:
drawbox=x=10:y=20:w=200:h=60:color=red@0.5
• Fill the box with pink color:
drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max
• Draw a 2-pixel red 2.40:1 mask:
drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red
drawgrid
Draw a grid on the input image.
It accepts the following parameters:
x
y The expressions which specify the coordinates of some point of grid intersection (meant to configure
offset). Both default to 0.
width, w
height, h
The expressions which specify the width and height of the grid cell, if 0 they are interpreted as the
input width and height, respectively, minus "thickness", so image gets framed. Default to 0.
color, c
Specify the color of the grid. For the general syntax of this option, check the "Color" section in
the ffmpeg-utils manual. If the special value "invert" is used, the grid color is the same as the
video with inverted luma.
thickness, t
The expression which sets the thickness of the grid line. Default value is 1.
See below for the list of accepted constants.
The parameters for x, y, w and h and t are expressions containing the following constants:
dar The input display aspect ratio, it is the same as (w / h) * sar.
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2
and vsub is 1.
in_h, ih
in_w, iw
The input grid cell width and height.
sar The input sample aspect ratio.
x
y The x and y coordinates of some point of grid intersection (meant to configure offset).
w
h The width and height of the drawn cell.
t The thickness of the drawn cell.
These constants allow the x, y, w, h and t expressions to refer to each other, so you may for example
specify "y=x/dar" or "h=w/dar".
Examples
• Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an opacity of 50%:
drawgrid=width=100:height=100:thickness=2:color=red@0.5
• Draw a white 3x3 grid with an opacity of 50%:
drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5
drawtext
Draw a text string or text from a specified file on top of a video, using the libfreetype library.
To enable compilation of this filter, you need to configure FFmpeg with "--enable-libfreetype". To
enable default font fallback and the font option you need to configure FFmpeg with
"--enable-libfontconfig". To enable the text_shaping option, you need to configure FFmpeg with
"--enable-libfribidi".
Syntax
It accepts the following parameters:
box Used to draw a box around text using the background color. The value must be either 1 (enable) or 0
(disable). The default value of box is 0.
boxborderw
Set the width of the border to be drawn around the box using boxcolor. The default value of
boxborderw is 0.
boxcolor
The color to be used for drawing box around text. For the syntax of this option, check the "Color"
section in the ffmpeg-utils manual.
The default value of boxcolor is "white".
line_spacing
Set the line spacing in pixels of the border to be drawn around the box using box. The default value
of line_spacing is 0.
borderw
Set the width of the border to be drawn around the text using bordercolor. The default value of
borderw is 0.
bordercolor
Set the color to be used for drawing border around text. For the syntax of this option, check the
"Color" section in the ffmpeg-utils manual.
The default value of bordercolor is "black".
expansion
Select how the text is expanded. Can be either "none", "strftime" (deprecated) or "normal" (default).
See the drawtext_expansion, Text expansion section below for details.
basetime
Set a start time for the count. Value is in microseconds. Only applied in the deprecated strftime
expansion mode. To emulate in normal expansion mode use the "pts" function, supplying the start time
(in seconds) as the second argument.
fix_bounds
If true, check and fix text coords to avoid clipping.
fontcolor
The color to be used for drawing fonts. For the syntax of this option, check the "Color" section in
the ffmpeg-utils manual.
The default value of fontcolor is "black".
fontcolor_expr
String which is expanded the same way as text to obtain dynamic fontcolor value. By default this
option has empty value and is not processed. When this option is set, it overrides fontcolor option.
font
The font family to be used for drawing text. By default Sans.
fontfile
The font file to be used for drawing text. The path must be included. This parameter is mandatory if
the fontconfig support is disabled.
alpha
Draw the text applying alpha blending. The value can be a number between 0.0 and 1.0. The expression
accepts the same variables x, y as well. The default value is 1. Please see fontcolor_expr.
fontsize
The font size to be used for drawing text. The default value of fontsize is 16.
text_shaping
If set to 1, attempt to shape the text (for example, reverse the order of right-to-left text and join
Arabic characters) before drawing it. Otherwise, just draw the text exactly as given. By default 1
(if supported).
ft_load_flags
The flags to be used for loading the fonts.
The flags map the corresponding flags supported by libfreetype, and are a combination of the
following values:
default
no_scale
no_hinting
render
no_bitmap
vertical_layout
force_autohint
crop_bitmap
pedantic
ignore_global_advance_width
no_recurse
ignore_transform
monochrome
linear_design
no_autohint
Default value is "default".
For more information consult the documentation for the FT_LOAD_* libfreetype flags.
shadowcolor
The color to be used for drawing a shadow behind the drawn text. For the syntax of this option, check
the "Color" section in the ffmpeg-utils manual.
The default value of shadowcolor is "black".
shadowx
shadowy
The x and y offsets for the text shadow position with respect to the position of the text. They can
be either positive or negative values. The default value for both is "0".
start_number
The starting frame number for the n/frame_num variable. The default value is "0".
tabsize
The size in number of spaces to use for rendering the tab. Default value is 4.
timecode
Set the initial timecode representation in "hh:mm:ss[:;.]ff" format. It can be used with or without
text parameter. timecode_rate option must be specified.
timecode_rate, rate, r
Set the timecode frame rate (timecode only).
tc24hmax
If set to 1, the output of the timecode option will wrap around at 24 hours. Default is 0
(disabled).
text
The text string to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter
is mandatory if no file is specified with the parameter textfile.
textfile
A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded characters.
This parameter is mandatory if no text string is specified with the parameter text.
If both text and textfile are specified, an error is thrown.
reload
If set to 1, the textfile will be reloaded before each frame. Be sure to update it atomically, or it
may be read partially, or even fail.
x
y The expressions which specify the offsets where text will be drawn within the video frame. They are
relative to the top/left border of the output image.
The default value of x and y is "0".
See below for the list of accepted constants and functions.
The parameters for x and y are expressions containing the following constants and functions:
dar input display aspect ratio, it is the same as (w / h) * sar
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2
and vsub is 1.
line_h, lh
the height of each text line
main_h, h, H
the input height
main_w, w, W
the input width
max_glyph_a, ascent
the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph
outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation
with the Y axis upwards.
max_glyph_d, descent
the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline
point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with
the Y axis upwards.
max_glyph_h
maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text,
it is equivalent to ascent - descent.
max_glyph_w
maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
n the number of input frame, starting from 0
rand(min, max)
return a random number included between min and max
sar The input sample aspect ratio.
t timestamp expressed in seconds, NAN if the input timestamp is unknown
text_h, th
the height of the rendered text
text_w, tw
the width of the rendered text
x
y the x and y offset coordinates where the text is drawn.
These parameters allow the x and y expressions to refer each other, so you can for example specify
"y=x/dar".
Text expansion
If expansion is set to "strftime", the filter recognizes strftime() sequences in the provided text and
expands them accordingly. Check the documentation of strftime(). This feature is deprecated.
If expansion is set to "none", the text is printed verbatim.
If expansion is set to "normal" (which is the default), the following expansion mechanism is used.
The backslash character \, followed by any character, always expands to the second character.
Sequences of the form "%{...}" are expanded. The text between the braces is a function name, possibly
followed by arguments separated by ':'. If the arguments contain special characters or delimiters (':'
or '}'), they should be escaped.
Note that they probably must also be escaped as the value for the text option in the filter argument
string and as the filter argument in the filtergraph description, and possibly also for the shell, that
makes up to four levels of escaping; using a text file avoids these problems.
The following functions are available:
expr, e
The expression evaluation result.
It must take one argument specifying the expression to be evaluated, which accepts the same constants
and functions as the x and y values. Note that not all constants should be used, for example the text
size is not known when evaluating the expression, so the constants text_w and text_h will have an
undefined value.
expr_int_format, eif
Evaluate the expression's value and output as formatted integer.
The first argument is the expression to be evaluated, just as for the expr function. The second
argument specifies the output format. Allowed values are x, X, d and u. They are treated exactly as
in the "printf" function. The third parameter is optional and sets the number of positions taken by
the output. It can be used to add padding with zeros from the left.
gmtime
The time at which the filter is running, expressed in UTC. It can accept an argument: a strftime()
format string.
localtime
The time at which the filter is running, expressed in the local time zone. It can accept an
argument: a strftime() format string.
metadata
Frame metadata. Takes one or two arguments.
The first argument is mandatory and specifies the metadata key.
The second argument is optional and specifies a default value, used when the metadata key is not
found or empty.
n, frame_num
The frame number, starting from 0.
pict_type
A 1 character description of the current picture type.
pts The timestamp of the current frame. It can take up to three arguments.
The first argument is the format of the timestamp; it defaults to "flt" for seconds as a decimal
number with microsecond accuracy; "hms" stands for a formatted [-]HH:MM:SS.mmm timestamp with
millisecond accuracy. "gmtime" stands for the timestamp of the frame formatted as UTC time;
"localtime" stands for the timestamp of the frame formatted as local time zone time.
The second argument is an offset added to the timestamp.
If the format is set to "localtime" or "gmtime", a third argument may be supplied: a strftime()
format string. By default, YYYY-MM-DD HH:MM:SS format will be used.
Examples
• Draw "Test Text" with font FreeSerif, using the default values for the optional parameters.
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"
• Draw 'Test Text' with font FreeSerif of size 24 at position x=100 and y=50 (counting from the top-
left corner of the screen), text is yellow with a red box around it. Both the text and the box have
an opacity of 20%.
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"
Note that the double quotes are not necessary if spaces are not used within the parameter list.
• Show the text at the center of the video frame:
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"
• Show the text at a random position, switching to a new position every 30 seconds:
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"
• Show a text line sliding from right to left in the last row of the video frame. The file LONG_LINE is
assumed to contain a single line with no newlines.
drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"
• Show the content of file CREDITS off the bottom of the frame and scroll up.
drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"
• Draw a single green letter "g", at the center of the input video. The glyph baseline is placed at
half screen height.
drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"
• Show text for 1 second every 3 seconds:
drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"
• Use fontconfig to set the font. Note that the colons need to be escaped.
drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'
• Print the date of a real-time encoding (see strftime(3)):
drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'
• Show text fading in and out (appearing/disappearing):
#!/bin/sh
DS=1.0 # display start
DE=10.0 # display end
FID=1.5 # fade in duration
FOD=5 # fade out duration
ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"
• Horizontally align multiple separate texts. Note that max_glyph_a and the fontsize value are included
in the y offset.
drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a
For more information about libfreetype, check: <http://www.freetype.org/>.
For more information about fontconfig, check:
<http://freedesktop.org/software/fontconfig/fontconfig-user.html>.
For more information about libfribidi, check: <http://fribidi.org/>.
edgedetect
Detect and draw edges. The filter uses the Canny Edge Detection algorithm.
The filter accepts the following options:
low
high
Set low and high threshold values used by the Canny thresholding algorithm.
The high threshold selects the "strong" edge pixels, which are then connected through 8-connectivity
with the "weak" edge pixels selected by the low threshold.
low and high threshold values must be chosen in the range [0,1], and low should be lesser or equal to
high.
Default value for low is "20/255", and default value for high is "50/255".
mode
Define the drawing mode.
wires
Draw white/gray wires on black background.
colormix
Mix the colors to create a paint/cartoon effect.
Default value is wires.
Examples
• Standard edge detection with custom values for the hysteresis thresholding:
edgedetect=low=0.1:high=0.4
• Painting effect without thresholding:
edgedetect=mode=colormix:high=0
eq
Set brightness, contrast, saturation and approximate gamma adjustment.
The filter accepts the following options:
contrast
Set the contrast expression. The value must be a float value in range "-2.0" to 2.0. The default
value is "1".
brightness
Set the brightness expression. The value must be a float value in range "-1.0" to 1.0. The default
value is "0".
saturation
Set the saturation expression. The value must be a float in range 0.0 to 3.0. The default value is
"1".
gamma
Set the gamma expression. The value must be a float in range 0.1 to 10.0. The default value is "1".
gamma_r
Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The default value
is "1".
gamma_g
Set the gamma expression for green. The value must be a float in range 0.1 to 10.0. The default value
is "1".
gamma_b
Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The default value
is "1".
gamma_weight
Set the gamma weight expression. It can be used to reduce the effect of a high gamma value on bright
image areas, e.g. keep them from getting overamplified and just plain white. The value must be a
float in range 0.0 to 1.0. A value of 0.0 turns the gamma correction all the way down while 1.0
leaves it at its full strength. Default is "1".
eval
Set when the expressions for brightness, contrast, saturation and gamma expressions are evaluated.
It accepts the following values:
init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame
Default value is init.
The expressions accept the following parameters:
n frame count of the input frame starting from 0
pos byte position of the corresponding packet in the input file, NAN if unspecified
r frame rate of the input video, NAN if the input frame rate is unknown
t timestamp expressed in seconds, NAN if the input timestamp is unknown
Commands
The filter supports the following commands:
contrast
Set the contrast expression.
brightness
Set the brightness expression.
saturation
Set the saturation expression.
gamma
Set the gamma expression.
gamma_r
Set the gamma_r expression.
gamma_g
Set gamma_g expression.
gamma_b
Set gamma_b expression.
gamma_weight
Set gamma_weight expression.
The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
erosion
Apply erosion effect to the video.
This filter replaces the pixel by the local(3x3) minimum.
It accepts the following options:
threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
coordinates
Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are used.
Flags to local 3x3 coordinates maps like this:
1 2 3
4 5
6 7 8
extractplanes
Extract color channel components from input video stream into separate grayscale video streams.
The filter accepts the following option:
planes
Set plane(s) to extract.
Available values for planes are:
y
u
v
a
r
g
b
Choosing planes not available in the input will result in an error. That means you cannot select
"r", "g", "b" planes with "y", "u", "v" planes at same time.
Examples
• Extract luma, u and v color channel component from input video frame into 3 grayscale outputs:
ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi
elbg
Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.
For each input image, the filter will compute the optimal mapping from the input to the output given the
codebook length, that is the number of distinct output colors.
This filter accepts the following options.
codebook_length, l
Set codebook length. The value must be a positive integer, and represents the number of distinct
output colors. Default value is 256.
nb_steps, n
Set the maximum number of iterations to apply for computing the optimal mapping. The higher the value
the better the result and the higher the computation time. Default value is 1.
seed, s
Set a random seed, must be an integer included between 0 and UINT32_MAX. If not specified, or if
explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
pal8
Set pal8 output pixel format. This option does not work with codebook length greater than 256.
fade
Apply a fade-in/out effect to the input video.
It accepts the following parameters:
type, t
The effect type can be either "in" for a fade-in, or "out" for a fade-out effect. Default is "in".
start_frame, s
Specify the number of the frame to start applying the fade effect at. Default is 0.
nb_frames, n
The number of frames that the fade effect lasts. At the end of the fade-in effect, the output video
will have the same intensity as the input video. At the end of the fade-out transition, the output
video will be filled with the selected color. Default is 25.
alpha
If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
start_time, st
Specify the timestamp (in seconds) of the frame to start to apply the fade effect. If both
start_frame and start_time are specified, the fade will start at whichever comes last. Default is 0.
duration, d
The number of seconds for which the fade effect has to last. At the end of the fade-in effect the
output video will have the same intensity as the input video, at the end of the fade-out transition
the output video will be filled with the selected color. If both duration and nb_frames are
specified, duration is used. Default is 0 (nb_frames is used by default).
color, c
Specify the color of the fade. Default is "black".
Examples
• Fade in the first 30 frames of video:
fade=in:0:30
The command above is equivalent to:
fade=t=in:s=0:n=30
• Fade out the last 45 frames of a 200-frame video:
fade=out:155:45
fade=type=out:start_frame=155:nb_frames=45
• Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video:
fade=in:0:25, fade=out:975:25
• Make the first 5 frames yellow, then fade in from frame 5-24:
fade=in:5:20:color=yellow
• Fade in alpha over first 25 frames of video:
fade=in:0:25:alpha=1
• Make the first 5.5 seconds black, then fade in for 0.5 seconds:
fade=t=in:st=5.5:d=0.5
fftfilt
Apply arbitrary expressions to samples in frequency domain
dc_Y
Adjust the dc value (gain) of the luma plane of the image. The filter accepts an integer value in
range 0 to 1000. The default value is set to 0.
dc_U
Adjust the dc value (gain) of the 1st chroma plane of the image. The filter accepts an integer value
in range 0 to 1000. The default value is set to 0.
dc_V
Adjust the dc value (gain) of the 2nd chroma plane of the image. The filter accepts an integer value
in range 0 to 1000. The default value is set to 0.
weight_Y
Set the frequency domain weight expression for the luma plane.
weight_U
Set the frequency domain weight expression for the 1st chroma plane.
weight_V
Set the frequency domain weight expression for the 2nd chroma plane.
eval
Set when the expressions are evaluated.
It accepts the following values:
init
Only evaluate expressions once during the filter initialization.
frame
Evaluate expressions for each incoming frame.
Default value is init.
The filter accepts the following variables:
X
Y The coordinates of the current sample.
W
H The width and height of the image.
N The number of input frame, starting from 0.
Examples
• High-pass:
fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'
• Low-pass:
fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'
• Sharpen:
fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'
• Blur:
fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'
field
Extract a single field from an interlaced image using stride arithmetic to avoid wasting CPU time. The
output frames are marked as non-interlaced.
The filter accepts the following options:
type
Specify whether to extract the top (if the value is 0 or "top") or the bottom field (if the value is
1 or "bottom").
fieldhint
Create new frames by copying the top and bottom fields from surrounding frames supplied as numbers by the
hint file.
hint
Set file containing hints: absolute/relative frame numbers.
There must be one line for each frame in a clip. Each line must contain two numbers separated by the
comma, optionally followed by "-" or "+". Numbers supplied on each line of file can not be out of
[N-1,N+1] where N is current frame number for "absolute" mode or out of [-1, 1] range for "relative"
mode. First number tells from which frame to pick up top field and second number tells from which
frame to pick up bottom field.
If optionally followed by "+" output frame will be marked as interlaced, else if followed by "-"
output frame will be marked as progressive, else it will be marked same as input frame. If line
starts with "#" or ";" that line is skipped.
mode
Can be item "absolute" or "relative". Default is "absolute".
Example of first several lines of "hint" file for "relative" mode:
0,0 - # first frame
1,0 - # second frame, use third's frame top field and second's frame bottom field
1,0 - # third frame, use fourth's frame top field and third's frame bottom field
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
0,0 -
1,0 -
1,0 -
1,0 -
0,0 -
fieldmatch
Field matching filter for inverse telecine. It is meant to reconstruct the progressive frames from a
telecined stream. The filter does not drop duplicated frames, so to achieve a complete inverse telecine
"fieldmatch" needs to be followed by a decimation filter such as decimate in the filtergraph.
The separation of the field matching and the decimation is notably motivated by the possibility of
inserting a de-interlacing filter fallback between the two. If the source has mixed telecined and real
interlaced content, "fieldmatch" will not be able to match fields for the interlaced parts. But these
remaining combed frames will be marked as interlaced, and thus can be de-interlaced by a later filter
such as yadif before decimation.
In addition to the various configuration options, "fieldmatch" can take an optional second stream,
activated through the ppsrc option. If enabled, the frames reconstruction will be based on the fields and
frames from this second stream. This allows the first input to be pre-processed in order to help the
various algorithms of the filter, while keeping the output lossless (assuming the fields are matched
properly). Typically, a field-aware denoiser, or brightness/contrast adjustments can help.
Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) and VIVTC/VFM (VapourSynth
project). The later is a light clone of TFM from which "fieldmatch" is based on. While the semantic and
usage are very close, some behaviour and options names can differ.
The decimate filter currently only works for constant frame rate input. If your input has mixed
telecined (30fps) and progressive content with a lower framerate like 24fps use the following filterchain
to produce the necessary cfr stream: "dejudder,fps=30000/1001,fieldmatch,decimate".
The filter accepts the following options:
order
Specify the assumed field order of the input stream. Available values are:
auto
Auto detect parity (use FFmpeg's internal parity value).
bff Assume bottom field first.
tff Assume top field first.
Note that it is sometimes recommended not to trust the parity announced by the stream.
Default value is auto.
mode
Set the matching mode or strategy to use. pc mode is the safest in the sense that it won't risk
creating jerkiness due to duplicate frames when possible, but if there are bad edits or blended
fields it will end up outputting combed frames when a good match might actually exist. On the other
hand, pcn_ub mode is the most risky in terms of creating jerkiness, but will almost always find a
good frame if there is one. The other values are all somewhere in between pc and pcn_ub in terms of
risking jerkiness and creating duplicate frames versus finding good matches in sections with bad
edits, orphaned fields, blended fields, etc.
More details about p/c/n/u/b are available in p/c/n/u/b meaning section.
Available values are:
pc 2-way matching (p/c)
pc_n
2-way matching, and trying 3rd match if still combed (p/c + n)
pc_u
2-way matching, and trying 3rd match (same order) if still combed (p/c + u)
pc_n_ub
2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if still combed (p/c
+ n + u/b)
pcn 3-way matching (p/c/n)
pcn_ub
3-way matching, and trying 4th/5th matches if all 3 of the original matches are detected as
combed (p/c/n + u/b)
The parenthesis at the end indicate the matches that would be used for that mode assuming order=tff
(and field on auto or top).
In terms of speed pc mode is by far the fastest and pcn_ub is the slowest.
Default value is pc_n.
ppsrc
Mark the main input stream as a pre-processed input, and enable the secondary input stream as the
clean source to pick the fields from. See the filter introduction for more details. It is similar to
the clip2 feature from VFM/TFM.
Default value is 0 (disabled).
field
Set the field to match from. It is recommended to set this to the same value as order unless you
experience matching failures with that setting. In certain circumstances changing the field that is
used to match from can have a large impact on matching performance. Available values are:
auto
Automatic (same value as order).
bottom
Match from the bottom field.
top Match from the top field.
Default value is auto.
mchroma
Set whether or not chroma is included during the match comparisons. In most cases it is recommended
to leave this enabled. You should set this to 0 only if your clip has bad chroma problems such as
heavy rainbowing or other artifacts. Setting this to 0 could also be used to speed things up at the
cost of some accuracy.
Default value is 1.
y0
y1 These define an exclusion band which excludes the lines between y0 and y1 from being included in the
field matching decision. An exclusion band can be used to ignore subtitles, a logo, or other things
that may interfere with the matching. y0 sets the starting scan line and y1 sets the ending line; all
lines in between y0 and y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the same value
will disable the feature. y0 and y1 defaults to 0.
scthresh
Set the scene change detection threshold as a percentage of maximum change on the luma plane. Good
values are in the "[8.0, 14.0]" range. Scene change detection is only relevant in case combmatch=sc.
The range for scthresh is "[0.0, 100.0]".
Default value is 12.0.
combmatch
When combatch is not none, "fieldmatch" will take into account the combed scores of matches when
deciding what match to use as the final match. Available values are:
none
No final matching based on combed scores.
sc Combed scores are only used when a scene change is detected.
full
Use combed scores all the time.
Default is sc.
combdbg
Force "fieldmatch" to calculate the combed metrics for certain matches and print them. This setting
is known as micout in TFM/VFM vocabulary. Available values are:
none
No forced calculation.
pcn Force p/c/n calculations.
pcnub
Force p/c/n/u/b calculations.
Default value is none.
cthresh
This is the area combing threshold used for combed frame detection. This essentially controls how
"strong" or "visible" combing must be to be detected. Larger values mean combing must be more
visible and smaller values mean combing can be less visible or strong and still be detected. Valid
settings are from "-1" (every pixel will be detected as combed) to 255 (no pixel will be detected as
combed). This is basically a pixel difference value. A good range is "[8, 12]".
Default value is 9.
chroma
Sets whether or not chroma is considered in the combed frame decision. Only disable this if your
source has chroma problems (rainbowing, etc.) that are causing problems for the combed frame
detection with chroma enabled. Actually, using chroma=0 is usually more reliable, except for the case
where there is chroma only combing in the source.
Default value is 0.
blockx
blocky
Respectively set the x-axis and y-axis size of the window used during combed frame detection. This
has to do with the size of the area in which combpel pixels are required to be detected as combed for
a frame to be declared combed. See the combpel parameter description for more info. Possible values
are any number that is a power of 2 starting at 4 and going up to 512.
Default value is 16.
combpel
The number of combed pixels inside any of the blocky by blockx size blocks on the frame for the frame
to be detected as combed. While cthresh controls how "visible" the combing must be, this setting
controls "how much" combing there must be in any localized area (a window defined by the blockx and
blocky settings) on the frame. Minimum value is 0 and maximum is "blocky x blockx" (at which point no
frames will ever be detected as combed). This setting is known as MI in TFM/VFM vocabulary.
Default value is 80.
p/c/n/u/b meaning
p/c/n
We assume the following telecined stream:
Top fields: 1 2 2 3 4
Bottom fields: 1 2 3 4 4
The numbers correspond to the progressive frame the fields relate to. Here, the first two frames are
progressive, the 3rd and 4th are combed, and so on.
When "fieldmatch" is configured to run a matching from bottom (field=bottom) this is how this input
stream get transformed:
Input stream:
T 1 2 2 3 4
B 1 2 3 4 4 <-- matching reference
Matches: c c n n c
Output stream:
T 1 2 3 4 4
B 1 2 3 4 4
As a result of the field matching, we can see that some frames get duplicated. To perform a complete
inverse telecine, you need to rely on a decimation filter after this operation. See for instance the
decimate filter.
The same operation now matching from top fields (field=top) looks like this:
Input stream:
T 1 2 2 3 4 <-- matching reference
B 1 2 3 4 4
Matches: c c p p c
Output stream:
T 1 2 2 3 4
B 1 2 2 3 4
In these examples, we can see what p, c and n mean; basically, they refer to the frame and field of the
opposite parity:
*<p matches the field of the opposite parity in the previous frame>
*<c matches the field of the opposite parity in the current frame>
*<n matches the field of the opposite parity in the next frame>
u/b
The u and b matching are a bit special in the sense that they match from the opposite parity flag. In the
following examples, we assume that we are currently matching the 2nd frame (Top:2, bottom:2). According
to the match, a 'x' is placed above and below each matched fields.
With bottom matching (field=bottom):
Match: c p n b u
x x x x x
Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2
Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3
x x x x x
Output frames:
2 1 2 2 2
2 2 2 1 3
With top matching (field=top):
Match: c p n b u
x x x x x
Top 1 2 2 1 2 2 1 2 2 1 2 2 1 2 2
Bottom 1 2 3 1 2 3 1 2 3 1 2 3 1 2 3
x x x x x
Output frames:
2 2 2 1 2
2 1 3 2 2
Examples
Simple IVTC of a top field first telecined stream:
fieldmatch=order=tff:combmatch=none, decimate
Advanced IVTC, with fallback on yadif for still combed frames:
fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate
fieldorder
Transform the field order of the input video.
It accepts the following parameters:
order
The output field order. Valid values are tff for top field first or bff for bottom field first.
The default value is tff.
The transformation is done by shifting the picture content up or down by one line, and filling the
remaining line with appropriate picture content. This method is consistent with most broadcast field
order converters.
If the input video is not flagged as being interlaced, or it is already flagged as being of the required
output field order, then this filter does not alter the incoming video.
It is very useful when converting to or from PAL DV material, which is bottom field first.
For example:
ffmpeg -i in.vob -vf "fieldorder=bff" out.dv
fifo, afifo
Buffer input images and send them when they are requested.
It is mainly useful when auto-inserted by the libavfilter framework.
It does not take parameters.
find_rect
Find a rectangular object
It accepts the following options:
object
Filepath of the object image, needs to be in gray8.
threshold
Detection threshold, default is 0.5.
mipmaps
Number of mipmaps, default is 3.
xmin, ymin, xmax, ymax
Specifies the rectangle in which to search.
Examples
• Generate a representative palette of a given video using ffmpeg:
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
cover_rect
Cover a rectangular object
It accepts the following options:
cover
Filepath of the optional cover image, needs to be in yuv420.
mode
Set covering mode.
It accepts the following values:
cover
cover it by the supplied image
blur
cover it by interpolating the surrounding pixels
Default value is blur.
Examples
• Generate a representative palette of a given video using ffmpeg:
ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv
floodfill
Flood area with values of same pixel components with another values.
It accepts the following options:
x Set pixel x coordinate.
y Set pixel y coordinate.
s0 Set source #0 component value.
s1 Set source #1 component value.
s2 Set source #2 component value.
s3 Set source #3 component value.
d0 Set destination #0 component value.
d1 Set destination #1 component value.
d2 Set destination #2 component value.
d3 Set destination #3 component value.
format
Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is
suitable as input to the next filter.
It accepts the following parameters:
pix_fmts
A '|'-separated list of pixel format names, such as "pix_fmts=yuv420p|monow|rgb24".
Examples
• Convert the input video to the yuv420p format
format=pix_fmts=yuv420p
Convert the input video to any of the formats in the list
format=pix_fmts=yuv420p|yuv444p|yuv410p
fps
Convert the video to specified constant frame rate by duplicating or dropping frames as necessary.
It accepts the following parameters:
fps The desired output frame rate. The default is 25.
start_time
Assume the first PTS should be the given value, in seconds. This allows for padding/trimming at the
start of stream. By default, no assumption is made about the first frame's expected PTS, so no
padding or trimming is done. For example, this could be set to 0 to pad the beginning with
duplicates of the first frame if a video stream starts after the audio stream or to trim any frames
with a negative PTS.
round
Timestamp (PTS) rounding method.
Possible values are:
zero
round towards 0
inf round away from 0
down
round towards -infinity
up round towards +infinity
near
round to nearest
The default is "near".
eof_action
Action performed when reading the last frame.
Possible values are:
round
Use same timestamp rounding method as used for other frames.
pass
Pass through last frame if input duration has not been reached yet.
The default is "round".
Alternatively, the options can be specified as a flat string: fps[:start_time[:round]].
See also the setpts filter.
Examples
• A typical usage in order to set the fps to 25:
fps=fps=25
• Sets the fps to 24, using abbreviation and rounding method to round to nearest:
fps=fps=film:round=near
framepack
Pack two different video streams into a stereoscopic video, setting proper metadata on supported codecs.
The two views should have the same size and framerate and processing will stop when the shorter video
ends. Please note that you may conveniently adjust view properties with the scale and fps filters.
It accepts the following parameters:
format
The desired packing format. Supported values are:
sbs The views are next to each other (default).
tab The views are on top of each other.
lines
The views are packed by line.
columns
The views are packed by column.
frameseq
The views are temporally interleaved.
Some examples:
# Convert left and right views into a frame-sequential video
ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT
# Convert views into a side-by-side video with the same output resolution as the input
ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT
framerate
Change the frame rate by interpolating new video output frames from the source frames.
This filter is not designed to function correctly with interlaced media. If you wish to change the frame
rate of interlaced media then you are required to deinterlace before this filter and re-interlace after
this filter.
A description of the accepted options follows.
fps Specify the output frames per second. This option can also be specified as a value alone. The default
is 50.
interp_start
Specify the start of a range where the output frame will be created as a linear interpolation of two
frames. The range is [0-255], the default is 15.
interp_end
Specify the end of a range where the output frame will be created as a linear interpolation of two
frames. The range is [0-255], the default is 240.
scene
Specify the level at which a scene change is detected as a value between 0 and 100 to indicate a new
scene; a low value reflects a low probability for the current frame to introduce a new scene, while a
higher value means the current frame is more likely to be one. The default is 7.
flags
Specify flags influencing the filter process.
Available value for flags is:
scene_change_detect, scd
Enable scene change detection using the value of the option scene. This flag is enabled by
default.
framestep
Select one frame every N-th frame.
This filter accepts the following option:
step
Select frame after every "step" frames. Allowed values are positive integers higher than 0. Default
value is 1.
frei0r
Apply a frei0r effect to the input video.
To enable the compilation of this filter, you need to install the frei0r header and configure FFmpeg with
"--enable-frei0r".
It accepts the following parameters:
filter_name
The name of the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r
effect is searched for in each of the directories specified by the colon-separated list in
FREI0R_PATH. Otherwise, the standard frei0r paths are searched, in this order: HOME/.frei0r-1/lib/,
/usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/.
filter_params
A '|'-separated list of parameters to pass to the frei0r effect.
A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified
as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or by a color
description specified in the "Color" section in the ffmpeg-utils manual), a position (specified as X/Y,
where X and Y are floating point numbers) and/or a string.
The number and types of parameters depend on the loaded effect. If an effect parameter is not specified,
the default value is set.
Examples
• Apply the distort0r effect, setting the first two double parameters:
frei0r=filter_name=distort0r:filter_params=0.5|0.01
• Apply the colordistance effect, taking a color as the first parameter:
frei0r=colordistance:0.2/0.3/0.4
frei0r=colordistance:violet
frei0r=colordistance:0x112233
• Apply the perspective effect, specifying the top left and top right image positions:
frei0r=perspective:0.2/0.2|0.8/0.2
For more information, see <http://frei0r.dyne.org>
fspp
Apply fast and simple postprocessing. It is a faster version of spp.
It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- processing filter, one of them
is performed once per block, not per pixel. This allows for much higher speed.
The filter accepts the following options:
quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the
range 4-5. Default value is 4.
qp Force a constant quantization parameter. It accepts an integer in range 0-63. If not set, the filter
will use the QP from the video stream (if available).
strength
Set filter strength. It accepts an integer in range -15 to 32. Lower values mean more details but
also more artifacts, while higher values make the image smoother but also blurrier. Default value is
0 X PSNR optimal.
use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the
B-Frames have often larger QP. Default is 0 (not enabled).
gblur
Apply Gaussian blur filter.
The filter accepts the following options:
sigma
Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.
steps
Set number of steps for Gaussian approximation. Defauls is 1.
planes
Set which planes to filter. By default all planes are filtered.
sigmaV
Set vertical sigma, if negative it will be same as "sigma". Default is "-1".
geq
The filter accepts the following options:
lum_expr, lum
Set the luminance expression.
cb_expr, cb
Set the chrominance blue expression.
cr_expr, cr
Set the chrominance red expression.
alpha_expr, a
Set the alpha expression.
red_expr, r
Set the red expression.
green_expr, g
Set the green expression.
blue_expr, b
Set the blue expression.
The colorspace is selected according to the specified options. If one of the lum_expr, cb_expr, or
cr_expr options is specified, the filter will automatically select a YCbCr colorspace. If one of the
red_expr, green_expr, or blue_expr options is specified, it will select an RGB colorspace.
If one of the chrominance expression is not defined, it falls back on the other one. If no alpha
expression is specified it will evaluate to opaque value. If none of chrominance expressions are
specified, they will evaluate to the luminance expression.
The expressions can use the following variables and functions:
N The sequential number of the filtered frame, starting from 0.
X
Y The coordinates of the current sample.
W
H The width and height of the image.
SW
SH Width and height scale depending on the currently filtered plane. It is the ratio between the
corresponding luma plane number of pixels and the current plane ones. E.g. for YUV4:2:0 the values
are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.
T Time of the current frame, expressed in seconds.
p(x, y)
Return the value of the pixel at location (x,y) of the current plane.
lum(x, y)
Return the value of the pixel at location (x,y) of the luminance plane.
cb(x, y)
Return the value of the pixel at location (x,y) of the blue-difference chroma plane. Return 0 if
there is no such plane.
cr(x, y)
Return the value of the pixel at location (x,y) of the red-difference chroma plane. Return 0 if there
is no such plane.
r(x, y)
g(x, y)
b(x, y)
Return the value of the pixel at location (x,y) of the red/green/blue component. Return 0 if there is
no such component.
alpha(x, y)
Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there is no such
plane.
For functions, if x and y are outside the area, the value will be automatically clipped to the closer
edge.
Examples
• Flip the image horizontally:
geq=p(W-X\,Y)
• Generate a bidimensional sine wave, with angle "PI/3" and a wavelength of 100 pixels:
geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128
• Generate a fancy enigmatic moving light:
nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128
• Generate a quick emboss effect:
format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'
• Modify RGB components depending on pixel position:
geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'
• Create a radial gradient that is the same size as the input (also see the vignette filter):
geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray
gradfun
Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8-bit
color depth. Interpolate the gradients that should go where the bands are, and dither them.
It is designed for playback only. Do not use it prior to lossy compression, because compression tends to
lose the dither and bring back the bands.
It accepts the following parameters:
strength
The maximum amount by which the filter will change any one pixel. This is also the threshold for
detecting nearly flat regions. Acceptable values range from .51 to 64; the default value is 1.2. Out-
of-range values will be clipped to the valid range.
radius
The neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also
prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32; the
default value is 16. Out-of-range values will be clipped to the valid range.
Alternatively, the options can be specified as a flat string: strength[:radius]
Examples
• Apply the filter with a 3.5 strength and radius of 8:
gradfun=3.5:8
• Specify radius, omitting the strength (which will fall-back to the default value):
gradfun=radius=8
haldclut
Apply a Hald CLUT to a video stream.
First input is the video stream to process, and second one is the Hald CLUT. The Hald CLUT input can be
a simple picture or a complete video stream.
The filter accepts the following options:
shortest
Force termination when the shortest input terminates. Default is 0.
repeatlast
Continue applying the last CLUT after the end of the stream. A value of 0 disable the filter after
the last frame of the CLUT is reached. Default is 1.
"haldclut" also has the same interpolation options as lut3d (both filters share the same internals).
More information about the Hald CLUT can be found on Eskil Steenberg's website (Hald CLUT author) at
<http://www.quelsolaar.com/technology/clut.html>.
Workflow examples
Hald CLUT video stream
Generate an identity Hald CLUT stream altered with various effects:
ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut
Note: make sure you use a lossless codec.
Then use it with "haldclut" to apply it on some random stream:
ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv
The Hald CLUT will be applied to the 10 first seconds (duration of clut.nut), then the latest picture of
that CLUT stream will be applied to the remaining frames of the "mandelbrot" stream.
Hald CLUT with preview
A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by "Level*Level*Level" pixels. For a
given Hald CLUT, FFmpeg will select the biggest possible square starting at the top left of the picture.
The remaining padding pixels (bottom or right) will be ignored. This area can be used to add a preview of
the Hald CLUT.
Typically, the following generated Hald CLUT will be supported by the "haldclut" filter:
ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
pad=iw+320 [padded_clut];
smptebars=s=320x256, split [a][b];
[padded_clut][a] overlay=W-320:h, curves=color_negative [main];
[main][b] overlay=W-320" -frames:v 1 clut.png
It contains the original and a preview of the effect of the CLUT: SMPTE color bars are displayed on the
right-top, and below the same color bars processed by the color changes.
Then, the effect of this Hald CLUT can be visualized with:
ffplay input.mkv -vf "movie=clut.png, [in] haldclut"
hflip
Flip the input video horizontally.
For example, to horizontally flip the input video with ffmpeg:
ffmpeg -i in.avi -vf "hflip" out.avi
histeq
This filter applies a global color histogram equalization on a per-frame basis.
It can be used to correct video that has a compressed range of pixel intensities. The filter
redistributes the pixel intensities to equalize their distribution across the intensity range. It may be
viewed as an "automatically adjusting contrast filter". This filter is useful only for correcting
degraded or poorly captured source video.
The filter accepts the following options:
strength
Determine the amount of equalization to be applied. As the strength is reduced, the distribution of
pixel intensities more-and-more approaches that of the input frame. The value must be a float number
in the range [0,1] and defaults to 0.200.
intensity
Set the maximum intensity that can generated and scale the output values appropriately. The strength
should be set as desired and then the intensity can be limited if needed to avoid washing-out. The
value must be a float number in the range [0,1] and defaults to 0.210.
antibanding
Set the antibanding level. If enabled the filter will randomly vary the luminance of output pixels by
a small amount to avoid banding of the histogram. Possible values are "none", "weak" or "strong". It
defaults to "none".
histogram
Compute and draw a color distribution histogram for the input video.
The computed histogram is a representation of the color component distribution in an image.
Standard histogram displays the color components distribution in an image. Displays color graph for each
color component. Shows distribution of the Y, U, V, A or R, G, B components, depending on input format,
in the current frame. Below each graph a color component scale meter is shown.
The filter accepts the following options:
level_height
Set height of level. Default value is 200. Allowed range is [50, 2048].
scale_height
Set height of color scale. Default value is 12. Allowed range is [0, 40].
display_mode
Set display mode. It accepts the following values:
stack
Per color component graphs are placed below each other.
parade
Per color component graphs are placed side by side.
overlay
Presents information identical to that in the "parade", except that the graphs representing color
components are superimposed directly over one another.
Default is "stack".
levels_mode
Set mode. Can be either "linear", or "logarithmic". Default is "linear".
components
Set what color components to display. Default is 7.
fgopacity
Set foreground opacity. Default is 0.7.
bgopacity
Set background opacity. Default is 0.5.
Examples
• Calculate and draw histogram:
ffplay -i input -vf histogram
hqdn3d
This is a high precision/quality 3d denoise filter. It aims to reduce image noise, producing smooth
images and making still images really still. It should enhance compressibility.
It accepts the following optional parameters:
luma_spatial
A non-negative floating point number which specifies spatial luma strength. It defaults to 4.0.
chroma_spatial
A non-negative floating point number which specifies spatial chroma strength. It defaults to
3.0*luma_spatial/4.0.
luma_tmp
A floating point number which specifies luma temporal strength. It defaults to 6.0*luma_spatial/4.0.
chroma_tmp
A floating point number which specifies chroma temporal strength. It defaults to
luma_tmp*chroma_spatial/luma_spatial.
hwdownload
Download hardware frames to system memory.
The input must be in hardware frames, and the output a non-hardware format. Not all formats will be
supported on the output - it may be necessary to insert an additional format filter immediately following
in the graph to get the output in a supported format.
hwmap
Map hardware frames to system memory or to another device.
This filter has several different modes of operation; which one is used depends on the input and output
formats:
• Hardware frame input, normal frame output
Map the input frames to system memory and pass them to the output. If the original hardware frame is
later required (for example, after overlaying something else on part of it), the hwmap filter can be
used again in the next mode to retrieve it.
• Normal frame input, hardware frame output
If the input is actually a software-mapped hardware frame, then unmap it - that is, return the
original hardware frame.
Otherwise, a device must be provided. Create new hardware surfaces on that device for the output,
then map them back to the software format at the input and give those frames to the preceding filter.
This will then act like the hwupload filter, but may be able to avoid an additional copy when the
input is already in a compatible format.
• Hardware frame input and output
A device must be supplied for the output, either directly or with the derive_device option. The
input and output devices must be of different types and compatible - the exact meaning of this is
system-dependent, but typically it means that they must refer to the same underlying hardware context
(for example, refer to the same graphics card).
If the input frames were originally created on the output device, then unmap to retrieve the original
frames.
Otherwise, map the frames to the output device - create new hardware frames on the output
corresponding to the frames on the input.
The following additional parameters are accepted:
mode
Set the frame mapping mode. Some combination of:
read
The mapped frame should be readable.
write
The mapped frame should be writeable.
overwrite
The mapping will always overwrite the entire frame.
This may improve performance in some cases, as the original contents of the frame need not be
loaded.
direct
The mapping must not involve any copying.
Indirect mappings to copies of frames are created in some cases where either direct mapping is
not possible or it would have unexpected properties. Setting this flag ensures that the mapping
is direct and will fail if that is not possible.
Defaults to read+write if not specified.
derive_device type
Rather than using the device supplied at initialisation, instead derive a new device of type type
from the device the input frames exist on.
reverse
In a hardware to hardware mapping, map in reverse - create frames in the sink and map them back to
the source. This may be necessary in some cases where a mapping in one direction is required but
only the opposite direction is supported by the devices being used.
This option is dangerous - it may break the preceding filter in undefined ways if there are any
additional constraints on that filter's output. Do not use it without fully understanding the
implications of its use.
hwupload
Upload system memory frames to hardware surfaces.
The device to upload to must be supplied when the filter is initialised. If using ffmpeg, select the
appropriate device with the -filter_hw_device option.
hwupload_cuda
Upload system memory frames to a CUDA device.
It accepts the following optional parameters:
device
The number of the CUDA device to use
hqx
Apply a high-quality magnification filter designed for pixel art. This filter was originally created by
Maxim Stepin.
It accepts the following option:
n Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for "hq4x". Default is 3.
hstack
Stack input videos horizontally.
All streams must be of same pixel format and of same height.
Note that this filter is faster than using overlay and pad filter to create same output.
The filter accept the following option:
inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
hue
Modify the hue and/or the saturation of the input.
It accepts the following parameters:
h Specify the hue angle as a number of degrees. It accepts an expression, and defaults to "0".
s Specify the saturation in the [-10,10] range. It accepts an expression and defaults to "1".
H Specify the hue angle as a number of radians. It accepts an expression, and defaults to "0".
b Specify the brightness in the [-10,10] range. It accepts an expression and defaults to "0".
h and H are mutually exclusive, and can't be specified at the same time.
The b, h, H and s option values are expressions containing the following constants:
n frame count of the input frame starting from 0
pts presentation timestamp of the input frame expressed in time base units
r frame rate of the input video, NAN if the input frame rate is unknown
t timestamp expressed in seconds, NAN if the input timestamp is unknown
tb time base of the input video
Examples
• Set the hue to 90 degrees and the saturation to 1.0:
hue=h=90:s=1
• Same command but expressing the hue in radians:
hue=H=PI/2:s=1
• Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:
hue="H=2*PI*t: s=sin(2*PI*t)+1"
• Apply a 3 seconds saturation fade-in effect starting at 0:
hue="s=min(t/3\,1)"
The general fade-in expression can be written as:
hue="s=min(0\, max((t-START)/DURATION\, 1))"
• Apply a 3 seconds saturation fade-out effect starting at 5 seconds:
hue="s=max(0\, min(1\, (8-t)/3))"
The general fade-out expression can be written as:
hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"
Commands
This filter supports the following commands:
b
s
h
H Modify the hue and/or the saturation and/or brightness of the input video. The command accepts the
same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
hysteresis
Grow first stream into second stream by connecting components. This makes it possible to build more
robust edge masks.
This filter accepts the following options:
planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream.
By default value 0xf, all planes will be processed.
threshold
Set threshold which is used in filtering. If pixel component value is higher than this value filter
algorithm for connecting components is activated. By default value is 0.
idet
Detect video interlacing type.
This filter tries to detect if the input frames are interlaced, progressive, top or bottom field first.
It will also try to detect fields that are repeated between adjacent frames (a sign of telecine).
Single frame detection considers only immediately adjacent frames when classifying each frame. Multiple
frame detection incorporates the classification history of previous frames.
The filter will log these metadata values:
single.current_frame
Detected type of current frame using single-frame detection. One of: ``tff'' (top field first),
``bff'' (bottom field first), ``progressive'', or ``undetermined''
single.tff
Cumulative number of frames detected as top field first using single-frame detection.
multiple.tff
Cumulative number of frames detected as top field first using multiple-frame detection.
single.bff
Cumulative number of frames detected as bottom field first using single-frame detection.
multiple.current_frame
Detected type of current frame using multiple-frame detection. One of: ``tff'' (top field first),
``bff'' (bottom field first), ``progressive'', or ``undetermined''
multiple.bff
Cumulative number of frames detected as bottom field first using multiple-frame detection.
single.progressive
Cumulative number of frames detected as progressive using single-frame detection.
multiple.progressive
Cumulative number of frames detected as progressive using multiple-frame detection.
single.undetermined
Cumulative number of frames that could not be classified using single-frame detection.
multiple.undetermined
Cumulative number of frames that could not be classified using multiple-frame detection.
repeated.current_frame
Which field in the current frame is repeated from the last. One of ``neither'', ``top'', or
``bottom''.
repeated.neither
Cumulative number of frames with no repeated field.
repeated.top
Cumulative number of frames with the top field repeated from the previous frame's top field.
repeated.bottom
Cumulative number of frames with the bottom field repeated from the previous frame's bottom field.
The filter accepts the following options:
intl_thres
Set interlacing threshold.
prog_thres
Set progressive threshold.
rep_thres
Threshold for repeated field detection.
half_life
Number of frames after which a given frame's contribution to the statistics is halved (i.e., it
contributes only 0.5 to its classification). The default of 0 means that all frames seen are given
full weight of 1.0 forever.
analyze_interlaced_flag
When this is not 0 then idet will use the specified number of frames to determine if the interlaced
flag is accurate, it will not count undetermined frames. If the flag is found to be accurate it will
be used without any further computations, if it is found to be inaccurate it will be cleared without
any further computations. This allows inserting the idet filter as a low computational method to
clean up the interlaced flag
il
Deinterleave or interleave fields.
This filter allows one to process interlaced images fields without deinterlacing them. Deinterleaving
splits the input frame into 2 fields (so called half pictures). Odd lines are moved to the top half of
the output image, even lines to the bottom half. You can process (filter) them independently and then
re-interleave them.
The filter accepts the following options:
luma_mode, l
chroma_mode, c
alpha_mode, a
Available values for luma_mode, chroma_mode and alpha_mode are:
none
Do nothing.
deinterleave, d
Deinterleave fields, placing one above the other.
interleave, i
Interleave fields. Reverse the effect of deinterleaving.
Default value is "none".
luma_swap, ls
chroma_swap, cs
alpha_swap, as
Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.
inflate
Apply inflate effect to the video.
This filter replaces the pixel by the local(3x3) average by taking into account only values higher than
the pixel.
It accepts the following options:
threshold0
threshold1
threshold2
threshold3
Limit the maximum change for each plane, default is 65535. If 0, plane will remain unchanged.
interlace
Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd
frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.
Original Original New Frame
Frame 'j' Frame 'j+1' (tff)
========== =========== ==================
Line 0 --------------------> Frame 'j' Line 0
Line 1 Line 1 ----> Frame 'j+1' Line 1
Line 2 ---------------------> Frame 'j' Line 2
Line 3 Line 3 ----> Frame 'j+1' Line 3
... ... ...
New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on
It accepts the following optional parameters:
scan
This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff)
lines of the progressive frame.
lowpass
Vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.
0, off
Disable vertical lowpass filter
1, linear
Enable linear filter (default)
2, complex
Enable complex filter. This will slightly less reduce twitter and moire but better retain detail
and subjective sharpness impression.
kerndeint
Deinterlace input video by applying Donald Graft's adaptive kernel deinterling. Work on interlaced parts
of a video to produce progressive frames.
The description of the accepted parameters follows.
thresh
Set the threshold which affects the filter's tolerance when determining if a pixel line must be
processed. It must be an integer in the range [0,255] and defaults to 10. A value of 0 will result in
applying the process on every pixels.
map Paint pixels exceeding the threshold value to white if set to 1. Default is 0.
order
Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.
sharp
Enable additional sharpening if set to 1. Default is 0.
twoway
Enable twoway sharpening if set to 1. Default is 0.
Examples
• Apply default values:
kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0
• Enable additional sharpening:
kerndeint=sharp=1
• Paint processed pixels in white:
kerndeint=map=1
lenscorrection
Correct radial lens distortion
This filter can be used to correct for radial distortion as can result from the use of wide angle lenses,
and thereby re-rectify the image. To find the right parameters one can use tools available for example as
part of opencv or simply trial-and-error. To use opencv use the calibration sample (under samples/cpp)
from the opencv sources and extract the k1 and k2 coefficients from the resulting matrix.
Note that effectively the same filter is available in the open-source tools Krita and Digikam from the
KDE project.
In contrast to the vignette filter, which can also be used to compensate lens errors, this filter
corrects the distortion of the image, whereas vignette corrects the brightness distribution, so you may
want to use both filters together in certain cases, though you will have to take care of ordering, i.e.
whether vignetting should be applied before or after lens correction.
Options
The filter accepts the following options:
cx Relative x-coordinate of the focal point of the image, and thereby the center of the distortion. This
value has a range [0,1] and is expressed as fractions of the image width.
cy Relative y-coordinate of the focal point of the image, and thereby the center of the distortion. This
value has a range [0,1] and is expressed as fractions of the image height.
k1 Coefficient of the quadratic correction term. 0.5 means no correction.
k2 Coefficient of the double quadratic correction term. 0.5 means no correction.
The formula that generates the correction is:
r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)
where r_0 is halve of the image diagonal and r_src and r_tgt are the distances from the focal point in
the source and target images, respectively.
libvmaf
Obtain the average VMAF (Video Multi-Method Assessment Fusion) score between two input videos.
This filter takes two input videos.
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 VMAF score is printed through the logging system.
It requires Netflix's vmaf library (libvmaf) as a pre-requisite. After installing the library it can be
enabled using: "./configure --enable-libvmaf". If no model path is specified it uses the default model:
"vmaf_v0.6.1.pkl".
On the below examples the input file main.mpg being processed is compared with the reference file
ref.mpg.
The filter has following options:
model_path
Set the model path which is to be used for SVM. Default value: "vmaf_v0.6.1.pkl"
log_path
Set the file path to be used to store logs.
log_fmt
Set the format of the log file (xml or json).
enable_transform
Enables transform for computing vmaf.
phone_model
Invokes the phone model which will generate VMAF scores higher than in the regular model, which is
more suitable for laptop, TV, etc. viewing conditions.
psnr
Enables computing psnr along with vmaf.
ssim
Enables computing ssim along with vmaf.
ms_ssim
Enables computing ms_ssim along with vmaf.
pool
Set the pool method to be used for computing vmaf.
This filter also supports the framesync options.
For example:
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -
Example with options:
ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -
limiter
Limits the pixel components values to the specified range [min, max].
The filter accepts the following options:
min Lower bound. Defaults to the lowest allowed value for the input.
max Upper bound. Defaults to the highest allowed value for the input.
planes
Specify which planes will be processed. Defaults to all available.
loop
Loop video frames.
The filter accepts the following options:
loop
Set the number of loops.
size
Set maximal size in number of frames.
start
Set first frame of loop.
lut3d
Apply a 3D LUT to an input video.
The filter accepts the following options:
file
Set the 3D LUT file name.
Currently supported formats:
3dl AfterEffects
cube
Iridas
dat DaVinci
m3d Pandora
interp
Select interpolation mode.
Available values are:
nearest
Use values from the nearest defined point.
trilinear
Interpolate values using the 8 points defining a cube.
tetrahedral
Interpolate values using a tetrahedron.
This filter also supports the framesync options.
lumakey
Turn certain luma values into transparency.
The filter accepts the following options:
threshold
Set the luma which will be used as base for transparency. Default value is 0.
tolerance
Set the range of luma values to be keyed out. Default value is 0.
softness
Set the range of softness. Default value is 0. Use this to control gradual transition from zero to
full transparency.
lut, lutrgb, lutyuv
Compute a look-up table for binding each pixel component input value to an output value, and apply it to
the input video.
lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.
These filters accept the following parameters:
c0 set first pixel component expression
c1 set second pixel component expression
c2 set third pixel component expression
c3 set fourth pixel component expression, corresponds to the alpha component
r set red component expression
g set green component expression
b set blue component expression
a alpha component expression
y set Y/luminance component expression
u set U/Cb component expression
v set V/Cr component expression
Each of them specifies the expression to use for computing the lookup table for the corresponding pixel
component values.
The exact component associated to each of the c* options depends on the format in input.
The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in
input, and lutyuv requires YUV.
The expressions can contain the following constants and functions:
w
h The input width and height.
val The input value for the pixel component.
clipval
The input value, clipped to the minval-maxval range.
maxval
The maximum value for the pixel component.
minval
The minimum value for the pixel component.
negval
The negated value for the pixel component value, clipped to the minval-maxval range; it corresponds
to the expression "maxval-clipval+minval".
clip(val)
The computed value in val, clipped to the minval-maxval range.
gammaval(gamma)
The computed gamma correction value of the pixel component value, clipped to the minval-maxval range.
It corresponds to the expression
"pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"
All expressions default to "val".
Examples
• Negate input video:
lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"
The above is the same as:
lutrgb="r=negval:g=negval:b=negval"
lutyuv="y=negval:u=negval:v=negval"
• Negate luminance:
lutyuv=y=negval
• Remove chroma components, turning the video into a graytone image:
lutyuv="u=128:v=128"
• Apply a luma burning effect:
lutyuv="y=2*val"
• Remove green and blue components:
lutrgb="g=0:b=0"
• Set a constant alpha channel value on input:
format=rgba,lutrgb=a="maxval-minval/2"
• Correct luminance gamma by a factor of 0.5:
lutyuv=y=gammaval(0.5)
• Discard least significant bits of luma:
lutyuv=y='bitand(val, 128+64+32)'
• Technicolor like effect:
lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'
lut2, tlut2
The "lut2" filter takes two input streams and outputs one stream.
The "tlut2" (time lut2) filter takes two consecutive frames from one single stream.
This filter accepts the following parameters:
c0 set first pixel component expression
c1 set second pixel component expression
c2 set third pixel component expression
c3 set fourth pixel component expression, corresponds to the alpha component
Each of them specifies the expression to use for computing the lookup table for the corresponding pixel
component values.
The exact component associated to each of the c* options depends on the format in inputs.
The expressions can contain the following constants:
w
h The input width and height.
x The first input value for the pixel component.
y The second input value for the pixel component.
bdx The first input video bit depth.
bdy The second input video bit depth.
All expressions default to "x".
Examples
• Highlight differences between two RGB video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'
• Highlight differences between two YUV video streams:
lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'
• Show max difference between two video streams:
lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'
maskedclamp
Clamp the first input stream with the second input and third input stream.
Returns the value of first stream to be between second input stream - "undershoot" and third input stream
+ "overshoot".
This filter accepts the following options:
undershoot
Default value is 0.
overshoot
Default value is 0.
planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream.
By default value 0xf, all planes will be processed.
maskedmerge
Merge the first input stream with the second input stream using per pixel weights in the third input
stream.
A value of 0 in the third stream pixel component means that pixel component from first stream is returned
unchanged, while maximum value (eg. 255 for 8-bit videos) means that pixel component from second stream
is returned unchanged. Intermediate values define the amount of merging between both input stream's pixel
components.
This filter accepts the following options:
planes
Set which planes will be processed as bitmap, unprocessed planes will be copied from first stream.
By default value 0xf, all planes will be processed.
mcdeint
Apply motion-compensation deinterlacing.
It needs one field per frame as input and must thus be used together with yadif=1/3 or equivalent.
This filter accepts the following options:
mode
Set the deinterlacing mode.
It accepts one of the following values:
fast
medium
slow
use iterative motion estimation
extra_slow
like slow, but use multiple reference frames.
Default value is fast.
parity
Set the picture field parity assumed for the input video. It must be one of the following values:
0, tff
assume top field first
1, bff
assume bottom field first
Default value is bff.
qp Set per-block quantization parameter (QP) used by the internal encoder.
Higher values should result in a smoother motion vector field but less optimal individual vectors.
Default value is 1.
mergeplanes
Merge color channel components from several video streams.
The filter accepts up to 4 input streams, and merge selected input planes to the output video.
This filter accepts the following options:
mapping
Set input to output plane mapping. Default is 0.
The mappings is specified as a bitmap. It should be specified as a hexadecimal number in the form
0xAa[Bb[Cc[Dd]]]. 'Aa' describes the mapping for the first plane of the output stream. 'A' sets the
number of the input stream to use (from 0 to 3), and 'a' the plane number of the corresponding input
to use (from 0 to 3). The rest of the mappings is similar, 'Bb' describes the mapping for the output
stream second plane, 'Cc' describes the mapping for the output stream third plane and 'Dd' describes
the mapping for the output stream fourth plane.
format
Set output pixel format. Default is "yuva444p".
Examples
• Merge three gray video streams of same width and height into single video stream:
[a0][a1][a2]mergeplanes=0x001020:yuv444p
• Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream:
[a0][a1]mergeplanes=0x00010210:yuva444p
• Swap Y and A plane in yuva444p stream:
format=yuva444p,mergeplanes=0x03010200:yuva444p
• Swap U and V plane in yuv420p stream:
format=yuv420p,mergeplanes=0x000201:yuv420p
• Cast a rgb24 clip to yuv444p:
format=rgb24,mergeplanes=0x000102:yuv444p
mestimate
Estimate and export motion vectors using block matching algorithms. Motion vectors are stored in frame
side data to be used by other filters.
This filter accepts the following options:
method
Specify the motion estimation method. Accepts one of the following values:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default value is esa.
mb_size
Macroblock size. Default 16.
search_param
Search parameter. Default 7.
midequalizer
Apply Midway Image Equalization effect using two video streams.
Midway Image Equalization adjusts a pair of images to have the same histogram, while maintaining their
dynamics as much as possible. It's useful for e.g. matching exposures from a pair of stereo cameras.
This filter has two inputs and one output, which must be of same pixel format, but may be of different
sizes. The output of filter is first input adjusted with midway histogram of both inputs.
This filter accepts the following option:
planes
Set which planes to process. Default is 15, which is all available planes.
minterpolate
Convert the video to specified frame rate using motion interpolation.
This filter accepts the following options:
fps Specify the output frame rate. This can be rational e.g. "60000/1001". Frames are dropped if fps is
lower than source fps. Default 60.
mi_mode
Motion interpolation mode. Following values are accepted:
dup Duplicate previous or next frame for interpolating new ones.
blend
Blend source frames. Interpolated frame is mean of previous and next frames.
mci Motion compensated interpolation. Following options are effective when this mode is selected:
mc_mode
Motion compensation mode. Following values are accepted:
obmc
Overlapped block motion compensation.
aobmc
Adaptive overlapped block motion compensation. Window weighting coefficients are
controlled adaptively according to the reliabilities of the neighboring motion vectors to
reduce oversmoothing.
Default mode is obmc.
me_mode
Motion estimation mode. Following values are accepted:
bidir
Bidirectional motion estimation. Motion vectors are estimated for each source frame in
both forward and backward directions.
bilat
Bilateral motion estimation. Motion vectors are estimated directly for interpolated
frame.
Default mode is bilat.
me The algorithm to be used for motion estimation. Following values are accepted:
esa Exhaustive search algorithm.
tss Three step search algorithm.
tdls
Two dimensional logarithmic search algorithm.
ntss
New three step search algorithm.
fss Four step search algorithm.
ds Diamond search algorithm.
hexbs
Hexagon-based search algorithm.
epzs
Enhanced predictive zonal search algorithm.
umh Uneven multi-hexagon search algorithm.
Default algorithm is epzs.
mb_size
Macroblock size. Default 16.
search_param
Motion estimation search parameter. Default 32.
vsbmc
Enable variable-size block motion compensation. Motion estimation is applied with smaller
block sizes at object boundaries in order to make the them less blur. Default is 0
(disabled).
scd Scene change detection method. Scene change leads motion vectors to be in random direction. Scene
change detection replace interpolated frames by duplicate ones. May not be needed for other modes.
Following values are accepted:
none
Disable scene change detection.
fdiff
Frame difference. Corresponding pixel values are compared and if it satisfies scd_threshold scene
change is detected.
Default method is fdiff.
scd_threshold
Scene change detection threshold. Default is 5.0.
mpdecimate
Drop frames that do not differ greatly from the previous frame in order to reduce frame rate.
The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it
could in theory be used for fixing movies that were inverse-telecined incorrectly.
A description of the accepted options follows.
max Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum
interval between dropped frames (if negative). If the value is 0, the frame is dropped disregarding
the number of previous sequentially dropped frames.
Default value is 0.
hi
lo
frac
Set the dropping threshold values.
Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a
threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out
differently over the block.
A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no
more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo.
Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33.
negate
Negate input video.
It accepts an integer in input; if non-zero it negates the alpha component (if available). The default
value in input is 0.
nlmeans
Denoise frames using Non-Local Means algorithm.
Each pixel is adjusted by looking for other pixels with similar contexts. This context similarity is
defined by comparing their surrounding patches of size pxp. Patches are searched in an area of rxr around
the pixel.
Note that the research area defines centers for patches, which means some patches will be made of pixels
outside that research area.
The filter accepts the following options.
s Set denoising strength.
p Set patch size.
pc Same as p but for chroma planes.
The default value is 0 and means automatic.
r Set research size.
rc Same as r but for chroma planes.
The default value is 0 and means automatic.
nnedi
Deinterlace video using neural network edge directed interpolation.
This filter accepts the following options:
weights
Mandatory option, without binary file filter can not work. Currently file can be found here:
https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin
deint
Set which frames to deinterlace, by default it is "all". Can be "all" or "interlaced".
field
Set mode of operation.
Can be one of the following:
af Use frame flags, both fields.
a Use frame flags, single field.
t Use top field only.
b Use bottom field only.
tf Use both fields, top first.
bf Use both fields, bottom first.
planes
Set which planes to process, by default filter process all frames.
nsize
Set size of local neighborhood around each pixel, used by the predictor neural network.
Can be one of the following:
s8x6
s16x6
s32x6
s48x6
s8x4
s16x4
s32x4
nns Set the number of neurons in predictor neural network. Can be one of the following:
n16
n32
n64
n128
n256
qual
Controls the number of different neural network predictions that are blended together to compute the
final output value. Can be "fast", default or "slow".
etype
Set which set of weights to use in the predictor. Can be one of the following:
a weights trained to minimize absolute error
s weights trained to minimize squared error
pscrn
Controls whether or not the prescreener neural network is used to decide which pixels should be
processed by the predictor neural network and which can be handled by simple cubic interpolation.
The prescreener is trained to know whether cubic interpolation will be sufficient for a pixel or
whether it should be predicted by the predictor nn. The computational complexity of the prescreener
nn is much less than that of the predictor nn. Since most pixels can be handled by cubic
interpolation, using the prescreener generally results in much faster processing. The prescreener is
pretty accurate, so the difference between using it and not using it is almost always unnoticeable.
Can be one of the following:
none
original
new
Default is "new".
fapprox
Set various debugging flags.
noformat
Force libavfilter not to use any of the specified pixel formats for the input to the next filter.
It accepts the following parameters:
pix_fmts
A '|'-separated list of pixel format names, such as pix_fmts=yuv420p|monow|rgb24".
Examples
• Force libavfilter to use a format different from yuv420p for the input to the vflip filter:
noformat=pix_fmts=yuv420p,vflip
• Convert the input video to any of the formats not contained in the list:
noformat=yuv420p|yuv444p|yuv410p
noise
Add noise on video input frame.
The filter accepts the following options:
all_seed
c0_seed
c1_seed
c2_seed
c3_seed
Set noise seed for specific pixel component or all pixel components in case of all_seed. Default
value is 123457.
all_strength, alls
c0_strength, c0s
c1_strength, c1s
c2_strength, c2s
c3_strength, c3s
Set noise strength for specific pixel component or all pixel components in case all_strength. Default
value is 0. Allowed range is [0, 100].
all_flags, allf
c0_flags, c0f
c1_flags, c1f
c2_flags, c2f
c3_flags, c3f
Set pixel component flags or set flags for all components if all_flags. Available values for
component flags are:
a averaged temporal noise (smoother)
p mix random noise with a (semi)regular pattern
t temporal noise (noise pattern changes between frames)
u uniform noise (gaussian otherwise)
Examples
Add temporal and uniform noise to input video:
noise=alls=20:allf=t+u
null
Pass the video source unchanged to the output.
ocr
Optical Character Recognition
This filter uses Tesseract for optical character recognition.
It accepts the following options:
datapath
Set datapath to tesseract data. Default is to use whatever was set at installation.
language
Set language, default is "eng".
whitelist
Set character whitelist.
blacklist
Set character blacklist.
The filter exports recognized text as the frame metadata "lavfi.ocr.text".
ocv
Apply a video transform using libopencv.
To enable this filter, install the libopencv library and headers and configure FFmpeg with
"--enable-libopencv".
It accepts the following parameters:
filter_name
The name of the libopencv filter to apply.
filter_params
The parameters to pass to the libopencv filter. If not specified, the default values are assumed.
Refer to the official libopencv documentation for more precise information:
<http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>
Several libopencv filters are supported; see the following subsections.
dilate
Dilate an image by using a specific structuring element. It corresponds to the libopencv function
"cvDilate".
It accepts the parameters: struct_el|nb_iterations.
struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape
cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y
the anchor point, and shape the shape for the structuring element. shape must be "rect", "cross",
"ellipse", or "custom".
If the value for shape is "custom", it must be followed by a string of the form "=filename". The file
with name filename is assumed to represent a binary image, with each printable character corresponding to
a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of
the read file are assumed instead.
The default value for struct_el is "3x3+0x0/rect".
nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.
Some examples:
# Use the default values
ocv=dilate
# Dilate using a structuring element with a 5x5 cross, iterating two times
ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2
# Read the shape from the file diamond.shape, iterating two times.
# The file diamond.shape may contain a pattern of characters like this
# *
# ***
# *****
# ***
# *
# The specified columns and rows are ignored
# but the anchor point coordinates are not
ocv=dilate:0x0+2x2/custom=diamond.shape|2
erode
Erode an image by using a specific structuring element. It corresponds to the libopencv function
"cvErode".
It accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate
filter.
smooth
Smooth the input video.
The filter takes the following parameters: type|param1|param2|param3|param4.
type is the type of smooth filter to apply, and must be one of the following values: "blur",
"blur_no_scale", "median", "gaussian", or "bilateral". The default value is "gaussian".
The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and param2 accept
integer positive values or 0. param3 and param4 accept floating point values.
The default value for param1 is 3. The default value for the other parameters is 0.
These parameters correspond to the parameters assigned to the libopencv function "cvSmooth".
oscilloscope
2D Video Oscilloscope.
Useful to measure spatial impulse, step responses, chroma delays, etc.
It accepts the following parameters:
x Set scope center x position.
y Set scope center y position.
s Set scope size, relative to frame diagonal.
t Set scope tilt/rotation.
o Set trace opacity.
tx Set trace center x position.
ty Set trace center y position.
tw Set trace width, relative to width of frame.
th Set trace height, relative to height of frame.
c Set which components to trace. By default it traces first three components.
g Draw trace grid. By default is enabled.
st Draw some statistics. By default is enabled.
sc Draw scope. By default is enabled.
Examples
• Inspect full first row of video frame.
oscilloscope=x=0.5:y=0:s=1
• Inspect full last row of video frame.
oscilloscope=x=0.5:y=1:s=1
• Inspect full 5th line of video frame of height 1080.
oscilloscope=x=0.5:y=5/1080:s=1
• Inspect full last column of video frame.
oscilloscope=x=1:y=0.5:s=1:t=1
overlay
Overlay one video on top of another.
It takes two inputs and has one output. The first input is the "main" video on which the second input is
overlaid.
It accepts the following parameters:
A description of the accepted options follows.
x
y Set the expression for the x and y coordinates of the overlaid video on the main video. Default value
is "0" for both expressions. In case the expression is invalid, it is set to a huge value (meaning
that the overlay will not be displayed within the output visible area).
eof_action
See framesync.
eval
Set when the expressions for x, and y are evaluated.
It accepts the following values:
init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame
Default value is frame.
shortest
See framesync.
format
Set the format for the output video.
It accepts the following values:
yuv420
force YUV420 output
yuv422
force YUV422 output
yuv444
force YUV444 output
rgb force packed RGB output
gbrp
force planar RGB output
auto
automatically pick format
Default value is yuv420.
repeatlast
See framesync.
The x, and y expressions can contain the following parameters.
main_w, W
main_h, H
The main input width and height.
overlay_w, w
overlay_h, h
The overlay input width and height.
x
y The computed values for x and y. They are evaluated for each new frame.
hsub
vsub
horizontal and vertical chroma subsample values of the output format. For example for the pixel
format "yuv422p" hsub is 2 and vsub is 1.
n the number of input frame, starting from 0
pos the position in the file of the input frame, NAN if unknown
t The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown.
This filter also supports the framesync options.
Note that the n, pos, t variables are available only when evaluation is done per frame, and will evaluate
to NAN when eval is set to init.
Be aware that frames are taken from each input video in timestamp order, hence, if their initial
timestamps differ, it is a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have
them begin in the same zero timestamp, as the example for the movie filter does.
You can chain together more overlays but you should test the efficiency of such approach.
Commands
This filter supports the following commands:
x
y Modify the x and y of the overlay input. The command accepts the same syntax of the corresponding
option.
If the specified expression is not valid, it is kept at its current value.
Examples
• Draw the overlay at 10 pixels from the bottom right corner of the main video:
overlay=main_w-overlay_w-10:main_h-overlay_h-10
Using named options the example above becomes:
overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10
• Insert a transparent PNG logo in the bottom left corner of the input, using the ffmpeg tool with the
"-filter_complex" option:
ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output
• Insert 2 different transparent PNG logos (second logo on bottom right corner) using the ffmpeg tool:
ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output
• Add a transparent color layer on top of the main video; "WxH" must specify the size of the main input
to the overlay filter:
color=color=red@.3:size=WxH [over]; [in][over] overlay [out]
• Play an original video and a filtered version (here with the deshake filter) side by side using the
ffplay tool:
ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'
The above command is the same as:
ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'
• Make a sliding overlay appearing from the left to the right top part of the screen starting since
time 2:
overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0
• Compose output by putting two input videos side to side:
ffmpeg -i left.avi -i right.avi -filter_complex "
nullsrc=size=200x100 [background];
[0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
[1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
[background][left] overlay=shortest=1 [background+left];
[background+left][right] overlay=shortest=1:x=100 [left+right]
"
• Mask 10-20 seconds of a video by applying the delogo filter to a section
ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
-vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
masked.avi
• Chain several overlays in cascade:
nullsrc=s=200x200 [bg];
testsrc=s=100x100, split=4 [in0][in1][in2][in3];
[in0] lutrgb=r=0, [bg] overlay=0:0 [mid0];
[in1] lutrgb=g=0, [mid0] overlay=100:0 [mid1];
[in2] lutrgb=b=0, [mid1] overlay=0:100 [mid2];
[in3] null, [mid2] overlay=100:100 [out0]
owdenoise
Apply Overcomplete Wavelet denoiser.
The filter accepts the following options:
depth
Set depth.
Larger depth values will denoise lower frequency components more, but slow down filtering.
Must be an int in the range 8-16, default is 8.
luma_strength, ls
Set luma strength.
Must be a double value in the range 0-1000, default is 1.0.
chroma_strength, cs
Set chroma strength.
Must be a double value in the range 0-1000, default is 1.0.
pad
Add paddings to the input image, and place the original input at the provided x, y coordinates.
It accepts the following parameters:
width, w
height, h
Specify an expression for the size of the output image with the paddings added. If the value for
width or height is 0, the corresponding input size is used for the output.
The width expression can reference the value set by the height expression, and vice versa.
The default value of width and height is 0.
x
y Specify the offsets to place the input image at within the padded area, with respect to the top/left
border of the output image.
The x expression can reference the value set by the y expression, and vice versa.
The default value of x and y is 0.
If x or y evaluate to a negative number, they'll be changed so the input image is centered on the
padded area.
color
Specify the color of the padded area. For the syntax of this option, check the "Color" section in the
ffmpeg-utils manual.
The default value of color is "black".
eval
Specify when to evaluate width, height, x and y expression.
It accepts the following values:
init
Only evaluate expressions once during the filter initialization or when a command is processed.
frame
Evaluate expressions for each incoming frame.
Default value is init.
aspect
Pad to aspect instead to a resolution.
The value for the width, height, x, and y options are expressions containing the following constants:
in_w
in_h
The input video width and height.
iw
ih These are the same as in_w and in_h.
out_w
out_h
The output width and height (the size of the padded area), as specified by the width and height
expressions.
ow
oh These are the same as out_w and out_h.
x
y The x and y offsets as specified by the x and y expressions, or NAN if not yet specified.
a same as iw / ih
sar input sample aspect ratio
dar input display aspect ratio, it is the same as (iw / ih) * sar
hsub
vsub
The horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub
is 2 and vsub is 1.
Examples
• Add paddings with the color "violet" to the input video. The output video size is 640x480, and the
top-left corner of the input video is placed at column 0, row 40
pad=640:480:0:40:violet
The example above is equivalent to the following command:
pad=width=640:height=480:x=0:y=40:color=violet
• Pad the input to get an output with dimensions increased by 3/2, and put the input video at the
center of the padded area:
pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"
• Pad the input to get a squared output with size equal to the maximum value between the input width
and height, and put the input video at the center of the padded area:
pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"
• Pad the input to get a final w/h ratio of 16:9:
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"
• In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to
use sar in the expression, according to the relation:
(ih * X / ih) * sar = output_dar
X = output_dar / sar
Thus the previous example needs to be modified to:
pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"
• Double the output size and put the input video in the bottom-right corner of the output padded area:
pad="2*iw:2*ih:ow-iw:oh-ih"
palettegen
Generate one palette for a whole video stream.
It accepts the following options:
max_colors
Set the maximum number of colors to quantize in the palette. Note: the palette will still contain
256 colors; the unused palette entries will be black.
reserve_transparent
Create a palette of 255 colors maximum and reserve the last one for transparency. Reserving the
transparency color is useful for GIF optimization. If not set, the maximum of colors in the palette
will be 256. You probably want to disable this option for a standalone image. Set by default.
stats_mode
Set statistics mode.
It accepts the following values:
full
Compute full frame histograms.
diff
Compute histograms only for the part that differs from previous frame. This might be relevant to
give more importance to the moving part of your input if the background is static.
single
Compute new histogram for each frame.
Default value is full.
The filter also exports the frame metadata "lavfi.color_quant_ratio" ("nb_color_in / nb_color_out") which
you can use to evaluate the degree of color quantization of the palette. This information is also visible
at info logging level.
Examples
• Generate a representative palette of a given video using ffmpeg:
ffmpeg -i input.mkv -vf palettegen palette.png
paletteuse
Use a palette to downsample an input video stream.
The filter takes two inputs: one video stream and a palette. The palette must be a 256 pixels image.
It accepts the following options:
dither
Select dithering mode. Available algorithms are:
bayer
Ordered 8x8 bayer dithering (deterministic)
heckbert
Dithering as defined by Paul Heckbert in 1982 (simple error diffusion). Note: this dithering is
sometimes considered "wrong" and is included as a reference.
floyd_steinberg
Floyd and Steingberg dithering (error diffusion)
sierra2
Frankie Sierra dithering v2 (error diffusion)
sierra2_4a
Frankie Sierra dithering v2 "Lite" (error diffusion)
Default is sierra2_4a.
bayer_scale
When bayer dithering is selected, this option defines the scale of the pattern (how much the
crosshatch pattern is visible). A low value means more visible pattern for less banding, and higher
value means less visible pattern at the cost of more banding.
The option must be an integer value in the range [0,5]. Default is 2.
diff_mode
If set, define the zone to process
rectangle
Only the changing rectangle will be reprocessed. This is similar to GIF cropping/offsetting
compression mechanism. This option can be useful for speed if only a part of the image is
changing, and has use cases such as limiting the scope of the error diffusal dither to the
rectangle that bounds the moving scene (it leads to more deterministic output if the scene
doesn't change much, and as a result less moving noise and better GIF compression).
Default is none.
new Take new palette for each output frame.
Examples
• Use a palette (generated for example with palettegen) to encode a GIF using ffmpeg:
ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif
perspective
Correct perspective of video not recorded perpendicular to the screen.
A description of the accepted parameters follows.
x0
y0
x1
y1
x2
y2
x3
y3 Set coordinates expression for top left, top right, bottom left and bottom right corners. Default
values are "0:0:W:0:0:H:W:H" with which perspective will remain unchanged. If the "sense" option is
set to "source", then the specified points will be sent to the corners of the destination. If the
"sense" option is set to "destination", then the corners of the source will be sent to the specified
coordinates.
The expressions can use the following variables:
W
H the width and height of video frame.
in Input frame count.
on Output frame count.
interpolation
Set interpolation for perspective correction.
It accepts the following values:
linear
cubic
Default value is linear.
sense
Set interpretation of coordinate options.
It accepts the following values:
0, source
Send point in the source specified by the given coordinates to the corners of the destination.
1, destination
Send the corners of the source to the point in the destination specified by the given
coordinates.
Default value is source.
eval
Set when the expressions for coordinates x0,y0,...x3,y3 are evaluated.
It accepts the following values:
init
only evaluate expressions once during the filter initialization or when a command is processed
frame
evaluate expressions for each incoming frame
Default value is init.
phase
Delay interlaced video by one field time so that the field order changes.
The intended use is to fix PAL movies that have been captured with the opposite field order to the film-
to-video transfer.
A description of the accepted parameters follows.
mode
Set phase mode.
It accepts the following values:
t Capture field order top-first, transfer bottom-first. Filter will delay the bottom field.
b Capture field order bottom-first, transfer top-first. Filter will delay the top field.
p Capture and transfer with the same field order. This mode only exists for the documentation of
the other options to refer to, but if you actually select it, the filter will faithfully do
nothing.
a Capture field order determined automatically by field flags, transfer opposite. Filter selects
among t and b modes on a frame by frame basis using field flags. If no field information is
available, then this works just like u.
u Capture unknown or varying, transfer opposite. Filter selects among t and b on a frame by frame
basis by analyzing the images and selecting the alternative that produces best match between the
fields.
T Capture top-first, transfer unknown or varying. Filter selects among t and p using image
analysis.
B Capture bottom-first, transfer unknown or varying. Filter selects among b and p using image
analysis.
A Capture determined by field flags, transfer unknown or varying. Filter selects among t, b and p
using field flags and image analysis. If no field information is available, then this works just
like U. This is the default mode.
U Both capture and transfer unknown or varying. Filter selects among t, b and p using image
analysis only.
pixdesctest
Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal
to the input video.
For example:
format=monow, pixdesctest
can be used to test the monowhite pixel format descriptor definition.
pixscope
Display sample values of color channels. Mainly useful for checking color and levels. Minimum supported
resolution is 640x480.
The filters accept the following options:
x Set scope X position, relative offset on X axis.
y Set scope Y position, relative offset on Y axis.
w Set scope width.
h Set scope height.
o Set window opacity. This window also holds statistics about pixel area.
wx Set window X position, relative offset on X axis.
wy Set window Y position, relative offset on Y axis.
pp
Enable the specified chain of postprocessing subfilters using libpostproc. This library should be
automatically selected with a GPL build ("--enable-gpl"). Subfilters must be separated by '/' and can be
disabled by prepending a '-'. Each subfilter and some options have a short and a long name that can be
used interchangeably, i.e. dr/dering are the same.
The filters accept the following options:
subfilters
Set postprocessing subfilters string.
All subfilters share common options to determine their scope:
a/autoq
Honor the quality commands for this subfilter.
c/chrom
Do chrominance filtering, too (default).
y/nochrom
Do luminance filtering only (no chrominance).
n/noluma
Do chrominance filtering only (no luminance).
These options can be appended after the subfilter name, separated by a '|'.
Available subfilters are:
hb/hdeblock[|difference[|flatness]]
Horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
vb/vdeblock[|difference[|flatness]]
Vertical deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
ha/hadeblock[|difference[|flatness]]
Accurate horizontal deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
va/vadeblock[|difference[|flatness]]
Accurate vertical deblocking filter
difference
Difference factor where higher values mean more deblocking (default: 32).
flatness
Flatness threshold where lower values mean more deblocking (default: 39).
The horizontal and vertical deblocking filters share the difference and flatness values so you cannot set
different horizontal and vertical thresholds.
h1/x1hdeblock
Experimental horizontal deblocking filter
v1/x1vdeblock
Experimental vertical deblocking filter
dr/dering
Deringing filter
tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer
threshold1
larger -> stronger filtering
threshold2
larger -> stronger filtering
threshold3
larger -> stronger filtering
al/autolevels[:f/fullyrange], automatic brightness / contrast correction
f/fullyrange
Stretch luminance to "0-255".
lb/linblenddeint
Linear blend deinterlacing filter that deinterlaces the given block by filtering all lines with a "(1
2 1)" filter.
li/linipoldeint
Linear interpolating deinterlacing filter that deinterlaces the given block by linearly interpolating
every second line.
ci/cubicipoldeint
Cubic interpolating deinterlacing filter deinterlaces the given block by cubically interpolating
every second line.
md/mediandeint
Median deinterlacing filter that deinterlaces the given block by applying a median filter to every
second line.
fd/ffmpegdeint
FFmpeg deinterlacing filter that deinterlaces the given block by filtering every second line with a
"(-1 4 2 4 -1)" filter.
l5/lowpass5
Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given block by filtering
all lines with a "(-1 2 6 2 -1)" filter.
fq/forceQuant[|quantizer]
Overrides the quantizer table from the input with the constant quantizer you specify.
quantizer
Quantizer to use
de/default
Default pp filter combination ("hb|a,vb|a,dr|a")
fa/fast
Fast pp filter combination ("h1|a,v1|a,dr|a")
ac High quality pp filter combination ("ha|a|128|7,va|a,dr|a")
Examples
• Apply horizontal and vertical deblocking, deringing and automatic brightness/contrast:
pp=hb/vb/dr/al
• Apply default filters without brightness/contrast correction:
pp=de/-al
• Apply default filters and temporal denoiser:
pp=default/tmpnoise|1|2|3
• Apply deblocking on luminance only, and switch vertical deblocking on or off automatically depending
on available CPU time:
pp=hb|y/vb|a
pp7
Apply Postprocessing filter 7. It is variant of the spp filter, similar to spp = 6 with 7 point DCT,
where only the center sample is used after IDCT.
The filter accepts the following options:
qp Force a constant quantization parameter. It accepts an integer in range 0 to 63. If not set, the
filter will use the QP from the video stream (if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding.
soft
Set soft thresholding (better de-ringing effect, but likely blurrier).
medium
Set medium thresholding (good results, default).
premultiply
Apply alpha premultiply effect to input video stream using first plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
inplace
Do not require 2nd input for processing, instead use alpha plane from input stream.
prewitt
Apply prewitt operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
pseudocolor
Alter frame colors in video with pseudocolors.
This filter accept the following options:
c0 set pixel first component expression
c1 set pixel second component expression
c2 set pixel third component expression
c3 set pixel fourth component expression, corresponds to the alpha component
i set component to use as base for altering colors
Each of them specifies the expression to use for computing the lookup table for the corresponding pixel
component values.
The expressions can contain the following constants and functions:
w
h The input width and height.
val The input value for the pixel component.
ymin, umin, vmin, amin
The minimum allowed component value.
ymax, umax, vmax, amax
The maximum allowed component value.
All expressions default to "val".
Examples
• Change too high luma values to gradient:
pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"
psnr
Obtain the average, maximum and minimum PSNR (Peak Signal to Noise Ratio) between two input videos.
This filter takes in input two input videos, the first input is considered the "main" source and is
passed unchanged to the output. The second input is used as a "reference" video for computing the PSNR.
Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also
it assumes that both inputs have the same number of frames, which are compared one by one.
The obtained average PSNR is printed through the logging system.
The filter stores the accumulated MSE (mean squared error) of each frame, and at the end of the
processing it is averaged across all frames equally, and the following formula is applied to obtain the
PSNR:
PSNR = 10*log10(MAX^2/MSE)
Where MAX is the average of the maximum values of each component of the image.
The description of the accepted parameters follows.
stats_file, f
If specified the filter will use the named file to save the PSNR of each individual frame. When
filename equals "-" the data is sent to standard output.
stats_version
Specifies which version of the stats file format to use. Details of each format are written below.
Default value is 1.
stats_add_max
Determines whether the max value is output to the stats log. Default value is 0. Requires
stats_version >= 2. If this is set and stats_version < 2, the filter will return an error.
This filter also supports the framesync options.
The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value
for each compared couple of frames.
If a stats_version greater than 1 is specified, a header line precedes the list of per-frame-pair stats,
with key value pairs following the frame format with the following parameters:
psnr_log_version
The version of the log file format. Will match stats_version.
fields
A comma separated list of the per-frame-pair parameters included in the log.
A description of each shown per-frame-pair parameter follows:
n sequential number of the input frame, starting from 1
mse_avg
Mean Square Error pixel-by-pixel average difference of the compared frames, averaged over all the
image components.
mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
Mean Square Error pixel-by-pixel average difference of the compared frames for the component
specified by the suffix.
psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
Peak Signal to Noise ratio of the compared frames for the component specified by the suffix.
max_avg, max_y, max_u, max_v
Maximum allowed value for each channel, and average over all channels.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] psnr="stats_file=stats.log" [out]
On this example the input file being processed is compared with the reference file ref_movie.mpg. The
PSNR of each individual frame is stored in stats.log.
pullup
Pulldown reversal (inverse telecine) filter, capable of handling mixed hard-telecine, 24000/1001 fps
progressive, and 30000/1001 fps progressive content.
The pullup filter is designed to take advantage of future context in making its decisions. This filter is
stateless in the sense that it does not lock onto a pattern to follow, but it instead looks forward to
the following fields in order to identify matches and rebuild progressive frames.
To produce content with an even framerate, insert the fps filter after pullup, use "fps=24000/1001" if
the input frame rate is 29.97fps, "fps=24" for 30fps and the (rare) telecined 25fps input.
The filter accepts the following options:
jl
jr
jt
jb These options set the amount of "junk" to ignore at the left, right, top, and bottom of the image,
respectively. Left and right are in units of 8 pixels, while top and bottom are in units of 2 lines.
The default is 8 pixels on each side.
sb Set the strict breaks. Setting this option to 1 will reduce the chances of filter generating an
occasional mismatched frame, but it may also cause an excessive number of frames to be dropped during
high motion sequences. Conversely, setting it to -1 will make filter match fields more easily. This
may help processing of video where there is slight blurring between the fields, but may also cause
there to be interlaced frames in the output. Default value is 0.
mp Set the metric plane to use. It accepts the following values:
l Use luma plane.
u Use chroma blue plane.
v Use chroma red plane.
This option may be set to use chroma plane instead of the default luma plane for doing filter's
computations. This may improve accuracy on very clean source material, but more likely will decrease
accuracy, especially if there is chroma noise (rainbow effect) or any grayscale video. The main
purpose of setting mp to a chroma plane is to reduce CPU load and make pullup usable in realtime on
slow machines.
For best results (without duplicated frames in the output file) it is necessary to change the output
frame rate. For example, to inverse telecine NTSC input:
ffmpeg -i input -vf pullup -r 24000/1001 ...
qp
Change video quantization parameters (QP).
The filter accepts the following option:
qp Set expression for quantization parameter.
The expression is evaluated through the eval API and can contain, among others, the following constants:
known
1 if index is not 129, 0 otherwise.
qp Sequential index starting from -129 to 128.
Examples
• Some equation like:
qp=2+2*sin(PI*qp)
random
Flush video frames from internal cache of frames into a random order. No frame is discarded. Inspired
by frei0r nervous filter.
frames
Set size in number of frames of internal cache, in range from 2 to 512. Default is 30.
seed
Set seed for random number generator, must be an integer included between 0 and "UINT32_MAX". If not
specified, or if explicitly set to less than 0, the filter will try to use a good random seed on a
best effort basis.
readeia608
Read closed captioning (EIA-608) information from the top lines of a video frame.
This filter adds frame metadata for "lavfi.readeia608.X.cc" and "lavfi.readeia608.X.line", where "X" is
the number of the identified line with EIA-608 data (starting from 0). A description of each metadata
value follows:
lavfi.readeia608.X.cc
The two bytes stored as EIA-608 data (printed in hexadecimal).
lavfi.readeia608.X.line
The number of the line on which the EIA-608 data was identified and read.
This filter accepts the following options:
scan_min
Set the line to start scanning for EIA-608 data. Default is 0.
scan_max
Set the line to end scanning for EIA-608 data. Default is 29.
mac Set minimal acceptable amplitude change for sync codes detection. Default is 0.2. Allowed range is
"[0.001 - 1]".
spw Set the ratio of width reserved for sync code detection. Default is 0.27. Allowed range is "[0.01 -
0.7]".
mhd Set the max peaks height difference for sync code detection. Default is 0.1. Allowed range is "[0.0
- 0.5]".
mpd Set max peaks period difference for sync code detection. Default is 0.1. Allowed range is "[0.0 -
0.5]".
msd Set the first two max start code bits differences. Default is 0.02. Allowed range is "[0.0 - 0.5]".
bhd Set the minimum ratio of bits height compared to 3rd start code bit. Default is 0.75. Allowed range
is "[0.01 - 1]".
th_w
Set the white color threshold. Default is 0.35. Allowed range is "[0.1 - 1]".
th_b
Set the black color threshold. Default is 0.15. Allowed range is "[0.0 - 0.5]".
chp Enable checking the parity bit. In the event of a parity error, the filter will output 0x00 for that
character. Default is false.
Examples
• Output a csv with presentation time and the first two lines of identified EIA-608 captioning data.
ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv
readvitc
Read vertical interval timecode (VITC) information from the top lines of a video frame.
The filter adds frame metadata key "lavfi.readvitc.tc_str" with the timecode value, if a valid timecode
has been detected. Further metadata key "lavfi.readvitc.found" is set to 0/1 depending on whether
timecode data has been found or not.
This filter accepts the following options:
scan_max
Set the maximum number of lines to scan for VITC data. If the value is set to "-1" the full video
frame is scanned. Default is 45.
thr_b
Set the luma threshold for black. Accepts float numbers in the range [0.0,1.0], default value is 0.2.
The value must be equal or less than "thr_w".
thr_w
Set the luma threshold for white. Accepts float numbers in the range [0.0,1.0], default value is 0.6.
The value must be equal or greater than "thr_b".
Examples
• Detect and draw VITC data onto the video frame; if no valid VITC is detected, draw "--:--:--:--" as a
placeholder:
ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'
remap
Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.
Destination pixel at position (X, Y) will be picked from source (x, y) position where x = Xmap(X, Y) and
y = Ymap(X, Y). If mapping values are out of range, zero value for pixel will be used for destination
pixel.
Xmap and Ymap input video streams must be of same dimensions. Output video stream will have Xmap/Ymap
video stream dimensions. Xmap and Ymap input video streams are 16bit depth, single channel.
removegrain
The removegrain filter is a spatial denoiser for progressive video.
m0 Set mode for the first plane.
m1 Set mode for the second plane.
m2 Set mode for the third plane.
m3 Set mode for the fourth plane.
Range of mode is from 0 to 24. Description of each mode follows:
0 Leave input plane unchanged. Default.
1 Clips the pixel with the minimum and maximum of the 8 neighbour pixels.
2 Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.
3 Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.
4 Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels. This is equivalent to
a median filter.
5 Line-sensitive clipping giving the minimal change.
6 Line-sensitive clipping, intermediate.
7 Line-sensitive clipping, intermediate.
8 Line-sensitive clipping, intermediate.
9 Line-sensitive clipping on a line where the neighbours pixels are the closest.
10 Replaces the target pixel with the closest neighbour.
11 [1 2 1] horizontal and vertical kernel blur.
12 Same as mode 11.
13 Bob mode, interpolates top field from the line where the neighbours pixels are the closest.
14 Bob mode, interpolates bottom field from the line where the neighbours pixels are the closest.
15 Bob mode, interpolates top field. Same as 13 but with a more complicated interpolation formula.
16 Bob mode, interpolates bottom field. Same as 14 but with a more complicated interpolation formula.
17 Clips the pixel with the minimum and maximum of respectively the maximum and minimum of each pair of
opposite neighbour pixels.
18 Line-sensitive clipping using opposite neighbours whose greatest distance from the current pixel is
minimal.
19 Replaces the pixel with the average of its 8 neighbours.
20 Averages the 9 pixels ([1 1 1] horizontal and vertical blur).
21 Clips pixels using the averages of opposite neighbour.
22 Same as mode 21 but simpler and faster.
23 Small edge and halo removal, but reputed useless.
24 Similar as 23.
removelogo
Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by
filling in the pixels that comprise the logo with neighboring pixels.
The filter accepts the following options:
filename, f
Set the filter bitmap file, which can be any image format supported by libavformat. The width and
height of the image file must match those of the video stream being processed.
Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero
pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest,
you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black
frame with the logo visible, and then using a threshold filter followed by the erode filter once or
twice.
If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the
filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much,
but it will increase the amount of blurring needed to cover over the image and will destroy more
information than necessary, and extra pixels will slow things down on a large logo.
repeatfields
This filter uses the repeat_field flag from the Video ES headers and hard repeats fields based on its
value.
reverse
Reverse a video clip.
Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.
Examples
• Take the first 5 seconds of a clip, and reverse it.
trim=end=5,reverse
roberts
Apply roberts cross operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
rotate
Rotate video by an arbitrary angle expressed in radians.
The filter accepts the following options:
A description of the optional parameters follows.
angle, a
Set an expression for the angle by which to rotate the input video clockwise, expressed as a number
of radians. A negative value will result in a counter-clockwise rotation. By default it is set to
"0".
This expression is evaluated for each frame.
out_w, ow
Set the output width expression, default value is "iw". This expression is evaluated just once
during configuration.
out_h, oh
Set the output height expression, default value is "ih". This expression is evaluated just once
during configuration.
bilinear
Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is 1.
fillcolor, c
Set the color used to fill the output area not covered by the rotated image. For the general syntax
of this option, check the "Color" section in the ffmpeg-utils manual. If the special value "none" is
selected then no background is printed (useful for example if the background is never shown).
Default value is "black".
The expressions for the angle and the output size can contain the following constants and functions:
n sequential number of the input frame, starting from 0. It is always NAN before the first frame is
filtered.
t time in seconds of the input frame, it is set to 0 when the filter is configured. It is always NAN
before the first frame is filtered.
hsub
vsub
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2
and vsub is 1.
in_w, iw
in_h, ih
the input video width and height
out_w, ow
out_h, oh
the output width and height, that is the size of the padded area as specified by the width and height
expressions
rotw(a)
roth(a)
the minimal width/height required for completely containing the input video rotated by a radians.
These are only available when computing the out_w and out_h expressions.
Examples
• Rotate the input by PI/6 radians clockwise:
rotate=PI/6
• Rotate the input by PI/6 radians counter-clockwise:
rotate=-PI/6
• Rotate the input by 45 degrees clockwise:
rotate=45*PI/180
• Apply a constant rotation with period T, starting from an angle of PI/3:
rotate=PI/3+2*PI*t/T
• Make the input video rotation oscillating with a period of T seconds and an amplitude of A radians:
rotate=A*sin(2*PI/T*t)
• Rotate the video, output size is chosen so that the whole rotating input video is always completely
contained in the output:
rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'
• Rotate the video, reduce the output size so that no background is ever shown:
rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none
Commands
The filter supports the following commands:
a, angle
Set the angle expression. The command accepts the same syntax of the corresponding option.
If the specified expression is not valid, it is kept at its current value.
sab
Apply Shape Adaptive Blur.
The filter accepts the following options:
luma_radius, lr
Set luma blur filter strength, must be a value in range 0.1-4.0, default value is 1.0. A greater
value will result in a more blurred image, and in slower processing.
luma_pre_filter_radius, lpfr
Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is 1.0.
luma_strength, ls
Set luma maximum difference between pixels to still be considered, must be a value in the 0.1-100.0
range, default value is 1.0.
chroma_radius, cr
Set chroma blur filter strength, must be a value in range -0.9-4.0. A greater value will result in a
more blurred image, and in slower processing.
chroma_pre_filter_radius, cpfr
Set chroma pre-filter radius, must be a value in the -0.9-2.0 range.
chroma_strength, cs
Set chroma maximum difference between pixels to still be considered, must be a value in the
-0.9-100.0 range.
Each chroma option value, if not explicitly specified, is set to the corresponding luma option value.
scale
Scale (resize) the input video, using the libswscale library.
The scale filter forces the output display aspect ratio to be the same of the input, by changing the
output sample aspect ratio.
If the input image format is different from the format requested by the next filter, the scale filter
will convert the input to the requested format.
Options
The filter accepts the following options, or any of the options supported by the libswscale scaler.
See the ffmpeg-scaler manual for the complete list of scaler options.
width, w
height, h
Set the output video dimension expression. Default value is the input dimension.
If the width or w value is 0, the input width is used for the output. If the height or h value is 0,
the input height is used for the output.
If one and only one of the values is -n with n >= 1, the scale filter will use a value that maintains
the aspect ratio of the input image, calculated from the other specified dimension. After that it
will, however, make sure that the calculated dimension is divisible by n and adjust the value if
necessary.
If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as
previously detailed.
See below for the list of accepted constants for use in the dimension expression.
eval
Specify when to evaluate width and height expression. It accepts the following values:
init
Only evaluate expressions once during the filter initialization or when a command is processed.
frame
Evaluate expressions for each incoming frame.
Default value is init.
interl
Set the interlacing mode. It accepts the following values:
1 Force interlaced aware scaling.
0 Do not apply interlaced scaling.
-1 Select interlaced aware scaling depending on whether the source frames are flagged as interlaced
or not.
Default value is 0.
flags
Set libswscale scaling flags. See the ffmpeg-scaler manual for the complete list of values. If not
explicitly specified the filter applies the default flags.
param0, param1
Set libswscale input parameters for scaling algorithms that need them. See the ffmpeg-scaler manual
for the complete documentation. If not explicitly specified the filter applies empty parameters.
size, s
Set the video size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual.
in_color_matrix
out_color_matrix
Set in/output YCbCr color space type.
This allows the autodetected value to be overridden as well as allows forcing a specific value used
for the output and encoder.
If not specified, the color space type depends on the pixel format.
Possible values:
auto
Choose automatically.
bt709
Format conforming to International Telecommunication Union (ITU) Recommendation BT.709.
fcc Set color space conforming to the United States Federal Communications Commission (FCC) Code of
Federal Regulations (CFR) Title 47 (2003) 73.682 (a).
bt601
Set color space conforming to:
• ITU Radiocommunication Sector (ITU-R) Recommendation BT.601
• ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G
• Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004
smpte240m
Set color space conforming to SMPTE ST 240:1999.
in_range
out_range
Set in/output YCbCr sample range.
This allows the autodetected value to be overridden as well as allows forcing a specific value used
for the output and encoder. If not specified, the range depends on the pixel format. Possible values:
auto
Choose automatically.
jpeg/full/pc
Set full range (0-255 in case of 8-bit luma).
mpeg/tv
Set "MPEG" range (16-235 in case of 8-bit luma).
force_original_aspect_ratio
Enable decreasing or increasing output video width or height if necessary to keep the original aspect
ratio. Possible values:
disable
Scale the video as specified and disable this feature.
decrease
The output video dimensions will automatically be decreased if needed.
increase
The output video dimensions will automatically be increased if needed.
One useful instance of this option is that when you know a specific device's maximum allowed
resolution, you can use this to limit the output video to that, while retaining the aspect ratio. For
example, device A allows 1280x720 playback, and your video is 1920x800. Using this option (set it to
decrease) and specifying 1280x720 to the command line makes the output 1280x533.
Please note that this is a different thing than specifying -1 for w or h, you still need to specify
the output resolution for this option to work.
The values of the w and h options are expressions containing the following constants:
in_w
in_h
The input width and height
iw
ih These are the same as in_w and in_h.
out_w
out_h
The output (scaled) width and height
ow
oh These are the same as out_w and out_h
a The same as iw / ih
sar input sample aspect ratio
dar The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p"
hsub is 2 and vsub is 1.
ohsub
ovsub
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p"
hsub is 2 and vsub is 1.
Examples
• Scale the input video to a size of 200x100
scale=w=200:h=100
This is equivalent to:
scale=200:100
or:
scale=200x100
• Specify a size abbreviation for the output size:
scale=qcif
which can also be written as:
scale=size=qcif
• Scale the input to 2x:
scale=w=2*iw:h=2*ih
• The above is the same as:
scale=2*in_w:2*in_h
• Scale the input to 2x with forced interlaced scaling:
scale=2*iw:2*ih:interl=1
• Scale the input to half size:
scale=w=iw/2:h=ih/2
• Increase the width, and set the height to the same size:
scale=3/2*iw:ow
• Seek Greek harmony:
scale=iw:1/PHI*iw
scale=ih*PHI:ih
• Increase the height, and set the width to 3/2 of the height:
scale=w=3/2*oh:h=3/5*ih
• Increase the size, making the size a multiple of the chroma subsample values:
scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"
• Increase the width to a maximum of 500 pixels, keeping the same aspect ratio as the input:
scale=w='min(500\, iw*3/2):h=-1'
Commands
This filter supports the following commands:
width, w
height, h
Set the output video dimension expression. The command accepts the same syntax of the corresponding
option.
If the specified expression is not valid, it is kept at its current value.
scale_npp
Use the NVIDIA Performance Primitives (libnpp) to perform scaling and/or pixel format conversion on CUDA
video frames. Setting the output width and height works in the same way as for the scale filter.
The following additional options are accepted:
format
The pixel format of the output CUDA frames. If set to the string "same" (the default), the input
format will be kept. Note that automatic format negotiation and conversion is not yet supported for
hardware frames
interp_algo
The interpolation algorithm used for resizing. One of the following:
nn Nearest neighbour.
linear
cubic
cubic2p_bspline
2-parameter cubic (B=1, C=0)
cubic2p_catmullrom
2-parameter cubic (B=0, C=1/2)
cubic2p_b05c03
2-parameter cubic (B=1/2, C=3/10)
super
Supersampling
lanczos
scale2ref
Scale (resize) the input video, based on a reference video.
See the scale filter for available options, scale2ref supports the same but uses the reference video
instead of the main input as basis. scale2ref also supports the following additional constants for the w
and h options:
main_w
main_h
The main input video's width and height
main_a
The same as main_w / main_h
main_sar
The main input video's sample aspect ratio
main_dar, mdar
The main input video's display aspect ratio. Calculated from "(main_w / main_h) * main_sar".
main_hsub
main_vsub
The main input video's horizontal and vertical chroma subsample values. For example for the pixel
format "yuv422p" hsub is 2 and vsub is 1.
Examples
• Scale a subtitle stream (b) to match the main video (a) in size before overlaying
'scale2ref[b][a];[a][b]overlay'
selectivecolor
Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of colors (such as "reds", "yellows",
"greens", "cyans", ...). The adjustment range is defined by the "purity" of the color (that is, how
saturated it already is).
This filter is similar to the Adobe Photoshop Selective Color tool.
The filter accepts the following options:
correction_method
Select color correction method.
Available values are:
absolute
Specified adjustments are applied "as-is" (added/subtracted to original pixel component value).
relative
Specified adjustments are relative to the original component value.
Default is "absolute".
reds
Adjustments for red pixels (pixels where the red component is the maximum)
yellows
Adjustments for yellow pixels (pixels where the blue component is the minimum)
greens
Adjustments for green pixels (pixels where the green component is the maximum)
cyans
Adjustments for cyan pixels (pixels where the red component is the minimum)
blues
Adjustments for blue pixels (pixels where the blue component is the maximum)
magentas
Adjustments for magenta pixels (pixels where the green component is the minimum)
whites
Adjustments for white pixels (pixels where all components are greater than 128)
neutrals
Adjustments for all pixels except pure black and pure white
blacks
Adjustments for black pixels (pixels where all components are lesser than 128)
psfile
Specify a Photoshop selective color file (".asv") to import the settings from.
All the adjustment settings (reds, yellows, ...) accept up to 4 space separated floating point adjustment
values in the [-1,1] range, respectively to adjust the amount of cyan, magenta, yellow and black for the
pixels of its range.
Examples
• Increase cyan by 50% and reduce yellow by 33% in every green areas, and increase magenta by 27% in
blue areas:
selectivecolor=greens=.5 0 -.33 0:blues=0 .27
• Use a Photoshop selective color preset:
selectivecolor=psfile=MySelectiveColorPresets/Misty.asv
separatefields
The "separatefields" takes a frame-based video input and splits each frame into its components fields,
producing a new half height clip with twice the frame rate and twice the frame count.
This filter use field-dominance information in frame to decide which of each pair of fields to place
first in the output. If it gets it wrong use setfield filter before "separatefields" filter.
setdar, setsar
The "setdar" filter sets the Display Aspect Ratio for the filter output video.
This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following
equation:
<DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>
Keep in mind that the "setdar" filter does not modify the pixel dimensions of the video frame. Also, the
display aspect ratio set by this filter may be changed by later filters in the filterchain, e.g. in case
of scaling or if another "setdar" or a "setsar" filter is applied.
The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the filter output video.
Note that as a consequence of the application of this filter, the output display aspect ratio will change
according to the equation above.
Keep in mind that the sample aspect ratio set by the "setsar" filter may be changed by later filters in
the filterchain, e.g. if another "setsar" or a "setdar" filter is applied.
It accepts the following parameters:
r, ratio, dar ("setdar" only), sar ("setsar" only)
Set the aspect ratio used by the filter.
The parameter can be a floating point number string, an expression, or a string of the form num:den,
where num and den are the numerator and denominator of the aspect ratio. If the parameter is not
specified, it is assumed the value "0". In case the form "num:den" is used, the ":" character should
be escaped.
max Set the maximum integer value to use for expressing numerator and denominator when reducing the
expressed aspect ratio to a rational. Default value is 100.
The parameter sar is an expression containing the following constants:
E, PI, PHI
These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and
phi (the golden ratio).
w, h
The input width and height.
a These are the same as w / h.
sar The input sample aspect ratio.
dar The input display aspect ratio. It is the same as (w / h) * sar.
hsub, vsub
Horizontal and vertical chroma subsample values. For example, for the pixel format "yuv422p" hsub is
2 and vsub is 1.
Examples
• To change the display aspect ratio to 16:9, specify one of the following:
setdar=dar=1.77777
setdar=dar=16/9
• To change the sample aspect ratio to 10:11, specify:
setsar=sar=10/11
• To set a display aspect ratio of 16:9, and specify a maximum integer value of 1000 in the aspect
ratio reduction, use the command:
setdar=ratio=16/9:max=1000
setfield
Force field for the output video frame.
The "setfield" filter marks the interlace type field for the output frames. It does not change the input
frame, but only sets the corresponding property, which affects how the frame is treated by following
filters (e.g. "fieldorder" or "yadif").
The filter accepts the following options:
mode
Available values are:
auto
Keep the same field property.
bff Mark the frame as bottom-field-first.
tff Mark the frame as top-field-first.
prog
Mark the frame as progressive.
showinfo
Show a line containing various information for each input video frame. The input video is not modified.
The shown line contains a sequence of key/value pairs of the form key:value.
The following values are shown in the output:
n The (sequential) number of the input frame, starting from 0.
pts The Presentation TimeStamp of the input frame, expressed as a number of time base units. The time
base unit depends on the filter input pad.
pts_time
The Presentation TimeStamp of the input frame, expressed as a number of seconds.
pos The position of the frame in the input stream, or -1 if this information is unavailable and/or
meaningless (for example in case of synthetic video).
fmt The pixel format name.
sar The sample aspect ratio of the input frame, expressed in the form num/den.
s The size of the input frame. For the syntax of this option, check the "Video size" section in the
ffmpeg-utils manual.
i The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field
first).
iskey
This is 1 if the frame is a key frame, 0 otherwise.
type
The picture type of the input frame ("I" for an I-frame, "P" for a P-frame, "B" for a B-frame, or "?"
for an unknown type). Also refer to the documentation of the "AVPictureType" enum and of the
"av_get_picture_type_char" function defined in libavutil/avutil.h.
checksum
The Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame.
plane_checksum
The Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the
form "[c0 c1 c2 c3]".
showpalette
Displays the 256 colors palette of each frame. This filter is only relevant for pal8 pixel format frames.
It accepts the following option:
s Set the size of the box used to represent one palette color entry. Default is 30 (for a "30x30" pixel
box).
shuffleframes
Reorder and/or duplicate and/or drop video frames.
It accepts the following parameters:
mapping
Set the destination indexes of input frames. This is space or '|' separated list of indexes that
maps input frames to output frames. Number of indexes also sets maximal value that each index may
have. '-1' index have special meaning and that is to drop frame.
The first frame has the index 0. The default is to keep the input unchanged.
Examples
• Swap second and third frame of every three frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT
• Swap 10th and 1st frame of every ten frames of the input:
ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT
shuffleplanes
Reorder and/or duplicate video planes.
It accepts the following parameters:
map0
The index of the input plane to be used as the first output plane.
map1
The index of the input plane to be used as the second output plane.
map2
The index of the input plane to be used as the third output plane.
map3
The index of the input plane to be used as the fourth output plane.
The first plane has the index 0. The default is to keep the input unchanged.
Examples
• Swap the second and third planes of the input:
ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT
signalstats
Evaluate various visual metrics that assist in determining issues associated with the digitization of
analog video media.
By default the filter will log these metadata values:
YMIN
Display the minimal Y value contained within the input frame. Expressed in range of [0-255].
YLOW
Display the Y value at the 10% percentile within the input frame. Expressed in range of [0-255].
YAVG
Display the average Y value within the input frame. Expressed in range of [0-255].
YHIGH
Display the Y value at the 90% percentile within the input frame. Expressed in range of [0-255].
YMAX
Display the maximum Y value contained within the input frame. Expressed in range of [0-255].
UMIN
Display the minimal U value contained within the input frame. Expressed in range of [0-255].
ULOW
Display the U value at the 10% percentile within the input frame. Expressed in range of [0-255].
UAVG
Display the average U value within the input frame. Expressed in range of [0-255].
UHIGH
Display the U value at the 90% percentile within the input frame. Expressed in range of [0-255].
UMAX
Display the maximum U value contained within the input frame. Expressed in range of [0-255].
VMIN
Display the minimal V value contained within the input frame. Expressed in range of [0-255].
VLOW
Display the V value at the 10% percentile within the input frame. Expressed in range of [0-255].
VAVG
Display the average V value within the input frame. Expressed in range of [0-255].
VHIGH
Display the V value at the 90% percentile within the input frame. Expressed in range of [0-255].
VMAX
Display the maximum V value contained within the input frame. Expressed in range of [0-255].
SATMIN
Display the minimal saturation value contained within the input frame. Expressed in range of
[0-~181.02].
SATLOW
Display the saturation value at the 10% percentile within the input frame. Expressed in range of
[0-~181.02].
SATAVG
Display the average saturation value within the input frame. Expressed in range of [0-~181.02].
SATHIGH
Display the saturation value at the 90% percentile within the input frame. Expressed in range of
[0-~181.02].
SATMAX
Display the maximum saturation value contained within the input frame. Expressed in range of
[0-~181.02].
HUEMED
Display the median value for hue within the input frame. Expressed in range of [0-360].
HUEAVG
Display the average value for hue within the input frame. Expressed in range of [0-360].
YDIF
Display the average of sample value difference between all values of the Y plane in the current frame
and corresponding values of the previous input frame. Expressed in range of [0-255].
UDIF
Display the average of sample value difference between all values of the U plane in the current frame
and corresponding values of the previous input frame. Expressed in range of [0-255].
VDIF
Display the average of sample value difference between all values of the V plane in the current frame
and corresponding values of the previous input frame. Expressed in range of [0-255].
YBITDEPTH
Display bit depth of Y plane in current frame. Expressed in range of [0-16].
UBITDEPTH
Display bit depth of U plane in current frame. Expressed in range of [0-16].
VBITDEPTH
Display bit depth of V plane in current frame. Expressed in range of [0-16].
The filter accepts the following options:
stat
out stat specify an additional form of image analysis. out output video with the specified type of pixel
highlighted.
Both options accept the following values:
tout
Identify temporal outliers pixels. A temporal outlier is a pixel unlike the neighboring pixels of
the same field. Examples of temporal outliers include the results of video dropouts, head clogs,
or tape tracking issues.
vrep
Identify vertical line repetition. Vertical line repetition includes similar rows of pixels
within a frame. In born-digital video vertical line repetition is common, but this pattern is
uncommon in video digitized from an analog source. When it occurs in video that results from the
digitization of an analog source it can indicate concealment from a dropout compensator.
brng
Identify pixels that fall outside of legal broadcast range.
color, c
Set the highlight color for the out option. The default color is yellow.
Examples
• Output data of various video metrics:
ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames
• Output specific data about the minimum and maximum values of the Y plane per frame:
ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN
• Playback video while highlighting pixels that are outside of broadcast range in red.
ffplay example.mov -vf signalstats="out=brng:color=red"
• Playback video with signalstats metadata drawn over the frame.
ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt
The contents of signalstat_drawtext.txt used in the command are:
time %{pts:hms}
Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
saturation maximum: %{metadata:lavfi.signalstats.SATMAX}
signature
Calculates the MPEG-7 Video Signature. The filter can handle more than one input. In this case the
matching between the inputs can be calculated additionally. The filter always passes through the first
input. The signature of each stream can be written into a file.
It accepts the following options:
detectmode
Enable or disable the matching process.
Available values are:
off Disable the calculation of a matching (default).
full
Calculate the matching for the whole video and output whether the whole video matches or only
parts.
fast
Calculate only until a matching is found or the video ends. Should be faster in some cases.
nb_inputs
Set the number of inputs. The option value must be a non negative integer. Default value is 1.
filename
Set the path to which the output is written. If there is more than one input, the path must be a
prototype, i.e. must contain %d or %0nd (where n is a positive integer), that will be replaced with
the input number. If no filename is specified, no output will be written. This is the default.
format
Choose the output format.
Available values are:
binary
Use the specified binary representation (default).
xml Use the specified xml representation.
th_d
Set threshold to detect one word as similar. The option value must be an integer greater than zero.
The default value is 9000.
th_dc
Set threshold to detect all words as similar. The option value must be an integer greater than zero.
The default value is 60000.
th_xh
Set threshold to detect frames as similar. The option value must be an integer greater than zero. The
default value is 116.
th_di
Set the minimum length of a sequence in frames to recognize it as matching sequence. The option value
must be a non negative integer value. The default value is 0.
th_it
Set the minimum relation, that matching frames to all frames must have. The option value must be a
double value between 0 and 1. The default value is 0.5.
Examples
• To calculate the signature of an input video and store it in signature.bin:
ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -
• To detect whether two videos match and store the signatures in XML format in signature0.xml and
signature1.xml:
ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -
smartblur
Blur the input video without impacting the outlines.
It accepts the following options:
luma_radius, lr
Set the luma radius. The option value must be a float number in the range [0.1,5.0] that specifies
the variance of the gaussian filter used to blur the image (slower if larger). Default value is 1.0.
luma_strength, ls
Set the luma strength. The option value must be a float number in the range [-1.0,1.0] that
configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included
in [-1.0,0.0] will sharpen the image. Default value is 1.0.
luma_threshold, lt
Set the luma threshold used as a coefficient to determine whether a pixel should be blurred or not.
The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a
value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges.
Default value is 0.
chroma_radius, cr
Set the chroma radius. The option value must be a float number in the range [0.1,5.0] that specifies
the variance of the gaussian filter used to blur the image (slower if larger). Default value is
luma_radius.
chroma_strength, cs
Set the chroma strength. The option value must be a float number in the range [-1.0,1.0] that
configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included
in [-1.0,0.0] will sharpen the image. Default value is luma_strength.
chroma_threshold, ct
Set the chroma threshold used as a coefficient to determine whether a pixel should be blurred or not.
The option value must be an integer in the range [-30,30]. A value of 0 will filter all the image, a
value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges.
Default value is luma_threshold.
If a chroma option is not explicitly set, the corresponding luma value is set.
ssim
Obtain the SSIM (Structural SImilarity Metric) between two input videos.
This filter takes in input two input videos, the first input is considered the "main" source and is
passed unchanged to the output. The second input is used as a "reference" video for computing the SSIM.
Both video inputs must have the same resolution and pixel format for this filter to work correctly. Also
it assumes that both inputs have the same number of frames, which are compared one by one.
The filter stores the calculated SSIM of each frame.
The description of the accepted parameters follows.
stats_file, f
If specified the filter will use the named file to save the SSIM of each individual frame. When
filename equals "-" the data is sent to standard output.
The file printed if stats_file is selected, contains a sequence of key/value pairs of the form key:value
for each compared couple of frames.
A description of each shown parameter follows:
n sequential number of the input frame, starting from 1
Y, U, V, R, G, B
SSIM of the compared frames for the component specified by the suffix.
All SSIM of the compared frames for the whole frame.
dB Same as above but in dB representation.
This filter also supports the framesync options.
For example:
movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
[main][ref] ssim="stats_file=stats.log" [out]
On this example the input file being processed is compared with the reference file ref_movie.mpg. The
SSIM of each individual frame is stored in stats.log.
Another example with both psnr and ssim at same time:
ffmpeg -i main.mpg -i ref.mpg -lavfi "ssim;[0:v][1:v]psnr" -f null -
stereo3d
Convert between different stereoscopic image formats.
The filters accept the following options:
in Set stereoscopic image format of input.
Available values for input image formats are:
sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye left, left eye right)
abl above-below (left eye above, right eye below)
abr above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right eye below)
ab2r
above-below with half height resolution (right eye above, left eye below)
al alternating frames (left eye first, right eye second)
ar alternating frames (right eye first, left eye second)
irl interleaved rows (left eye has top row, right eye starts on next row)
irr interleaved rows (right eye has top row, left eye starts on next row)
icl interleaved columns, left eye first
icr interleaved columns, right eye first
Default value is sbsl.
out Set stereoscopic image format of output.
sbsl
side by side parallel (left eye left, right eye right)
sbsr
side by side crosseye (right eye left, left eye right)
sbs2l
side by side parallel with half width resolution (left eye left, right eye right)
sbs2r
side by side crosseye with half width resolution (right eye left, left eye right)
abl above-below (left eye above, right eye below)
abr above-below (right eye above, left eye below)
ab2l
above-below with half height resolution (left eye above, right eye below)
ab2r
above-below with half height resolution (right eye above, left eye below)
al alternating frames (left eye first, right eye second)
ar alternating frames (right eye first, left eye second)
irl interleaved rows (left eye has top row, right eye starts on next row)
irr interleaved rows (right eye has top row, left eye starts on next row)
arbg
anaglyph red/blue gray (red filter on left eye, blue filter on right eye)
argg
anaglyph red/green gray (red filter on left eye, green filter on right eye)
arcg
anaglyph red/cyan gray (red filter on left eye, cyan filter on right eye)
arch
anaglyph red/cyan half colored (red filter on left eye, cyan filter on right eye)
arcc
anaglyph red/cyan color (red filter on left eye, cyan filter on right eye)
arcd
anaglyph red/cyan color optimized with the least squares projection of dubois (red filter on left
eye, cyan filter on right eye)
agmg
anaglyph green/magenta gray (green filter on left eye, magenta filter on right eye)
agmh
anaglyph green/magenta half colored (green filter on left eye, magenta filter on right eye)
agmc
anaglyph green/magenta colored (green filter on left eye, magenta filter on right eye)
agmd
anaglyph green/magenta color optimized with the least squares projection of dubois (green filter
on left eye, magenta filter on right eye)
aybg
anaglyph yellow/blue gray (yellow filter on left eye, blue filter on right eye)
aybh
anaglyph yellow/blue half colored (yellow filter on left eye, blue filter on right eye)
aybc
anaglyph yellow/blue colored (yellow filter on left eye, blue filter on right eye)
aybd
anaglyph yellow/blue color optimized with the least squares projection of dubois (yellow filter
on left eye, blue filter on right eye)
ml mono output (left eye only)
mr mono output (right eye only)
chl checkerboard, left eye first
chr checkerboard, right eye first
icl interleaved columns, left eye first
icr interleaved columns, right eye first
hdmi
HDMI frame pack
Default value is arcd.
Examples
• Convert input video from side by side parallel to anaglyph yellow/blue dubois:
stereo3d=sbsl:aybd
• Convert input video from above below (left eye above, right eye below) to side by side crosseye.
stereo3d=abl:sbsr
streamselect, astreamselect
Select video or audio streams.
The filter accepts the following options:
inputs
Set number of inputs. Default is 2.
map Set input indexes to remap to outputs.
Commands
The "streamselect" and "astreamselect" filter supports the following commands:
map Set input indexes to remap to outputs.
Examples
• Select first 5 seconds 1st stream and rest of time 2nd stream:
sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0
• Same as above, but for audio:
asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0
sobel
Apply sobel operator to input video stream.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
scale
Set value which will be multiplied with filtered result.
delta
Set value which will be added to filtered result.
spp
Apply a simple postprocessing filter that compresses and decompresses the image at several (or - in the
case of quality level 6 - all) shifts and average the results.
The filter accepts the following options:
quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the
range 0-6. If set to 0, the filter will have no effect. A value of 6 means the higher quality. For
each increment of that value the speed drops by a factor of approximately 2. Default value is 3.
qp Force a constant quantization parameter. If not set, the filter will use the QP from the video stream
(if available).
mode
Set thresholding mode. Available modes are:
hard
Set hard thresholding (default).
soft
Set soft thresholding (better de-ringing effect, but likely blurrier).
use_bframe_qp
Enable the use of the QP from the B-Frames if set to 1. Using this option may cause flicker since the
B-Frames have often larger QP. Default is 0 (not enabled).
subtitles
Draw subtitles on top of input video using the libass library.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libass". This filter
also requires a build with libavcodec and libavformat to convert the passed subtitles file to ASS
(Advanced Substation Alpha) subtitles format.
The filter accepts the following options:
filename, f
Set the filename of the subtitle file to read. It must be specified.
original_size
Specify the size of the original video, the video for which the ASS file was composed. For the syntax
of this option, check the "Video size" section in the ffmpeg-utils manual. Due to a misdesign in ASS
aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been
changed.
fontsdir
Set a directory path containing fonts that can be used by the filter. These fonts will be used in
addition to whatever the font provider uses.
alpha
Process alpha channel, by default alpha channel is untouched.
charenc
Set subtitles input character encoding. "subtitles" filter only. Only useful if not UTF-8.
stream_index, si
Set subtitles stream index. "subtitles" filter only.
force_style
Override default style or script info parameters of the subtitles. It accepts a string containing ASS
style format "KEY=VALUE" couples separated by ",".
If the first key is not specified, it is assumed that the first value specifies the filename.
For example, to render the file sub.srt on top of the input video, use the command:
subtitles=sub.srt
which is equivalent to:
subtitles=filename=sub.srt
To render the default subtitles stream from file video.mkv, use:
subtitles=video.mkv
To render the second subtitles stream from that file, use:
subtitles=video.mkv:si=1
To make the subtitles stream from sub.srt appear in transparent green "DejaVu Serif", use:
subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'
super2xsai
Scale the input by 2x and smooth using the Super2xSaI (Scale and Interpolate) pixel art scaling
algorithm.
Useful for enlarging pixel art images without reducing sharpness.
swaprect
Swap two rectangular objects in video.
This filter accepts the following options:
w Set object width.
h Set object height.
x1 Set 1st rect x coordinate.
y1 Set 1st rect y coordinate.
x2 Set 2nd rect x coordinate.
y2 Set 2nd rect y coordinate.
All expressions are evaluated once for each frame.
The all options are expressions containing the following constants:
w
h The input width and height.
a same as w / h
sar input sample aspect ratio
dar input display aspect ratio, it is the same as (w / h) * sar
n The number of the input frame, starting from 0.
t The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.
pos the position in the file of the input frame, NAN if unknown
swapuv
Swap U & V plane.
telecine
Apply telecine process to the video.
This filter accepts the following options:
first_field
top, t
top field first
bottom, b
bottom field first The default value is "top".
pattern
A string of numbers representing the pulldown pattern you wish to apply. The default value is 23.
Some typical patterns:
NTSC output (30i):
27.5p: 32222
24p: 23 (classic)
24p: 2332 (preferred)
20p: 33
18p: 334
16p: 3444
PAL output (25i):
27.5p: 12222
24p: 222222222223 ("Euro pulldown")
16.67p: 33
16p: 33333334
threshold
Apply threshold effect to video stream.
This filter needs four video streams to perform thresholding. First stream is stream we are filtering.
Second stream is holding threshold values, third stream is holding min values, and last, fourth stream is
holding max values.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
For example if first stream pixel's component value is less then threshold value of pixel component from
2nd threshold stream, third stream value will picked, otherwise fourth stream pixel component value will
be picked.
Using color source filter one can perform various types of thresholding:
Examples
• Binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi
• Inverted binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi
• Truncate binary threshold, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi
• Threshold to zero, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi
• Inverted threshold to zero, using gray color as threshold:
ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi
thumbnail
Select the most representative frame in a given sequence of consecutive frames.
The filter accepts the following options:
n Set the frames batch size to analyze; in a set of n frames, the filter will pick one of them, and
then handle the next batch of n frames until the end. Default is 100.
Since the filter keeps track of the whole frames sequence, a bigger n value will result in a higher
memory usage, so a high value is not recommended.
Examples
• Extract one picture each 50 frames:
thumbnail=50
• Complete example of a thumbnail creation with ffmpeg:
ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png
tile
Tile several successive frames together.
The filter accepts the following options:
layout
Set the grid size (i.e. the number of lines and columns). For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual.
nb_frames
Set the maximum number of frames to render in the given area. It must be less than or equal to wxh.
The default value is 0, meaning all the area will be used.
margin
Set the outer border margin in pixels.
padding
Set the inner border thickness (i.e. the number of pixels between frames). For more advanced padding
options (such as having different values for the edges), refer to the pad video filter.
color
Specify the color of the unused area. For the syntax of this option, check the "Color" section in the
ffmpeg-utils manual. The default value of color is "black".
Examples
• Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a movie:
ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png
The -vsync 0 is necessary to prevent ffmpeg from duplicating each output frame to accommodate the
originally detected frame rate.
• Display 5 pictures in an area of "3x2" frames, with 7 pixels between them, and 2 pixels of initial
margin, using mixed flat and named options:
tile=3x2:nb_frames=5:padding=7:margin=2
tinterlace
Perform various types of temporal field interlacing.
Frames are counted starting from 1, so the first input frame is considered odd.
The filter accepts the following options:
mode
Specify the mode of the interlacing. This option can also be specified as a value alone. See below
for a list of values for this option.
Available values are:
merge, 0
Move odd frames into the upper field, even into the lower field, generating a double height frame
at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444
11111 33333
22222 44444
drop_even, 1
Only output odd frames, even frames are dropped, generating a frame with unchanged height at half
frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333
11111 33333
11111 33333
11111 33333
drop_odd, 2
Only output even frames, odd frames are dropped, generating a frame with unchanged height at half
frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
22222 44444
22222 44444
22222 44444
22222 44444
pad, 3
Expand each frame to full height, but pad alternate lines with black, generating a frame with
double height at the same input frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
11111 ..... 33333 .....
..... 22222 ..... 44444
interleave_top, 4
Interleave the upper field from odd frames with the lower field from even frames, generating a
frame with unchanged height at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
Output:
11111 33333
22222 44444
11111 33333
22222 44444
interleave_bottom, 5
Interleave the lower field from odd frames with the upper field from even frames, generating a
frame with unchanged height at half frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
11111 22222<- 33333 44444<-
11111<- 22222 33333<- 44444
Output:
22222 44444
11111 33333
22222 44444
11111 33333
interlacex2, 6
Double frame rate with unchanged height. Frames are inserted each containing the second temporal
field from the previous input frame and the first temporal field from the next input frame. This
mode relies on the top_field_first flag. Useful for interlaced video displays with no field
synchronisation.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444
11111 22222 22222 33333 33333 44444 44444
11111 11111 22222 22222 33333 33333 44444
mergex2, 7
Move odd frames into the upper field, even into the lower field, generating a double height frame
at same frame rate.
------> time
Input:
Frame 1 Frame 2 Frame 3 Frame 4
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
11111 22222 33333 44444
Output:
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
11111 33333 33333 55555
22222 22222 44444 44444
Numeric values are deprecated but are accepted for backward compatibility reasons.
Default mode is "merge".
flags
Specify flags influencing the filter process.
Available value for flags is:
low_pass_filter, vlfp
Enable linear vertical low-pass filtering in the filter. Vertical low-pass filtering is required
when creating an interlaced destination from a progressive source which contains high-frequency
vertical detail. Filtering will reduce interlace 'twitter' and Moire patterning.
complex_filter, cvlfp
Enable complex vertical low-pass filtering. This will slightly less reduce interlace 'twitter'
and Moire patterning but better retain detail and subjective sharpness impression.
Vertical low-pass filtering can only be enabled for mode interleave_top and interleave_bottom.
tonemap
Tone map colors from different dynamic ranges.
This filter expects data in single precision floating point, as it needs to operate on (and can output)
out-of-range values. Another filter, such as zscale, is needed to convert the resulting frame to a usable
format.
The tonemapping algorithms implemented only work on linear light, so input data should be linearized
beforehand (and possibly correctly tagged).
ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT
Options
The filter accepts the following options.
tonemap
Set the tone map algorithm to use.
Possible values are:
none
Do not apply any tone map, only desaturate overbright pixels.
clip
Hard-clip any out-of-range values. Use it for perfect color accuracy for in-range values, while
distorting out-of-range values.
linear
Stretch the entire reference gamut to a linear multiple of the display.
gamma
Fit a logarithmic transfer between the tone curves.
reinhard
Preserve overall image brightness with a simple curve, using nonlinear contrast, which results in
flattening details and degrading color accuracy.
hable
Preserve both dark and bright details better than reinhard, at the cost of slightly darkening
everything. Use it when detail preservation is more important than color and brightness accuracy.
mobius
Smoothly map out-of-range values, while retaining contrast and colors for in-range material as
much as possible. Use it when color accuracy is more important than detail preservation.
Default is none.
param
Tune the tone mapping algorithm.
This affects the following algorithms:
none
Ignored.
linear
Specifies the scale factor to use while stretching. Default to 1.0.
gamma
Specifies the exponent of the function. Default to 1.8.
clip
Specify an extra linear coefficient to multiply into the signal before clipping. Default to 1.0.
reinhard
Specify the local contrast coefficient at the display peak. Default to 0.5, which means that in-
gamut values will be about half as bright as when clipping.
hable
Ignored.
mobius
Specify the transition point from linear to mobius transform. Every value below this point is
guaranteed to be mapped 1:1. The higher the value, the more accurate the result will be, at the
cost of losing bright details. Default to 0.3, which due to the steep initial slope still
preserves in-range colors fairly accurately.
desat
Apply desaturation for highlights that exceed this level of brightness. The higher the parameter, the
more color information will be preserved. This setting helps prevent unnaturally blown-out colors for
super-highlights, by (smoothly) turning into white instead. This makes images feel more natural, at
the cost of reducing information about out-of-range colors.
The default of 2.0 is somewhat conservative and will mostly just apply to skies or directly sunlit
surfaces. A setting of 0.0 disables this option.
This option works only if the input frame has a supported color tag.
peak
Override signal/nominal/reference peak with this value. Useful when the embedded peak information in
display metadata is not reliable or when tone mapping from a lower range to a higher range.
transpose
Transpose rows with columns in the input video and optionally flip it.
It accepts the following parameters:
dir Specify the transposition direction.
Can assume the following values:
0, 4, cclock_flip
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:
L.R L.l
. . -> . .
l.r R.r
1, 5, clock
Rotate by 90 degrees clockwise, that is:
L.R l.L
. . -> . .
l.r r.R
2, 6, cclock
Rotate by 90 degrees counterclockwise, that is:
L.R R.r
. . -> . .
l.r L.l
3, 7, clock_flip
Rotate by 90 degrees clockwise and vertically flip, that is:
L.R r.R
. . -> . .
l.r l.L
For values between 4-7, the transposition is only done if the input video geometry is portrait and
not landscape. These values are deprecated, the "passthrough" option should be used instead.
Numerical values are deprecated, and should be dropped in favor of symbolic constants.
passthrough
Do not apply the transposition if the input geometry matches the one specified by the specified
value. It accepts the following values:
none
Always apply transposition.
portrait
Preserve portrait geometry (when height >= width).
landscape
Preserve landscape geometry (when width >= height).
Default value is "none".
For example to rotate by 90 degrees clockwise and preserve portrait layout:
transpose=dir=1:passthrough=portrait
The command above can also be specified as:
transpose=1:portrait
trim
Trim the input so that the output contains one continuous subpart of the input.
It accepts the following parameters:
start
Specify the time of the start of the kept section, i.e. the frame with the timestamp start will be
the first frame in the output.
end Specify the time of the first frame that will be dropped, i.e. the frame immediately preceding the
one with the timestamp end will be the last frame in the output.
start_pts
This is the same as start, except this option sets the start timestamp in timebase units instead of
seconds.
end_pts
This is the same as end, except this option sets the end timestamp in timebase units instead of
seconds.
duration
The maximum duration of the output in seconds.
start_frame
The number of the first frame that should be passed to the output.
end_frame
The number of the first frame that should be dropped.
start, end, and duration are expressed as time duration specifications; see the Time duration section in
the ffmpeg-utils(1) manual for the accepted syntax.
Note that the first two sets of the start/end options and the duration option look at the frame
timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that
this filter does not modify the timestamps. If you wish for the output timestamps to start at zero,
insert a setpts filter after the trim filter.
If multiple start or end options are set, this filter tries to be greedy and keep all the frames that
match at least one of the specified constraints. To keep only the part that matches all the constraints
at once, chain multiple trim filters.
The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to
keep everything before the specified time.
Examples:
• Drop everything except the second minute of input:
ffmpeg -i INPUT -vf trim=60:120
• Keep only the first second:
ffmpeg -i INPUT -vf trim=duration=1
unpremultiply
Apply alpha unpremultiply effect to input video stream using first plane of second stream as alpha.
Both streams must have same dimensions and same pixel format.
The filter accepts the following option:
planes
Set which planes will be processed, unprocessed planes will be copied. By default value 0xf, all
planes will be processed.
If the format has 1 or 2 components, then luma is bit 0. If the format has 3 or 4 components: for
RGB formats bit 0 is green, bit 1 is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
chroma-U and bit 2 is chroma-V. If present, the alpha channel is always the last bit.
inplace
Do not require 2nd input for processing, instead use alpha plane from input stream.
unsharp
Sharpen or blur the input video.
It accepts the following parameters:
luma_msize_x, lx
Set the luma matrix horizontal size. It must be an odd integer between 3 and 23. The default value is
5.
luma_msize_y, ly
Set the luma matrix vertical size. It must be an odd integer between 3 and 23. The default value is
5.
luma_amount, la
Set the luma effect strength. It must be a floating point number, reasonable values lay between -1.5
and 1.5.
Negative values will blur the input video, while positive values will sharpen it, a value of zero
will disable the effect.
Default value is 1.0.
chroma_msize_x, cx
Set the chroma matrix horizontal size. It must be an odd integer between 3 and 23. The default value
is 5.
chroma_msize_y, cy
Set the chroma matrix vertical size. It must be an odd integer between 3 and 23. The default value is
5.
chroma_amount, ca
Set the chroma effect strength. It must be a floating point number, reasonable values lay between
-1.5 and 1.5.
Negative values will blur the input video, while positive values will sharpen it, a value of zero
will disable the effect.
Default value is 0.0.
opencl
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'.
Examples
• Apply strong luma sharpen effect:
unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
• Apply a strong blur of both luma and chroma parameters:
unsharp=7:7:-2:7:7:-2
uspp
Apply ultra slow/simple postprocessing filter that compresses and decompresses the image at several (or -
in the case of quality level 8 - all) shifts and average the results.
The way this differs from the behavior of spp is that uspp actually encodes & decodes each case with
libavcodec Snow, whereas spp uses a simplified intra only 8x8 DCT similar to MJPEG.
The filter accepts the following options:
quality
Set quality. This option defines the number of levels for averaging. It accepts an integer in the
range 0-8. If set to 0, the filter will have no effect. A value of 8 means the higher quality. For
each increment of that value the speed drops by a factor of approximately 2. Default value is 3.
qp Force a constant quantization parameter. If not set, the filter will use the QP from the video stream
(if available).
vaguedenoiser
Apply a wavelet based denoiser.
It transforms each frame from the video input into the wavelet domain, using Cohen-Daubechies-Feauveau
9/7. Then it applies some filtering to the obtained coefficients. It does an inverse wavelet transform
after. Due to wavelet properties, it should give a nice smoothed result, and reduced noise, without
blurring picture features.
This filter accepts the following options:
threshold
The filtering strength. The higher, the more filtered the video will be. Hard thresholding can use a
higher threshold than soft thresholding before the video looks overfiltered. Default value is 2.
method
The filtering method the filter will use.
It accepts the following values:
hard
All values under the threshold will be zeroed.
soft
All values under the threshold will be zeroed. All values above will be reduced by the threshold.
garrote
Scales or nullifies coefficients - intermediary between (more) soft and (less) hard thresholding.
Default is garrote.
nsteps
Number of times, the wavelet will decompose the picture. Picture can't be decomposed beyond a
particular point (typically, 8 for a 640x480 frame - as 2^9 = 512 > 480). Valid values are integers
between 1 and 32. Default value is 6.
percent
Partial of full denoising (limited coefficients shrinking), from 0 to 100. Default value is 85.
planes
A list of the planes to process. By default all planes are processed.
vectorscope
Display 2 color component values in the two dimensional graph (which is called a vectorscope).
This filter accepts the following options:
mode, m
Set vectorscope mode.
It accepts the following values:
gray
Gray values are displayed on graph, higher brightness means more pixels have same component color
value on location in graph. This is the default mode.
color
Gray values are displayed on graph. Surrounding pixels values which are not present in video
frame are drawn in gradient of 2 color components which are set by option "x" and "y". The 3rd
color component is static.
color2
Actual color components values present in video frame are displayed on graph.
color3
Similar as color2 but higher frequency of same values "x" and "y" on graph increases value of
another color component, which is luminance by default values of "x" and "y".
color4
Actual colors present in video frame are displayed on graph. If two different colors map to same
position on graph then color with higher value of component not present in graph is picked.
color5
Gray values are displayed on graph. Similar to "color" but with 3rd color component picked from
radial gradient.
x Set which color component will be represented on X-axis. Default is 1.
y Set which color component will be represented on Y-axis. Default is 2.
intensity, i
Set intensity, used by modes: gray, color, color3 and color5 for increasing brightness of color
component which represents frequency of (X, Y) location in graph.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, even darkest single pixel will be clearly highlighted.
peak
Hold maximum and minimum values presented in graph over time. This way you can still spot out of
range values without constantly looking at vectorscope.
peak+instant
Peak and instant envelope combined together.
graticule, g
Set what kind of graticule to draw.
none
green
color
opacity, o
Set graticule opacity.
flags, f
Set graticule flags.
white
Draw graticule for white point.
black
Draw graticule for black point.
name
Draw color points short names.
bgopacity, b
Set background opacity.
lthreshold, l
Set low threshold for color component not represented on X or Y axis. Values lower than this value
will be ignored. Default is 0. Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and low threshold value of 0.1 actual threshold is 0.1 *
255 = 25.
hthreshold, h
Set high threshold for color component not represented on X or Y axis. Values higher than this value
will be ignored. Default is 1. Note this value is multiplied with actual max possible value one
pixel component can have. So for 8-bit input and high threshold value of 0.9 actual threshold is 0.9
* 255 = 230.
colorspace, c
Set what kind of colorspace to use when drawing graticule.
auto
601
709
Default is auto.
vidstabdetect
Analyze video stabilization/deshaking. Perform pass 1 of 2, see vidstabtransform for pass 2.
This filter generates a file with relative translation and rotation transform information about
subsequent frames, which is then used by the vidstabtransform filter.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libvidstab".
This filter accepts the following options:
result
Set the path to the file used to write the transforms information. Default value is transforms.trf.
shakiness
Set how shaky the video is and how quick the camera is. It accepts an integer in the range 1-10, a
value of 1 means little shakiness, a value of 10 means strong shakiness. Default value is 5.
accuracy
Set the accuracy of the detection process. It must be a value in the range 1-15. A value of 1 means
low accuracy, a value of 15 means high accuracy. Default value is 15.
stepsize
Set stepsize of the search process. The region around minimum is scanned with 1 pixel resolution.
Default value is 6.
mincontrast
Set minimum contrast. Below this value a local measurement field is discarded. Must be a floating
point value in the range 0-1. Default value is 0.3.
tripod
Set reference frame number for tripod mode.
If enabled, the motion of the frames is compared to a reference frame in the filtered stream,
identified by the specified number. The idea is to compensate all movements in a more-or-less static
scene and keep the camera view absolutely still.
If set to 0, it is disabled. The frames are counted starting from 1.
show
Show fields and transforms in the resulting frames. It accepts an integer in the range 0-2. Default
value is 0, which disables any visualization.
Examples
• Use default values:
vidstabdetect
• Analyze strongly shaky movie and put the results in file mytransforms.trf:
vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"
• Visualize the result of internal transformations in the resulting video:
vidstabdetect=show=1
• Analyze a video with medium shakiness using ffmpeg:
ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi
vidstabtransform
Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass 1.
Read a file with transform information for each frame and apply/compensate them. Together with the
vidstabdetect filter this can be used to deshake videos. See also <http://public.hronopik.de/vid.stab>.
It is important to also use the unsharp filter, see below.
To enable compilation of this filter you need to configure FFmpeg with "--enable-libvidstab".
Options
input
Set path to the file used to read the transforms. Default value is transforms.trf.
smoothing
Set the number of frames (value*2 + 1) used for lowpass filtering the camera movements. Default value
is 10.
For example a number of 10 means that 21 frames are used (10 in the past and 10 in the future) to
smoothen the motion in the video. A larger value leads to a smoother video, but limits the
acceleration of the camera (pan/tilt movements). 0 is a special case where a static camera is
simulated.
optalgo
Set the camera path optimization algorithm.
Accepted values are:
gauss
gaussian kernel low-pass filter on camera motion (default)
avg averaging on transformations
maxshift
Set maximal number of pixels to translate frames. Default value is -1, meaning no limit.
maxangle
Set maximal angle in radians (degree*PI/180) to rotate frames. Default value is -1, meaning no limit.
crop
Specify how to deal with borders that may be visible due to movement compensation.
Available values are:
keep
keep image information from previous frame (default)
black
fill the border black
invert
Invert transforms if set to 1. Default value is 0.
relative
Consider transforms as relative to previous frame if set to 1, absolute if set to 0. Default value is
0.
zoom
Set percentage to zoom. A positive value will result in a zoom-in effect, a negative value in a zoom-
out effect. Default value is 0 (no zoom).
optzoom
Set optimal zooming to avoid borders.
Accepted values are:
0 disabled
1 optimal static zoom value is determined (only very strong movements will lead to visible borders)
(default)
2 optimal adaptive zoom value is determined (no borders will be visible), see zoomspeed
Note that the value given at zoom is added to the one calculated here.
zoomspeed
Set percent to zoom maximally each frame (enabled when optzoom is set to 2). Range is from 0 to 5,
default value is 0.25.
interpol
Specify type of interpolation.
Available values are:
no no interpolation
linear
linear only horizontal
bilinear
linear in both directions (default)
bicubic
cubic in both directions (slow)
tripod
Enable virtual tripod mode if set to 1, which is equivalent to "relative=0:smoothing=0". Default
value is 0.
Use also "tripod" option of vidstabdetect.
debug
Increase log verbosity if set to 1. Also the detected global motions are written to the temporary
file global_motions.trf. Default value is 0.
Examples
• Use ffmpeg for a typical stabilization with default values:
ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg
Note the use of the unsharp filter which is always recommended.
• Zoom in a bit more and load transform data from a given file:
vidstabtransform=zoom=5:input="mytransforms.trf"
• Smoothen the video even more:
vidstabtransform=smoothing=30
vflip
Flip the input video vertically.
For example, to vertically flip a video with ffmpeg:
ffmpeg -i in.avi -vf "vflip" out.avi
vignette
Make or reverse a natural vignetting effect.
The filter accepts the following options:
angle, a
Set lens angle expression as a number of radians.
The value is clipped in the "[0,PI/2]" range.
Default value: "PI/5"
x0
y0 Set center coordinates expressions. Respectively "w/2" and "h/2" by default.
mode
Set forward/backward mode.
Available modes are:
forward
The larger the distance from the central point, the darker the image becomes.
backward
The larger the distance from the central point, the brighter the image becomes. This can be used
to reverse a vignette effect, though there is no automatic detection to extract the lens angle
and other settings (yet). It can also be used to create a burning effect.
Default value is forward.
eval
Set evaluation mode for the expressions (angle, x0, y0).
It accepts the following values:
init
Evaluate expressions only once during the filter initialization.
frame
Evaluate expressions for each incoming frame. This is way slower than the init mode since it
requires all the scalers to be re-computed, but it allows advanced dynamic expressions.
Default value is init.
dither
Set dithering to reduce the circular banding effects. Default is 1 (enabled).
aspect
Set vignette aspect. This setting allows one to adjust the shape of the vignette. Setting this value
to the SAR of the input will make a rectangular vignetting following the dimensions of the video.
Default is "1/1".
Expressions
The alpha, x0 and y0 expressions can contain the following parameters.
w
h input width and height
n the number of input frame, starting from 0
pts the PTS (Presentation TimeStamp) time of the filtered video frame, expressed in TB units, NAN if
undefined
r frame rate of the input video, NAN if the input frame rate is unknown
t the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
tb time base of the input video
Examples
• Apply simple strong vignetting effect:
vignette=PI/4
• Make a flickering vignetting:
vignette='PI/4+random(1)*PI/50':eval=frame
vmafmotion
Obtain the average vmaf motion score of a video. It is one of the component filters of VMAF.
The obtained average motion score is printed through the logging system.
In the below example the input file ref.mpg is being processed and score is computed.
ffmpeg -i ref.mpg -lavfi vmafmotion -f null -
vstack
Stack input videos vertically.
All streams must be of same pixel format and of same width.
Note that this filter is faster than using overlay and pad filter to create same output.
The filter accept the following option:
inputs
Set number of input streams. Default is 2.
shortest
If set to 1, force the output to terminate when the shortest input terminates. Default value is 0.
w3fdif
Deinterlace the input video ("w3fdif" stands for "Weston 3 Field Deinterlacing Filter").
Based on the process described by Martin Weston for BBC R&D, and implemented based on the de-interlace
algorithm written by Jim Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses filter
coefficients calculated by BBC R&D.
There are two sets of filter coefficients, so called "simple": and "complex". Which set of filter
coefficients is used can be set by passing an optional parameter:
filter
Set the interlacing filter coefficients. Accepts one of the following values:
simple
Simple filter coefficient set.
complex
More-complex filter coefficient set.
Default value is complex.
deint
Specify which frames to deinterlace. Accept one of the following values:
all Deinterlace all frames,
interlaced
Only deinterlace frames marked as interlaced.
Default value is all.
waveform
Video waveform monitor.
The waveform monitor plots color component intensity. By default luminance only. Each column of the
waveform corresponds to a column of pixels in the source video.
It accepts the following options:
mode, m
Can be either "row", or "column". Default is "column". In row mode, the graph on the left side
represents color component value 0 and the right side represents value = 255. In column mode, the top
side represents color component value = 0 and bottom side represents value = 255.
intensity, i
Set intensity. Smaller values are useful to find out how many values of the same luminance are
distributed across input rows/columns. Default value is 0.04. Allowed range is [0, 1].
mirror, r
Set mirroring mode. 0 means unmirrored, 1 means mirrored. In mirrored mode, higher values will be
represented on the left side for "row" mode and at the top for "column" mode. Default is 1
(mirrored).
display, d
Set display mode. It accepts the following values:
overlay
Presents information identical to that in the "parade", except that the graphs representing color
components are superimposed directly over one another.
This display mode makes it easier to spot relative differences or similarities in overlapping
areas of the color components that are supposed to be identical, such as neutral whites, grays,
or blacks.
stack
Display separate graph for the color components side by side in "row" mode or one below the other
in "column" mode.
parade
Display separate graph for the color components side by side in "column" mode or one below the
other in "row" mode.
Using this display mode makes it easy to spot color casts in the highlights and shadows of an
image, by comparing the contours of the top and the bottom graphs of each waveform. Since whites,
grays, and blacks are characterized by exactly equal amounts of red, green, and blue, neutral
areas of the picture should display three waveforms of roughly equal width/height. If not, the
correction is easy to perform by making level adjustments the three waveforms.
Default is "stack".
components, c
Set which color components to display. Default is 1, which means only luminance or red color
component if input is in RGB colorspace. If is set for example to 7 it will display all 3 (if)
available color components.
envelope, e
none
No envelope, this is default.
instant
Instant envelope, minimum and maximum values presented in graph will be easily visible even with
small "step" value.
peak
Hold minimum and maximum values presented in graph across time. This way you can still spot out
of range values without constantly looking at waveforms.
peak+instant
Peak and instant envelope combined together.
filter, f
lowpass
No filtering, this is default.
flat
Luma and chroma combined together.
aflat
Similar as above, but shows difference between blue and red chroma.
chroma
Displays only chroma.
color
Displays actual color value on waveform.
acolor
Similar as above, but with luma showing frequency of chroma values.
graticule, g
Set which graticule to display.
none
Do not display graticule.
green
Display green graticule showing legal broadcast ranges.
opacity, o
Set graticule opacity.
flags, fl
Set graticule flags.
numbers
Draw numbers above lines. By default enabled.
dots
Draw dots instead of lines.
scale, s
Set scale used for displaying graticule.
digital
millivolts
ire
Default is digital.
bgopacity, b
Set background opacity.
weave, doubleweave
The "weave" takes a field-based video input and join each two sequential fields into single frame,
producing a new double height clip with half the frame rate and half the frame count.
The "doubleweave" works same as "weave" but without halving frame rate and frame count.
It accepts the following option:
first_field
Set first field. Available values are:
top, t
Set the frame as top-field-first.
bottom, b
Set the frame as bottom-field-first.
Examples
• Interlace video using select and separatefields filter:
separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave
xbr
Apply the xBR high-quality magnification filter which is designed for pixel art. It follows a set of
edge-detection rules, see <http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.
It accepts the following option:
n Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for "4xBR". Default is 3.
yadif
Deinterlace the input video ("yadif" means "yet another deinterlacing filter").
It accepts the following parameters:
mode
The interlacing mode to adopt. It accepts one of the following values:
0, send_frame
Output one frame for each frame.
1, send_field
Output one frame for each field.
2, send_frame_nospatial
Like "send_frame", but it skips the spatial interlacing check.
3, send_field_nospatial
Like "send_field", but it skips the spatial interlacing check.
The default value is "send_frame".
parity
The picture field parity assumed for the input interlaced video. It accepts one of the following
values:
0, tff
Assume the top field is first.
1, bff
Assume the bottom field is first.
-1, auto
Enable automatic detection of field parity.
The default value is "auto". If the interlacing is unknown or the decoder does not export this
information, top field first will be assumed.
deint
Specify which frames to deinterlace. Accept one of the following values:
0, all
Deinterlace all frames.
1, interlaced
Only deinterlace frames marked as interlaced.
The default value is "all".
zoompan
Apply Zoom & Pan effect.
This filter accepts the following options:
zoom, z
Set the zoom expression. Default is 1.
x
y Set the x and y expression. Default is 0.
d Set the duration expression in number of frames. This sets for how many number of frames effect will
last for single input image.
s Set the output image size, default is 'hd720'.
fps Set the output frame rate, default is '25'.
Each expression can contain the following constants:
in_w, iw
Input width.
in_h, ih
Input height.
out_w, ow
Output width.
out_h, oh
Output height.
in Input frame count.
on Output frame count.
x
y Last calculated 'x' and 'y' position from 'x' and 'y' expression for current input frame.
px
py 'x' and 'y' of last output frame of previous input frame or 0 when there was not yet such frame
(first input frame).
zoom
Last calculated zoom from 'z' expression for current input frame.
pzoom
Last calculated zoom of last output frame of previous input frame.
duration
Number of output frames for current input frame. Calculated from 'd' expression for each input frame.
pduration
number of output frames created for previous input frame
a Rational number: input width / input height
sar sample aspect ratio
dar display aspect ratio
Examples
• Zoom-in up to 1.5 and pan at same time to some spot near center of picture:
zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360
• Zoom-in up to 1.5 and pan always at center of picture:
zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
• Same as above but without pausing:
zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'
zscale
Scale (resize) the input video, using the z.lib library: https://github.com/sekrit-twc/zimg.
The zscale filter forces the output display aspect ratio to be the same as the input, by changing the
output sample aspect ratio.
If the input image format is different from the format requested by the next filter, the zscale filter
will convert the input to the requested format.
Options
The filter accepts the following options.
width, w
height, h
Set the output video dimension expression. Default value is the input dimension.
If the width or w value is 0, the input width is used for the output. If the height or h value is 0,
the input height is used for the output.
If one and only one of the values is -n with n >= 1, the zscale filter will use a value that
maintains the aspect ratio of the input image, calculated from the other specified dimension. After
that it will, however, make sure that the calculated dimension is divisible by n and adjust the value
if necessary.
If both values are -n with n >= 1, the behavior will be identical to both values being set to 0 as
previously detailed.
See below for the list of accepted constants for use in the dimension expression.
size, s
Set the video size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual.
dither, d
Set the dither type.
Possible values are:
none
ordered
random
error_diffusion
Default is none.
filter, f
Set the resize filter type.
Possible values are:
point
bilinear
bicubic
spline16
spline36
lanczos
Default is bilinear.
range, r
Set the color range.
Possible values are:
input
limited
full
Default is same as input.
primaries, p
Set the color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transfer, t
Set the transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
smpte2084
iec61966-2-1
arib-std-b67
Default is same as input.
matrix, m
Set the colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
Default is same as input.
rangein, rin
Set the input color range.
Possible values are:
input
limited
full
Default is same as input.
primariesin, pin
Set the input color primaries.
Possible values are:
input
709
unspecified
170m
240m
2020
Default is same as input.
transferin, tin
Set the input transfer characteristics.
Possible values are:
input
709
unspecified
601
linear
2020_10
2020_12
Default is same as input.
matrixin, min
Set the input colorspace matrix.
Possible value are:
input
709
unspecified
470bg
170m
2020_ncl
2020_cl
chromal, c
Set the output chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
chromalin, cin
Set the input chroma location.
Possible values are:
input
left
center
topleft
top
bottomleft
bottom
npl Set the nominal peak luminance.
The values of the w and h options are expressions containing the following constants:
in_w
in_h
The input width and height
iw
ih These are the same as in_w and in_h.
out_w
out_h
The output (scaled) width and height
ow
oh These are the same as out_w and out_h
a The same as iw / ih
sar input sample aspect ratio
dar The input display aspect ratio. Calculated from "(iw / ih) * sar".
hsub
vsub
horizontal and vertical input chroma subsample values. For example for the pixel format "yuv422p"
hsub is 2 and vsub is 1.
ohsub
ovsub
horizontal and vertical output chroma subsample values. For example for the pixel format "yuv422p"
hsub is 2 and vsub is 1.
VIDEO SOURCES
Below is a description of the currently available video sources.
buffer
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in
libavfilter/vsrc_buffer.h.
It accepts the following parameters:
video_size
Specify the size (width and height) of the buffered video frames. For the syntax of this option,
check the "Video size" section in the ffmpeg-utils manual.
width
The input video width.
height
The input video height.
pix_fmt
A string representing the pixel format of the buffered video frames. It may be a number
corresponding to a pixel format, or a pixel format name.
time_base
Specify the timebase assumed by the timestamps of the buffered frames.
frame_rate
Specify the frame rate expected for the video stream.
pixel_aspect, sar
The sample (pixel) aspect ratio of the input video.
sws_param
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.
hw_frames_ctx
When using a hardware pixel format, this should be a reference to an AVHWFramesContext describing
input frames.
For example:
buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1
will instruct the source to accept video frames with size 320x240 and with format "yuv410p", assuming
1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). Since the pixel format with
name "yuv410p" corresponds to the number 6 (check the enum AVPixelFormat definition in
libavutil/pixfmt.h), this example corresponds to:
buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1
Alternatively, the options can be specified as a flat string, but this syntax is deprecated:
width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]
cellauto
Create a pattern generated by an elementary cellular automaton.
The initial state of the cellular automaton can be defined through the filename and pattern options. If
such options are not specified an initial state is created randomly.
At each new frame a new row in the video is filled with the result of the cellular automaton next
generation. The behavior when the whole frame is filled is defined by the scroll option.
This source accepts the following options:
filename, f
Read the initial cellular automaton state, i.e. the starting row, from the specified file. In the
file, each non-whitespace character is considered an alive cell, a newline will terminate the row,
and further characters in the file will be ignored.
pattern, p
Read the initial cellular automaton state, i.e. the starting row, from the specified string.
Each non-whitespace character in the string is considered an alive cell, a newline will terminate the
row, and further characters in the string will be ignored.
rate, r
Set the video rate, that is the number of frames generated per second. Default is 25.
random_fill_ratio, ratio
Set the random fill ratio for the initial cellular automaton row. It is a floating point number value
ranging from 0 to 1, defaults to 1/PHI.
This option is ignored when a file or a pattern is specified.
random_seed, seed
Set the seed for filling randomly the initial row, must be an integer included between 0 and
UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random
seed on a best effort basis.
rule
Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.
size, s
Set the size of the output video. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.
If filename or pattern is specified, the size is set by default to the width of the specified initial
state row, and the height is set to width * PHI.
If size is set, it must contain the width of the specified pattern string, and the specified pattern
will be centered in the larger row.
If a filename or a pattern string is not specified, the size value defaults to "320x518" (used for a
randomly generated initial state).
scroll
If set to 1, scroll the output upward when all the rows in the output have been already filled. If
set to 0, the new generated row will be written over the top row just after the bottom row is filled.
Defaults to 1.
start_full, full
If set to 1, completely fill the output with generated rows before outputting the first frame. This
is the default behavior, for disabling set the value to 0.
stitch
If set to 1, stitch the left and right row edges together. This is the default behavior, for
disabling set the value to 0.
Examples
• Read the initial state from pattern, and specify an output of size 200x400.
cellauto=f=pattern:s=200x400
• Generate a random initial row with a width of 200 cells, with a fill ratio of 2/3:
cellauto=ratio=2/3:s=200x200
• Create a pattern generated by rule 18 starting by a single alive cell centered on an initial row with
width 100:
cellauto=p=@s=100x400:full=0:rule=18
• Specify a more elaborated initial pattern:
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18
coreimagesrc
Video source generated on GPU using Apple's CoreImage API on OSX.
This video source is a specialized version of the coreimage video filter. Use a core image generator at
the beginning of the applied filterchain to generate the content.
The coreimagesrc video source accepts the following options:
list_generators
List all available generators along with all their respective options as well as possible minimum and
maximum values along with the default values.
list_generators=true
size, s
Specify the size of the sourced video. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. The default value is "320x240".
rate, r
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to
be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number
or a valid video frame rate abbreviation. The default value is "25".
sar Set the sample aspect ratio of the sourced video.
duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.
If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.
Additionally, all options of the coreimage video filter are accepted. A complete filterchain can be used
for further processing of the generated input without CPU-HOST transfer. See coreimage documentation and
examples for details.
Examples
• Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage, given as complete and escaped
command-line for Apple's standard bash shell:
ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png
This example is equivalent to the QRCode example of coreimage without the need for a nullsrc video
source.
mandelbrot
Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and
start_y.
This source accepts the following options:
end_pts
Set the terminal pts value. Default value is 400.
end_scale
Set the terminal scale value. Must be a floating point value. Default value is 0.3.
inner
Set the inner coloring mode, that is the algorithm used to draw the Mandelbrot fractal internal
region.
It shall assume one of the following values:
black
Set black mode.
convergence
Show time until convergence.
mincol
Set color based on point closest to the origin of the iterations.
period
Set period mode.
Default value is mincol.
bailout
Set the bailout value. Default value is 10.0.
maxiter
Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.
outer
Set outer coloring mode. It shall assume one of following values:
iteration_count
Set iteration cound mode.
normalized_iteration_count
set normalized iteration count mode.
Default value is normalized_iteration_count.
rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".
size, s
Set frame size. For the syntax of this option, check the "Video size" section in the ffmpeg-utils
manual. Default value is "640x480".
start_scale
Set the initial scale value. Default value is 3.0.
start_x
Set the initial x position. Must be a floating point value between -100 and 100. Default value is
-0.743643887037158704752191506114774.
start_y
Set the initial y position. Must be a floating point value between -100 and 100. Default value is
-0.131825904205311970493132056385139.
mptestsrc
Generate various test patterns, as generated by the MPlayer test filter.
The size of the generated video is fixed, and is 256x256. This source is useful in particular for
testing encoding features.
This source accepts the following options:
rate, r
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to
be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number
or a valid video frame rate abbreviation. The default value is "25".
duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.
If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.
test, t
Set the number or the name of the test to perform. Supported tests are:
dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma
cbp
mv
ring1
ring2
all
Default value is "all", which will cycle through the list of all tests.
Some examples:
mptestsrc=t=dc_luma
will generate a "dc_luma" test pattern.
frei0r_src
Provide a frei0r source.
To enable compilation of this filter you need to install the frei0r header and configure FFmpeg with
"--enable-frei0r".
This source accepts the following parameters:
size
The size of the video to generate. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.
framerate
The framerate of the generated video. It may be a string of the form num/den or a frame rate
abbreviation.
filter_name
The name to the frei0r source to load. For more information regarding frei0r and how to set the
parameters, read the frei0r section in the video filters documentation.
filter_params
A '|'-separated list of parameters to pass to the frei0r source.
For example, to generate a frei0r partik0l source with size 200x200 and frame rate 10 which is overlaid
on the overlay filter main input:
frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay
life
Generate a life pattern.
This source is based on a generalization of John Conway's life game.
The sourced input represents a life grid, each pixel represents a cell which can be in one of two
possible states, alive or dead. Every cell interacts with its eight neighbours, which are the cells that
are horizontally, vertically, or diagonally adjacent.
At each interaction the grid evolves according to the adopted rule, which specifies the number of
neighbor alive cells which will make a cell stay alive or born. The rule option allows one to specify the
rule to adopt.
This source accepts the following options:
filename, f
Set the file from which to read the initial grid state. In the file, each non-whitespace character is
considered an alive cell, and newline is used to delimit the end of each row.
If this option is not specified, the initial grid is generated randomly.
rate, r
Set the video rate, that is the number of frames generated per second. Default is 25.
random_fill_ratio, ratio
Set the random fill ratio for the initial random grid. It is a floating point number value ranging
from 0 to 1, defaults to 1/PHI. It is ignored when a file is specified.
random_seed, seed
Set the seed for filling the initial random grid, must be an integer included between 0 and
UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random
seed on a best effort basis.
rule
Set the life rule.
A rule can be specified with a code of the kind "SNS/BNB", where NS and NB are sequences of numbers
in the range 0-8, NS specifies the number of alive neighbor cells which make a live cell stay alive,
and NB the number of alive neighbor cells which make a dead cell to become alive (i.e. to "born").
"s" and "b" can be used in place of "S" and "B", respectively.
Alternatively a rule can be specified by an 18-bits integer. The 9 high order bits are used to encode
the next cell state if it is alive for each number of neighbor alive cells, the low order bits
specify the rule for "borning" new cells. Higher order bits encode for an higher number of neighbor
cells. For example the number 6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born rule
of 9, which corresponds to "S23/B03".
Default value is "S23/B3", which is the original Conway's game of life rule, and will keep a cell
alive if it has 2 or 3 neighbor alive cells, and will born a new cell if there are three alive cells
around a dead cell.
size, s
Set the size of the output video. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual.
If filename is specified, the size is set by default to the same size of the input file. If size is
set, it must contain the size specified in the input file, and the initial grid defined in that file
is centered in the larger resulting area.
If a filename is not specified, the size value defaults to "320x240" (used for a randomly generated
initial grid).
stitch
If set to 1, stitch the left and right grid edges together, and the top and bottom edges also.
Defaults to 1.
mold
Set cell mold speed. If set, a dead cell will go from death_color to mold_color with a step of mold.
mold can have a value from 0 to 255.
life_color
Set the color of living (or new born) cells.
death_color
Set the color of dead cells. If mold is set, this is the first color used to represent a dead cell.
mold_color
Set mold color, for definitely dead and moldy cells.
For the syntax of these 3 color options, check the "Color" section in the ffmpeg-utils manual.
Examples
• Read a grid from pattern, and center it on a grid of size 300x300 pixels:
life=f=pattern:s=300x300
• Generate a random grid of size 200x200, with a fill ratio of 2/3:
life=ratio=2/3:s=200x200
• Specify a custom rule for evolving a randomly generated grid:
life=rule=S14/B34
• Full example with slow death effect (mold) using ffplay:
ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16
allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars, smptehdbars, testsrc, testsrc2,
yuvtestsrc
The "allrgb" source returns frames of size 4096x4096 of all rgb colors.
The "allyuv" source returns frames of size 4096x4096 of all yuv colors.
The "color" source provides an uniformly colored input.
The "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 "testsrc2" source is similar to testsrc, but supports more pixel formats instead of just "rgb24".
This allows using it as an input for other tests without requiring a format conversion.
The "yuvtestsrc" source generates an YUV test pattern. You should see a y, cb and cr stripe from top to
bottom.
The sources accept the following parameters:
alpha
Specify the alpha (opacity) of the background, only available in the "testsrc2" source. The value
must be between 0 (fully transparent) and 255 (fully opaque, the default).
color, c
Specify the color of the source, only available in the "color" source. For the syntax of this option,
check the "Color" section in the ffmpeg-utils manual.
level
Specify the level of the Hald CLUT, only available in the "haldclutsrc" source. A level of "N"
generates a picture of "N*N*N" by "N*N*N" pixels to be used as identity matrix for 3D lookup tables.
Each component is coded on a "1/(N*N)" scale.
size, s
Specify the size of the sourced video. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. The default value is "320x240".
This option is not available with the "haldclutsrc" filter.
rate, r
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to
be a string in the format frame_rate_num/frame_rate_den, an integer number, a floating point number
or a valid video frame rate abbreviation. The default value is "25".
sar Set the sample aspect ratio of the sourced video.
duration, d
Set the duration of the sourced video. See the Time duration section in the ffmpeg-utils(1) manual
for the accepted syntax.
If not specified, or the expressed duration is negative, the video is supposed to be generated
forever.
decimals, n
Set the number of decimals to show in the timestamp, only available in the "testsrc" source.
The displayed timestamp value will correspond to the original timestamp value multiplied by the power
of 10 of the specified value. Default value is 0.
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
Commands
The "color" source supports the following commands:
c, color
Set the color of the created image. Accepts the same syntax of the corresponding color option.
VIDEO SINKS
Below is a description of the currently available video sinks.
buffersink
Buffer video frames, and make them available to the end of the filter graph.
This sink is mainly intended for programmatic use, in particular through the interface defined in
libavfilter/buffersink.h or the options system.
It accepts a pointer to an AVBufferSinkContext structure, which defines the incoming buffers' formats, to
be passed as the opaque parameter to "avfilter_init_filter" for initialization.
nullsink
Null video sink: do absolutely nothing with the input video. It is mainly useful as a template and for
use in analysis / debugging tools.
MULTIMEDIA FILTERS
Below is a description of the currently available multimedia filters.
abitscope
Convert input audio to a video output, displaying the audio bit scope.
The filter accepts the following options:
rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "1024x256".
colors
Specify list of colors separated by space or by '|' which will be used to draw channels. Unrecognized
or missing colors will be replaced by white color.
ahistogram
Convert input audio to a video output, displaying the volume histogram.
The filter accepts the following options:
dmode
Specify how histogram is calculated.
It accepts the following values:
single
Use single histogram for all channels.
separate
Use separate histogram for each channel.
Default is "single".
rate, r
Set frame rate, expressed as number of frames per second. Default value is "25".
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "hd720".
scale
Set display scale.
It accepts the following values:
log logarithmic
sqrt
square root
cbrt
cubic root
lin linear
rlog
reverse logarithmic
Default is "log".
ascale
Set amplitude scale.
It accepts the following values:
log logarithmic
lin linear
Default is "log".
acount
Set how much frames to accumulate in histogram. Defauls is 1. Setting this to -1 accumulates all
frames.
rheight
Set histogram ratio of window height.
slide
Set sonogram sliding.
It accepts the following values:
replace
replace old rows with new ones.
scroll
scroll from top to bottom.
Default is "replace".
aphasemeter
Convert input audio to a video output, displaying the audio phase.
The filter accepts the following options:
rate, r
Set the output frame rate. Default value is 25.
size, s
Set the video size for the output. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual. Default value is "800x400".
rc
gc
bc Specify the red, green, blue contrast. Default values are 2, 7 and 1. Allowed range is "[0, 255]".
mpc Set color which will be used for drawing median phase. If color is "none" which is default, no median
phase value will be drawn.
video
Enable video output. Default is enabled.
The filter also exports the frame metadata "lavfi.aphasemeter.phase" which represents mean phase of
current audio frame. Value is in range "[-1, 1]". The "-1" means left and right channels are completely
out of phase and 1 means channels are in phase.
avectorscope
Convert input audio to a video output, representing the audio vector scope.
The filter is used to measure the difference between channels of stereo audio stream. A 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:
mode, m
Set the vectorscope mode.
Available values are:
lissajous
Lissajous rotated by 45 degrees.
lissajous_xy
Same as above but not rotated.
polar
Shape resembling half of circle.
Default value is lissajous.
size, s
Set the video size for the output. For the syntax of this option, check the "Video size" section in
the ffmpeg-utils manual. Default value is "400x400".
rate, r
Set the output frame rate. Default value is 25.
rc
gc
bc
ac Specify the red, green, blue and alpha contrast. Default values are 40, 160, 80 and 255. Allowed
range is "[0, 255]".
rf
gf
bf
af Specify the red, green, blue and alpha fade. Default values are 15, 10, 5 and 5. Allowed range is
"[0, 255]".
zoom
Set the zoom factor. Default value is 1. Allowed range is "[0, 10]". Values lower than 1 will auto
adjust zoom factor to maximal possible value.
draw
Set the vectorscope drawing mode.
Available values are:
dot Draw dot for each sample.
line
Draw line between previous and current sample.
Default value is dot.
scale
Specify amplitude scale of audio samples.
Available values are:
lin Linear.
sqrt
Square root.
cbrt
Cubic root.
log Logarithmic.
Examples
• Complete example using ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'
bench, abench
Benchmark part of a filtergraph.
The filter accepts the following options:
action
Start or stop a timer.
Available values are:
start
Get the current time, set it as frame metadata (using the key "lavfi.bench.start_time"), and
forward the frame to the next filter.
stop
Get the current time and fetch the "lavfi.bench.start_time" metadata from the input frame
metadata to get the time difference. Time difference, average, maximum and minimum time
(respectively "t", "avg", "max" and "min") are then printed. The timestamps are expressed in
seconds.
Examples
• Benchmark selectivecolor filter:
bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop
concat
Concatenate audio and video streams, joining them together one after the other.
The filter works on segments of synchronized video and audio streams. All segments must have the same
number of streams of each type, and that will also be the number of streams at output.
The filter accepts the following options:
n Set the number of segments. Default is 2.
v Set the number of output video streams, that is also the number of video streams in each segment.
Default is 1.
a Set the number of output audio streams, that is also the number of audio streams in each segment.
Default is 0.
unsafe
Activate unsafe mode: do not fail if segments have a different format.
The filter has v+a outputs: first v video outputs, then a audio outputs.
There are nx(v+a) inputs: first the inputs for the first segment, in the same order as the outputs, then
the inputs for the second segment, etc.
Related streams do not always have exactly the same duration, for various reasons including codec frame
size or sloppy authoring. For that reason, related synchronized streams (e.g. a video and its audio
track) should be concatenated at once. The concat filter will use the duration of the longest stream in
each segment (except the last one), and if necessary pad shorter audio streams with silence.
For this filter to work correctly, all segments must start at timestamp 0.
All corresponding streams must have the same parameters in all segments; the filtering system will
automatically select a common pixel format for video streams, and a common sample format, sample rate and
channel layout for audio streams, but other settings, such as resolution, must be converted explicitly by
the user.
Different frame rates are acceptable but will result in variable frame rate at output; be sure to
configure the output file to handle it.
Examples
• Concatenate an opening, an episode and an ending, all in bilingual version (video in stream 0, audio
in streams 1 and 2):
ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
'[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
concat=n=3:v=1:a=2 [v] [a1] [a2]' \
-map '[v]' -map '[a1]' -map '[a2]' output.mkv
• Concatenate two parts, handling audio and video separately, using the (a)movie sources, and adjusting
the resolution:
movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
[v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]
Note that a desync will happen at the stitch if the audio and video streams do not have exactly the
same duration in the first file.
drawgraph, adrawgraph
Draw a graph using input video or audio metadata.
It accepts the following parameters:
m1 Set 1st frame metadata key from which metadata values will be used to draw a graph.
fg1 Set 1st foreground color expression.
m2 Set 2nd frame metadata key from which metadata values will be used to draw a graph.
fg2 Set 2nd foreground color expression.
m3 Set 3rd frame metadata key from which metadata values will be used to draw a graph.
fg3 Set 3rd foreground color expression.
m4 Set 4th frame metadata key from which metadata values will be used to draw a graph.
fg4 Set 4th foreground color expression.
min Set minimal value of metadata value.
max Set maximal value of metadata value.
bg Set graph background color. Default is white.
mode
Set graph mode.
Available values for mode is:
bar
dot
line
Default is "line".
slide
Set slide mode.
Available values for slide is:
frame
Draw new frame when right border is reached.
replace
Replace old columns with new ones.
scroll
Scroll from right to left.
rscroll
Scroll from left to right.
picture
Draw single picture.
Default is "frame".
size
Set size of graph video. For the syntax of this option, check the "Video size" section in the ffmpeg-
utils manual. The default value is "900x256".
The foreground color expressions can use the following variables:
MIN Minimal value of metadata value.
MAX Maximal value of metadata value.
VAL Current metadata key value.
The color is defined as 0xAABBGGRR.
Example using metadata from signalstats filter:
signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255
Example using metadata from ebur128 filter:
ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5
ebur128
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:
video
Activate the video output. The audio stream is passed unchanged whether this option is set or no. The
video stream will be the first output stream if activated. Default is 0.
size
Set the video size. This option is for video only. For the syntax of this option, check the "Video
size" section in the ffmpeg-utils manual. Default and minimum resolution is "640x480".
meter
Set the EBU scale meter. Default is 9. Common values are 9 and 18, respectively for EBU scale meter
+9 and EBU scale meter +18. Any other integer value between this range is allowed.
metadata
Set metadata injection. If set to 1, the audio input will be segmented into 100ms output frames, each
of them containing various loudness information in metadata. All the metadata keys are prefixed with
"lavfi.r128.".
Default is 0.
framelog
Force the frame logging level.
Available values are:
info
information logging level
verbose
verbose logging level
By default, the logging level is set to info. If the video or the metadata options are set, it
switches to verbose.
peak
Set peak mode(s).
Available modes can be cumulated (the option is a "flag" type). Possible values are:
none
Disable any peak mode (default).
sample
Enable sample-peak mode.
Simple peak mode looking for the higher sample value. It logs a message for sample-peak
(identified by "SPK").
true
Enable true-peak mode.
If enabled, the peak lookup is done on an over-sampled version of the input stream for better
peak accuracy. It logs a message for true-peak. (identified by "TPK") and true-peak per frame
(identified by "FTPK"). This mode requires a build with "libswresample".
dualmono
Treat mono input files as "dual mono". If a mono file is intended for playback on a stereo system,
its EBU R128 measurement will be perceptually incorrect. If set to "true", this option will
compensate for this effect. Multi-channel input files are not affected by this option.
panlaw
Set a specific pan law to be used for the measurement of dual mono files. This parameter is
optional, and has a default value of -3.01dB.
Examples
• Real-time graph using ffplay, with a EBU scale meter +18:
ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"
• Run an analysis with ffmpeg:
ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -
interleave, ainterleave
Temporally interleave frames from several inputs.
"interleave" works with video inputs, "ainterleave" with audio.
These filters read frames from several inputs and send the oldest queued frame to the output.
Input streams must have well defined, monotonically increasing frame timestamp values.
In order to submit one frame to output, these filters need to enqueue at least one frame for each input,
so they cannot work in case one input is not yet terminated and will not receive incoming frames.
For example consider the case when one input is a "select" filter which always drops input frames. The
"interleave" filter will keep reading from that input, but it will never be able to send new frames to
output until the input sends an end-of-stream signal.
Also, depending on inputs synchronization, the filters will drop frames in case one input receives more
frames than the other ones, and the queue is already filled.
These filters accept the following options:
nb_inputs, n
Set the number of different inputs, it is 2 by default.
Examples
• Interleave frames belonging to different streams using ffmpeg:
ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi
• Add flickering blur effect:
select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave
metadata, ametadata
Manipulate frame metadata.
This filter accepts the following options:
mode
Set mode of operation of the filter.
Can be one of the following:
select
If both "value" and "key" is set, select frames which have such metadata. If only "key" is set,
select every frame that has such key in metadata.
add Add new metadata "key" and "value". If key is already available do nothing.
modify
Modify value of already present key.
delete
If "value" is set, delete only keys that have such value. Otherwise, delete key. If "key" is not
set, delete all metadata values in the frame.
print
Print key and its value if metadata was found. If "key" is not set print all metadata values
available in frame.
key Set key used with all modes. Must be set for all modes except "print" and "delete".
value
Set metadata value which will be used. This option is mandatory for "modify" and "add" mode.
function
Which function to use when comparing metadata value and "value".
Can be one of following:
same_str
Values are interpreted as strings, returns true if metadata value is same as "value".
starts_with
Values are interpreted as strings, returns true if metadata value starts with the "value" option
string.
less
Values are interpreted as floats, returns true if metadata value is less than "value".
equal
Values are interpreted as floats, returns true if "value" is equal with metadata value.
greater
Values are interpreted as floats, returns true if metadata value is greater than "value".
expr
Values are interpreted as floats, returns true if expression from option "expr" evaluates to
true.
expr
Set expression which is used when "function" is set to "expr". The expression is evaluated through
the eval API and can contain the following constants:
VALUE1
Float representation of "value" from metadata key.
VALUE2
Float representation of "value" as supplied by user in "value" option.
file
If specified in "print" mode, output is written to the named file. Instead of plain filename any
writable url can be specified. Filename ``-'' is a shorthand for standard output. If "file" option is
not set, output is written to the log with AV_LOG_INFO loglevel.
Examples
• Print all metadata values for frames with key "lavfi.singnalstats.YDIF" with values between 0 and 1.
signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'
• Print silencedetect output to file metadata.txt.
silencedetect,ametadata=mode=print:file=metadata.txt
• Direct all metadata to a pipe with file descriptor 4.
metadata=mode=print:file='pipe\:4'
perms, aperms
Set read/write permissions for the output frames.
These filters are mainly aimed at developers to test direct path in the following filter in the
filtergraph.
The filters accept the following options:
mode
Select the permissions mode.
It accepts the following values:
none
Do nothing. This is the default.
ro Set all the output frames read-only.
rw Set all the output frames directly writable.
toggle
Make the frame read-only if writable, and writable if read-only.
random
Set each output frame read-only or writable randomly.
seed
Set the seed for the random mode, must be an integer included between 0 and "UINT32_MAX". If not
specified, or if explicitly set to "-1", the filter will try to use a good random seed on a best
effort basis.
Note: in case of auto-inserted filter between the permission filter and the following one, the permission
might not be received as expected in that following filter. Inserting a format or aformat filter before
the perms/aperms filter can avoid this problem.
realtime, arealtime
Slow down filtering to match real time approximately.
These filters will pause the filtering for a variable amount of time to match the output rate with the
input timestamps. They are similar to the re option to "ffmpeg".
They accept the following options:
limit
Time limit for the pauses. Any pause longer than that will be considered a timestamp discontinuity
and reset the timer. Default is 2 seconds.
select, aselect
Select frames to pass in output.
This filter accepts the following options:
expr, e
Set expression, which is evaluated for each input frame.
If the expression is evaluated to zero, the frame is discarded.
If the evaluation result is negative or NaN, the frame is sent to the first output; otherwise it is
sent to the output with index "ceil(val)-1", assuming that the input index starts from 0.
For example a value of 1.2 corresponds to the output with index "ceil(1.2)-1 = 2-1 = 1", that is the
second output.
outputs, n
Set the number of outputs. The output to which to send the selected frame is based on the result of
the evaluation. Default value is 1.
The expression can contain the following constants:
n The (sequential) number of the filtered frame, starting from 0.
selected_n
The (sequential) number of the selected frame, starting from 0.
prev_selected_n
The sequential number of the last selected frame. It's NAN if undefined.
TB The timebase of the input timestamps.
pts The PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units. It's NAN if
undefined.
t The PTS of the filtered video frame, expressed in seconds. It's NAN if undefined.
prev_pts
The PTS of the previously filtered video frame. It's NAN if undefined.
prev_selected_pts
The PTS of the last previously filtered video frame. It's NAN if undefined.
prev_selected_t
The PTS of the last previously selected video frame. It's NAN if undefined.
start_pts
The PTS of the first video frame in the video. It's NAN if undefined.
start_t
The time of the first video frame in the video. It's NAN if undefined.
pict_type (video only)
The type of the filtered frame. It can assume one of the following values:
I
P
B
S
SI
SP
BI
interlace_type (video only)
The frame interlace type. It can assume one of the following values:
PROGRESSIVE
The frame is progressive (not interlaced).
TOPFIRST
The frame is top-field-first.
BOTTOMFIRST
The frame is bottom-field-first.
consumed_sample_n (audio only)
the number of selected samples before the current frame
samples_n (audio only)
the number of samples in the current frame
sample_rate (audio only)
the input sample rate
key This is 1 if the filtered frame is a key-frame, 0 otherwise.
pos the position in the file of the filtered frame, -1 if the information is not available (e.g. for
synthetic video)
scene (video only)
value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current
frame to introduce a new scene, while a higher value means the current frame is more likely to be one
(see the example below)
concatdec_select
The concat demuxer can select only part of a concat input file by setting an inpoint and an outpoint,
but the output packets may not be entirely contained in the selected interval. By using this
variable, it is possible to skip frames generated by the concat demuxer which are not exactly
contained in the selected interval.
This works by comparing the frame pts against the lavf.concat.start_time and the lavf.concat.duration
packet metadata values which are also present in the decoded frames.
The concatdec_select variable is -1 if the frame pts is at least start_time and either the duration
metadata is missing or the frame pts is less than start_time + duration, 0 otherwise, and NaN if the
start_time metadata is missing.
That basically means that an input frame is selected if its pts is within the interval set by the
concat demuxer.
The default value of the select expression is "1".
Examples
• Select all frames in input:
select
The example above is the same as:
select=1
• Skip all frames:
select=0
• Select only I-frames:
select='eq(pict_type\,I)'
• Select one frame every 100:
select='not(mod(n\,100))'
• Select only frames contained in the 10-20 time interval:
select=between(t\,10\,20)
• Select only I-frames contained in the 10-20 time interval:
select=between(t\,10\,20)*eq(pict_type\,I)
• Select frames with a minimum distance of 10 seconds:
select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'
• Use aselect to select only audio frames with samples number > 100:
aselect='gt(samples_n\,100)'
• Create a mosaic of the first scenes:
ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png
Comparing scene against a value between 0.3 and 0.5 is generally a sane choice.
• Send even and odd frames to separate outputs, and compose them:
select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h
• Select useful frames from an ffconcat file which is using inpoints and outpoints but where the source
files are not intra frame only.
ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi
sendcmd, asendcmd
Send commands to filters in the filtergraph.
These filters read commands to be sent to other filters in the filtergraph.
"sendcmd" must be inserted between two video filters, "asendcmd" must be inserted between two audio
filters, but apart from that they act the same way.
The specification of commands can be provided in the filter arguments with the commands option, or in a
file specified by the filename option.
These filters accept the following options:
commands, c
Set the commands to be read and sent to the other filters.
filename, f
Set the filename of the commands to be read and sent to the other filters.
Commands syntax
A commands description consists of a sequence of interval specifications, comprising a list of commands
to be executed when a particular event related to that interval occurs. The occurring event is typically
the current frame time entering or leaving a given time interval.
An interval is specified by the following syntax:
<START>[-<END>] <COMMANDS>;
The time interval is specified by the START and END times. END is optional and defaults to the maximum
time.
The current frame time is considered within the specified interval if it is included in the interval
[START, END), that is when the time is greater or equal to START and is lesser than END.
COMMANDS consists of a sequence of one or more command specifications, separated by ",", relating to that
interval. The syntax of a command specification is given by:
[<FLAGS>] <TARGET> <COMMAND> <ARG>
FLAGS is optional and specifies the type of events relating to the time interval which enable sending the
specified command, and must be a non-null sequence of identifier flags separated by "+" or "|" and
enclosed between "[" and "]".
The following flags are recognized:
enter
The command is sent when the current frame timestamp enters the specified interval. In other words,
the command is sent when the previous frame timestamp was not in the given interval, and the current
is.
leave
The command is sent when the current frame timestamp leaves the specified interval. In other words,
the command is sent when the previous frame timestamp was in the given interval, and the current is
not.
If FLAGS is not specified, a default value of "[enter]" is assumed.
TARGET specifies the target of the command, usually the name of the filter class or a specific filter
instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional list of argument for the given COMMAND.
Between one interval specification and another, whitespaces, or sequences of characters starting with "#"
until the end of line, are ignored and can be used to annotate comments.
A simplified BNF description of the commands specification syntax follows:
<COMMAND_FLAG> ::= "enter" | "leave"
<COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
<COMMAND> ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
<COMMANDS> ::= <COMMAND> [,<COMMANDS>]
<INTERVAL> ::= <START>[-<END>] <COMMANDS>
<INTERVALS> ::= <INTERVAL>[;<INTERVALS>]
Examples
• Specify audio tempo change at second 4:
asendcmd=c='4.0 atempo tempo 1.5',atempo
• Target a specific filter instance:
asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my
• Specify a list of drawtext and hue commands in a file.
# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';
# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue s 0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue s 1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';
# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s exp(25-t)
A filtergraph allowing to read and process the above command list stored in a file test.cmd, can be
specified with:
sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue
setpts, asetpts
Change the PTS (presentation timestamp) of the input frames.
"setpts" works on video frames, "asetpts" on audio frames.
This filter accepts the following options:
expr
The expression which is evaluated for each frame to construct its timestamp.
The expression is evaluated through the eval API and can contain the following constants:
FRAME_RATE
frame rate, only defined for constant frame-rate video
PTS The presentation timestamp in input
N The count of the input frame for video or the number of consumed samples, not including the current
frame for audio, starting from 0.
NB_CONSUMED_SAMPLES
The number of consumed samples, not including the current frame (only audio)
NB_SAMPLES, S
The number of samples in the current frame (only audio)
SAMPLE_RATE, SR
The audio sample rate.
STARTPTS
The PTS of the first frame.
STARTT
the time in seconds of the first frame
INTERLACED
State whether the current frame is interlaced.
T the time in seconds of the current frame
POS original position in the file of the frame, or undefined if undefined for the current frame
PREV_INPTS
The previous input PTS.
PREV_INT
previous input time in seconds
PREV_OUTPTS
The previous output PTS.
PREV_OUTT
previous output time in seconds
RTCTIME
The wallclock (RTC) time in microseconds. This is deprecated, use time(0) instead.
RTCSTART
The wallclock (RTC) time at the start of the movie in microseconds.
TB The timebase of the input timestamps.
Examples
• Start counting PTS from zero
setpts=PTS-STARTPTS
• Apply fast motion effect:
setpts=0.5*PTS
• Apply slow motion effect:
setpts=2.0*PTS
• Set fixed rate of 25 frames per second:
setpts=N/(25*TB)
• Set fixed rate 25 fps with some jitter:
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
• Apply an offset of 10 seconds to the input PTS:
setpts=PTS+10/TB
• Generate timestamps from a "live source" and rebase onto the current timebase:
setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'
• Generate timestamps by counting samples:
asetpts=N/SR/TB
settb, asettb
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase
configuration.
It accepts the following parameters:
expr, tb
The expression which is evaluated into the output timebase.
The value for tb is an arithmetic expression representing a rational. The expression can contain the
constants "AVTB" (the default timebase), "intb" (the input timebase) and "sr" (the sample rate, audio
only). Default value is "intb".
Examples
• Set the timebase to 1/25:
settb=expr=1/25
• Set the timebase to 1/10:
settb=expr=0.1
• Set the timebase to 1001/1000:
settb=1+0.001
• Set the timebase to 2*intb:
settb=2*intb
• Set the default timebase value:
settb=AVTB
showcqt
Convert input audio to a video output representing frequency spectrum logarithmically using Brown-
Puckette constant Q transform algorithm with direct frequency domain coefficient calculation (but the
transform itself is not really constant Q, instead the Q factor is actually variable/clamped), with
musical tone scale, from E0 to D#10.
The filter accepts the following options:
size, s
Specify the video size for the output. It must be even. For the syntax of this option, check the
"Video size" section in the ffmpeg-utils manual. Default value is "1920x1080".
fps, rate, r
Set the output frame rate. Default value is 25.
bar_h
Set the bargraph height. It must be even. Default value is "-1" which computes the bargraph height
automatically.
axis_h
Set the axis height. It must be even. Default value is "-1" which computes the axis height
automatically.
sono_h
Set the sonogram height. It must be even. Default value is "-1" which computes the sonogram height
automatically.
fullhd
Set the fullhd resolution. This option is deprecated, use size, s instead. Default value is 1.
sono_v, volume
Specify the sonogram volume expression. It can contain variables:
bar_v
the bar_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
a_weighting(f)
A-weighting of equal loudness
b_weighting(f)
B-weighting of equal loudness
c_weighting(f)
C-weighting of equal loudness.
Default value is 16.
bar_v, volume2
Specify the bargraph volume expression. It can contain variables:
sono_v
the sono_v evaluated expression
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
a_weighting(f)
A-weighting of equal loudness
b_weighting(f)
B-weighting of equal loudness
c_weighting(f)
C-weighting of equal loudness.
Default value is "sono_v".
sono_g, gamma
Specify the sonogram gamma. Lower gamma makes the spectrum more contrast, higher gamma makes the
spectrum having more range. Default value is 3. Acceptable range is "[1, 7]".
bar_g, gamma2
Specify the bargraph gamma. Default value is 1. Acceptable range is "[1, 7]".
bar_t
Specify the bargraph transparency level. Lower value makes the bargraph sharper. Default value is 1.
Acceptable range is "[0, 1]".
timeclamp, tc
Specify the transform timeclamp. At low frequency, there is trade-off between accuracy in time domain
and frequency domain. If timeclamp is lower, event in time domain is represented more accurately
(such as fast bass drum), otherwise event in frequency domain is represented more accurately (such as
bass guitar). Acceptable range is "[0.002, 1]". Default value is 0.17.
attack
Set attack time in seconds. The default is 0 (disabled). Otherwise, it limits future samples by
applying asymmetric windowing in time domain, useful when low latency is required. Accepted range is
"[0, 1]".
basefreq
Specify the transform base frequency. Default value is 20.01523126408007475, which is frequency 50
cents below E0. Acceptable range is "[10, 100000]".
endfreq
Specify the transform end frequency. Default value is 20495.59681441799654, which is frequency 50
cents above D#10. Acceptable range is "[10, 100000]".
coeffclamp
This option is deprecated and ignored.
tlength
Specify the transform length in time domain. Use this option to control accuracy trade-off between
time domain and frequency domain at every frequency sample. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option.
Default value is "384*tc/(384+tc*f)".
count
Specify the transform count for every video frame. Default value is 6. Acceptable range is "[1,
30]".
fcount
Specify the transform count for every single pixel. Default value is 0, which makes it computed
automatically. Acceptable range is "[0, 10]".
fontfile
Specify font file for use with freetype to draw the axis. If not specified, use embedded font. Note
that drawing with font file or embedded font is not implemented with custom basefreq and endfreq, use
axisfile option instead.
font
Specify fontconfig pattern. This has lower priority than fontfile. The : in the pattern may be
replaced by | to avoid unnecessary escaping.
fontcolor
Specify font color expression. This is arithmetic expression that should return integer value
0xRRGGBB. It can contain variables:
frequency, freq, f
the frequency where it is evaluated
timeclamp, tc
the value of timeclamp option
and functions:
midi(f)
midi number of frequency f, some midi numbers: E0(16), C1(24), C2(36), A4(69)
r(x), g(x), b(x)
red, green, and blue value of intensity x.
Default value is "st(0, (midi(f)-59.5)/12); st(1, if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)),
0)); r(1-ld(1)) + b(ld(1))".
axisfile
Specify image file to draw the axis. This option override fontfile and fontcolor option.
axis, text
Enable/disable drawing text to the axis. If it is set to 0, drawing to the axis is disabled, ignoring
fontfile and axisfile option. Default value is 1.
csp Set colorspace. The accepted values are:
unspecified
Unspecified (default)
bt709
BT.709
fcc FCC
bt470bg
BT.470BG or BT.601-6 625
smpte170m
SMPTE-170M or BT.601-6 525
smpte240m
SMPTE-240M
bt2020ncl
BT.2020 with non-constant luminance
cscheme
Set spectrogram color scheme. This is list of floating point values with format
"left_r|left_g|left_b|right_r|right_g|right_b". The default is "1|0.5|0|0|0.5|1".
Examples
• Playing audio while showing the spectrum:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'
• Same as above, but with frame rate 30 fps:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'
• Playing at 1280x720:
ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'
• Disable sonogram display:
sono_h=0
• A1 and its harmonics: A1, A2, (near)E3, A3:
ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt [out0]'
• Same as above, but with more accuracy in frequency domain:
ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'
• Custom volume:
bar_v=10:sono_v=bar_v*a_weighting(f)
• Custom gamma, now spectrum is linear to the amplitude.
bar_g=2:sono_g=2
• Custom tlength equation:
tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'
• Custom fontcolor and fontfile, C-note is colored green, others are colored blue:
fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf
• Custom font using fontconfig:
font='Courier New,Monospace,mono|bold'
• Custom frequency range with custom axis using image file:
axisfile=myaxis.png:basefreq=40:endfreq=10000
showfreqs
Convert input audio to video output representing the audio power spectrum. Audio amplitude is on Y-axis
while frequency is on X-axis.
The filter accepts the following options:
size, s
Specify size of video. For the syntax of this option, check the "Video size" section in the ffmpeg-
utils manual. Default is "1024x512".
mode
Set display mode. This set how each frequency bin will be represented.
It accepts the following values:
line
bar
dot
Default is "bar".
ascale
Set amplitude scale.
It accepts the following values:
lin Linear scale.
sqrt
Square root scale.
cbrt
Cubic root scale.
log Logarithmic scale.
Default is "log".
fscale
Set frequency scale.
It accepts the following values:
lin Linear scale.
log Logarithmic scale.
rlog
Reverse logarithmic scale.
Default is "lin".
win_size
Set window size.
It accepts the following values:
w16
w32
w64
w128
w256
w512
w1024
w2048
w4096
w8192
w16384
w32768
w65536
Default is "w2048"
win_func
Set windowing function.
It accepts the following values:
rect
bartlett
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default is "hanning".
overlap
Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window
function will be picked.
averaging
Set time averaging. Setting this to 0 will display current maximal peaks. Default is 1, which means
time averaging is disabled.
colors
Specify list of colors separated by space or by '|' which will be used to draw channel frequencies.
Unrecognized or missing colors will be replaced by white color.
cmode
Set channel display mode.
It accepts the following values:
combined
separate
Default is "combined".
minamp
Set minimum amplitude used in "log" amplitude scaler.
showspectrum
Convert input audio to a video output, representing the audio frequency spectrum.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "640x512".
slide
Specify how the spectrum should slide along the window.
It accepts the following values:
replace
the samples start again on the left when they reach the right
scroll
the samples scroll from right to left
fullframe
frames are only produced when the samples reach the right
rscroll
the samples scroll from left to right
Default value is "replace".
mode
Specify display mode.
It accepts the following values:
combined
all channels are displayed in the same row
separate
all channels are displayed in separate rows
Default value is combined.
color
Specify display color mode.
It accepts the following values:
channel
each channel is displayed in a separate color
intensity
each channel is displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool color scheme
Default value is channel.
scale
Specify scale used for calculating intensity color values.
It accepts the following values:
lin linear
sqrt
square root, default
cbrt
cubic root
log logarithmic
4thrt
4th root
5thrt
5th root
Default value is sqrt.
saturation
Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is
no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.
win_func
Set window function.
It accepts the following values:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".
overlap
Set ratio of overlap window. Default value is 0. When value is 1 overlap is set to recommended size
for specific window function currently used.
gain
Set scale gain for calculating intensity color values. Default value is 1.
data
Set which data to display. Can be "magnitude", default or "phase".
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.
The usage is very similar to the showwaves filter; see the examples in that section.
Examples
• Large window with logarithmic color scaling:
showspectrum=s=1280x480:scale=log
• Complete example for a colored and sliding spectrum per channel using ffplay:
ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'
showspectrumpic
Convert input audio to a single video frame, representing the audio frequency spectrum.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "4096x2048".
mode
Specify display mode.
It accepts the following values:
combined
all channels are displayed in the same row
separate
all channels are displayed in separate rows
Default value is combined.
color
Specify display color mode.
It accepts the following values:
channel
each channel is displayed in a separate color
intensity
each channel is displayed using the same color scheme
rainbow
each channel is displayed using the rainbow color scheme
moreland
each channel is displayed using the moreland color scheme
nebulae
each channel is displayed using the nebulae color scheme
fire
each channel is displayed using the fire color scheme
fiery
each channel is displayed using the fiery color scheme
fruit
each channel is displayed using the fruit color scheme
cool
each channel is displayed using the cool color scheme
Default value is intensity.
scale
Specify scale used for calculating intensity color values.
It accepts the following values:
lin linear
sqrt
square root, default
cbrt
cubic root
log logarithmic
4thrt
4th root
5thrt
5th root
Default value is log.
saturation
Set saturation modifier for displayed colors. Negative values provide alternative color scheme. 0 is
no saturation at all. Saturation must be in [-10.0, 10.0] range. Default value is 1.
win_func
Set window function.
It accepts the following values:
rect
bartlett
hann
hanning
hamming
blackman
welch
flattop
bharris
bnuttall
bhann
sine
nuttall
lanczos
gauss
tukey
dolph
cauchy
parzen
poisson
Default value is "hann".
orientation
Set orientation of time vs frequency axis. Can be "vertical" or "horizontal". Default is "vertical".
gain
Set scale gain for calculating intensity color values. Default value is 1.
legend
Draw time and frequency axes and legends. Default is enabled.
rotation
Set color rotation, must be in [-1.0, 1.0] range. Default value is 0.
Examples
• Extract an audio spectrogram of a whole audio track in a 1024x1024 picture using ffmpeg:
ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png
showvolume
Convert input audio volume to a video output.
The filter accepts the following options:
rate, r
Set video rate.
b Set border width, allowed range is [0, 5]. Default is 1.
w Set channel width, allowed range is [80, 8192]. Default is 400.
h Set channel height, allowed range is [1, 900]. Default is 20.
f Set fade, allowed range is [0.001, 1]. Default is 0.95.
c Set volume color expression.
The expression can use the following variables:
VOLUME
Current max volume of channel in dB.
PEAK
Current peak.
CHANNEL
Current channel number, starting from 0.
t If set, displays channel names. Default is enabled.
v If set, displays volume values. Default is enabled.
o Set orientation, can be "horizontal" or "vertical", default is "horizontal".
s Set step size, allowed range s [0, 5]. Default is 0, which means step is disabled.
showwaves
Convert input audio to a video output, representing the samples waves.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "600x240".
mode
Set display mode.
Available values are:
point
Draw a point for each sample.
line
Draw a vertical line for each sample.
p2p Draw a point for each sample and a line between them.
cline
Draw a centered vertical line for each sample.
Default value is "point".
n Set the number of samples which are printed on the same column. A larger value will decrease the
frame rate. Must be a positive integer. This option can be set only if the value for rate is not
explicitly specified.
rate, r
Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".
split_channels
Set if channels should be drawn separately or overlap. Default value is 0.
colors
Set colors separated by '|' which are going to be used for drawing of each channel.
scale
Set amplitude scale.
Available values are:
lin Linear.
log Logarithmic.
sqrt
Square root.
cbrt
Cubic root.
Default is linear.
Examples
• Output the input file audio and the corresponding video representation at the same time:
amovie=a.mp3,asplit[out0],showwaves[out1]
• Create a synthetic signal and show it with showwaves, forcing a frame rate of 30 frames per second:
aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]
showwavespic
Convert input audio to a single video frame, representing the samples waves.
The filter accepts the following options:
size, s
Specify the video size for the output. For the syntax of this option, check the "Video size" section
in the ffmpeg-utils manual. Default value is "600x240".
split_channels
Set if channels should be drawn separately or overlap. Default value is 0.
colors
Set colors separated by '|' which are going to be used for drawing of each channel.
scale
Set amplitude scale.
Available values are:
lin Linear.
log Logarithmic.
sqrt
Square root.
cbrt
Cubic root.
Default is linear.
Examples
• Extract a channel split representation of the wave form of a whole audio track in a 1024x800 picture
using ffmpeg:
ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png
sidedata, asidedata
Delete frame side data, or select frames based on it.
This filter accepts the following options:
mode
Set mode of operation of the filter.
Can be one of the following:
select
Select every frame with side data of "type".
delete
Delete side data of "type". If "type" is not set, delete all side data in the frame.
type
Set side data type used with all modes. Must be set for "select" mode. For the list of frame side
data types, refer to the "AVFrameSideDataType" enum in libavutil/frame.h. For example, to choose
"AV_FRAME_DATA_PANSCAN" side data, you must specify "PANSCAN".
spectrumsynth
Sythesize audio from 2 input video spectrums, first input stream represents magnitude across time and
second represents phase across time. The filter will transform from frequency domain as displayed in
videos back to time domain as presented in audio output.
This filter is primarily created for reversing processed showspectrum filter outputs, but can synthesize
sound from other spectrograms too. But in such case results are going to be poor if the phase data is
not available, because in such cases phase data need to be recreated, usually its just recreated from
random noise. For best results use gray only output ("channel" color mode in showspectrum filter) and
"log" scale for magnitude video and "lin" scale for phase video. To produce phase, for 2nd video, use
"data" option. Inputs videos should generally use "fullframe" slide mode as that saves resources needed
for decoding video.
The filter accepts the following options:
sample_rate
Specify sample rate of output audio, the sample rate of audio from which spectrum was generated may
differ.
channels
Set number of channels represented in input video spectrums.
scale
Set scale which was used when generating magnitude input spectrum. Can be "lin" or "log". Default is
"log".
slide
Set slide which was used when generating inputs spectrums. Can be "replace", "scroll", "fullframe"
or "rscroll". Default is "fullframe".
win_func
Set window function used for resynthesis.
overlap
Set window overlap. In range "[0, 1]". Default is 1, which means optimal overlap for selected window
function will be picked.
orientation
Set orientation of input videos. Can be "vertical" or "horizontal". Default is "vertical".
Examples
• First create magnitude and phase videos from audio, assuming audio is stereo with 44100 sample rate,
then resynthesize videos back to audio with spectrumsynth:
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac
split, asplit
Split input into several identical outputs.
"asplit" works with audio input, "split" with video.
The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults
to 2.
Examples
• Create two separate outputs from the same input:
[in] split [out0][out1]
• To create 3 or more outputs, you need to specify the number of outputs, like in:
[in] asplit=3 [out0][out1][out2]
• Create two separate outputs from the same input, one cropped and one padded:
[in] split [splitout1][splitout2];
[splitout1] crop=100:100:0:0 [cropout];
[splitout2] pad=200:200:100:100 [padout];
• Create 5 copies of the input audio with ffmpeg:
ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT
zmq, azmq
Receive commands sent through a libzmq client, and forward them to filters in the filtergraph.
"zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted between two video filters, "azmq"
between two audio filters.
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.
Examples
Look at tools/zmqsend for an example of a zmq client which can be used to send commands processed by
these filters.
Consider the following filtergraph generated by ffplay
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
MULTIMEDIA SOURCES
Below is a description of the currently available multimedia sources.
amovie
This is the same as movie source, except it selects an audio stream by default.
movie
Read audio and/or video stream(s) from a movie container.
It accepts the following parameters:
filename
The name of the resource to read (not necessarily a file; it can also be a device or a stream
accessed through some protocol).
format_name, f
Specifies the format assumed for the movie to read, and can be either the name of a container or an
input device. If not specified, the format is guessed from movie_name or by probing.
seek_point, sp
Specifies the seek point in seconds. The frames will be output starting from this seek point. The
parameter is evaluated with "av_strtod", so the numerical value may be suffixed by an IS postfix. The
default value is "0".
streams, s
Specifies the streams to read. Several streams can be specified, separated by "+". The source will
then have as many outputs, in the same order. The syntax is explained in the ``Stream specifiers''
section in the ffmpeg manual. Two special names, "dv" and "da" specify respectively the default (best
suited) video and audio stream. Default is "dv", or "da" if the filter is called as "amovie".
stream_index, si
Specifies the index of the video stream to read. If the value is -1, the most suitable video stream
will be automatically selected. The default value is "-1". Deprecated. If the filter is called
"amovie", it will select audio instead of video.
loop
Specifies how many times to read the stream in sequence. If the value is 0, the stream will be
looped infinitely. Default value is "1".
Note that when the movie is looped the source timestamps are not changed, so it will generate non
monotonically increasing timestamps.
discontinuity
Specifies the time difference between frames above which the point is considered a timestamp
discontinuity which is removed by adjusting the later timestamps.
It allows overlaying a second video on top of the main input of a filtergraph, as shown in this graph:
input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+
Examples
• Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it on top of the input labelled
"in":
movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]
• Read from a video4linux2 device, and overlay it on top of the input labelled "in":
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
[in] setpts=PTS-STARTPTS [main];
[main][over] overlay=16:16 [out]
• Read the first video stream and the audio stream with id 0x81 from dvd.vob; the video is connected to
the pad named "video" and the audio is connected to the pad named "audio":
movie=dvd.vob:s=v:0+#0x81 [video] [audio]
Commands
Both movie and amovie support the following commands:
seek
Perform seek using "av_seek_frame". The syntax is: seek stream_index|timestamp|flags
• stream_index: If stream_index is -1, a default stream is selected, and timestamp is automatically
converted from AV_TIME_BASE units to the stream specific time_base.
• timestamp: Timestamp in AVStream.time_base units or, if no stream is specified, in AV_TIME_BASE
units.
• flags: Flags which select direction and seeking mode.
get_duration
Get movie duration in AV_TIME_BASE units.
SEE ALSO
ffserver(1), the doc/ffserver.conf example, ffmpeg(1), ffplay(1), ffprobe(1), ffmpeg-utils(1),
ffmpeg-scaler(1), ffmpeg-resampler(1), ffmpeg-codecs(1), ffmpeg-bitstream-filters(1), ffmpeg-formats(1),
ffmpeg-devices(1), ffmpeg-protocols(1), ffmpeg-filters(1)
AUTHORS
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.
FFSERVER-ALL(1)