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FFSERVER-ALL(1) FFSERVER-ALL(1)

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 widthx height, 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 "%0 Nd", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0 Nd" 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 "%0 Nd", 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 "%0 Nd", 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=0x CRC, 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 "%0 Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0 Nd" 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 "%0 Nd", 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 "%0 Nd", 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
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 "out N", 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 widthx height 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 -
    
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 widthx height 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: "c chn 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.
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_