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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

       o   Allow/deny access to address.

                   ACL ALLOW <address>
                   ACL DENY <address>

       o   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

       o   Multipart JPEG

                   <Stream test.mjpg>
                   Feed feed1.ffm
                   Format mpjpeg
                   VideoFrameRate 2
                   VideoIntraOnly
                   NoAudio
                   Strict -1
                   </Stream>

       o   Single JPEG

                   <Stream test.jpg>
                   Feed feed1.ffm
                   Format jpeg
                   VideoFrameRate 2
                   VideoIntraOnly
                   VideoSize 352x240
                   NoAudio
                   Strict -1
                   </Stream>

       o   Flash

                   <Stream test.swf>
                   Feed feed1.ffm
                   Format swf
                   VideoFrameRate 2
                   VideoIntraOnly
                   NoAudio
                   </Stream>

       o   ASF compatible

                   <Stream test.asf>
                   Feed feed1.ffm
                   Format asf
                   VideoFrameRate 15
                   VideoSize 352x240
                   VideoBitRate 256
                   VideoBufferSize 40
                   VideoGopSize 30
                   AudioBitRate 64
                   StartSendOnKey
                   </Stream>

       o   MP3 audio

                   <Stream test.mp3>
                   Feed feed1.ffm
                   Format mp2
                   AudioCodec mp3
                   AudioBitRate 64
                   AudioChannels 1
                   AudioSampleRate 44100
                   NoVideo
                   </Stream>

       o   Ogg Vorbis audio

                   <Stream test.ogg>
                   Feed feed1.ffm
                   Metadata title "Stream title"
                   AudioBitRate 64
                   AudioChannels 2
                   AudioSampleRate 44100
                   NoVideo
                   </Stream>

       o   Real with audio only at 32 kbits

                   <Stream test.ra>
                   Feed feed1.ffm
                   Format rm
                   AudioBitRate 32
                   NoVideo
                   </Stream>

       o   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>

       o   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:

       o   ' 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.

       o   A special character is escaped by prefixing it with a \.

       o   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.

       o   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

       o   Escape the string "Crime d'Amour" containing the "'" special
           character:

                   Crime d\'Amour

       o   The string above contains a quote, so the "'" needs to be escaped
           when quoting it:

                   'Crime d'\''Amour'

       o   Include leading or trailing whitespaces using quoting:

                   '  this string starts and ends with whitespaces  '

       o   Escaping and quoting can be mixed together:

                   ' The string '\'string\'' is a string '

       o   To include a literal \ you can use either escaping or quoting:

                   'c:\foo' can be written as c:\\foo

   Date
       The accepted syntax is:

               [(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z]
               now

       If the value is "now" it takes the current time.

       Time is local time unless Z is appended, in which case it is
       interpreted as UTC.  If the year-month-day part is not specified it
       takes the current year-month-day.

   Time duration
       There are two accepted syntaxes for expressing time duration.

               [-][<HH>:]<MM>:<SS>[.<m>...]

       HH expresses the number of hours, MM the number of minutes for a
       maximum of 2 digits, and SS the number of seconds for a maximum of 2
       digits. The m at the end expresses decimal value for SS.

       or

               [-]<S>+[.<m>...]

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

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

       12:03:45
           12 hours, 03 minutes and 45 seconds

       23.189
           23.189 seconds

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

       The following abbreviations are recognized:

       ntsc
           720x480

       pal 720x576

       qntsc
           352x240

       qpal
           352x288

       sntsc
           640x480

       spal
           768x576

       film
           352x240

       ntsc-film
           352x240

       sqcif
           128x96

       qcif
           176x144

       cif 352x288

       4cif
           704x576

       16cif
           1408x1152

       qqvga
           160x120

       qvga
           320x240

       vga 640x480

       svga
           800x600

       xga 1024x768

       uxga
           1600x1200

       qxga
           2048x1536

       sxga
           1280x1024

       qsxga
           2560x2048

       hsxga
           5120x4096

       wvga
           852x480

       wxga
           1366x768

       wsxga
           1600x1024

       wuxga
           1920x1200

       woxga
           2560x1600

       wqsxga
           3200x2048

       wquxga
           3840x2400

       whsxga
           6400x4096

       whuxga
           7680x4800

       cga 320x200

       ega 640x350

       hd480
           852x480

       hd720
           1280x720

       hd1080
           1920x1080

       2k  2048x1080

       2kflat
           1998x1080

       2kscope
           2048x858

       4k  4096x2160

       4kflat
           3996x2160

       4kscope
           4096x1716

       nhd 640x360

       hqvga
           240x160

       wqvga
           400x240

       fwqvga
           432x240

       hvga
           480x320

       qhd 960x540

       2kdci
           2048x1080

       4kdci
           4096x2160

       uhd2160
           3840x2160

       uhd4320
           7680x4320

   Video rate
       Specify the frame rate of a video, expressed as the number of frames
       generated per second. It has to be a string in the format
       frame_rate_num/frame_rate_den, an integer number, a float number or a
       valid video frame rate abbreviation.

       The following abbreviations are recognized:

       ntsc
           30000/1001

       pal 25/1

       qntsc
           30000/1001

       qpal
           25/1

       sntsc
           30000/1001

       spal
           25/1

       film
           24/1

       ntsc-film
           24000/1001

   Ratio
       A ratio can be expressed as an expression, or in the form
       numerator:denominator.

       Note that a ratio with infinite (1/0) or negative value is considered
       valid, so you should check on the returned value if you want to exclude
       those values.

       The undefined value can be expressed using the "0:0" string.

   Color
       It can be the name of a color as defined below (case insensitive match)
       or a "[0x|#]RRGGBB[AA]" sequence, possibly followed by @ and a string
       representing the alpha component.

       The alpha component may be a string composed by "0x" followed by an
       hexadecimal number or a decimal number between 0.0 and 1.0, which
       represents the opacity value (0x00 or 0.0 means completely transparent,
       0xff or 1.0 completely opaque). If the alpha component is not specified
       then 0xff is assumed.

       The string random will result in a random color.

       The following names of colors are recognized:

       AliceBlue
           0xF0F8FF

       AntiqueWhite
           0xFAEBD7

       Aqua
           0x00FFFF

       Aquamarine
           0x7FFFD4

       Azure
           0xF0FFFF

       Beige
           0xF5F5DC

       Bisque
           0xFFE4C4

       Black
           0x000000

       BlanchedAlmond
           0xFFEBCD

       Blue
           0x0000FF

       BlueViolet
           0x8A2BE2

       Brown
           0xA52A2A

       BurlyWood
           0xDEB887

       CadetBlue
           0x5F9EA0

       Chartreuse
           0x7FFF00

       Chocolate
           0xD2691E

       Coral
           0xFF7F50

       CornflowerBlue
           0x6495ED

       Cornsilk
           0xFFF8DC

       Crimson
           0xDC143C

       Cyan
           0x00FFFF

       DarkBlue
           0x00008B

       DarkCyan
           0x008B8B

       DarkGoldenRod
           0xB8860B

       DarkGray
           0xA9A9A9

       DarkGreen
           0x006400

       DarkKhaki
           0xBDB76B

       DarkMagenta
           0x8B008B

       DarkOliveGreen
           0x556B2F

       Darkorange
           0xFF8C00

       DarkOrchid
           0x9932CC

       DarkRed
           0x8B0000

       DarkSalmon
           0xE9967A

       DarkSeaGreen
           0x8FBC8F

       DarkSlateBlue
           0x483D8B

       DarkSlateGray
           0x2F4F4F

       DarkTurquoise
           0x00CED1

       DarkViolet
           0x9400D3

       DeepPink
           0xFF1493

       DeepSkyBlue
           0x00BFFF

       DimGray
           0x696969

       DodgerBlue
           0x1E90FF

       FireBrick
           0xB22222

       FloralWhite
           0xFFFAF0

       ForestGreen
           0x228B22

       Fuchsia
           0xFF00FF

       Gainsboro
           0xDCDCDC

       GhostWhite
           0xF8F8FF

       Gold
           0xFFD700

       GoldenRod
           0xDAA520

       Gray
           0x808080

       Green
           0x008000

       GreenYellow
           0xADFF2F

       HoneyDew
           0xF0FFF0

       HotPink
           0xFF69B4

       IndianRed
           0xCD5C5C

       Indigo
           0x4B0082

       Ivory
           0xFFFFF0

       Khaki
           0xF0E68C

       Lavender
           0xE6E6FA

       LavenderBlush
           0xFFF0F5

       LawnGreen
           0x7CFC00

       LemonChiffon
           0xFFFACD

       LightBlue
           0xADD8E6

       LightCoral
           0xF08080

       LightCyan
           0xE0FFFF

       LightGoldenRodYellow
           0xFAFAD2

       LightGreen
           0x90EE90

       LightGrey
           0xD3D3D3

       LightPink
           0xFFB6C1

       LightSalmon
           0xFFA07A

       LightSeaGreen
           0x20B2AA

       LightSkyBlue
           0x87CEFA

       LightSlateGray
           0x778899

       LightSteelBlue
           0xB0C4DE

       LightYellow
           0xFFFFE0

       Lime
           0x00FF00

       LimeGreen
           0x32CD32

       Linen
           0xFAF0E6

       Magenta
           0xFF00FF

       Maroon
           0x800000

       MediumAquaMarine
           0x66CDAA

       MediumBlue
           0x0000CD

       MediumOrchid
           0xBA55D3

       MediumPurple
           0x9370D8

       MediumSeaGreen
           0x3CB371

       MediumSlateBlue
           0x7B68EE

       MediumSpringGreen
           0x00FA9A

       MediumTurquoise
           0x48D1CC

       MediumVioletRed
           0xC71585

       MidnightBlue
           0x191970

       MintCream
           0xF5FFFA

       MistyRose
           0xFFE4E1

       Moccasin
           0xFFE4B5

       NavajoWhite
           0xFFDEAD

       Navy
           0x000080

       OldLace
           0xFDF5E6

       Olive
           0x808000

       OliveDrab
           0x6B8E23

       Orange
           0xFFA500

       OrangeRed
           0xFF4500

       Orchid
           0xDA70D6

       PaleGoldenRod
           0xEEE8AA

       PaleGreen
           0x98FB98

       PaleTurquoise
           0xAFEEEE

       PaleVioletRed
           0xD87093

       PapayaWhip
           0xFFEFD5

       PeachPuff
           0xFFDAB9

       Peru
           0xCD853F

       Pink
           0xFFC0CB

       Plum
           0xDDA0DD

       PowderBlue
           0xB0E0E6

       Purple
           0x800080

       Red 0xFF0000

       RosyBrown
           0xBC8F8F

       RoyalBlue
           0x4169E1

       SaddleBrown
           0x8B4513

       Salmon
           0xFA8072

       SandyBrown
           0xF4A460

       SeaGreen
           0x2E8B57

       SeaShell
           0xFFF5EE

       Sienna
           0xA0522D

       Silver
           0xC0C0C0

       SkyBlue
           0x87CEEB

       SlateBlue
           0x6A5ACD

       SlateGray
           0x708090

       Snow
           0xFFFAFA

       SpringGreen
           0x00FF7F

       SteelBlue
           0x4682B4

       Tan 0xD2B48C

       Teal
           0x008080

       Thistle
           0xD8BFD8

       Tomato
           0xFF6347

       Turquoise
           0x40E0D0

       Violet
           0xEE82EE

       Wheat
           0xF5DEB3

       White
           0xFFFFFF

       WhiteSmoke
           0xF5F5F5

       Yellow
           0xFFFF00

       YellowGreen
           0x9ACD32

   Channel Layout
       A channel layout specifies the spatial disposition of the channels in a
       multi-channel audio stream. To specify a channel layout, FFmpeg makes
       use of a special syntax.

       Individual channels are identified by an id, as given by the table
       below:

       FL  front left

       FR  front right

       FC  front center

       LFE low frequency

       BL  back left

       BR  back right

       FLC front left-of-center

       FRC front right-of-center

       BC  back center

       SL  side left

       SR  side right

       TC  top center

       TFL top front left

       TFC top front center

       TFR top front right

       TBL top back left

       TBC top back center

       TBR top back right

       DL  downmix left

       DR  downmix right

       WL  wide left

       WR  wide right

       SDL surround direct left

       SDR surround direct right

       LFE2
           low frequency 2

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

       mono
           FC

       stereo
           FL+FR

       2.1 FL+FR+LFE

       3.0 FL+FR+FC

       3.0(back)
           FL+FR+BC

       4.0 FL+FR+FC+BC

       quad
           FL+FR+BL+BR

       quad(side)
           FL+FR+SL+SR

       3.1 FL+FR+FC+LFE

       5.0 FL+FR+FC+BL+BR

       5.0(side)
           FL+FR+FC+SL+SR

       4.1 FL+FR+FC+LFE+BC

       5.1 FL+FR+FC+LFE+BL+BR

       5.1(side)
           FL+FR+FC+LFE+SL+SR

       6.0 FL+FR+FC+BC+SL+SR

       6.0(front)
           FL+FR+FLC+FRC+SL+SR

       hexagonal
           FL+FR+FC+BL+BR+BC

       6.1 FL+FR+FC+LFE+BC+SL+SR

       6.1 FL+FR+FC+LFE+BL+BR+BC

       6.1(front)
           FL+FR+LFE+FLC+FRC+SL+SR

       7.0 FL+FR+FC+BL+BR+SL+SR

       7.0(front)
           FL+FR+FC+FLC+FRC+SL+SR

       7.1 FL+FR+FC+LFE+BL+BR+SL+SR

       7.1(wide)
           FL+FR+FC+LFE+BL+BR+FLC+FRC

       7.1(wide-side)
           FL+FR+FC+LFE+FLC+FRC+SL+SR

       octagonal
           FL+FR+FC+BL+BR+BC+SL+SR

       downmix
           DL+DR

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

       o   the name of a standard channel layout (e.g. mono, stereo, 4.0,
           quad, 5.0, etc.)

       o   the name of a single channel (e.g. FL, FR, FC, LFE, etc.)

       o   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.

       o   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.

       o   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

       o   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

       o   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

       o   Set maximum constant quality (VBR) encoding with ffmpeg:

                   ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg

       o   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:

       o   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.

       o   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:

       o   g/gop_size -> GopPicSize

       o   bf/max_b_frames+1 -> GopRefDist

       o   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       o   slices -> NumSlice

       o   refs -> NumRefFrame

       o   b_strategy/b_frame_strategy -> BRefType

       o   cgop/CLOSED_GOP codec flag -> GopOptFlag

       o   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.

       o   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:

       o   g / gop_size

       o   bf / max_b_frames

       o   profile

       o   level

       o   b / bit_rate

       o   maxrate / rc_max_rate

       o   bufsize / rc_buffer_size

       o   rc_init_occupancy / rc_initial_buffer_occupancy

       o   compression_level

           Speed / quality tradeoff: higher values are faster / worse quality.

       o   q / global_quality

           Size / quality tradeoff: higher values are smaller / worse quality.

       o   qmin (only: qmax is not supported)

       o   i_qfactor / i_quant_factor

       o   i_qoffset / i_quant_offset

       o   b_qfactor / b_quant_factor

       o   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

       o   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'

       o   Allow for input format auto-probing, use safe filenames and set the
           duration of the first file:

                   ffconcat version 1.0

                   file file-1.wav
                   duration 20.0

                   file subdir/file-2.wav

   flv, live_flv
       Adobe Flash Video Format demuxer.

       This demuxer is used to demux FLV files and RTMP network streams. In
       case of live network streams, if you force format, you may use live_flv
       option instead of flv to survive timestamp discontinuities.

               ffmpeg -f flv -i myfile.flv ...
               ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....

       -flv_metadata bool
           Allocate the streams according to the onMetaData array content.

   gif
       Animated GIF demuxer.

       It accepts the following options:

       min_delay
           Set the minimum valid delay between frames in hundredths of
           seconds.  Range is 0 to 6000. Default value is 2.

       max_gif_delay
           Set the maximum valid delay between frames in hundredth of seconds.
           Range is 0 to 65535. Default value is 65535 (nearly eleven
           minutes), the maximum value allowed by the specification.

       default_delay
           Set the default delay between frames in hundredths of seconds.
           Range is 0 to 6000. Default value is 10.

       ignore_loop
           GIF files can contain information to loop a certain number of times
           (or infinitely). If ignore_loop is set to 1, then the loop setting
           from the input will be ignored and looping will not occur. If set
           to 0, then looping will occur and will cycle the number of times
           according to the GIF. Default value is 1.

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

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

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

   hls
       HLS demuxer

       It accepts the following options:

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

       allowed_extensions
           ',' separated list of file extensions that hls is allowed to
           access.

       max_reload
           Maximum number of times a insufficient list is attempted to be
           reloaded.  Default value is 1000.

   image2
       Image file demuxer.

       This demuxer reads from a list of image files specified by a pattern.
       The syntax and meaning of the pattern is specified by the option
       pattern_type.

       The pattern may contain a suffix which is used to automatically
       determine the format of the images contained in the files.

       The size, the pixel format, and the format of each image must be the
       same for all the files in the sequence.

       This demuxer accepts the following options:

       framerate
           Set the frame rate for the video stream. It defaults to 25.

       loop
           If set to 1, loop over the input. Default value is 0.

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

           pattern_type accepts one of the following values.

           none
               Disable pattern matching, therefore the video will only contain
               the specified image. You should use this option if you do not
               want to create sequences from multiple images and your
               filenames may contain special pattern characters.

           sequence
               Select a sequence pattern type, used to specify a sequence of
               files indexed by sequential numbers.

               A sequence pattern may contain the string "%d" or "%0Nd", which
               specifies the position of the characters representing a
               sequential number in each filename matched by the pattern. If
               the form "%d0Nd" is used, the string representing the number in
               each filename is 0-padded and N is the total number of 0-padded
               digits representing the number. The literal character '%' can
               be specified in the pattern with the string "%%".

               If the sequence pattern contains "%d" or "%0Nd", the first
               filename of the file list specified by the pattern must contain
               a number inclusively contained between start_number and
               start_number+start_number_range-1, and all the following
               numbers must be sequential.

               For example the pattern "img-%03d.bmp" will match a sequence of
               filenames of the form img-001.bmp, img-002.bmp, ...,
               img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a
               sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg,
               ..., i%m%g-10.jpg, etc.

               Note that the pattern must not necessarily contain "%d" or
               "%0Nd", for example to convert a single image file img.jpeg you
               can employ the command:

                       ffmpeg -i img.jpeg img.png

           glob
               Select a glob wildcard pattern type.

               The pattern is interpreted like a "glob()" pattern. This is
               only selectable if libavformat was compiled with globbing
               support.

           glob_sequence (deprecated, will be removed)
               Select a mixed glob wildcard/sequence pattern.

               If your version of libavformat was compiled with globbing
               support, and the provided pattern contains at least one glob
               meta character among "%*?[]{}" that is preceded by an unescaped
               "%", the pattern is interpreted like a "glob()" pattern,
               otherwise it is interpreted like a sequence pattern.

               All glob special characters "%*?[]{}" must be prefixed with
               "%". To escape a literal "%" you shall use "%%".

               For example the pattern "foo-%*.jpeg" will match all the
               filenames prefixed by "foo-" and terminating with ".jpeg", and
               "foo-%?%?%?.jpeg" will match all the filenames prefixed with
               "foo-", followed by a sequence of three characters, and
               terminating with ".jpeg".

               This pattern type is deprecated in favor of glob and sequence.

           Default value is glob_sequence.

       pixel_format
           Set the pixel format of the images to read. If not specified the
           pixel format is guessed from the first image file in the sequence.

       start_number
           Set the index of the file matched by the image file pattern to
           start to read from. Default value is 0.

       start_number_range
           Set the index interval range to check when looking for the first
           image file in the sequence, starting from start_number. Default
           value is 5.

       ts_from_file
           If set to 1, will set frame timestamp to modification time of image
           file. Note that monotonity of timestamps is not provided: images go
           in the same order as without this option. Default value is 0.  If
           set to 2, will set frame timestamp to the modification time of the
           image file in nanosecond precision.

       video_size
           Set the video size of the images to read. If not specified the
           video size is guessed from the first image file in the sequence.

       Examples

       o   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

       o   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

       o   Read images matching the "*.png" glob pattern , that is all the
           files terminating with the ".png" suffix:

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

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

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

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

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

   libopenmpt
       libopenmpt based module demuxer

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

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

       It accepts the following options:

       subsong
           Set the subsong index. This can be either  'all', 'auto', or the
           index of the subsong. Subsong indexes start at 0. The default is
           'auto'.

           The default value is to let libopenmpt choose.

       layout
           Set the channel layout. Valid values are 1, 2, and 4 channel
           layouts.  The default value is STEREO.

       sample_rate
           Set the sample rate for libopenmpt to output.  Range is from 1000
           to INT_MAX. The value default is 48000.

   mov/mp4/3gp/QuickTime
       QuickTime / MP4 demuxer.

       This demuxer accepts the following options:

       enable_drefs
           Enable loading of external tracks, disabled by default.  Enabling
           this can theoretically leak information in some use cases.

       use_absolute_path
           Allows loading of external tracks via absolute paths, disabled by
           default.  Enabling this poses a security risk. It should only be
           enabled if the source is known to be non malicious.

   mpegts
       MPEG-2 transport stream demuxer.

       This demuxer accepts the following options:

       resync_size
           Set size limit for looking up a new synchronization. Default value
           is 65536.

       fix_teletext_pts
           Override teletext packet PTS and DTS values with the timestamps
           calculated from the PCR of the first program which the teletext
           stream is part of and is not discarded. Default value is 1, set
           this option to 0 if you want your teletext packet PTS and DTS
           values untouched.

       ts_packetsize
           Output option carrying the raw packet size in bytes.  Show the
           detected raw packet size, cannot be set by the user.

       scan_all_pmts
           Scan and combine all PMTs. The value is an integer with value from
           -1 to 1 (-1 means automatic setting, 1 means enabled, 0 means
           disabled). Default value is -1.

   mpjpeg
       MJPEG encapsulated in multi-part MIME demuxer.

       This demuxer allows reading of MJPEG, where each frame is represented
       as a part of multipart/x-mixed-replace stream.

       strict_mime_boundary
           Default implementation applies a relaxed standard to multi-part
           MIME boundary detection, to prevent regression with numerous
           existing endpoints not generating a proper MIME MJPEG stream.
           Turning this option on by setting it to 1 will result in a stricter
           check of the boundary value.

   rawvideo
       Raw video demuxer.

       This demuxer allows one to read raw video data. Since there is no
       header specifying the assumed video parameters, the user must specify
       them in order to be able to decode the data correctly.

       This demuxer accepts the following options:

       framerate
           Set input video frame rate. Default value is 25.

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

       video_size
           Set the input video size. This value must be specified explicitly.

       For example to read a rawvideo file input.raw with ffplay, assuming a
       pixel format of "rgb24", a video size of "320x240", and a frame rate of
       10 images per second, use the command:

               ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

   sbg
       SBaGen script demuxer.

       This demuxer reads the script language used by SBaGen
       <http://uazu.net/sbagen/> to generate binaural beats sessions. A SBG
       script looks like that:

               -SE
               a: 300-2.5/3 440+4.5/0
               b: 300-2.5/0 440+4.5/3
               off: -
               NOW      == a
               +0:07:00 == b
               +0:14:00 == a
               +0:21:00 == b
               +0:30:00    off

       A SBG script can mix absolute and relative timestamps. If the script
       uses either only absolute timestamps (including the script start time)
       or only relative ones, then its layout is fixed, and the conversion is
       straightforward. On the other hand, if the script mixes both kind of
       timestamps, then the NOW reference for relative timestamps will be
       taken from the current time of day at the time the script is read, and
       the script layout will be frozen according to that reference. That
       means that if the script is directly played, the actual times will
       match the absolute timestamps up to the sound controller's clock
       accuracy, but if the user somehow pauses the playback or seeks, all
       times will be shifted accordingly.

   tedcaptions
       JSON captions used for <http://www.ted.com/>.

       TED does not provide links to the captions, but they can be guessed
       from the page. The file tools/bookmarklets.html from the FFmpeg source
       tree contains a bookmarklet to expose them.

       This demuxer accepts the following option:

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

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

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


MUXERS

       Muxers are configured elements in FFmpeg which allow writing multimedia
       streams to a particular type of file.

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

       You can disable all the muxers with the configure option
       "--disable-muxers" and selectively enable / disable single muxers with
       the options "--enable-muxer=MUXER" / "--disable-muxer=MUXER".

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

       A description of some of the currently available muxers follows.

   aiff
       Audio Interchange File Format muxer.

       Options

       It accepts the following options:

       write_id3v2
           Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).

       id3v2_version
           Select ID3v2 version to write. Currently only version 3 and 4 (aka.
           ID3v2.3 and ID3v2.4) are supported. The default is version 4.

   asf
       Advanced Systems Format muxer.

       Note that Windows Media Audio (wma) and Windows Media Video (wmv) use
       this muxer too.

       Options

       It accepts the following options:

       packet_size
           Set the muxer packet size. By tuning this setting you may reduce
           data fragmentation or muxer overhead depending on your source.
           Default value is 3200, minimum is 100, maximum is 64k.

   avi
       Audio Video Interleaved muxer.

       Options

       It accepts the following options:

       reserve_index_space
           Reserve the specified amount of bytes for the OpenDML master index
           of each stream within the file header. By default additional master
           indexes are embedded within the data packets if there is no space
           left in the first master index and are linked together as a chain
           of indexes. This index structure can cause problems for some use
           cases, e.g. third-party software strictly relying on the OpenDML
           index specification or when file seeking is slow. Reserving enough
           index space in the file header avoids these problems.

           The required index space depends on the output file size and should
           be about 16 bytes per gigabyte. When this option is omitted or set
           to zero the necessary index space is guessed.

       write_channel_mask
           Write the channel layout mask into the audio stream header.

           This option is enabled by default. Disabling the channel mask can
           be useful in specific scenarios, e.g. when merging multiple audio
           streams into one for compatibility with software that only supports
           a single audio stream in AVI (see the "amerge" section in the
           ffmpeg-filters manual).

   chromaprint
       Chromaprint fingerprinter

       This muxer feeds audio data to the Chromaprint library, which generates
       a fingerprint for the provided audio data. It takes a single signed
       native-endian 16-bit raw audio stream.

       Options

       silence_threshold
           Threshold for detecting silence, ranges from 0 to 32767. -1 for
           default (required for use with the AcoustID service).

       algorithm
           Algorithm index to fingerprint with.

       fp_format
           Format to output the fingerprint as. Accepts the following options:

           raw Binary raw fingerprint

           compressed
               Binary compressed fingerprint

           base64
               Base64 compressed fingerprint

   crc
       CRC (Cyclic Redundancy Check) testing format.

       This muxer computes and prints the Adler-32 CRC of all the input audio
       and video frames. By default audio frames are converted to signed
       16-bit raw audio and video frames to raw video before computing the
       CRC.

       The output of the muxer consists of a single line of the form:
       CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits
       containing the CRC for all the decoded input frames.

       See also the framecrc muxer.

       Examples

       For example to compute the CRC of the input, and store it in the file
       out.crc:

               ffmpeg -i INPUT -f crc out.crc

       You can print the CRC to stdout with the command:

               ffmpeg -i INPUT -f crc -

       You can select the output format of each frame with ffmpeg by
       specifying the audio and video codec and format. For example to compute
       the CRC of the input audio converted to PCM unsigned 8-bit and the
       input video converted to MPEG-2 video, use the command:

               ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -

   flv
       Adobe Flash Video Format muxer.

       This muxer accepts the following options:

       flvflags flags
           Possible values:

           aac_seq_header_detect
               Place AAC sequence header based on audio stream data.

           no_sequence_end
               Disable sequence end tag.

           no_metadata
               Disable metadata tag.

           no_duration_filesize
               Disable duration and filesize in metadata when they are equal
               to zero at the end of stream. (Be used to non-seekable living
               stream).

           add_keyframe_index
               Used to facilitate seeking; particularly for HTTP pseudo
               streaming.

   dash
       Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments
       and manifest files according to the MPEG-DASH standard ISO/IEC
       23009-1:2014.

       For more information see:

       o   ISO DASH Specification:
           <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

       o   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:

       o   Size cannot exceed 256 pixels in any dimension

       o   Only BMP and PNG images can be stored

       o   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

       o   If a BMP image is used, it must use the BITMAPINFOHEADER DIB header

       o   If a PNG image is used, it must use the rgba pixel format

   image2
       Image file muxer.

       The image file muxer writes video frames to image files.

       The output filenames are specified by a pattern, which can be used to
       produce sequentially numbered series of files.  The pattern may contain
       the string "%d" or "%0Nd", this string specifies the position of the
       characters representing a numbering in the filenames. If the form
       "%0Nd" is used, the string representing the number in each filename is
       0-padded to N digits. The literal character '%' can be specified in the
       pattern with the string "%%".

       If the pattern contains "%d" or "%0Nd", the first filename of the file
       list specified will contain the number 1, all the following numbers
       will be sequential.

       The pattern may contain a suffix which is used to automatically
       determine the format of the image files to write.

       For example the pattern "img-%03d.bmp" will specify a sequence of
       filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.
       The pattern "img%%-%d.jpg" will specify a sequence of filenames of the
       form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.

       Examples

       The following example shows how to use ffmpeg for creating a sequence
       of files img-001.jpeg, img-002.jpeg, ..., taking one image every second
       from the input video:

               ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'

       Note that with ffmpeg, if the format is not specified with the "-f"
       option and the output filename specifies an image file format, the
       image2 muxer is automatically selected, so the previous command can be
       written as:

               ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'

       Note also that the pattern must not necessarily contain "%d" or "%0Nd",
       for example to create a single image file img.jpeg from the start of
       the input video you can employ the command:

               ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg

       The strftime option allows you to expand the filename with date and
       time information. Check the documentation of the "strftime()" function
       for the syntax.

       For example to generate image files from the "strftime()"
       "%Y-%m-%d_%H-%M-%S" pattern, the following ffmpeg command can be used:

               ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"

       Options

       start_number
           Start the sequence from the specified number. Default value is 1.

       update
           If set to 1, the filename will always be interpreted as just a
           filename, not a pattern, and the corresponding file will be
           continuously overwritten with new images. Default value is 0.

       strftime
           If set to 1, expand the filename with date and time information
           from "strftime()". Default value is 0.

       The image muxer supports the .Y.U.V image file format. This format is
       special in that that each image frame consists of three files, for each
       of the YUV420P components. To read or write this image file format,
       specify the name of the '.Y' file. The muxer will automatically open
       the '.U' and '.V' files as required.

   matroska
       Matroska container muxer.

       This muxer implements the matroska and webm container specs.

       Metadata

       The recognized metadata settings in this muxer are:

       title
           Set title name provided to a single track.

       language
           Specify the language of the track in the Matroska languages form.

           The language can be either the 3 letters bibliographic ISO-639-2
           (ISO 639-2/B) form (like "fre" for French), or a language code
           mixed with a country code for specialities in languages (like "fre-
           ca" for Canadian French).

       stereo_mode
           Set stereo 3D video layout of two views in a single video track.

           The following values are recognized:

           mono
               video is not stereo

           left_right
               Both views are arranged side by side, Left-eye view is on the
               left

           bottom_top
               Both views are arranged in top-bottom orientation, Left-eye
               view is at bottom

           top_bottom
               Both views are arranged in top-bottom orientation, Left-eye
               view is on top

           checkerboard_rl
               Each view is arranged in a checkerboard interleaved pattern,
               Left-eye view being first

           checkerboard_lr
               Each view is arranged in a checkerboard interleaved pattern,
               Right-eye view being first

           row_interleaved_rl
               Each view is constituted by a row based interleaving, Right-eye
               view is first row

           row_interleaved_lr
               Each view is constituted by a row based interleaving, Left-eye
               view is first row

           col_interleaved_rl
               Both views are arranged in a column based interleaving manner,
               Right-eye view is first column

           col_interleaved_lr
               Both views are arranged in a column based interleaving manner,
               Left-eye view is first column

           anaglyph_cyan_red
               All frames are in anaglyph format viewable through red-cyan
               filters

           right_left
               Both views are arranged side by side, Right-eye view is on the
               left

           anaglyph_green_magenta
               All frames are in anaglyph format viewable through green-
               magenta filters

           block_lr
               Both eyes laced in one Block, Left-eye view is first

           block_rl
               Both eyes laced in one Block, Right-eye view is first

       For example a 3D WebM clip can be created using the following command
       line:

               ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm

       Options

       This muxer supports the following options:

       reserve_index_space
           By default, this muxer writes the index for seeking (called cues in
           Matroska terms) at the end of the file, because it cannot know in
           advance how much space to leave for the index at the beginning of
           the file. However for some use cases -- e.g.  streaming where
           seeking is possible but slow -- it is useful to put the index at
           the beginning of the file.

           If this option is set to a non-zero value, the muxer will reserve a
           given amount of space in the file header and then try to write the
           cues there when the muxing finishes. If the available space does
           not suffice, muxing will fail. A safe size for most use cases
           should be about 50kB per hour of video.

           Note that cues are only written if the output is seekable and this
           option will have no effect if it is not.

   md5
       MD5 testing format.

       This is a variant of the hash muxer. Unlike that muxer, it defaults to
       using the MD5 hash function.

       Examples

       To compute the MD5 hash of the input converted to raw audio and video,
       and store it in the file out.md5:

               ffmpeg -i INPUT -f md5 out.md5

       You can print the MD5 to stdout with the command:

               ffmpeg -i INPUT -f md5 -

       See also the hash and framemd5 muxers.

   mov, mp4, ismv
       MOV/MP4/ISMV (Smooth Streaming) muxer.

       The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file
       has all the metadata about all packets stored in one location (written
       at the end of the file, it can be moved to the start for better
       playback by adding faststart to the movflags, or using the qt-faststart
       tool). A fragmented file consists of a number of fragments, where
       packets and metadata about these packets are stored together. Writing a
       fragmented file has the advantage that the file is decodable even if
       the writing is interrupted (while a normal MOV/MP4 is undecodable if it
       is not properly finished), and it requires less memory when writing
       very long files (since writing normal MOV/MP4 files stores info about
       every single packet in memory until the file is closed). The downside
       is that it is less compatible with other applications.

       Options

       Fragmentation is enabled by setting one of the AVOptions that define
       how to cut the file into fragments:

       -moov_size bytes
           Reserves space for the moov atom at the beginning of the file
           instead of placing the moov atom at the end. If the space reserved
           is insufficient, muxing will fail.

       -movflags frag_keyframe
           Start a new fragment at each video keyframe.

       -frag_duration duration
           Create fragments that are duration microseconds long.

       -frag_size size
           Create fragments that contain up to size bytes of payload data.

       -movflags frag_custom
           Allow the caller to manually choose when to cut fragments, by
           calling "av_write_frame(ctx, NULL)" to write a fragment with the
           packets written so far. (This is only useful with other
           applications integrating libavformat, not from ffmpeg.)

       -min_frag_duration duration
           Don't create fragments that are shorter than duration microseconds
           long.

       If more than one condition is specified, fragments are cut when one of
       the specified conditions is fulfilled. The exception to this is
       "-min_frag_duration", which has to be fulfilled for any of the other
       conditions to apply.

       Additionally, the way the output file is written can be adjusted
       through a few other options:

       -movflags empty_moov
           Write an initial moov atom directly at the start of the file,
           without describing any samples in it. Generally, an mdat/moov pair
           is written at the start of the file, as a normal MOV/MP4 file,
           containing only a short portion of the file. With this option set,
           there is no initial mdat atom, and the moov atom only describes the
           tracks but has a zero duration.

           This option is implicitly set when writing ismv (Smooth Streaming)
           files.

       -movflags separate_moof
           Write a separate moof (movie fragment) atom for each track.
           Normally, packets for all tracks are written in a moof atom (which
           is slightly more efficient), but with this option set, the muxer
           writes one moof/mdat pair for each track, making it easier to
           separate tracks.

           This option is implicitly set when writing ismv (Smooth Streaming)
           files.

       -movflags faststart
           Run a second pass moving the index (moov atom) to the beginning of
           the file.  This operation can take a while, and will not work in
           various situations such as fragmented output, thus it is not
           enabled by default.

       -movflags rtphint
           Add RTP hinting tracks to the output file.

       -movflags disable_chpl
           Disable Nero chapter markers (chpl atom).  Normally, both Nero
           chapters and a QuickTime chapter track are written to the file.
           With this option set, only the QuickTime chapter track will be
           written. Nero chapters can cause failures when the file is
           reprocessed with certain tagging programs, like mp3Tag 2.61a and
           iTunes 11.3, most likely other versions are affected as well.

       -movflags omit_tfhd_offset
           Do not write any absolute base_data_offset in tfhd atoms. This
           avoids tying fragments to absolute byte positions in the
           file/streams.

       -movflags default_base_moof
           Similarly to the omit_tfhd_offset, this flag avoids writing the
           absolute base_data_offset field in tfhd atoms, but does so by using
           the new default-base-is-moof flag instead. This flag is new from
           14496-12:2012. This may make the fragments easier to parse in
           certain circumstances (avoiding basing track fragment location
           calculations on the implicit end of the previous track fragment).

       -write_tmcd
           Specify "on" to force writing a timecode track, "off" to disable it
           and "auto" to write a timecode track only for mov and mp4 output
           (default).

       -movflags negative_cts_offsets
           Enables utilization of version 1 of the CTTS box, in which the CTS
           offsets can be negative. This enables the initial sample to have
           DTS/CTS of zero, and reduces the need for edit lists for some cases
           such as video tracks with B-frames. Additionally, eases conformance
           with the DASH-IF interoperability guidelines.

       Example

       Smooth Streaming content can be pushed in real time to a publishing
       point on IIS with this muxer. Example:

               ffmpeg -re <<normal input/transcoding options>> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)

       Audible AAX

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

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

   mp3
       The MP3 muxer writes a raw MP3 stream with the following optional
       features:

       o   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.

       o   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.

       o   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

       o   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

       o   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

       o   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

       o   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.

       o   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

       o   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

       o   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,

       o   output can be transparently restarted with configurable delay
           between retries based on real time or time of the processed stream.

       o   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

       o   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

       o   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/"

       o   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/"

       o   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"

       o   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:

       o   WebM DASH Specification:
           <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

       o   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.

       o   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

       o   Watch a stream tunneled over HTTP:

                   ffplay -rtsp_transport http rtsp://server/video.mp4

       o   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

       o   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

       o   Use ffmpeg to stream over UDP to a remote endpoint:

                   ffmpeg -i <input> -f <format> udp://<hostname>:<port>

       o   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

       o   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

       o   Print the list of AVFoundation supported devices and exit:

                   $ ffmpeg -f avfoundation -list_devices true -i ""

       o   Record video from video device 0 and audio from audio device 0 into
           out.avi:

                   $ ffmpeg -f avfoundation -i "0:0" out.avi

       o   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

       o   Record video from the system default video device using the pixel
           format bgr0 and do not record any audio into out.avi:

                   $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

   bktr
       BSD video input device.

       Options

       framerate
           Set the frame rate.

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

       standard
           Available values are:

           pal
           ntsc
           secam
           paln
           palm
           ntscj

   decklink
       The decklink input device provides capture capabilities for Blackmagic
       DeckLink devices.

       To enable this input device, you need the Blackmagic DeckLink SDK and
       you need to configure with the appropriate "--extra-cflags" and
       "--extra-ldflags".  On Windows, you need to run the IDL files through
       widl.

       DeckLink is very picky about the formats it supports. Pixel format of
       the input can be set with raw_format.  Framerate and video size must be
       determined for your device with -list_formats 1. Audio sample rate is
       always 48 kHz and the number of channels can be 2, 8 or 16. Note that
       all audio channels are bundled in one single audio track.

       Options

       list_devices
           If set to true, print a list of devices and exit.  Defaults to
           false.

       list_formats
           If set to true, print a list of supported formats and exit.
           Defaults to false.

       format_code <FourCC>
           This sets the input video format to the format given by the FourCC.
           To see the supported values of your device(s) use list_formats.
           Note that there is a FourCC 'pal ' that can also be used as pal (3
           letters).

       bm_v210
           This is a deprecated option, you can use raw_format instead.  If
           set to 1, video is captured in 10 bit v210 instead of uyvy422. Not
           all Blackmagic devices support this option.

       raw_format
           Set the pixel format of the captured video.  Available values are:

           uyvy422
           yuv422p10
           argb
           bgra
           rgb10
       teletext_lines
           If set to nonzero, an additional teletext stream will be captured
           from the vertical ancillary data. Both SD PAL (576i) and HD (1080i
           or 1080p) sources are supported. In case of HD sources, OP47
           packets are decoded.

           This option is a bitmask of the SD PAL VBI lines captured,
           specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB
           in the mask. Selected lines which do not contain teletext
           information will be ignored. You can use the special all constant
           to select all possible lines, or standard to skip lines 6, 318 and
           319, which are not compatible with all receivers.

           For SD sources, ffmpeg needs to be compiled with
           "--enable-libzvbi". For HD sources, on older (pre-4K) DeckLink card
           models you have to capture in 10 bit mode.

       channels
           Defines number of audio channels to capture. Must be 2, 8 or 16.
           Defaults to 2.

       duplex_mode
           Sets the decklink device duplex mode. Must be unset, half or full.
           Defaults to unset.

       video_input
           Sets the video input source. Must be unset, sdi, hdmi, optical_sdi,
           component, composite or s_video.  Defaults to unset.

       audio_input
           Sets the audio input source. Must be unset, embedded, aes_ebu,
           analog, analog_xlr, analog_rca or microphone. Defaults to unset.

       video_pts
           Sets the video packet timestamp source. Must be video, audio,
           reference or wallclock. Defaults to video.

       audio_pts
           Sets the audio packet timestamp source. Must be video, audio,
           reference or wallclock. Defaults to audio.

       draw_bars
           If set to true, color bars are drawn in the event of a signal loss.
           Defaults to true.

       queue_size
           Sets maximum input buffer size in bytes. If the buffering reaches
           this value, incoming frames will be dropped.  Defaults to
           1073741824.

       Examples

       o   List input devices:

                   ffmpeg -f decklink -list_devices 1 -i dummy

       o   List supported formats:

                   ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'

       o   Capture video clip at 1080i50:

                   ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi

       o   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

       o   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

       o   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

       o   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

       o   List input devices:

                   ffmpeg -f libndi_newtek -find_sources 1 -i dummy

       o   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

       o   Print the list of DirectShow supported devices and exit:

                   $ ffmpeg -list_devices true -f dshow -i dummy

       o   Open video device Camera:

                   $ ffmpeg -f dshow -i video="Camera"

       o   Open second video device with name Camera:

                   $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"

       o   Open video device Camera and audio device Microphone:

                   $ ffmpeg -f dshow -i video="Camera":audio="Microphone"

       o   Print the list of supported options in selected device and exit:

                   $ ffmpeg -list_options true -f dshow -i video="Camera"

       o   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"

       o   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

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

                   ffplay -f iec61883 -i auto

       o   Grab and record the input of a FireWire DV/HDV device, using a
           packet buffer of 100000 packets if the source is HDV.

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

   jack
       JACK input device.

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

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

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

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

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

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

               # Create a JACK writable client with name "ffmpeg".
               $ ffmpeg -f jack -i ffmpeg -y out.wav

               # Start the sample jack_metro readable client.
               $ jack_metro -b 120 -d 0.2 -f 4000

               # List the current JACK clients.
               $ jack_lsp -c
               system:capture_1
               system:capture_2
               system:playback_1
               system:playback_2
               ffmpeg:input_1
               metro:120_bpm

               # Connect metro to the ffmpeg writable client.
               $ jack_connect metro:120_bpm ffmpeg:input_1

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

       Options

       channels
           Set the number of channels. Default is 2.

   lavfi
       Libavfilter input virtual device.

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

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

       Options

       graph
           Specify the filtergraph to use as input. Each video open output
           must be labelled by a unique string of the form "outN", where N is
           a number starting from 0 corresponding to the mapped input stream
           generated by the device.  The first unlabelled output is
           automatically assigned to the "out0" label, but all the others need
           to be specified explicitly.

           The suffix "+subcc" can be appended to the output label to create
           an extra stream with the closed captions packets attached to that
           output (experimental; only for EIA-608 / CEA-708 for now).  The
           subcc streams are created after all the normal streams, in the
           order of the corresponding stream.  For example, if there is
           "out19+subcc", "out7+subcc" and up to "out42", the stream #43 is
           subcc for stream #7 and stream #44 is subcc for stream #19.

           If not specified defaults to the filename specified for the input
           device.

       graph_file
           Set the filename of the filtergraph to be read and sent to the
           other filters. Syntax of the filtergraph is the same as the one
           specified by the option graph.

       dumpgraph
           Dump graph to stderr.

       Examples

       o   Create a color video stream and play it back with ffplay:

                   ffplay -f lavfi -graph "color=c=pink [out0]" dummy

       o   As the previous example, but use filename for specifying the graph
           description, and omit the "out0" label:

                   ffplay -f lavfi color=c=pink

       o   Create three different video test filtered sources and play them:

                   ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

       o   Read an audio stream from a file using the amovie source and play
           it back with ffplay:

                   ffplay -f lavfi "amovie=test.wav"

       o   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]"

       o   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:

       o   List supported formats for a video4linux2 device:

                   ffplay -f video4linux2 -list_formats all /dev/video0

       o   Grab and show the input of a video4linux2 device:

                   ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

       o   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

       o   Play a file on default ALSA device:

                   ffmpeg -i INPUT -f alsa default

       o   Play a file on soundcard 1, audio device 7:

                   ffmpeg -i INPUT -f alsa hw:1,7

   caca
       CACA output device.

       This output device allows one to show a video stream in CACA window.
       Only one CACA window is allowed per application, so you can have only
       one instance of this output device in an application.

       To enable this output device you need to configure FFmpeg with
       "--enable-libcaca".  libcaca is a graphics library that outputs text
       instead of pixels.

       For more information about libcaca, check:
       <http://caca.zoy.org/wiki/libcaca>

       Options

       window_title
           Set the CACA window title, if not specified default to the filename
           specified for the output device.

       window_size
           Set the CACA window size, can be a string of the form widthxheight
           or a video size abbreviation.  If not specified it defaults to the
           size of the input video.

       driver
           Set display driver.

       algorithm
           Set dithering algorithm. Dithering is necessary because the picture
           being rendered has usually far more colours than the available
           palette.  The accepted values are listed with "-list_dither
           algorithms".

       antialias
           Set antialias method. Antialiasing smoothens the rendered image and
           avoids the commonly seen staircase effect.  The accepted values are
           listed with "-list_dither antialiases".

       charset
           Set which characters are going to be used when rendering text.  The
           accepted values are listed with "-list_dither charsets".

       color
           Set color to be used when rendering text.  The accepted values are
           listed with "-list_dither colors".

       list_drivers
           If set to true, print a list of available drivers and exit.

       list_dither
           List available dither options related to the argument.  The
           argument must be one of "algorithms", "antialiases", "charsets",
           "colors".

       Examples

       o   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 -

       o   Show the list of available drivers and exit:

                   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -

       o   Show the list of available dither colors and exit:

                   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -

   decklink
       The decklink output device provides playback capabilities for
       Blackmagic DeckLink devices.

       To enable this output device, you need the Blackmagic DeckLink SDK and
       you need to configure with the appropriate "--extra-cflags" and
       "--extra-ldflags".  On Windows, you need to run the IDL files through
       widl.

       DeckLink is very picky about the formats it supports. Pixel format is
       always uyvy422, framerate, field order and video size must be
       determined for your device with -list_formats 1. Audio sample rate is
       always 48 kHz.

       Options

       list_devices
           If set to true, print a list of devices and exit.  Defaults to
           false.

       list_formats
           If set to true, print a list of supported formats and exit.
           Defaults to false.

       preroll
           Amount of time to preroll video in seconds.  Defaults to 0.5.

       Examples

       o   List output devices:

                   ffmpeg -i test.avi -f decklink -list_devices 1 dummy

       o   List supported formats:

                   ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'

       o   Play video clip:

                   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'

       o   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

       o   Play video clip:

                   ffmpeg -i "udp://@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1

   fbdev
       Linux framebuffer output device.

       The Linux framebuffer is a graphic hardware-independent abstraction
       layer to show graphics on a computer monitor, typically on the console.
       It is accessed through a file device node, usually /dev/fb0.

       For more detailed information read the file
       Documentation/fb/framebuffer.txt included in the Linux source tree.

       Options

       xoffset
       yoffset
           Set x/y coordinate of top left corner. Default is 0.

       Examples

       Play a file on framebuffer device /dev/fb0.  Required pixel format
       depends on current framebuffer settings.

               ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0

       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

   opengl
       OpenGL output device.

       To enable this output device you need to configure FFmpeg with
       "--enable-opengl".

       This output device allows one to render to OpenGL context.  Context may
       be provided by application or default SDL window is created.

       When device renders to external context, application must implement
       handlers for following messages: "AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" -
       create OpenGL context on current thread.
       "AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
       "AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers.
       "AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER" - destroy OpenGL context.
       Application is also required to inform a device about current
       resolution by sending "AV_APP_TO_DEV_WINDOW_SIZE" message.

       Options

       background
           Set background color. Black is a default.

       no_window
           Disables default SDL window when set to non-zero value.
           Application must provide OpenGL context and both "window_size_cb"
           and "window_swap_buffers_cb" callbacks when set.

       window_title
           Set the SDL window title, if not specified default to the filename
           specified for the output device.  Ignored when no_window is set.

       window_size
           Set preferred window size, can be a string of the form widthxheight
           or a video size abbreviation.  If not specified it defaults to the
           size of the input video, downscaled according to the aspect ratio.
           Mostly usable when no_window is not set.

       Examples

       Play a file on SDL window using OpenGL rendering:

               ffmpeg  -i INPUT -f opengl "window title"

   oss
       OSS (Open Sound System) output device.

   pulse
       PulseAudio output device.

       To enable this output device you need to configure FFmpeg with
       "--enable-libpulse".

       More information about PulseAudio can be found on
       <http://www.pulseaudio.org>

       Options

       server
           Connect to a specific PulseAudio server, specified by an IP
           address.  Default server is used when not provided.

       name
           Specify the application name PulseAudio will use when showing
           active clients, by default it is the "LIBAVFORMAT_IDENT" string.

       stream_name
           Specify the stream name PulseAudio will use when showing active
           streams, by default it is set to the specified output name.

       device
           Specify the device to use. Default device is used when not
           provided.  List of output devices can be obtained with command
           pactl list sinks.

       buffer_size
       buffer_duration
           Control the size and duration of the PulseAudio buffer. A small
           buffer gives more control, but requires more frequent updates.

           buffer_size specifies size in bytes while buffer_duration specifies
           duration in milliseconds.

           When both options are provided then the highest value is used
           (duration is recalculated to bytes using stream parameters). If
           they are set to 0 (which is default), the device will use the
           default PulseAudio duration value. By default PulseAudio set buffer
           duration to around 2 seconds.

       prebuf
           Specify pre-buffering size in bytes. The server does not start with
           playback before at least prebuf bytes are available in the buffer.
           By default this option is initialized to the same value as
           buffer_size or buffer_duration (whichever is bigger).

       minreq
           Specify minimum request size in bytes. The server does not request
           less than minreq bytes from the client, instead waits until the
           buffer is free enough to request more bytes at once. It is
           recommended to not set this option, which will initialize this to a
           value that is deemed sensible by the server.

       Examples

       Play a file on default device on default server:

               ffmpeg  -i INPUT -f pulse "stream name"

   sdl
       SDL (Simple DirectMedia Layer) output device.

       This output device allows one to show a video stream in an SDL window.
       Only one SDL window is allowed per application, so you can have only
       one instance of this output device in an application.

       To enable this output device you need libsdl installed on your system
       when configuring your build.

       For more information about SDL, check: <http://www.libsdl.org/>

       Options

       window_title
           Set the SDL window title, if not specified default to the filename
           specified for the output device.

       icon_title
           Set the name of the iconified SDL window, if not specified it is
           set to the same value of window_title.

       window_size
           Set the SDL window size, can be a string of the form widthxheight
           or a video size abbreviation.  If not specified it defaults to the
           size of the input video, downscaled according to the aspect ratio.

       window_fullscreen
           Set fullscreen mode when non-zero value is provided.  Default value
           is zero.

       Interactive commands

       The window created by the device can be controlled through the
       following interactive commands.

       q, ESC
           Quit the device immediately.

       Examples

       The following command shows the ffmpeg output is an SDL window, forcing
       its size to the qcif format:

               ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

   sndio
       sndio audio output device.

   xv
       XV (XVideo) output device.

       This output device allows one to show a video stream in a X Window
       System window.

       Options

       display_name
           Specify the hardware display name, which determines the display and
           communications domain to be used.

           The display name or DISPLAY environment variable can be a string in
           the format hostname[:number[.screen_number]].

           hostname specifies the name of the host machine on which the
           display is physically attached. number specifies the number of the
           display server on that host machine. screen_number specifies the
           screen to be used on that server.

           If unspecified, it defaults to the value of the DISPLAY environment
           variable.

           For example, "dual-headed:0.1" would specify screen 1 of display 0
           on the machine named ``dual-headed''.

           Check the X11 specification for more detailed information about the
           display name format.

       window_id
           When set to non-zero value then device doesn't create new window,
           but uses existing one with provided window_id. By default this
           options is set to zero and device creates its own window.

       window_size
           Set the created window size, can be a string of the form
           widthxheight or a video size abbreviation. If not specified it
           defaults to the size of the input video.  Ignored when window_id is
           set.

       window_x
       window_y
           Set the X and Y window offsets for the created window. They are
           both set to 0 by default. The values may be ignored by the window
           manager.  Ignored when window_id is set.

       window_title
           Set the window title, if not specified default to the filename
           specified for the output device. Ignored when window_id is set.

       For more information about XVideo see <http://www.x.org/>.

       Examples

       o   Decode, display and encode video input with ffmpeg at the same
           time:

                   ffmpeg -i INPUT OUTPUT -f xv display

       o   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:

       o   A ':'-separated list of key=value pairs.

       o   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.

       o   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

       o   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

       o   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

       o   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

       o   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

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

                   aecho=0.8:0.88:60:0.4

       o   If delay is very short, then it sound like a (metallic) robot
           playing music:

                   aecho=0.8:0.88:6:0.4

       o   A longer delay will sound like an open air concert in the
           mountains:

                   aecho=0.8:0.9:1000:0.3

       o   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

       o   Half volume:

                   aeval=val(ch)/2:c=same

       o   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

       o   Fade in first 15 seconds of audio:

                   afade=t=in:ss=0:d=15

       o   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

       o   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

       o   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

       o   Merge two mono files into a stereo stream:

                   amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

       o   Multiple merges assuming 1 video stream and 6 audio streams in
           input.mkv:

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

   amix
       Mixes multiple audio inputs into a single output.

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

       For example

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

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

       It accepts the following parameters:

       inputs
           The number of inputs. If unspecified, it defaults to 2.

       duration
           How to determine the end-of-stream.

           longest
               The duration of the longest input. (default)

           shortest
               The duration of the shortest input.

           first
               The duration of the first input.

       dropout_transition
           The transition time, in seconds, for volume renormalization when an
           input stream ends. The default value is 2 seconds.

   anequalizer
       High-order parametric multiband equalizer for each channel.

       It accepts the following parameters:

       params
           This option string is in format: "cchn f=cf w=w g=g t=f | ..."
           Each equalizer band is separated by '|'.

           chn Set channel number to which equalization will be applied.  If
               input doesn't have that channel the entry is ignored.

           f   Set central frequency for band.  If input doesn't have that
               frequency the entry is ignored.

           w   Set band width in hertz.

           g   Set band gain in dB.

           t   Set filter type for band, optional, can be:

               0   Butterworth, this is default.

               1   Chebyshev type 1.

               2   Chebyshev type 2.

       curves
           With this option activated frequency response of anequalizer is
           displayed in video stream.

       size
           Set video stream size. Only useful if curves option is activated.

       mgain
           Set max gain that will be displayed. Only useful if curves option
           is activated.  Setting this to a reasonable value makes it possible
           to display gain which is derived from neighbour bands which are too
           close to each other and thus produce higher gain when both are
           activated.

       fscale
           Set frequency scale used to draw frequency response in video
           output.  Can be linear or logarithmic. Default is logarithmic.

       colors
           Set color for each channel curve which is going to be displayed in
           video stream.  This is list of color names separated by space or by
           '|'.  Unrecognised or missing colors will be replaced by white
           color.

       Examples

       o   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

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

                   apad=pad_len=1024

       o   Make sure the audio output will contain at least 10000 samples, pad
           the input with silence if required:

                   apad=whole_len=10000

       o   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

       o   Resample the input audio to 44100Hz:

                   aresample=44100

       o   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

       o   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

       o   Slow down audio to 80% tempo:

                   atempo=0.8

       o   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:

       o   Drop everything except the second minute of input:

                   ffmpeg -i INPUT -af atrim=60:120

       o   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

       o   A single delay:

                   chorus=0.7:0.9:55:0.4:0.25:2

       o   Two delays:

                   chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

       o   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

       o   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

       o   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

       o   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

       o   2:1 compression starting at -6dB:

                   compand=points=-80/-80|-6/-6|0/-3.8|20/3.5

       o   2:1 compression starting at -9dB:

                   compand=points=-80/-80|-9/-9|0/-5.3|20/2.9

       o   2:1 compression starting at -12dB:

                   compand=points=-80/-80|-12/-12|0/-6.8|20/1.9

       o   2:1 compression starting at -18dB:

                   compand=points=-80/-80|-18/-18|0/-9.8|20/0.7

       o   3:1 compression starting at -15dB:

                   compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2

       o   Compressor/Gate:

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

       o   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

       o   Hard limiter at -6dB:

                   compand=attacks=0:points=-80/-80|-6/-6|20/-6

       o   Hard limiter at -12dB:

                   compand=attacks=0:points=-80/-80|-12/-12|20/-12

       o   Hard noise gate at -35 dB:

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

       o   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

       o   Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

                   equalizer=f=1000:t=h:width=200:g=-10

       o   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

       o   lowpass at 1000 Hz:

                   firequalizer=gain='if(lt(f,1000), 0, -INF)'

       o   lowpass at 1000 Hz with gain_entry:

                   firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'

       o   custom equalization:

                   firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'

       o   higher delay with zero phase to compensate delay:

                   firequalizer=delay=0.1:fixed=on:zero_phase=on

       o   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

       o   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

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

                   ladspa=file=amp

       o   List all available controls and their valid ranges for "vcf_notch"
           plugin from "VCF" library:

                   ladspa=f=vcf:p=vcf_notch:c=help

       o   Simulate low quality audio equipment using "Computer Music Toolkit"
           (CMT) plugin library:

                   ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

       o   Add reverberation to the audio using TAP-plugins (Tom's Audio
           Processing plugins):

                   ladspa=file=tap_reverb:tap_reverb

       o   Generate white noise, with 0.2 amplitude:

                   ladspa=file=cmt:noise_source_white:c=c0=.2

       o   Generate 20 bpm clicks using plugin "C* Click - Metronome" from the
           "C* Audio Plugin Suite" (CAPS) library:

                   ladspa=file=caps:Click:c=c1=20'

       o   Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

                   ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

       o   Increase volume by 20dB using fast lookahead limiter from Steve
           Harris "SWH Plugins" collection:

                   ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

       o   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

       o   Reduce stereo image using "Narrower" from the "C* Audio Plugin
           Suite" (CAPS) library:

                   ladspa=caps:Narrower

       o   Another white noise, now using "C* Audio Plugin Suite" (CAPS)
           library:

                   ladspa=caps:White:.2

       o   Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:

                   ladspa=caps:Fractal:c=c1=1

       o   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

       o   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

       o   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

       o   Detect 5 seconds of silence with -50dB noise tolerance:

                   silencedetect=n=-50dB:d=5

       o   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

       o   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

       o   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

       o   Using ClubFritz6 sofa file:

                   sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1

       o   Using ClubFritz12 sofa file and bigger radius with small rotation:

                   sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5

       o   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

       o   Apply karaoke like effect:

                   stereotools=mlev=0.015625

       o   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

       o   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

       o   Increase input audio power by 6 decibels using fixed-point
           precision:

                   volume=volume=6dB:precision=fixed

       o   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:

       o   The mean square energy is approximately -27 dB, or 10^-2.7.

       o   The largest sample is at -4 dB, or more precisely between -4 dB and
           -5 dB.

       o   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

       o   Generate silence:

                   aevalsrc=0

       o   Generate a sin signal with frequency of 440 Hz, set sample rate to
           8000 Hz:

                   aevalsrc="sin(440*2*PI*t):s=8000"

       o   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"

       o   Generate white noise:

                   aevalsrc="-2+random(0)"

       o   Generate an amplitude modulated signal:

                   aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

       o   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

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

                   anullsrc=r=48000:cl=4

       o   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

       o   Read from file speech.txt, and synthesize the text using the
           standard flite voice:

                   flite=textfile=speech.txt

       o   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

       o   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

       o   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

       o   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

       o   Generate a simple 440 Hz sine wave:

                   sine

       o   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

       o   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

       o   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)))'

       o   Apply linear horizontal transition from top layer to bottom layer:

                   blend=all_expr='A*(X/W)+B*(1-X/W)'

       o   Apply 1x1 checkerboard effect:

                   blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

       o   Apply uncover left effect:

                   blend=all_expr='if(gte(N*SW+X,W),A,B)'

       o   Apply uncover down effect:

                   blend=all_expr='if(gte(Y-N*SH,0),A,B)'

       o   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)'

       o   Split diagonally video and shows top and bottom layer on each side:

                   blend=all_expr='if(gt(X,Y*(W/H)),A,B)'

       o   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

       o   Apply a boxblur filter with the luma, chroma, and alpha radii set
           to 2:

                   boxblur=luma_radius=2:luma_power=1
                   boxblur=2:1

       o   Set the luma radius to 2, and alpha and chroma radius to 0:

                   boxblur=2:1:cr=0:ar=0

       o   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

       o   Make every green pixel in the input image transparent:

                   ffmpeg -i input.png -vf chromakey=green out.png

       o   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

       o   Visualize forward predicted MVs of all frames using ffplay:

                   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp

       o   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

       o   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

       o   Make every green pixel in the input image transparent:

                   ffmpeg -i input.png -vf colorkey=green out.png

       o   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

       o   Make video output darker:

                   colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

       o   Increase contrast:

                   colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

       o   Make video output lighter:

                   colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

       o   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

       o   Convert source to grayscale:

                   colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

       o   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 33 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

       o   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"

       o   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"

       o   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"

       o   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"

       o   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"

       o   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

       o   List all filters available:

                   coreimage=list_filters=true

       o   Use the CIBoxBlur filter with default options to blur an image:

                   coreimage=filter=CIBoxBlur@default

       o   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

       o   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

       o   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

       o   Crop the central input area with size 100x100:

                   crop=100:100

       o   Crop the central input area with size 2/3 of the input video:

                   crop=2/3*in_w:2/3*in_h

       o   Crop the input video central square:

                   crop=out_w=in_h
                   crop=in_h

       o   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

       o   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

       o   Keep only the bottom right quarter of the input image:

                   crop=in_w/2:in_h/2:in_w/2:in_h/2

       o   Crop height for getting Greek harmony:

                   crop=in_w:1/PHI*in_w

       o   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)

       o   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)"

       o   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

       o   Increase slightly the middle level of blue:

                   curves=blue='0/0 0.5/0.58 1/1'

       o   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)"

       o   The previous example can also be achieved with the associated
           built-in preset:

                   curves=preset=vintage

       o   Or simply:

                   curves=vintage

       o   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'

       o   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(33) average by taking into
       account only values lower than the pixel.

       It accepts the following options:

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

   deflicker
       Remove temporal frame luminance variations.

       It accepts the following options:

       size, s
           Set moving-average filter size in frames. Default is 5. Allowed
           range is 2 - 129.

       mode, m
           Set averaging mode to smooth temporal luminance variations.

           Available values are:

           am  Arithmetic mean

           gm  Geometric mean

           hm  Harmonic mean

           qm  Quadratic mean

           cm  Cubic mean

           pm  Power mean

           median
               Median

       bypass
           Do not actually modify frame. Useful when one only wants metadata.

   dejudder
       Remove judder produced by partially interlaced telecined content.

       Judder can be introduced, for instance, by pullup filter. If the
       original source was partially telecined content then the output of
       "pullup,dejudder" will have a variable frame rate. May change the
       recorded frame rate of the container. Aside from that change, this
       filter will not affect constant frame rate video.

       The option available in this filter is:

       cycle
           Specify the length of the window over which the judder repeats.

           Accepts any integer greater than 1. Useful values are:

           4   If the original was telecined from 24 to 30 fps (Film to NTSC).

           5   If the original was telecined from 25 to 30 fps (PAL to NTSC).

           20  If a mixture of the two.

           The default is 4.

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

       It accepts the following parameters:

       x
       y   Specify the top left corner coordinates of the logo. They must be
           specified.

       w
       h   Specify the width and height of the logo to clear. They must be
           specified.

       band, t
           Specify the thickness of the fuzzy edge of the rectangle (added to
           w and h). The default value is 1. This option is deprecated,
           setting higher values should no longer be necessary and is not
           recommended.

       show
           When set to 1, a green rectangle is drawn on the screen to simplify
           finding the right x, y, w, and h parameters.  The default value is
           0.

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

       Examples

       o   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(33) 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 33 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

       o   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

       o   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

       o   Draw a black box around the edge of the input image:

                   drawbox

       o   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

       o   Fill the box with pink color:

                   drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=max

       o   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

       o   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

       o   Draw a white 33 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

       o   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'"

       o   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.

       o   Show the text at the center of the video frame:

                   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"

       o   Show the text at a random position, switching to a new position
           every 30 seconds:

                   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"

       o   Show a text line sliding from right to left in the last row of the
           video frame. The file LONG_LINE is assumed to contain a single line
           with no newlines.

                   drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"

       o   Show the content of file CREDITS off the bottom of the frame and
           scroll up.

                   drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"

       o   Draw a single green letter "g", at the center of the input video.
           The glyph baseline is placed at half screen height.

                   drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"

       o   Show text for 1 second every 3 seconds:

                   drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"

       o   Use fontconfig to set the font. Note that the colons need to be
           escaped.

                   drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'

       o   Print the date of a real-time encoding (see strftime(3)):

                   drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

       o   Show text fading in and out (appearing/disappearing):

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

       o   Horizontally align multiple separate texts. Note that max_glyph_a
           and the fontsize value are included in the y offset.

                   drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
                   drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a

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

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

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

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

       The filter accepts the following options:

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

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

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

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

       mode
           Define the drawing mode.

           wires
               Draw white/gray wires on black background.

           colormix
               Mix the colors to create a paint/cartoon effect.

           Default value is wires.

       Examples

       o   Standard edge detection with custom values for the hysteresis
           thresholding:

                   edgedetect=low=0.1:high=0.4

       o   Painting effect without thresholding:

                   edgedetect=mode=colormix:high=0

   eq
       Set brightness, contrast, saturation and approximate gamma adjustment.

       The filter accepts the following options:

       contrast
           Set the contrast expression. The value must be a float value in
           range "-2.0" to 2.0. The default value is "1".

       brightness
           Set the brightness expression. The value must be a float value in
           range "-1.0" to 1.0. The default value is "0".

       saturation
           Set the saturation expression. The value must be a float in range
           0.0 to 3.0. The default value is "1".

       gamma
           Set the gamma expression. The value must be a float in range 0.1 to
           10.0.  The default value is "1".

       gamma_r
           Set the gamma expression for red. The value must be a float in
           range 0.1 to 10.0. The default value is "1".

       gamma_g
           Set the gamma expression for green. The value must be a float in
           range 0.1 to 10.0. The default value is "1".

       gamma_b
           Set the gamma expression for blue. The value must be a float in
           range 0.1 to 10.0. The default value is "1".

       gamma_weight
           Set the gamma weight expression. It can be used to reduce the
           effect of a high gamma value on bright image areas, e.g. keep them
           from getting overamplified and just plain white. The value must be
           a float in range 0.0 to 1.0. A value of 0.0 turns the gamma
           correction all the way down while 1.0 leaves it at its full
           strength. Default is "1".

       eval
           Set when the expressions for brightness, contrast, saturation and
           gamma expressions are evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization
               or when a command is processed

           frame
               evaluate expressions for each incoming frame

           Default value is init.

       The expressions accept the following parameters:

       n   frame count of the input frame starting from 0

       pos byte position of the corresponding packet in the input file, NAN if
           unspecified

       r   frame rate of the input video, NAN if the input frame rate is
           unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is
           unknown

       Commands

       The filter supports the following commands:

       contrast
           Set the contrast expression.

       brightness
           Set the brightness expression.

       saturation
           Set the saturation expression.

       gamma
           Set the gamma expression.

       gamma_r
           Set the gamma_r expression.

       gamma_g
           Set gamma_g expression.

       gamma_b
           Set gamma_b expression.

       gamma_weight
           Set gamma_weight expression.

           The command accepts the same syntax of the corresponding option.

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

   erosion
       Apply erosion effect to the video.

       This filter replaces the pixel by the local(33) minimum.

       It accepts the following options:

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

       coordinates
           Flag which specifies the pixel to refer to. Default is 255 i.e. all
           eight pixels are used.

           Flags to local 33 coordinates maps like this:

               1 2 3
               4   5
               6 7 8

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

       The filter accepts the following option:

       planes
           Set plane(s) to extract.

           Available values for planes are:

           y
           u
           v
           a
           r
           g
           b

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

       Examples

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

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

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

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

       This filter accepts the following options.

       codebook_length, l
           Set codebook length. The value must be a positive integer, and
           represents the number of distinct output colors. Default value is
           256.

       nb_steps, n
           Set the maximum number of iterations to apply for computing the
           optimal mapping. The higher the value the better the result and the
           higher the computation time. Default value is 1.

       seed, s
           Set a random seed, must be an integer included between 0 and
           UINT32_MAX. If not specified, or if explicitly set to -1, the
           filter will try to use a good random seed on a best effort basis.

       pal8
           Set pal8 output pixel format. This option does not work with
           codebook length greater than 256.

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

       It accepts the following parameters:

       type, t
           The effect type can be either "in" for a fade-in, or "out" for a
           fade-out effect.  Default is "in".

       start_frame, s
           Specify the number of the frame to start applying the fade effect
           at. Default is 0.

       nb_frames, n
           The number of frames that the fade effect lasts. At the end of the
           fade-in effect, the output video will have the same intensity as
           the input video.  At the end of the fade-out transition, the output
           video will be filled with the selected color.  Default is 25.

       alpha
           If set to 1, fade only alpha channel, if one exists on the input.
           Default value is 0.

       start_time, st
           Specify the timestamp (in seconds) of the frame to start to apply
           the fade effect. If both start_frame and start_time are specified,
           the fade will start at whichever comes last.  Default is 0.

       duration, d
           The number of seconds for which the fade effect has to last. At the
           end of the fade-in effect the output video will have the same
           intensity as the input video, at the end of the fade-out transition
           the output video will be filled with the selected color.  If both
           duration and nb_frames are specified, duration is used. Default is
           0 (nb_frames is used by default).

       color, c
           Specify the color of the fade. Default is "black".

       Examples

       o   Fade in the first 30 frames of video:

                   fade=in:0:30

           The command above is equivalent to:

                   fade=t=in:s=0:n=30

       o   Fade out the last 45 frames of a 200-frame video:

                   fade=out:155:45
                   fade=type=out:start_frame=155:nb_frames=45

       o   Fade in the first 25 frames and fade out the last 25 frames of a
           1000-frame video:

                   fade=in:0:25, fade=out:975:25

       o   Make the first 5 frames yellow, then fade in from frame 5-24:

                   fade=in:5:20:color=yellow

       o   Fade in alpha over first 25 frames of video:

                   fade=in:0:25:alpha=1

       o   Make the first 5.5 seconds black, then fade in for 0.5 seconds:

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

   fftfilt
       Apply arbitrary expressions to samples in frequency domain

       dc_Y
           Adjust the dc value (gain) of the luma plane of the image. The
           filter accepts an integer value in range 0 to 1000. The default
           value is set to 0.

       dc_U
           Adjust the dc value (gain) of the 1st chroma plane of the image.
           The filter accepts an integer value in range 0 to 1000. The default
           value is set to 0.

       dc_V
           Adjust the dc value (gain) of the 2nd chroma plane of the image.
           The filter accepts an integer value in range 0 to 1000. The default
           value is set to 0.

       weight_Y
           Set the frequency domain weight expression for the luma plane.

       weight_U
           Set the frequency domain weight expression for the 1st chroma
           plane.

       weight_V
           Set the frequency domain weight expression for the 2nd chroma
           plane.

       eval
           Set when the expressions are evaluated.

           It accepts the following values:

           init
               Only evaluate expressions once during the filter
               initialization.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

           The filter accepts the following variables:

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       N   The number of input frame, starting from 0.

       Examples

       o   High-pass:

                   fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

       o   Low-pass:

                   fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

       o   Sharpen:

                   fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

       o   Blur:

                   fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'

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

       The filter accepts the following options:

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

   fieldhint
       Create new frames by copying the top and bottom fields from surrounding
       frames supplied as numbers by the hint file.

       hint
           Set file containing hints: absolute/relative frame numbers.

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

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

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

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

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

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

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

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

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

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

       The filter accepts the following options:

       order
           Specify the assumed field order of the input stream. Available
           values are:

           auto
               Auto detect parity (use FFmpeg's internal parity value).

           bff Assume bottom field first.

           tff Assume top field first.

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

           Default value is auto.

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

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

           Available values are:

           pc  2-way matching (p/c)

           pc_n
               2-way matching, and trying 3rd match if still combed (p/c + n)

           pc_u
               2-way matching, and trying 3rd match (same order) if still
               combed (p/c + u)

           pc_n_ub
               2-way matching, trying 3rd match if still combed, and trying
               4th/5th matches if still combed (p/c + n + u/b)

           pcn 3-way matching (p/c/n)

           pcn_ub
               3-way matching, and trying 4th/5th matches if all 3 of the
               original matches are detected as combed (p/c/n + u/b)

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

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

           Default value is pc_n.

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

           Default value is 0 (disabled).

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

           auto
               Automatic (same value as order).

           bottom
               Match from the bottom field.

           top Match from the top field.

           Default value is auto.

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

           Default value is 1.

       y0
       y1  These define an exclusion band which excludes the lines between y0
           and y1 from being included in the field matching decision. An
           exclusion band can be used to ignore subtitles, a logo, or other
           things that may interfere with the matching. y0 sets the starting
           scan line and y1 sets the ending line; all lines in between y0 and
           y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the
           same value will disable the feature.  y0 and y1 defaults to 0.

       scthresh
           Set the scene change detection threshold as a percentage of maximum
           change on the luma plane. Good values are in the "[8.0, 14.0]"
           range. Scene change detection is only relevant in case
           combmatch=sc.  The range for scthresh is "[0.0, 100.0]".

           Default value is 12.0.

       combmatch
           When combatch is not none, "fieldmatch" will take into account the
           combed scores of matches when deciding what match to use as the
           final match. Available values are:

           none
               No final matching based on combed scores.

           sc  Combed scores are only used when a scene change is detected.

           full
               Use combed scores all the time.

           Default is sc.

       combdbg
           Force "fieldmatch" to calculate the combed metrics for certain
           matches and print them. This setting is known as micout in TFM/VFM
           vocabulary.  Available values are:

           none
               No forced calculation.

           pcn Force p/c/n calculations.

           pcnub
               Force p/c/n/u/b calculations.

           Default value is none.

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

           Default value is 9.

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

           Default value is 0.

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

           Default value is 16.

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

           Default value is 80.

       p/c/n/u/b meaning

       p/c/n

       We assume the following telecined stream:

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

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

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

               Input stream:
                               T     1 2 2 3 4
                               B     1 2 3 4 4   <-- matching reference

               Matches:              c c n n c

               Output stream:
                               T     1 2 3 4 4
                               B     1 2 3 4 4

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

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

               Input stream:
                               T     1 2 2 3 4   <-- matching reference
                               B     1 2 3 4 4

               Matches:              c c p p c

               Output stream:
                               T     1 2 2 3 4
                               B     1 2 2 3 4

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

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

       u/b

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

       With bottom matching (field=bottom):

               Match:           c         p           n          b          u

                                x       x               x        x          x
                 Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
                 Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                                x         x           x        x              x

               Output frames:
                                2          1          2          2          2
                                2          2          2          1          3

       With top matching (field=top):

               Match:           c         p           n          b          u

                                x         x           x        x              x
                 Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
                 Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                                x       x               x        x          x

               Output frames:
                                2          2          2          1          2
                                2          1          3          2          2

       Examples

       Simple IVTC of a top field first telecined stream:

               fieldmatch=order=tff:combmatch=none, decimate

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

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

   fieldorder
       Transform the field order of the input video.

       It accepts the following parameters:

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

       The default value is tff.

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

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

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

       For example:

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

   fifo, afifo
       Buffer input images and send them when they are requested.

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

       It does not take parameters.

   find_rect
       Find a rectangular object

       It accepts the following options:

       object
           Filepath of the object image, needs to be in gray8.

       threshold
           Detection threshold, default is 0.5.

       mipmaps
           Number of mipmaps, default is 3.

       xmin, ymin, xmax, ymax
           Specifies the rectangle in which to search.

       Examples

       o   Generate a representative palette of a given video using ffmpeg:

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

   cover_rect
       Cover a rectangular object

       It accepts the following options:

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

       mode
           Set covering mode.

           It accepts the following values:

           cover
               cover it by the supplied image

           blur
               cover it by interpolating the surrounding pixels

           Default value is blur.

       Examples

       o   Generate a representative palette of a given video using ffmpeg:

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

   floodfill
       Flood area with values of same pixel components with another values.

       It accepts the following options:

       x   Set pixel x coordinate.

       y   Set pixel y coordinate.

       s0  Set source #0 component value.

       s1  Set source #1 component value.

       s2  Set source #2 component value.

       s3  Set source #3 component value.

       d0  Set destination #0 component value.

       d1  Set destination #1 component value.

       d2  Set destination #2 component value.

       d3  Set destination #3 component value.

   format
       Convert the input video to one of the specified pixel formats.
       Libavfilter will try to pick one that is suitable as input to the next
       filter.

       It accepts the following parameters:

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

       Examples

       o   Convert the input video to the yuv420p format

                   format=pix_fmts=yuv420p

           Convert the input video to any of the formats in the list

                   format=pix_fmts=yuv420p|yuv444p|yuv410p

   fps
       Convert the video to specified constant frame rate by duplicating or
       dropping frames as necessary.

       It accepts the following parameters:

       fps The desired output frame rate. The default is 25.

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

       round
           Timestamp (PTS) rounding method.

           Possible values are:

           zero
               round towards 0

           inf round away from 0

           down
               round towards -infinity

           up  round towards +infinity

           near
               round to nearest

           The default is "near".

       eof_action
           Action performed when reading the last frame.

           Possible values are:

           round
               Use same timestamp rounding method as used for other frames.

           pass
               Pass through last frame if input duration has not been reached
               yet.

           The default is "round".

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

       See also the setpts filter.

       Examples

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

                   fps=fps=25

       o   Sets the fps to 24, using abbreviation and rounding method to round
           to nearest:

                   fps=fps=film:round=near

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

       It accepts the following parameters:

       format
           The desired packing format. Supported values are:

           sbs The views are next to each other (default).

           tab The views are on top of each other.

           lines
               The views are packed by line.

           columns
               The views are packed by column.

           frameseq
               The views are temporally interleaved.

       Some examples:

               # Convert left and right views into a frame-sequential video
               ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

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

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

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

       A description of the accepted options follows.

       fps Specify the output frames per second. This option can also be
           specified as a value alone. The default is 50.

       interp_start
           Specify the start of a range where the output frame will be created
           as a linear interpolation of two frames. The range is [0-255], the
           default is 15.

       interp_end
           Specify the end of a range where the output frame will be created
           as a linear interpolation of two frames. The range is [0-255], the
           default is 240.

       scene
           Specify the level at which a scene change is detected as a value
           between 0 and 100 to indicate a new scene; a low value reflects a
           low probability for the current frame to introduce a new scene,
           while a higher value means the current frame is more likely to be
           one.  The default is 7.

       flags
           Specify flags influencing the filter process.

           Available value for flags is:

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

   framestep
       Select one frame every N-th frame.

       This filter accepts the following option:

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

   frei0r
       Apply a frei0r effect to the input video.

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

       It accepts the following parameters:

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

       filter_params
           A '|'-separated list of parameters to pass to the frei0r effect.

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

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

       Examples

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

                   frei0r=filter_name=distort0r:filter_params=0.5|0.01

       o   Apply the colordistance effect, taking a color as the first
           parameter:

                   frei0r=colordistance:0.2/0.3/0.4
                   frei0r=colordistance:violet
                   frei0r=colordistance:0x112233

       o   Apply the perspective effect, specifying the top left and top right
           image positions:

                   frei0r=perspective:0.2/0.2|0.8/0.2

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

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

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

       The filter accepts the following options:

       quality
           Set quality. This option defines the number of levels for
           averaging. It accepts an integer in the range 4-5. Default value is
           4.

       qp  Force a constant quantization parameter. It accepts an integer in
           range 0-63.  If not set, the filter will use the QP from the video
           stream (if available).

       strength
           Set filter strength. It accepts an integer in range -15 to 32.
           Lower values mean more details but also more artifacts, while
           higher values make the image smoother but also blurrier. Default
           value is 0 X PSNR optimal.

       use_bframe_qp
           Enable the use of the QP from the B-Frames if set to 1. Using this
           option may cause flicker since the B-Frames have often larger QP.
           Default is 0 (not enabled).

   gblur
       Apply Gaussian blur filter.

       The filter accepts the following options:

       sigma
           Set horizontal sigma, standard deviation of Gaussian blur. Default
           is 0.5.

       steps
           Set number of steps for Gaussian approximation. Defauls is 1.

       planes
           Set which planes to filter. By default all planes are filtered.

       sigmaV
           Set vertical sigma, if negative it will be same as "sigma".
           Default is "-1".

   geq
       The filter accepts the following options:

       lum_expr, lum
           Set the luminance expression.

       cb_expr, cb
           Set the chrominance blue expression.

       cr_expr, cr
           Set the chrominance red expression.

       alpha_expr, a
           Set the alpha expression.

       red_expr, r
           Set the red expression.

       green_expr, g
           Set the green expression.

       blue_expr, b
           Set the blue expression.

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

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

       The expressions can use the following variables and functions:

       N   The sequential number of the filtered frame, starting from 0.

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       SW
       SH  Width and height scale depending on the currently filtered plane.
           It is the ratio between the corresponding luma plane number of
           pixels and the current plane ones. E.g. for YUV4:2:0 the values are
           "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

       T   Time of the current frame, expressed in seconds.

       p(x, y)
           Return the value of the pixel at location (x,y) of the current
           plane.

       lum(x, y)
           Return the value of the pixel at location (x,y) of the luminance
           plane.

       cb(x, y)
           Return the value of the pixel at location (x,y) of the blue-
           difference chroma plane. Return 0 if there is no such plane.

       cr(x, y)
           Return the value of the pixel at location (x,y) of the red-
           difference chroma plane. Return 0 if there is no such plane.

       r(x, y)
       g(x, y)
       b(x, y)
           Return the value of the pixel at location (x,y) of the
           red/green/blue component. Return 0 if there is no such component.

       alpha(x, y)
           Return the value of the pixel at location (x,y) of the alpha plane.
           Return 0 if there is no such plane.

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

       Examples

       o   Flip the image horizontally:

                   geq=p(W-X\,Y)

       o   Generate a bidimensional sine wave, with angle "PI/3" and a
           wavelength of 100 pixels:

                   geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

       o   Generate a fancy enigmatic moving light:

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

       o   Generate a quick emboss effect:

                   format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

       o   Modify RGB components depending on pixel position:

                   geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

       o   Create a radial gradient that is the same size as the input (also
           see the vignette filter):

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

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

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

       It accepts the following parameters:

       strength
           The maximum amount by which the filter will change any one pixel.
           This is also the threshold for detecting nearly flat regions.
           Acceptable values range from .51 to 64; the default value is 1.2.
           Out-of-range values will be clipped to the valid range.

       radius
           The neighborhood to fit the gradient to. A larger radius makes for
           smoother gradients, but also prevents the filter from modifying the
           pixels near detailed regions. Acceptable values are 8-32; the
           default value is 16. Out-of-range values will be clipped to the
           valid range.

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

       Examples

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

                   gradfun=3.5:8

       o   Specify radius, omitting the strength (which will fall-back to the
           default value):

                   gradfun=radius=8

   haldclut
       Apply a Hald CLUT to a video stream.

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

       The filter accepts the following options:

       shortest
           Force termination when the shortest input terminates. Default is 0.

       repeatlast
           Continue applying the last CLUT after the end of the stream. A
           value of 0 disable the filter after the last frame of the CLUT is
           reached.  Default is 1.

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

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

       Workflow examples

       Hald CLUT video stream

       Generate an identity Hald CLUT stream altered with various effects:

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

       Note: make sure you use a lossless codec.

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

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

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

       Hald CLUT with preview

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

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

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

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

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

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

   hflip
       Flip the input video horizontally.

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

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

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

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

       The filter accepts the following options:

       strength
           Determine the amount of equalization to be applied.  As the
           strength is reduced, the distribution of pixel intensities more-
           and-more approaches that of the input frame. The value must be a
           float number in the range [0,1] and defaults to 0.200.

       intensity
           Set the maximum intensity that can generated and scale the output
           values appropriately.  The strength should be set as desired and
           then the intensity can be limited if needed to avoid washing-out.
           The value must be a float number in the range [0,1] and defaults to
           0.210.

       antibanding
           Set the antibanding level. If enabled the filter will randomly vary
           the luminance of output pixels by a small amount to avoid banding
           of the histogram. Possible values are "none", "weak" or "strong".
           It defaults to "none".

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

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

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

       The filter accepts the following options:

       level_height
           Set height of level. Default value is 200.  Allowed range is [50,
           2048].

       scale_height
           Set height of color scale. Default value is 12.  Allowed range is
           [0, 40].

       display_mode
           Set display mode.  It accepts the following values:

           stack
               Per color component graphs are placed below each other.

           parade
               Per color component graphs are placed side by side.

           overlay
               Presents information identical to that in the "parade", except
               that the graphs representing color components are superimposed
               directly over one another.

           Default is "stack".

       levels_mode
           Set mode. Can be either "linear", or "logarithmic".  Default is
           "linear".

       components
           Set what color components to display.  Default is 7.

       fgopacity
           Set foreground opacity. Default is 0.7.

       bgopacity
           Set background opacity. Default is 0.5.

       Examples

       o   Calculate and draw histogram:

                   ffplay -i input -vf histogram

   hqdn3d
       This is a high precision/quality 3d denoise filter. It aims to reduce
       image noise, producing smooth images and making still images really
       still. It should enhance compressibility.

       It accepts the following optional parameters:

       luma_spatial
           A non-negative floating point number which specifies spatial luma
           strength.  It defaults to 4.0.

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

       luma_tmp
           A floating point number which specifies luma temporal strength. It
           defaults to 6.0*luma_spatial/4.0.

       chroma_tmp
           A floating point number which specifies chroma temporal strength.
           It defaults to luma_tmp*chroma_spatial/luma_spatial.

   hwdownload
       Download hardware frames to system memory.

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

   hwmap
       Map hardware frames to system memory or to another device.

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

       o   Hardware frame input, normal frame output

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

       o   Normal frame input, hardware frame output

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

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

       o   Hardware frame input and output

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

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

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

       The following additional parameters are accepted:

       mode
           Set the frame mapping mode.  Some combination of:

           read
               The mapped frame should be readable.

           write
               The mapped frame should be writeable.

           overwrite
               The mapping will always overwrite the entire frame.

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

           direct
               The mapping must not involve any copying.

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

           Defaults to read+write if not specified.

       derive_device type
           Rather than using the device supplied at initialisation, instead
           derive a new device of type type from the device the input frames
           exist on.

       reverse
           In a hardware to hardware mapping, map in reverse - create frames
           in the sink and map them back to the source.  This may be necessary
           in some cases where a mapping in one direction is required but only
           the opposite direction is supported by the devices being used.

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

   hwupload
       Upload system memory frames to hardware surfaces.

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

   hwupload_cuda
       Upload system memory frames to a CUDA device.

       It accepts the following optional parameters:

       device
           The number of the CUDA device to use

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

       It accepts the following option:

       n   Set the scaling dimension: 2 for "hq2x", 3 for "hq3" and 4 for
           "hq4x".  Default is 3.

   hstack
       Stack input videos horizontally.

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

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

       The filter accept the following option:

       inputs
           Set number of input streams. Default is 2.

       shortest
           If set to 1, force the output to terminate when the shortest input
           terminates. Default value is 0.

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

       It accepts the following parameters:

       h   Specify the hue angle as a number of degrees. It accepts an
           expression, and defaults to "0".

       s   Specify the saturation in the [-10,10] range. It accepts an
           expression and defaults to "1".

       H   Specify the hue angle as a number of radians. It accepts an
           expression, and defaults to "0".

       b   Specify the brightness in the [-10,10] range. It accepts an
           expression and defaults to "0".

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

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

       n   frame count of the input frame starting from 0

       pts presentation timestamp of the input frame expressed in time base
           units

       r   frame rate of the input video, NAN if the input frame rate is
           unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is
           unknown

       tb  time base of the input video

       Examples

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

                   hue=h=90:s=1

       o   Same command but expressing the hue in radians:

                   hue=H=PI/2:s=1

       o   Rotate hue and make the saturation swing between 0 and 2 over a
           period of 1 second:

                   hue="H=2*PI*t: s=sin(2*PI*t)+1"

       o   Apply a 3 seconds saturation fade-in effect starting at 0:

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

           The general fade-in expression can be written as:

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

       o   Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

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

           The general fade-out expression can be written as:

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

       Commands

       This filter supports the following commands:

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

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

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

       This filter accepts the following options:

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

       threshold
           Set threshold which is used in filtering. If pixel component value
           is higher than this value filter algorithm for connecting
           components is activated.  By default value is 0.

   idet
       Detect video interlacing type.

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

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

       The filter will log these metadata values:

       single.current_frame
           Detected type of current frame using single-frame detection. One
           of: ``tff'' (top field first), ``bff'' (bottom field first),
           ``progressive'', or ``undetermined''

       single.tff
           Cumulative number of frames detected as top field first using
           single-frame detection.

       multiple.tff
           Cumulative number of frames detected as top field first using
           multiple-frame detection.

       single.bff
           Cumulative number of frames detected as bottom field first using
           single-frame detection.

       multiple.current_frame
           Detected type of current frame using multiple-frame detection. One
           of: ``tff'' (top field first), ``bff'' (bottom field first),
           ``progressive'', or ``undetermined''

       multiple.bff
           Cumulative number of frames detected as bottom field first using
           multiple-frame detection.

       single.progressive
           Cumulative number of frames detected as progressive using single-
           frame detection.

       multiple.progressive
           Cumulative number of frames detected as progressive using multiple-
           frame detection.

       single.undetermined
           Cumulative number of frames that could not be classified using
           single-frame detection.

       multiple.undetermined
           Cumulative number of frames that could not be classified using
           multiple-frame detection.

       repeated.current_frame
           Which field in the current frame is repeated from the last. One of
           ``neither'', ``top'', or ``bottom''.

       repeated.neither
           Cumulative number of frames with no repeated field.

       repeated.top
           Cumulative number of frames with the top field repeated from the
           previous frame's top field.

       repeated.bottom
           Cumulative number of frames with the bottom field repeated from the
           previous frame's bottom field.

       The filter accepts the following options:

       intl_thres
           Set interlacing threshold.

       prog_thres
           Set progressive threshold.

       rep_thres
           Threshold for repeated field detection.

       half_life
           Number of frames after which a given frame's contribution to the
           statistics is halved (i.e., it contributes only 0.5 to its
           classification). The default of 0 means that all frames seen are
           given full weight of 1.0 forever.

       analyze_interlaced_flag
           When this is not 0 then idet will use the specified number of
           frames to determine if the interlaced flag is accurate, it will not
           count undetermined frames.  If the flag is found to be accurate it
           will be used without any further computations, if it is found to be
           inaccurate it will be cleared without any further computations.
           This allows inserting the idet filter as a low computational method
           to clean up the interlaced flag

   il
       Deinterleave or interleave fields.

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

       The filter accepts the following options:

       luma_mode, l
       chroma_mode, c
       alpha_mode, a
           Available values for luma_mode, chroma_mode and alpha_mode are:

           none
               Do nothing.

           deinterleave, d
               Deinterleave fields, placing one above the other.

           interleave, i
               Interleave fields. Reverse the effect of deinterleaving.

           Default value is "none".

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

   inflate
       Apply inflate effect to the video.

       This filter replaces the pixel by the local(33) average by taking into
       account only values higher than the pixel.

       It accepts the following options:

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

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

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

       It accepts the following optional parameters:

       scan
           This determines whether the interlaced frame is taken from the even
           (tff - default) or odd (bff) lines of the progressive frame.

       lowpass
           Vertical lowpass filter to avoid twitter interlacing and reduce
           moire patterns.

           0, off
               Disable vertical lowpass filter

           1, linear
               Enable linear filter (default)

           2, complex
               Enable complex filter. This will slightly less reduce twitter
               and moire but better retain detail and subjective sharpness
               impression.

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

       The description of the accepted parameters follows.

       thresh
           Set the threshold which affects the filter's tolerance when
           determining if a pixel line must be processed. It must be an
           integer in the range [0,255] and defaults to 10. A value of 0 will
           result in applying the process on every pixels.

       map Paint pixels exceeding the threshold value to white if set to 1.
           Default is 0.

       order
           Set the fields order. Swap fields if set to 1, leave fields alone
           if 0. Default is 0.

       sharp
           Enable additional sharpening if set to 1. Default is 0.

       twoway
           Enable twoway sharpening if set to 1. Default is 0.

       Examples

       o   Apply default values:

                   kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

       o   Enable additional sharpening:

                   kerndeint=sharp=1

       o   Paint processed pixels in white:

                   kerndeint=map=1

   lenscorrection
       Correct radial lens distortion

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

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

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

       Options

       The filter accepts the following options:

       cx  Relative x-coordinate of the focal point of the image, and thereby
           the center of the distortion. This value has a range [0,1] and is
           expressed as fractions of the image width.

       cy  Relative y-coordinate of the focal point of the image, and thereby
           the center of the distortion. This value has a range [0,1] and is
           expressed as fractions of the image height.

       k1  Coefficient of the quadratic correction term. 0.5 means no
           correction.

       k2  Coefficient of the double quadratic correction term. 0.5 means no
           correction.

       The formula that generates the correction is:

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

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

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

       This filter takes two input videos.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average VMAF score is printed through the logging system.

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

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

       The filter has following options:

       model_path
           Set the model path which is to be used for SVM.  Default value:
           "vmaf_v0.6.1.pkl"

       log_path
           Set the file path to be used to store logs.

       log_fmt
           Set the format of the log file (xml or json).

       enable_transform
           Enables transform for computing vmaf.

       phone_model
           Invokes the phone model which will generate VMAF scores higher than
           in the regular model, which is more suitable for laptop, TV, etc.
           viewing conditions.

       psnr
           Enables computing psnr along with vmaf.

       ssim
           Enables computing ssim along with vmaf.

       ms_ssim
           Enables computing ms_ssim along with vmaf.

       pool
           Set the pool method to be used for computing vmaf.

       This filter also supports the framesync options.

       For example:

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

       Example with options:

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

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

       The filter accepts the following options:

       min Lower bound. Defaults to the lowest allowed value for the input.

       max Upper bound. Defaults to the highest allowed value for the input.

       planes
           Specify which planes will be processed. Defaults to all available.

   loop
       Loop video frames.

       The filter accepts the following options:

       loop
           Set the number of loops.

       size
           Set maximal size in number of frames.

       start
           Set first frame of loop.

   lut3d
       Apply a 3D LUT to an input video.

       The filter accepts the following options:

       file
           Set the 3D LUT file name.

           Currently supported formats:

           3dl AfterEffects

           cube
               Iridas

           dat DaVinci

           m3d Pandora

       interp
           Select interpolation mode.

           Available values are:

           nearest
               Use values from the nearest defined point.

           trilinear
               Interpolate values using the 8 points defining a cube.

           tetrahedral
               Interpolate values using a tetrahedron.

       This filter also supports the framesync options.

   lumakey
       Turn certain luma values into transparency.

       The filter accepts the following options:

       threshold
           Set the luma which will be used as base for transparency.  Default
           value is 0.

       tolerance
           Set the range of luma values to be keyed out.  Default value is 0.

       softness
           Set the range of softness. Default value is 0.  Use this to control
           gradual transition from zero to full transparency.

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

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

       These filters accept the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
           component

       r   set red component expression

       g   set green component expression

       b   set blue component expression

       a   alpha component expression

       y   set Y/luminance component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

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

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

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

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       clipval
           The input value, clipped to the minval-maxval range.

       maxval
           The maximum value for the pixel component.

       minval
           The minimum value for the pixel component.

       negval
           The negated value for the pixel component value, clipped to the
           minval-maxval range; it corresponds to the expression
           "maxval-clipval+minval".

       clip(val)
           The computed value in val, clipped to the minval-maxval range.

       gammaval(gamma)
           The computed gamma correction value of the pixel component value,
           clipped to the minval-maxval range. It corresponds to the
           expression
           "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "val".

       Examples

       o   Negate input video:

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

           The above is the same as:

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

       o   Negate luminance:

                   lutyuv=y=negval

       o   Remove chroma components, turning the video into a graytone image:

                   lutyuv="u=128:v=128"

       o   Apply a luma burning effect:

                   lutyuv="y=2*val"

       o   Remove green and blue components:

                   lutrgb="g=0:b=0"

       o   Set a constant alpha channel value on input:

                   format=rgba,lutrgb=a="maxval-minval/2"

       o   Correct luminance gamma by a factor of 0.5:

                   lutyuv=y=gammaval(0.5)

       o   Discard least significant bits of luma:

                   lutyuv=y='bitand(val, 128+64+32)'

       o   Technicolor like effect:

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

   lut2, tlut2
       The "lut2" filter takes two input streams and outputs one stream.

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

       This filter accepts the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
           component

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

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

       The expressions can contain the following constants:

       w
       h   The input width and height.

       x   The first input value for the pixel component.

       y   The second input value for the pixel component.

       bdx The first input video bit depth.

       bdy The second input video bit depth.

       All expressions default to "x".

       Examples

       o   Highlight differences between two RGB video streams:

                   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

       o   Highlight differences between two YUV video streams:

                   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

       o   Show max difference between two video streams:

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

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

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

       This filter accepts the following options:

       undershoot
           Default value is 0.

       overshoot
           Default value is 0.

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

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

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

       This filter accepts the following options:

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

   mcdeint
       Apply motion-compensation deinterlacing.

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

       This filter accepts the following options:

       mode
           Set the deinterlacing mode.

           It accepts one of the following values:

           fast
           medium
           slow
               use iterative motion estimation

           extra_slow
               like slow, but use multiple reference frames.

           Default value is fast.

       parity
           Set the picture field parity assumed for the input video. It must
           be one of the following values:

           0, tff
               assume top field first

           1, bff
               assume bottom field first

           Default value is bff.

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

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

   mergeplanes
       Merge color channel components from several video streams.

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

       This filter accepts the following options:

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

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

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

       Examples

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

                   [a0][a1][a2]mergeplanes=0x001020:yuv444p

       o   Merge 1st yuv444p stream and 2nd gray video stream into yuva444p
           video stream:

                   [a0][a1]mergeplanes=0x00010210:yuva444p

       o   Swap Y and A plane in yuva444p stream:

                   format=yuva444p,mergeplanes=0x03010200:yuva444p

       o   Swap U and V plane in yuv420p stream:

                   format=yuv420p,mergeplanes=0x000201:yuv420p

       o   Cast a rgb24 clip to yuv444p:

                   format=rgb24,mergeplanes=0x000102:yuv444p

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

       This filter accepts the following options:

       method
           Specify the motion estimation method. Accepts one of the following
           values:

           esa Exhaustive search algorithm.

           tss Three step search algorithm.

           tdls
               Two dimensional logarithmic search algorithm.

           ntss
               New three step search algorithm.

           fss Four step search algorithm.

           ds  Diamond search algorithm.

           hexbs
               Hexagon-based search algorithm.

           epzs
               Enhanced predictive zonal search algorithm.

           umh Uneven multi-hexagon search algorithm.

           Default value is esa.

       mb_size
           Macroblock size. Default 16.

       search_param
           Search parameter. Default 7.

   midequalizer
       Apply Midway Image Equalization effect using two video streams.

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

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

       This filter accepts the following option:

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

   minterpolate
       Convert the video to specified frame rate using motion interpolation.

       This filter accepts the following options:

       fps Specify the output frame rate. This can be rational e.g.
           "60000/1001". Frames are dropped if fps is lower than source fps.
           Default 60.

       mi_mode
           Motion interpolation mode. Following values are accepted:

           dup Duplicate previous or next frame for interpolating new ones.

           blend
               Blend source frames. Interpolated frame is mean of previous and
               next frames.

           mci Motion compensated interpolation. Following options are
               effective when this mode is selected:

               mc_mode
                   Motion compensation mode. Following values are accepted:

                   obmc
                       Overlapped block motion compensation.

                   aobmc
                       Adaptive overlapped block motion compensation. Window
                       weighting coefficients are controlled adaptively
                       according to the reliabilities of the neighboring
                       motion vectors to reduce oversmoothing.

                   Default mode is obmc.

               me_mode
                   Motion estimation mode. Following values are accepted:

                   bidir
                       Bidirectional motion estimation. Motion vectors are
                       estimated for each source frame in both forward and
                       backward directions.

                   bilat
                       Bilateral motion estimation. Motion vectors are
                       estimated directly for interpolated frame.

                   Default mode is bilat.

               me  The algorithm to be used for motion estimation. Following
                   values are accepted:

                   esa Exhaustive search algorithm.

                   tss Three step search algorithm.

                   tdls
                       Two dimensional logarithmic search algorithm.

                   ntss
                       New three step search algorithm.

                   fss Four step search algorithm.

                   ds  Diamond search algorithm.

                   hexbs
                       Hexagon-based search algorithm.

                   epzs
                       Enhanced predictive zonal search algorithm.

                   umh Uneven multi-hexagon search algorithm.

                   Default algorithm is epzs.

               mb_size
                   Macroblock size. Default 16.

               search_param
                   Motion estimation search parameter. Default 32.

               vsbmc
                   Enable variable-size block motion compensation. Motion
                   estimation is applied with smaller block sizes at object
                   boundaries in order to make the them less blur. Default is
                   0 (disabled).

       scd Scene change detection method. Scene change leads motion vectors to
           be in random direction. Scene change detection replace interpolated
           frames by duplicate ones. May not be needed for other modes.
           Following values are accepted:

           none
               Disable scene change detection.

           fdiff
               Frame difference. Corresponding pixel values are compared and
               if it satisfies scd_threshold scene change is detected.

           Default method is fdiff.

       scd_threshold
           Scene change detection threshold. Default is 5.0.

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

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

       A description of the accepted options follows.

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

           Default value is 0.

       hi
       lo
       frac
           Set the dropping threshold values.

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

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

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

   negate
       Negate input video.

       It accepts an integer in input; if non-zero it negates the alpha
       component (if available). The default value in input is 0.

   nlmeans
       Denoise frames using Non-Local Means algorithm.

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

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

       The filter accepts the following options.

       s   Set denoising strength.

       p   Set patch size.

       pc  Same as p but for chroma planes.

           The default value is 0 and means automatic.

       r   Set research size.

       rc  Same as r but for chroma planes.

           The default value is 0 and means automatic.

   nnedi
       Deinterlace video using neural network edge directed interpolation.

       This filter accepts the following options:

       weights
           Mandatory option, without binary file filter can not work.
           Currently file can be found here:
           https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

       deint
           Set which frames to deinterlace, by default it is "all".  Can be
           "all" or "interlaced".

       field
           Set mode of operation.

           Can be one of the following:

           af  Use frame flags, both fields.

           a   Use frame flags, single field.

           t   Use top field only.

           b   Use bottom field only.

           tf  Use both fields, top first.

           bf  Use both fields, bottom first.

       planes
           Set which planes to process, by default filter process all frames.

       nsize
           Set size of local neighborhood around each pixel, used by the
           predictor neural network.

           Can be one of the following:

           s8x6
           s16x6
           s32x6
           s48x6
           s8x4
           s16x4
           s32x4
       nns Set the number of neurons in predictor neural network.  Can be one
           of the following:

           n16
           n32
           n64
           n128
           n256
       qual
           Controls the number of different neural network predictions that
           are blended together to compute the final output value. Can be
           "fast", default or "slow".

       etype
           Set which set of weights to use in the predictor.  Can be one of
           the following:

           a   weights trained to minimize absolute error

           s   weights trained to minimize squared error

       pscrn
           Controls whether or not the prescreener neural network is used to
           decide which pixels should be processed by the predictor neural
           network and which can be handled by simple cubic interpolation.
           The prescreener is trained to know whether cubic interpolation will
           be sufficient for a pixel or whether it should be predicted by the
           predictor nn.  The computational complexity of the prescreener nn
           is much less than that of the predictor nn. Since most pixels can
           be handled by cubic interpolation, using the prescreener generally
           results in much faster processing.  The prescreener is pretty
           accurate, so the difference between using it and not using it is
           almost always unnoticeable.

           Can be one of the following:

           none
           original
           new

           Default is "new".

       fapprox
           Set various debugging flags.

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

       It accepts the following parameters:

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

       Examples

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

                   noformat=pix_fmts=yuv420p,vflip

       o   Convert the input video to any of the formats not contained in the
           list:

                   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
           Set noise seed for specific pixel component or all pixel components
           in case of all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
           Set noise strength for specific pixel component or all pixel
           components in case all_strength. Default value is 0. Allowed range
           is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
           Set pixel component flags or set flags for all components if
           all_flags.  Available values for component flags are:

           a   averaged temporal noise (smoother)

           p   mix random noise with a (semi)regular pattern

           t   temporal noise (noise pattern changes between frames)

           u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

               noise=alls=20:allf=t+u

   null
       Pass the video source unchanged to the output.

   ocr
       Optical Character Recognition

       This filter uses Tesseract for optical character recognition.

       It accepts the following options:

       datapath
           Set datapath to tesseract data. Default is to use whatever was set
           at installation.

       language
           Set language, default is "eng".

       whitelist
           Set character whitelist.

       blacklist
           Set character blacklist.

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

   ocv
       Apply a video transform using libopencv.

       To enable this filter, install the libopencv library and headers and
       configure FFmpeg with "--enable-libopencv".

       It accepts the following parameters:

       filter_name
           The name of the libopencv filter to apply.

       filter_params
           The parameters to pass to the libopencv filter. If not specified,
           the default values are assumed.

       Refer to the official libopencv documentation for more precise
       information:
       <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

       Several libopencv filters are supported; see the following subsections.

       dilate

       Dilate an image by using a specific structuring element.  It
       corresponds to the libopencv function "cvDilate".

       It accepts the parameters: struct_el|nb_iterations.

       struct_el represents a structuring element, and has the syntax:
       colsxrows+anchor_xxanchor_y/shape

       cols and rows represent the number of columns and rows of the
       structuring element, anchor_x and anchor_y the anchor point, and shape
       the shape for the structuring element. shape must be "rect", "cross",
       "ellipse", or "custom".

       If the value for shape is "custom", it must be followed by a string of
       the form "=filename". The file with name filename is assumed to
       represent a binary image, with each printable character corresponding
       to a bright pixel. When a custom shape is used, cols and rows are
       ignored, the number or columns and rows of the read file are assumed
       instead.

       The default value for struct_el is "33+0x0/rect".

       nb_iterations specifies the number of times the transform is applied to
       the image, and defaults to 1.

       Some examples:

               # Use the default values
               ocv=dilate

               # Dilate using a structuring element with a 5x5 cross, iterating two times
               ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

               # Read the shape from the file diamond.shape, iterating two times.
               # The file diamond.shape may contain a pattern of characters like this
               #   *
               #  ***
               # *****
               #  ***
               #   *
               # The specified columns and rows are ignored
               # but the anchor point coordinates are not
               ocv=dilate:0x0+2x2/custom=diamond.shape|2

       erode

       Erode an image by using a specific structuring element.  It corresponds
       to the libopencv function "cvErode".

       It accepts the parameters: struct_el:nb_iterations, with the same
       syntax and semantics as the dilate filter.

       smooth

       Smooth the input video.

       The filter takes the following parameters:
       type|param1|param2|param3|param4.

       type is the type of smooth filter to apply, and must be one of the
       following values: "blur", "blur_no_scale", "median", "gaussian", or
       "bilateral". The default value is "gaussian".

       The meaning of param1, param2, param3, and param4 depend on the smooth
       type. param1 and param2 accept integer positive values or 0. param3 and
       param4 accept floating point values.

       The default value for param1 is 3. The default value for the other
       parameters is 0.

       These parameters correspond to the parameters assigned to the libopencv
       function "cvSmooth".

   oscilloscope
       2D Video Oscilloscope.

       Useful to measure spatial impulse, step responses, chroma delays, etc.

       It accepts the following parameters:

       x   Set scope center x position.

       y   Set scope center y position.

       s   Set scope size, relative to frame diagonal.

       t   Set scope tilt/rotation.

       o   Set trace opacity.

       tx  Set trace center x position.

       ty  Set trace center y position.

       tw  Set trace width, relative to width of frame.

       th  Set trace height, relative to height of frame.

       c   Set which components to trace. By default it traces first three
           components.

       g   Draw trace grid. By default is enabled.

       st  Draw some statistics. By default is enabled.

       sc  Draw scope. By default is enabled.

       Examples

       o   Inspect full first row of video frame.

                   oscilloscope=x=0.5:y=0:s=1

       o   Inspect full last row of video frame.

                   oscilloscope=x=0.5:y=1:s=1

       o   Inspect full 5th line of video frame of height 1080.

                   oscilloscope=x=0.5:y=5/1080:s=1

       o   Inspect full last column of video frame.

                   oscilloscope=x=1:y=0.5:s=1:t=1

   overlay
       Overlay one video on top of another.

       It takes two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.

       It accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and y coordinates of the overlaid
           video on the main video. Default value is "0" for both expressions.
           In case the expression is invalid, it is set to a huge value
           (meaning that the overlay will not be displayed within the output
           visible area).

       eof_action
           See framesync.

       eval
           Set when the expressions for x, and y are evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization
               or when a command is processed

           frame
               evaluate expressions for each incoming frame

           Default value is frame.

       shortest
           See framesync.

       format
           Set the format for the output video.

           It accepts the following values:

           yuv420
               force YUV420 output

           yuv422
               force YUV422 output

           yuv444
               force YUV444 output

           rgb force packed RGB output

           gbrp
               force planar RGB output

           auto
               automatically pick format

           Default value is yuv420.

       repeatlast
           See framesync.

       The x, and y expressions can contain the following parameters.

       main_w, W
       main_h, H
           The main input width and height.

       overlay_w, w
       overlay_h, h
           The overlay input width and height.

       x
       y   The computed values for x and y. They are evaluated for each new
           frame.

       hsub
       vsub
           horizontal and vertical chroma subsample values of the output
           format. For example for the pixel format "yuv422p" hsub is 2 and
           vsub is 1.

       n   the number of input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       t   The timestamp, expressed in seconds. It's NAN if the input
           timestamp is unknown.

       This filter also supports the framesync options.

       Note that the n, pos, t variables are available only when evaluation is
       done per frame, and will evaluate to NAN when eval is set to init.

       Be aware that frames are taken from each input video in timestamp
       order, hence, if their initial timestamps differ, it is a good idea to
       pass the two inputs through a setpts=PTS-STARTPTS filter to have them
       begin in the same zero timestamp, as the example for the movie filter
       does.

       You can chain together more overlays but you should test the efficiency
       of such approach.

       Commands

       This filter supports the following commands:

       x
       y   Modify the x and y of the overlay input.  The command accepts the
           same syntax of the corresponding option.

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

       Examples

       o   Draw the overlay at 10 pixels from the bottom right corner of the
           main video:

                   overlay=main_w-overlay_w-10:main_h-overlay_h-10

           Using named options the example above becomes:

                   overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

       o   Insert a transparent PNG logo in the bottom left corner of the
           input, using the ffmpeg tool with the "-filter_complex" option:

                   ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

       o   Insert 2 different transparent PNG logos (second logo on bottom
           right corner) using the ffmpeg tool:

                   ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

       o   Add a transparent color layer on top of the main video; "WxH" must
           specify the size of the main input to the overlay filter:

                   color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

       o   Play an original video and a filtered version (here with the
           deshake filter) side by side using the ffplay tool:

                   ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

           The above command is the same as:

                   ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

       o   Make a sliding overlay appearing from the left to the right top
           part of the screen starting since time 2:

                   overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

       o   Compose output by putting two input videos side to side:

                   ffmpeg -i left.avi -i right.avi -filter_complex "
                   nullsrc=size=200x100 [background];
                   [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
                   [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
                   [background][left]       overlay=shortest=1       [background+left];
                   [background+left][right] overlay=shortest=1:x=100 [left+right]
                   "

       o   Mask 10-20 seconds of a video by applying the delogo filter to a
           section

                   ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
                   -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
                   masked.avi

       o   Chain several overlays in cascade:

                   nullsrc=s=200x200 [bg];
                   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
                   [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
                   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
                   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
                   [in3] null,       [mid2] overlay=100:100 [out0]

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
           Set depth.

           Larger depth values will denoise lower frequency components more,
           but slow down filtering.

           Must be an int in the range 8-16, default is 8.

       luma_strength, ls
           Set luma strength.

           Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
           Set chroma strength.

           Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add paddings to the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following parameters:

       width, w
       height, h
           Specify an expression for the size of the output image with the
           paddings added. If the value for width or height is 0, the
           corresponding input size is used for the output.

           The width expression can reference the value set by the height
           expression, and vice versa.

           The default value of width and height is 0.

       x
       y   Specify the offsets to place the input image at within the padded
           area, with respect to the top/left border of the output image.

           The x expression can reference the value set by the y expression,
           and vice versa.

           The default value of x and y is 0.

           If x or y evaluate to a negative number, they'll be changed so the
           input image is centered on the padded area.

       color
           Specify the color of the padded area. For the syntax of this
           option, check the "Color" section in the ffmpeg-utils manual.

           The default value of color is "black".

       eval
           Specify when to evaluate  width, height, x and y expression.

           It accepts the following values:

           init
               Only evaluate expressions once during the filter initialization
               or when a command is processed.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

       aspect
           Pad to aspect instead to a resolution.

       The value for the width, height, x, and y options are expressions
       containing the following constants:

       in_w
       in_h
           The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output width and height (the size of the padded area), as
           specified by the width and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN
           if not yet specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
           The horizontal and vertical chroma subsample values. For example
           for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       o   Add paddings with the color "violet" to the input video. The output
           video size is 640x480, and the top-left corner of the input video
           is placed at column 0, row 40

                   pad=640:480:0:40:violet

           The example above is equivalent to the following command:

                   pad=width=640:height=480:x=0:y=40:color=violet

       o   Pad the input to get an output with dimensions increased by 3/2,
           and put the input video at the center of the padded area:

                   pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

       o   Pad the input to get a squared output with size equal to the
           maximum value between the input width and height, and put the input
           video at the center of the padded area:

                   pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

       o   Pad the input to get a final w/h ratio of 16:9:

                   pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

       o   In case of anamorphic video, in order to set the output display
           aspect correctly, it is necessary to use sar in the expression,
           according to the relation:

                   (ih * X / ih) * sar = output_dar
                   X = output_dar / sar

           Thus the previous example needs to be modified to:

                   pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

       o   Double the output size and put the input video in the bottom-right
           corner of the output padded area:

                   pad="2*iw:2*ih:ow-iw:oh-ih"

   palettegen
       Generate one palette for a whole video stream.

       It accepts the following options:

       max_colors
           Set the maximum number of colors to quantize in the palette.  Note:
           the palette will still contain 256 colors; the unused palette
           entries will be black.

       reserve_transparent
           Create a palette of 255 colors maximum and reserve the last one for
           transparency. Reserving the transparency color is useful for GIF
           optimization.  If not set, the maximum of colors in the palette
           will be 256. You probably want to disable this option for a
           standalone image.  Set by default.

       stats_mode
           Set statistics mode.

           It accepts the following values:

           full
               Compute full frame histograms.

           diff
               Compute histograms only for the part that differs from previous
               frame. This might be relevant to give more importance to the
               moving part of your input if the background is static.

           single
               Compute new histogram for each frame.

           Default value is full.

       The filter also exports the frame metadata "lavfi.color_quant_ratio"
       ("nb_color_in / nb_color_out") which you can use to evaluate the degree
       of color quantization of the palette. This information is also visible
       at info logging level.

       Examples

       o   Generate a representative palette of a given video using ffmpeg:

                   ffmpeg -i input.mkv -vf palettegen palette.png

   paletteuse
       Use a palette to downsample an input video stream.

       The filter takes two inputs: one video stream and a palette. The
       palette must be a 256 pixels image.

       It accepts the following options:

       dither
           Select dithering mode. Available algorithms are:

           bayer
               Ordered 8x8 bayer dithering (deterministic)

           heckbert
               Dithering as defined by Paul Heckbert in 1982 (simple error
               diffusion).  Note: this dithering is sometimes considered
               "wrong" and is included as a reference.

           floyd_steinberg
               Floyd and Steingberg dithering (error diffusion)

           sierra2
               Frankie Sierra dithering v2 (error diffusion)

           sierra2_4a
               Frankie Sierra dithering v2 "Lite" (error diffusion)

           Default is sierra2_4a.

       bayer_scale
           When bayer dithering is selected, this option defines the scale of
           the pattern (how much the crosshatch pattern is visible). A low
           value means more visible pattern for less banding, and higher value
           means less visible pattern at the cost of more banding.

           The option must be an integer value in the range [0,5]. Default is
           2.

       diff_mode
           If set, define the zone to process

           rectangle
               Only the changing rectangle will be reprocessed. This is
               similar to GIF cropping/offsetting compression mechanism. This
               option can be useful for speed if only a part of the image is
               changing, and has use cases such as limiting the scope of the
               error diffusal dither to the rectangle that bounds the moving
               scene (it leads to more deterministic output if the scene
               doesn't change much, and as a result less moving noise and
               better GIF compression).

           Default is none.

       new Take new palette for each output frame.

       Examples

       o   Use a palette (generated for example with palettegen) to encode a
           GIF using ffmpeg:

                   ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

   perspective
       Correct perspective of video not recorded perpendicular to the screen.

       A description of the accepted parameters follows.

       x0
       y0
       x1
       y1
       x2
       y2
       x3
       y3  Set coordinates expression for top left, top right, bottom left and
           bottom right corners.  Default values are "0:0:W:0:0:H:W:H" with
           which perspective will remain unchanged.  If the "sense" option is
           set to "source", then the specified points will be sent to the
           corners of the destination. If the "sense" option is set to
           "destination", then the corners of the source will be sent to the
           specified coordinates.

           The expressions can use the following variables:

           W
           H   the width and height of video frame.

           in  Input frame count.

           on  Output frame count.

       interpolation
           Set interpolation for perspective correction.

           It accepts the following values:

           linear
           cubic

           Default value is linear.

       sense
           Set interpretation of coordinate options.

           It accepts the following values:

           0, source
               Send point in the source specified by the given coordinates to
               the corners of the destination.

           1, destination
               Send the corners of the source to the point in the destination
               specified by the given coordinates.

               Default value is source.

       eval
           Set when the expressions for coordinates x0,y0,...x3,y3 are
           evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization
               or when a command is processed

           frame
               evaluate expressions for each incoming frame

           Default value is init.

   phase
       Delay interlaced video by one field time so that the field order
       changes.

       The intended use is to fix PAL movies that have been captured with the
       opposite field order to the film-to-video transfer.

       A description of the accepted parameters follows.

       mode
           Set phase mode.

           It accepts the following values:

           t   Capture field order top-first, transfer bottom-first.  Filter
               will delay the bottom field.

           b   Capture field order bottom-first, transfer top-first.  Filter
               will delay the top field.

           p   Capture and transfer with the same field order. This mode only
               exists for the documentation of the other options to refer to,
               but if you actually select it, the filter will faithfully do
               nothing.

           a   Capture field order determined automatically by field flags,
               transfer opposite.  Filter selects among t and b modes on a
               frame by frame basis using field flags. If no field information
               is available, then this works just like u.

           u   Capture unknown or varying, transfer opposite.  Filter selects
               among t and b on a frame by frame basis by analyzing the images
               and selecting the alternative that produces best match between
               the fields.

           T   Capture top-first, transfer unknown or varying.  Filter selects
               among t and p using image analysis.

           B   Capture bottom-first, transfer unknown or varying.  Filter
               selects among b and p using image analysis.

           A   Capture determined by field flags, transfer unknown or varying.
               Filter selects among t, b and p using field flags and image
               analysis. If no field information is available, then this works
               just like U. This is the default mode.

           U   Both capture and transfer unknown or varying.  Filter selects
               among t, b and p using image analysis only.

   pixdesctest
       Pixel format descriptor test filter, mainly useful for internal
       testing. The output video should be equal to the input video.

       For example:

               format=monow, pixdesctest

       can be used to test the monowhite pixel format descriptor definition.

   pixscope
       Display sample values of color channels. Mainly useful for checking
       color and levels. Minimum supported resolution is 640x480.

       The filters accept the following options:

       x   Set scope X position, relative offset on X axis.

       y   Set scope Y position, relative offset on Y axis.

       w   Set scope width.

       h   Set scope height.

       o   Set window opacity. This window also holds statistics about pixel
           area.

       wx  Set window X position, relative offset on X axis.

       wy  Set window Y position, relative offset on Y axis.

   pp
       Enable the specified chain of postprocessing subfilters using
       libpostproc. This library should be automatically selected with a GPL
       build ("--enable-gpl").  Subfilters must be separated by '/' and can be
       disabled by prepending a '-'.  Each subfilter and some options have a
       short and a long name that can be used interchangeably, i.e. dr/dering
       are the same.

       The filters accept the following options:

       subfilters
           Set postprocessing subfilters string.

       All subfilters share common options to determine their scope:

       a/autoq
           Honor the quality commands for this subfilter.

       c/chrom
           Do chrominance filtering, too (default).

       y/nochrom
           Do luminance filtering only (no chrominance).

       n/noluma
           Do chrominance filtering only (no luminance).

       These options can be appended after the subfilter name, separated by a
       '|'.

       Available subfilters are:

       hb/hdeblock[|difference[|flatness]]
           Horizontal deblocking filter

           difference
               Difference factor where higher values mean more deblocking
               (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking
               (default: 39).

       vb/vdeblock[|difference[|flatness]]
           Vertical deblocking filter

           difference
               Difference factor where higher values mean more deblocking
               (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking
               (default: 39).

       ha/hadeblock[|difference[|flatness]]
           Accurate horizontal deblocking filter

           difference
               Difference factor where higher values mean more deblocking
               (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking
               (default: 39).

       va/vadeblock[|difference[|flatness]]
           Accurate vertical deblocking filter

           difference
               Difference factor where higher values mean more deblocking
               (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking
               (default: 39).

       The horizontal and vertical deblocking filters share the difference and
       flatness values so you cannot set different horizontal and vertical
       thresholds.

       h1/x1hdeblock
           Experimental horizontal deblocking filter

       v1/x1vdeblock
           Experimental vertical deblocking filter

       dr/dering
           Deringing filter

       tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
       reducer
           threshold1
               larger -> stronger filtering

           threshold2
               larger -> stronger filtering

           threshold3
               larger -> stronger filtering

       al/autolevels[:f/fullyrange], automatic brightness / contrast
       correction
           f/fullyrange
               Stretch luminance to "0-255".

       lb/linblenddeint
           Linear blend deinterlacing filter that deinterlaces the given block
           by filtering all lines with a "(1 2 1)" filter.

       li/linipoldeint
           Linear interpolating deinterlacing filter that deinterlaces the
           given block by linearly interpolating every second line.

       ci/cubicipoldeint
           Cubic interpolating deinterlacing filter deinterlaces the given
           block by cubically interpolating every second line.

       md/mediandeint
           Median deinterlacing filter that deinterlaces the given block by
           applying a median filter to every second line.

       fd/ffmpegdeint
           FFmpeg deinterlacing filter that deinterlaces the given block by
           filtering every second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
           Vertically applied FIR lowpass deinterlacing filter that
           deinterlaces the given block by filtering all lines with a "(-1 2 6
           2 -1)" filter.

       fq/forceQuant[|quantizer]
           Overrides the quantizer table from the input with the constant
           quantizer you specify.

           quantizer
               Quantizer to use

       de/default
           Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
           Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       o   Apply horizontal and vertical deblocking, deringing and automatic
           brightness/contrast:

                   pp=hb/vb/dr/al

       o   Apply default filters without brightness/contrast correction:

                   pp=de/-al

       o   Apply default filters and temporal denoiser:

                   pp=default/tmpnoise|1|2|3

       o   Apply deblocking on luminance only, and switch vertical deblocking
           on or off automatically depending on available CPU time:

                   pp=hb|y/vb|a

   pp7
       Apply Postprocessing filter 7. It is variant of the spp filter, similar
       to spp = 6 with 7 point DCT, where only the center sample is used after
       IDCT.

       The filter accepts the following options:

       qp  Force a constant quantization parameter. It accepts an integer in
           range 0 to 63. If not set, the filter will use the QP from the
           video stream (if available).

       mode
           Set thresholding mode. Available modes are:

           hard
               Set hard thresholding.

           soft
               Set soft thresholding (better de-ringing effect, but likely
               blurrier).

           medium
               Set medium thresholding (good results, default).

   premultiply
       Apply alpha premultiply effect to input video stream using first plane
       of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

       inplace
           Do not require 2nd input for processing, instead use alpha plane
           from input stream.

   prewitt
       Apply prewitt operator to input video stream.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

       scale
           Set value which will be multiplied with filtered result.

       delta
           Set value which will be added to filtered result.

   pseudocolor
       Alter frame colors in video with pseudocolors.

       This filter accept the following options:

       c0  set pixel first component expression

       c1  set pixel second component expression

       c2  set pixel third component expression

       c3  set pixel fourth component expression, corresponds to the alpha
           component

       i   set component to use as base for altering colors

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

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       ymin, umin, vmin, amin
           The minimum allowed component value.

       ymax, umax, vmax, amax
           The maximum allowed component value.

       All expressions default to "val".

       Examples

       o   Change too high luma values to gradient:

                   pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"

   psnr
       Obtain the average, maximum and minimum PSNR (Peak Signal to Noise
       Ratio) between two input videos.

       This filter takes in input two input videos, the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the PSNR.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average PSNR is printed through the logging system.

       The filter stores the accumulated MSE (mean squared error) of each
       frame, and at the end of the processing it is averaged across all
       frames equally, and the following formula is applied to obtain the
       PSNR:

               PSNR = 10*log10(MAX^2/MSE)

       Where MAX is the average of the maximum values of each component of the
       image.

       The description of the accepted parameters follows.

       stats_file, f
           If specified the filter will use the named file to save the PSNR of
           each individual frame. When filename equals "-" the data is sent to
           standard output.

       stats_version
           Specifies which version of the stats file format to use. Details of
           each format are written below.  Default value is 1.

       stats_add_max
           Determines whether the max value is output to the stats log.
           Default value is 0.  Requires stats_version >= 2. If this is set
           and stats_version < 2, the filter will return an error.

       This filter also supports the framesync options.

       The file printed if stats_file is selected, contains a sequence of
       key/value pairs of the form key:value for each compared couple of
       frames.

       If a stats_version greater than 1 is specified, a header line precedes
       the list of per-frame-pair stats, with key value pairs following the
       frame format with the following parameters:

       psnr_log_version
           The version of the log file format. Will match stats_version.

       fields
           A comma separated list of the per-frame-pair parameters included in
           the log.

       A description of each shown per-frame-pair parameter follows:

       n   sequential number of the input frame, starting from 1

       mse_avg
           Mean Square Error pixel-by-pixel average difference of the compared
           frames, averaged over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
           Mean Square Error pixel-by-pixel average difference of the compared
           frames for the component specified by the suffix.

       psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
           Peak Signal to Noise ratio of the compared frames for the component
           specified by the suffix.

       max_avg, max_y, max_u, max_v
           Maximum allowed value for each channel, and average over all
           channels.

       For example:

               movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
               [main][ref] psnr="stats_file=stats.log" [out]

       On this example the input file being processed is compared with the
       reference file ref_movie.mpg. The PSNR of each individual frame is
       stored in stats.log.

   pullup
       Pulldown reversal (inverse telecine) filter, capable of handling mixed
       hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps
       progressive content.

       The pullup filter is designed to take advantage of future context in
       making its decisions. This filter is stateless in the sense that it
       does not lock onto a pattern to follow, but it instead looks forward to
       the following fields in order to identify matches and rebuild
       progressive frames.

       To produce content with an even framerate, insert the fps filter after
       pullup, use "fps=24000/1001" if the input frame rate is 29.97fps,
       "fps=24" for 30fps and the (rare) telecined 25fps input.

       The filter accepts the following options:

       jl
       jr
       jt
       jb  These options set the amount of "junk" to ignore at the left,
           right, top, and bottom of the image, respectively. Left and right
           are in units of 8 pixels, while top and bottom are in units of 2
           lines.  The default is 8 pixels on each side.

       sb  Set the strict breaks. Setting this option to 1 will reduce the
           chances of filter generating an occasional mismatched frame, but it
           may also cause an excessive number of frames to be dropped during
           high motion sequences.  Conversely, setting it to -1 will make
           filter match fields more easily.  This may help processing of video
           where there is slight blurring between the fields, but may also
           cause there to be interlaced frames in the output.  Default value
           is 0.

       mp  Set the metric plane to use. It accepts the following values:

           l   Use luma plane.

           u   Use chroma blue plane.

           v   Use chroma red plane.

           This option may be set to use chroma plane instead of the default
           luma plane for doing filter's computations. This may improve
           accuracy on very clean source material, but more likely will
           decrease accuracy, especially if there is chroma noise (rainbow
           effect) or any grayscale video.  The main purpose of setting mp to
           a chroma plane is to reduce CPU load and make pullup usable in
           realtime on slow machines.

       For best results (without duplicated frames in the output file) it is
       necessary to change the output frame rate. For example, to inverse
       telecine NTSC input:

               ffmpeg -i input -vf pullup -r 24000/1001 ...

   qp
       Change video quantization parameters (QP).

       The filter accepts the following option:

       qp  Set expression for quantization parameter.

       The expression is evaluated through the eval API and can contain, among
       others, the following constants:

       known
           1 if index is not 129, 0 otherwise.

       qp  Sequential index starting from -129 to 128.

       Examples

       o   Some equation like:

                   qp=2+2*sin(PI*qp)

   random
       Flush video frames from internal cache of frames into a random order.
       No frame is discarded.  Inspired by frei0r nervous filter.

       frames
           Set size in number of frames of internal cache, in range from 2 to
           512. Default is 30.

       seed
           Set seed for random number generator, must be an integer included
           between 0 and "UINT32_MAX". If not specified, or if explicitly set
           to less than 0, the filter will try to use a good random seed on a
           best effort basis.

   readeia608
       Read closed captioning (EIA-608) information from the top lines of a
       video frame.

       This filter adds frame metadata for "lavfi.readeia608.X.cc" and
       "lavfi.readeia608.X.line", where "X" is the number of the identified
       line with EIA-608 data (starting from 0). A description of each
       metadata value follows:

       lavfi.readeia608.X.cc
           The two bytes stored as EIA-608 data (printed in hexadecimal).

       lavfi.readeia608.X.line
           The number of the line on which the EIA-608 data was identified and
           read.

       This filter accepts the following options:

       scan_min
           Set the line to start scanning for EIA-608 data. Default is 0.

       scan_max
           Set the line to end scanning for EIA-608 data. Default is 29.

       mac Set minimal acceptable amplitude change for sync codes detection.
           Default is 0.2. Allowed range is "[0.001 - 1]".

       spw Set the ratio of width reserved for sync code detection.  Default
           is 0.27. Allowed range is "[0.01 - 0.7]".

       mhd Set the max peaks height difference for sync code detection.
           Default is 0.1. Allowed range is "[0.0 - 0.5]".

       mpd Set max peaks period difference for sync code detection.  Default
           is 0.1. Allowed range is "[0.0 - 0.5]".

       msd Set the first two max start code bits differences.  Default is
           0.02. Allowed range is "[0.0 - 0.5]".

       bhd Set the minimum ratio of bits height compared to 3rd start code
           bit.  Default is 0.75. Allowed range is "[0.01 - 1]".

       th_w
           Set the white color threshold. Default is 0.35. Allowed range is
           "[0.1 - 1]".

       th_b
           Set the black color threshold. Default is 0.15. Allowed range is
           "[0.0 - 0.5]".

       chp Enable checking the parity bit. In the event of a parity error, the
           filter will output 0x00 for that character. Default is false.

       Examples

       o   Output a csv with presentation time and the first two lines of
           identified EIA-608 captioning data.

                   ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv

   readvitc
       Read vertical interval timecode (VITC) information from the top lines
       of a video frame.

       The filter adds frame metadata key "lavfi.readvitc.tc_str" with the
       timecode value, if a valid timecode has been detected. Further metadata
       key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
       data has been found or not.

       This filter accepts the following options:

       scan_max
           Set the maximum number of lines to scan for VITC data. If the value
           is set to "-1" the full video frame is scanned. Default is 45.

       thr_b
           Set the luma threshold for black. Accepts float numbers in the
           range [0.0,1.0], default value is 0.2. The value must be equal or
           less than "thr_w".

       thr_w
           Set the luma threshold for white. Accepts float numbers in the
           range [0.0,1.0], default value is 0.6. The value must be equal or
           greater than "thr_b".

       Examples

       o   Detect and draw VITC data onto the video frame; if no valid VITC is
           detected, draw "--:--:--:--" as a placeholder:

                   ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

   remap
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x, y)
       position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
       out of range, zero value for pixel will be used for destination pixel.

       Xmap and Ymap input video streams must be of same dimensions. Output
       video stream will have Xmap/Ymap video stream dimensions.  Xmap and
       Ymap input video streams are 16bit depth, single channel.

   removegrain
       The removegrain filter is a spatial denoiser for progressive video.

       m0  Set mode for the first plane.

       m1  Set mode for the second plane.

       m2  Set mode for the third plane.

       m3  Set mode for the fourth plane.

       Range of mode is from 0 to 24. Description of each mode follows:

       0   Leave input plane unchanged. Default.

       1   Clips the pixel with the minimum and maximum of the 8 neighbour
           pixels.

       2   Clips the pixel with the second minimum and maximum of the 8
           neighbour pixels.

       3   Clips the pixel with the third minimum and maximum of the 8
           neighbour pixels.

       4   Clips the pixel with the fourth minimum and maximum of the 8
           neighbour pixels.  This is equivalent to a median filter.

       5   Line-sensitive clipping giving the minimal change.

       6   Line-sensitive clipping, intermediate.

       7   Line-sensitive clipping, intermediate.

       8   Line-sensitive clipping, intermediate.

       9   Line-sensitive clipping on a line where the neighbours pixels are
           the closest.

       10  Replaces the target pixel with the closest neighbour.

       11  [1 2 1] horizontal and vertical kernel blur.

       12  Same as mode 11.

       13  Bob mode, interpolates top field from the line where the neighbours
           pixels are the closest.

       14  Bob mode, interpolates bottom field from the line where the
           neighbours pixels are the closest.

       15  Bob mode, interpolates top field. Same as 13 but with a more
           complicated interpolation formula.

       16  Bob mode, interpolates bottom field. Same as 14 but with a more
           complicated interpolation formula.

       17  Clips the pixel with the minimum and maximum of respectively the
           maximum and minimum of each pair of opposite neighbour pixels.

       18  Line-sensitive clipping using opposite neighbours whose greatest
           distance from the current pixel is minimal.

       19  Replaces the pixel with the average of its 8 neighbours.

       20  Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

       21  Clips pixels using the averages of opposite neighbour.

       22  Same as mode 21 but simpler and faster.

       23  Small edge and halo removal, but reputed useless.

       24  Similar as 23.

   removelogo
       Suppress a TV station logo, using an image file to determine which
       pixels comprise the logo. It works by filling in the pixels that
       comprise the logo with neighboring pixels.

       The filter accepts the following options:

       filename, f
           Set the filter bitmap file, which can be any image format supported
           by libavformat. The width and height of the image file must match
           those of the video stream being processed.

       Pixels in the provided bitmap image with a value of zero are not
       considered part of the logo, non-zero pixels are considered part of the
       logo. If you use white (255) for the logo and black (0) for the rest,
       you will be safe. For making the filter bitmap, it is recommended to
       take a screen capture of a black frame with the logo visible, and then
       using a threshold filter followed by the erode filter once or twice.

       If needed, little splotches can be fixed manually. Remember that if
       logo pixels are not covered, the filter quality will be much reduced.
       Marking too many pixels as part of the logo does not hurt as much, but
       it will increase the amount of blurring needed to cover over the image
       and will destroy more information than necessary, and extra pixels will
       slow things down on a large logo.

   repeatfields
       This filter uses the repeat_field flag from the Video ES headers and
       hard repeats fields based on its value.

   reverse
       Reverse a video clip.

       Warning: This filter requires memory to buffer the entire clip, so
       trimming is suggested.

       Examples

       o   Take the first 5 seconds of a clip, and reverse it.

                   trim=end=5,reverse

   roberts
       Apply roberts cross operator to input video stream.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

       scale
           Set value which will be multiplied with filtered result.

       delta
           Set value which will be added to filtered result.

   rotate
       Rotate video by an arbitrary angle expressed in radians.

       The filter accepts the following options:

       A description of the optional parameters follows.

       angle, a
           Set an expression for the angle by which to rotate the input video
           clockwise, expressed as a number of radians. A negative value will
           result in a counter-clockwise rotation. By default it is set to
           "0".

           This expression is evaluated for each frame.

       out_w, ow
           Set the output width expression, default value is "iw".  This
           expression is evaluated just once during configuration.

       out_h, oh
           Set the output height expression, default value is "ih".  This
           expression is evaluated just once during configuration.

       bilinear
           Enable bilinear interpolation if set to 1, a value of 0 disables
           it. Default value is 1.

       fillcolor, c
           Set the color used to fill the output area not covered by the
           rotated image. For the general syntax of this option, check the
           "Color" section in the ffmpeg-utils manual. If the special value
           "none" is selected then no background is printed (useful for
           example if the background is never shown).

           Default value is "black".

       The expressions for the angle and the output size can contain the
       following constants and functions:

       n   sequential number of the input frame, starting from 0. It is always
           NAN before the first frame is filtered.

       t   time in seconds of the input frame, it is set to 0 when the filter
           is configured. It is always NAN before the first frame is filtered.

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for
           the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_w, iw
       in_h, ih
           the input video width and height

       out_w, ow
       out_h, oh
           the output width and height, that is the size of the padded area as
           specified by the width and height expressions

       rotw(a)
       roth(a)
           the minimal width/height required for completely containing the
           input video rotated by a radians.

           These are only available when computing the out_w and out_h
           expressions.

       Examples

       o   Rotate the input by PI/6 radians clockwise:

                   rotate=PI/6

       o   Rotate the input by PI/6 radians counter-clockwise:

                   rotate=-PI/6

       o   Rotate the input by 45 degrees clockwise:

                   rotate=45*PI/180

       o   Apply a constant rotation with period T, starting from an angle of
           PI/3:

                   rotate=PI/3+2*PI*t/T

       o   Make the input video rotation oscillating with a period of T
           seconds and an amplitude of A radians:

                   rotate=A*sin(2*PI/T*t)

       o   Rotate the video, output size is chosen so that the whole rotating
           input video is always completely contained in the output:

                   rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

       o   Rotate the video, reduce the output size so that no background is
           ever shown:

                   rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

       Commands

       The filter supports the following commands:

       a, angle
           Set the angle expression.  The command accepts the same syntax of
           the corresponding option.

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

   sab
       Apply Shape Adaptive Blur.

       The filter accepts the following options:

       luma_radius, lr
           Set luma blur filter strength, must be a value in range 0.1-4.0,
           default value is 1.0. A greater value will result in a more blurred
           image, and in slower processing.

       luma_pre_filter_radius, lpfr
           Set luma pre-filter radius, must be a value in the 0.1-2.0 range,
           default value is 1.0.

       luma_strength, ls
           Set luma maximum difference between pixels to still be considered,
           must be a value in the 0.1-100.0 range, default value is 1.0.

       chroma_radius, cr
           Set chroma blur filter strength, must be a value in range -0.9-4.0.
           A greater value will result in a more blurred image, and in slower
           processing.

       chroma_pre_filter_radius, cpfr
           Set chroma pre-filter radius, must be a value in the -0.9-2.0
           range.

       chroma_strength, cs
           Set chroma maximum difference between pixels to still be
           considered, must be a value in the -0.9-100.0 range.

       Each chroma option value, if not explicitly specified, is set to the
       corresponding luma option value.

   scale
       Scale (resize) the input video, using the libswscale library.

       The scale filter forces the output display aspect ratio to be the same
       of the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the scale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options, or any of the options
       supported by the libswscale scaler.

       See the ffmpeg-scaler manual for the complete list of scaler options.

       width, w
       height, h
           Set the output video dimension expression. Default value is the
           input dimension.

           If the width or w value is 0, the input width is used for the
           output. If the height or h value is 0, the input height is used for
           the output.

           If one and only one of the values is -n with n >= 1, the scale
           filter will use a value that maintains the aspect ratio of the
           input image, calculated from the other specified dimension. After
           that it will, however, make sure that the calculated dimension is
           divisible by n and adjust the value if necessary.

           If both values are -n with n >= 1, the behavior will be identical
           to both values being set to 0 as previously detailed.

           See below for the list of accepted constants for use in the
           dimension expression.

       eval
           Specify when to evaluate width and height expression. It accepts
           the following values:

           init
               Only evaluate expressions once during the filter initialization
               or when a command is processed.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

       interl
           Set the interlacing mode. It accepts the following values:

           1   Force interlaced aware scaling.

           0   Do not apply interlaced scaling.

           -1  Select interlaced aware scaling depending on whether the source
               frames are flagged as interlaced or not.

           Default value is 0.

       flags
           Set libswscale scaling flags. See the ffmpeg-scaler manual for the
           complete list of values. If not explicitly specified the filter
           applies the default flags.

       param0, param1
           Set libswscale input parameters for scaling algorithms that need
           them. See the ffmpeg-scaler manual for the complete documentation.
           If not explicitly specified the filter applies empty parameters.

       size, s
           Set the video size. For the syntax of this option, check the "Video
           size" section in the ffmpeg-utils manual.

       in_color_matrix
       out_color_matrix
           Set in/output YCbCr color space type.

           This allows the autodetected value to be overridden as well as
           allows forcing a specific value used for the output and encoder.

           If not specified, the color space type depends on the pixel format.

           Possible values:

           auto
               Choose automatically.

           bt709
               Format conforming to International Telecommunication Union
               (ITU) Recommendation BT.709.

           fcc Set color space conforming to the United States Federal
               Communications Commission (FCC) Code of Federal Regulations
               (CFR) Title 47 (2003) 73.682 (a).

           bt601
               Set color space conforming to:

               o   ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

               o   ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

               o   Society of Motion Picture and Television Engineers (SMPTE)
                   ST 170:2004

           smpte240m
               Set color space conforming to SMPTE ST 240:1999.

       in_range
       out_range
           Set in/output YCbCr sample range.

           This allows the autodetected value to be overridden as well as
           allows forcing a specific value used for the output and encoder. If
           not specified, the range depends on the pixel format. Possible
           values:

           auto
               Choose automatically.

           jpeg/full/pc
               Set full range (0-255 in case of 8-bit luma).

           mpeg/tv
               Set "MPEG" range (16-235 in case of 8-bit luma).

       force_original_aspect_ratio
           Enable decreasing or increasing output video width or height if
           necessary to keep the original aspect ratio. Possible values:

           disable
               Scale the video as specified and disable this feature.

           decrease
               The output video dimensions will automatically be decreased if
               needed.

           increase
               The output video dimensions will automatically be increased if
               needed.

           One useful instance of this option is that when you know a specific
           device's maximum allowed resolution, you can use this to limit the
           output video to that, while retaining the aspect ratio. For
           example, device A allows 1280x720 playback, and your video is
           1920x800. Using this option (set it to decrease) and specifying
           1280x720 to the command line makes the output 1280x533.

           Please note that this is a different thing than specifying -1 for w
           or h, you still need to specify the output resolution for this
           option to work.

       The values of the w and h options are expressions containing the
       following constants:

       in_w
       in_h
           The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
           horizontal and vertical input chroma subsample values. For example
           for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
           horizontal and vertical output chroma subsample values. For example
           for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       o   Scale the input video to a size of 200x100

                   scale=w=200:h=100

           This is equivalent to:

                   scale=200:100

           or:

                   scale=200x100

       o   Specify a size abbreviation for the output size:

                   scale=qcif

           which can also be written as:

                   scale=size=qcif

       o   Scale the input to 2x:

                   scale=w=2*iw:h=2*ih

       o   The above is the same as:

                   scale=2*in_w:2*in_h

       o   Scale the input to 2x with forced interlaced scaling:

                   scale=2*iw:2*ih:interl=1

       o   Scale the input to half size:

                   scale=w=iw/2:h=ih/2

       o   Increase the width, and set the height to the same size:

                   scale=3/2*iw:ow

       o   Seek Greek harmony:

                   scale=iw:1/PHI*iw
                   scale=ih*PHI:ih

       o   Increase the height, and set the width to 3/2 of the height:

                   scale=w=3/2*oh:h=3/5*ih

       o   Increase the size, making the size a multiple of the chroma
           subsample values:

                   scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

       o   Increase the width to a maximum of 500 pixels, keeping the same
           aspect ratio as the input:

                   scale=w='min(500\, iw*3/2):h=-1'

       Commands

       This filter supports the following commands:

       width, w
       height, h
           Set the output video dimension expression.  The command accepts the
           same syntax of the corresponding option.

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

   scale_npp
       Use the NVIDIA Performance Primitives (libnpp) to perform scaling
       and/or pixel format conversion on CUDA video frames. Setting the output
       width and height works in the same way as for the scale filter.

       The following additional options are accepted:

       format
           The pixel format of the output CUDA frames. If set to the string
           "same" (the default), the input format will be kept. Note that
           automatic format negotiation and conversion is not yet supported
           for hardware frames

       interp_algo
           The interpolation algorithm used for resizing. One of the
           following:

           nn  Nearest neighbour.

           linear
           cubic
           cubic2p_bspline
               2-parameter cubic (B=1, C=0)

           cubic2p_catmullrom
               2-parameter cubic (B=0, C=1/2)

           cubic2p_b05c03
               2-parameter cubic (B=1/2, C=3/10)

           super
               Supersampling

           lanczos

   scale2ref
       Scale (resize) the input video, based on a reference video.

       See the scale filter for available options, scale2ref supports the same
       but uses the reference video instead of the main input as basis.
       scale2ref also supports the following additional constants for the w
       and h options:

       main_w
       main_h
           The main input video's width and height

       main_a
           The same as main_w / main_h

       main_sar
           The main input video's sample aspect ratio

       main_dar, mdar
           The main input video's display aspect ratio. Calculated from
           "(main_w / main_h) * main_sar".

       main_hsub
       main_vsub
           The main input video's horizontal and vertical chroma subsample
           values.  For example for the pixel format "yuv422p" hsub is 2 and
           vsub is 1.

       Examples

       o   Scale a subtitle stream (b) to match the main video (a) in size
           before overlaying

                   'scale2ref[b][a];[a][b]overlay'

   selectivecolor
       Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of
       colors (such as "reds", "yellows", "greens", "cyans", ...). The
       adjustment range is defined by the "purity" of the color (that is, how
       saturated it already is).

       This filter is similar to the Adobe Photoshop Selective Color tool.

       The filter accepts the following options:

       correction_method
           Select color correction method.

           Available values are:

           absolute
               Specified adjustments are applied "as-is" (added/subtracted to
               original pixel component value).

           relative
               Specified adjustments are relative to the original component
               value.

           Default is "absolute".

       reds
           Adjustments for red pixels (pixels where the red component is the
           maximum)

       yellows
           Adjustments for yellow pixels (pixels where the blue component is
           the minimum)

       greens
           Adjustments for green pixels (pixels where the green component is
           the maximum)

       cyans
           Adjustments for cyan pixels (pixels where the red component is the
           minimum)

       blues
           Adjustments for blue pixels (pixels where the blue component is the
           maximum)

       magentas
           Adjustments for magenta pixels (pixels where the green component is
           the minimum)

       whites
           Adjustments for white pixels (pixels where all components are
           greater than 128)

       neutrals
           Adjustments for all pixels except pure black and pure white

       blacks
           Adjustments for black pixels (pixels where all components are
           lesser than 128)

       psfile
           Specify a Photoshop selective color file (".asv") to import the
           settings from.

       All the adjustment settings (reds, yellows, ...) accept up to 4 space
       separated floating point adjustment values in the [-1,1] range,
       respectively to adjust the amount of cyan, magenta, yellow and black
       for the pixels of its range.

       Examples

       o   Increase cyan by 50% and reduce yellow by 33% in every green areas,
           and increase magenta by 27% in blue areas:

                   selectivecolor=greens=.5 0 -.33 0:blues=0 .27

       o   Use a Photoshop selective color preset:

                   selectivecolor=psfile=MySelectiveColorPresets/Misty.asv

   separatefields
       The "separatefields" takes a frame-based video input and splits each
       frame into its components fields, producing a new half height clip with
       twice the frame rate and twice the frame count.

       This filter use field-dominance information in frame to decide which of
       each pair of fields to place first in the output.  If it gets it wrong
       use setfield filter before "separatefields" filter.

   setdar, setsar
       The "setdar" filter sets the Display Aspect Ratio for the filter output
       video.

       This is done by changing the specified Sample (aka Pixel) Aspect Ratio,
       according to the following equation:

               <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

       Keep in mind that the "setdar" filter does not modify the pixel
       dimensions of the video frame. Also, the display aspect ratio set by
       this filter may be changed by later filters in the filterchain, e.g. in
       case of scaling or if another "setdar" or a "setsar" filter is applied.

       The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the
       filter output video.

       Note that as a consequence of the application of this filter, the
       output display aspect ratio will change according to the equation
       above.

       Keep in mind that the sample aspect ratio set by the "setsar" filter
       may be changed by later filters in the filterchain, e.g. if another
       "setsar" or a "setdar" filter is applied.

       It accepts the following parameters:

       r, ratio, dar ("setdar" only), sar ("setsar" only)
           Set the aspect ratio used by the filter.

           The parameter can be a floating point number string, an expression,
           or a string of the form num:den, where num and den are the
           numerator and denominator of the aspect ratio. If the parameter is
           not specified, it is assumed the value "0".  In case the form
           "num:den" is used, the ":" character should be escaped.

       max Set the maximum integer value to use for expressing numerator and
           denominator when reducing the expressed aspect ratio to a rational.
           Default value is 100.

       The parameter sar is an expression containing the following constants:

       E, PI, PHI
           These are approximated values for the mathematical constants e
           (Euler's number), pi (Greek pi), and phi (the golden ratio).

       w, h
           The input width and height.

       a   These are the same as w / h.

       sar The input sample aspect ratio.

       dar The input display aspect ratio. It is the same as (w / h) * sar.

       hsub, vsub
           Horizontal and vertical chroma subsample values. For example, for
           the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       o   To change the display aspect ratio to 16:9, specify one of the
           following:

                   setdar=dar=1.77777
                   setdar=dar=16/9

       o   To change the sample aspect ratio to 10:11, specify:

                   setsar=sar=10/11

       o   To set a display aspect ratio of 16:9, and specify a maximum
           integer value of 1000 in the aspect ratio reduction, use the
           command:

                   setdar=ratio=16/9:max=1000

   setfield
       Force field for the output video frame.

       The "setfield" filter marks the interlace type field for the output
       frames. It does not change the input frame, but only sets the
       corresponding property, which affects how the frame is treated by
       following filters (e.g. "fieldorder" or "yadif").

       The filter accepts the following options:

       mode
           Available values are:

           auto
               Keep the same field property.

           bff Mark the frame as bottom-field-first.

           tff Mark the frame as top-field-first.

           prog
               Mark the frame as progressive.

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

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

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The Presentation TimeStamp of the input frame, expressed as a
           number of time base units. The time base unit depends on the filter
           input pad.

       pts_time
           The Presentation TimeStamp of the input frame, expressed as a
           number of seconds.

       pos The position of the frame in the input stream, or -1 if this
           information is unavailable and/or meaningless (for example in case
           of synthetic video).

       fmt The pixel format name.

       sar The sample aspect ratio of the input frame, expressed in the form
           num/den.

       s   The size of the input frame. For the syntax of this option, check
           the "Video size" section in the ffmpeg-utils manual.

       i   The type of interlaced mode ("P" for "progressive", "T" for top
           field first, "B" for bottom field first).

       iskey
           This is 1 if the frame is a key frame, 0 otherwise.

       type
           The picture type of the input frame ("I" for an I-frame, "P" for a
           P-frame, "B" for a B-frame, or "?" for an unknown type).  Also
           refer to the documentation of the "AVPictureType" enum and of the
           "av_get_picture_type_char" function defined in libavutil/avutil.h.

       checksum
           The Adler-32 checksum (printed in hexadecimal) of all the planes of
           the input frame.

       plane_checksum
           The Adler-32 checksum (printed in hexadecimal) of each plane of the
           input frame, expressed in the form "[c0 c1 c2 c3]".

   showpalette
       Displays the 256 colors palette of each frame. This filter is only
       relevant for pal8 pixel format frames.

       It accepts the following option:

       s   Set the size of the box used to represent one palette color entry.
           Default is 30 (for a "30x30" pixel box).

   shuffleframes
       Reorder and/or duplicate and/or drop video frames.

       It accepts the following parameters:

       mapping
           Set the destination indexes of input frames.  This is space or '|'
           separated list of indexes that maps input frames to output frames.
           Number of indexes also sets maximal value that each index may have.
           '-1' index have special meaning and that is to drop frame.

       The first frame has the index 0. The default is to keep the input
       unchanged.

       Examples

       o   Swap second and third frame of every three frames of the input:

                   ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT

       o   Swap 10th and 1st frame of every ten frames of the input:

                   ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT

   shuffleplanes
       Reorder and/or duplicate video planes.

       It accepts the following parameters:

       map0
           The index of the input plane to be used as the first output plane.

       map1
           The index of the input plane to be used as the second output plane.

       map2
           The index of the input plane to be used as the third output plane.

       map3
           The index of the input plane to be used as the fourth output plane.

       The first plane has the index 0. The default is to keep the input
       unchanged.

       Examples

       o   Swap the second and third planes of the input:

                   ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

   signalstats
       Evaluate various visual metrics that assist in determining issues
       associated with the digitization of analog video media.

       By default the filter will log these metadata values:

       YMIN
           Display the minimal Y value contained within the input frame.
           Expressed in range of [0-255].

       YLOW
           Display the Y value at the 10% percentile within the input frame.
           Expressed in range of [0-255].

       YAVG
           Display the average Y value within the input frame. Expressed in
           range of [0-255].

       YHIGH
           Display the Y value at the 90% percentile within the input frame.
           Expressed in range of [0-255].

       YMAX
           Display the maximum Y value contained within the input frame.
           Expressed in range of [0-255].

       UMIN
           Display the minimal U value contained within the input frame.
           Expressed in range of [0-255].

       ULOW
           Display the U value at the 10% percentile within the input frame.
           Expressed in range of [0-255].

       UAVG
           Display the average U value within the input frame. Expressed in
           range of [0-255].

       UHIGH
           Display the U value at the 90% percentile within the input frame.
           Expressed in range of [0-255].

       UMAX
           Display the maximum U value contained within the input frame.
           Expressed in range of [0-255].

       VMIN
           Display the minimal V value contained within the input frame.
           Expressed in range of [0-255].

       VLOW
           Display the V value at the 10% percentile within the input frame.
           Expressed in range of [0-255].

       VAVG
           Display the average V value within the input frame. Expressed in
           range of [0-255].

       VHIGH
           Display the V value at the 90% percentile within the input frame.
           Expressed in range of [0-255].

       VMAX
           Display the maximum V value contained within the input frame.
           Expressed in range of [0-255].

       SATMIN
           Display the minimal saturation value contained within the input
           frame.  Expressed in range of [0-~181.02].

       SATLOW
           Display the saturation value at the 10% percentile within the input
           frame.  Expressed in range of [0-~181.02].

       SATAVG
           Display the average saturation value within the input frame.
           Expressed in range of [0-~181.02].

       SATHIGH
           Display the saturation value at the 90% percentile within the input
           frame.  Expressed in range of [0-~181.02].

       SATMAX
           Display the maximum saturation value contained within the input
           frame.  Expressed in range of [0-~181.02].

       HUEMED
           Display the median value for hue within the input frame. Expressed
           in range of [0-360].

       HUEAVG
           Display the average value for hue within the input frame. Expressed
           in range of [0-360].

       YDIF
           Display the average of sample value difference between all values
           of the Y plane in the current frame and corresponding values of the
           previous input frame.  Expressed in range of [0-255].

       UDIF
           Display the average of sample value difference between all values
           of the U plane in the current frame and corresponding values of the
           previous input frame.  Expressed in range of [0-255].

       VDIF
           Display the average of sample value difference between all values
           of the V plane in the current frame and corresponding values of the
           previous input frame.  Expressed in range of [0-255].

       YBITDEPTH
           Display bit depth of Y plane in current frame.  Expressed in range
           of [0-16].

       UBITDEPTH
           Display bit depth of U plane in current frame.  Expressed in range
           of [0-16].

       VBITDEPTH
           Display bit depth of V plane in current frame.  Expressed in range
           of [0-16].

       The filter accepts the following options:

       stat
       out stat specify an additional form of image analysis.  out output
           video with the specified type of pixel highlighted.

           Both options accept the following values:

           tout
               Identify temporal outliers pixels. A temporal outlier is a
               pixel unlike the neighboring pixels of the same field. Examples
               of temporal outliers include the results of video dropouts,
               head clogs, or tape tracking issues.

           vrep
               Identify vertical line repetition. Vertical line repetition
               includes similar rows of pixels within a frame. In born-digital
               video vertical line repetition is common, but this pattern is
               uncommon in video digitized from an analog source. When it
               occurs in video that results from the digitization of an analog
               source it can indicate concealment from a dropout compensator.

           brng
               Identify pixels that fall outside of legal broadcast range.

       color, c
           Set the highlight color for the out option. The default color is
           yellow.

       Examples

       o   Output data of various video metrics:

                   ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

       o   Output specific data about the minimum and maximum values of the Y
           plane per frame:

                   ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

       o   Playback video while highlighting pixels that are outside of
           broadcast range in red.

                   ffplay example.mov -vf signalstats="out=brng:color=red"

       o   Playback video with signalstats metadata drawn over the frame.

                   ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

           The contents of signalstat_drawtext.txt used in the command are:

                   time %{pts:hms}
                   Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
                   U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
                   V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
                   saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

   signature
       Calculates the MPEG-7 Video Signature. The filter can handle more than
       one input. In this case the matching between the inputs can be
       calculated additionally.  The filter always passes through the first
       input. The signature of each stream can be written into a file.

       It accepts the following options:

       detectmode
           Enable or disable the matching process.

           Available values are:

           off Disable the calculation of a matching (default).

           full
               Calculate the matching for the whole video and output whether
               the whole video matches or only parts.

           fast
               Calculate only until a matching is found or the video ends.
               Should be faster in some cases.

       nb_inputs
           Set the number of inputs. The option value must be a non negative
           integer.  Default value is 1.

       filename
           Set the path to which the output is written. If there is more than
           one input, the path must be a prototype, i.e. must contain %d or
           %0nd (where n is a positive integer), that will be replaced with
           the input number. If no filename is specified, no output will be
           written. This is the default.

       format
           Choose the output format.

           Available values are:

           binary
               Use the specified binary representation (default).

           xml Use the specified xml representation.

       th_d
           Set threshold to detect one word as similar. The option value must
           be an integer greater than zero. The default value is 9000.

       th_dc
           Set threshold to detect all words as similar. The option value must
           be an integer greater than zero. The default value is 60000.

       th_xh
           Set threshold to detect frames as similar. The option value must be
           an integer greater than zero. The default value is 116.

       th_di
           Set the minimum length of a sequence in frames to recognize it as
           matching sequence. The option value must be a non negative integer
           value.  The default value is 0.

       th_it
           Set the minimum relation, that matching frames to all frames must
           have.  The option value must be a double value between 0 and 1. The
           default value is 0.5.

       Examples

       o   To calculate the signature of an input video and store it in
           signature.bin:

                   ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -

       o   To detect whether two videos match and store the signatures in XML
           format in signature0.xml and signature1.xml:

                   ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -

   smartblur
       Blur the input video without impacting the outlines.

       It accepts the following options:

       luma_radius, lr
           Set the luma radius. The option value must be a float number in the
           range [0.1,5.0] that specifies the variance of the gaussian filter
           used to blur the image (slower if larger). Default value is 1.0.

       luma_strength, ls
           Set the luma strength. The option value must be a float number in
           the range [-1.0,1.0] that configures the blurring. A value included
           in [0.0,1.0] will blur the image whereas a value included in
           [-1.0,0.0] will sharpen the image. Default value is 1.0.

       luma_threshold, lt
           Set the luma threshold used as a coefficient to determine whether a
           pixel should be blurred or not. The option value must be an integer
           in the range [-30,30]. A value of 0 will filter all the image, a
           value included in [0,30] will filter flat areas and a value
           included in [-30,0] will filter edges. Default value is 0.

       chroma_radius, cr
           Set the chroma radius. The option value must be a float number in
           the range [0.1,5.0] that specifies the variance of the gaussian
           filter used to blur the image (slower if larger). Default value is
           luma_radius.

       chroma_strength, cs
           Set the chroma strength. The option value must be a float number in
           the range [-1.0,1.0] that configures the blurring. A value included
           in [0.0,1.0] will blur the image whereas a value included in
           [-1.0,0.0] will sharpen the image. Default value is luma_strength.

       chroma_threshold, ct
           Set the chroma threshold used as a coefficient to determine whether
           a pixel should be blurred or not. The option value must be an
           integer in the range [-30,30]. A value of 0 will filter all the
           image, a value included in [0,30] will filter flat areas and a
           value included in [-30,0] will filter edges. Default value is
           luma_threshold.

       If a chroma option is not explicitly set, the corresponding luma value
       is set.

   ssim
       Obtain the SSIM (Structural SImilarity Metric) between two input
       videos.

       This filter takes in input two input videos, the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the SSIM.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The filter stores the calculated SSIM of each frame.

       The description of the accepted parameters follows.

       stats_file, f
           If specified the filter will use the named file to save the SSIM of
           each individual frame. When filename equals "-" the data is sent to
           standard output.

       The file printed if stats_file is selected, contains a sequence of
       key/value pairs of the form key:value for each compared couple of
       frames.

       A description of each shown parameter follows:

       n   sequential number of the input frame, starting from 1

       Y, U, V, R, G, B
           SSIM of the compared frames for the component specified by the
           suffix.

       All SSIM of the compared frames for the whole frame.

       dB  Same as above but in dB representation.

       This filter also supports the framesync options.

       For example:

               movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
               [main][ref] ssim="stats_file=stats.log" [out]

       On this example the input file being processed is compared with the
       reference file ref_movie.mpg. The SSIM of each individual frame is
       stored in stats.log.

       Another example with both psnr and ssim at same time:

               ffmpeg -i main.mpg -i ref.mpg -lavfi  "ssim;[0:v][1:v]psnr" -f null -

   stereo3d
       Convert between different stereoscopic image formats.

       The filters accept the following options:

       in  Set stereoscopic image format of input.

           Available values for input image formats are:

           sbsl
               side by side parallel (left eye left, right eye right)

           sbsr
               side by side crosseye (right eye left, left eye right)

           sbs2l
               side by side parallel with half width resolution (left eye
               left, right eye right)

           sbs2r
               side by side crosseye with half width resolution (right eye
               left, left eye right)

           abl above-below (left eye above, right eye below)

           abr above-below (right eye above, left eye below)

           ab2l
               above-below with half height resolution (left eye above, right
               eye below)

           ab2r
               above-below with half height resolution (right eye above, left
               eye below)

           al  alternating frames (left eye first, right eye second)

           ar  alternating frames (right eye first, left eye second)

           irl interleaved rows (left eye has top row, right eye starts on
               next row)

           irr interleaved rows (right eye has top row, left eye starts on
               next row)

           icl interleaved columns, left eye first

           icr interleaved columns, right eye first

               Default value is sbsl.

       out Set stereoscopic image format of output.

           sbsl
               side by side parallel (left eye left, right eye right)

           sbsr
               side by side crosseye (right eye left, left eye right)

           sbs2l
               side by side parallel with half width resolution (left eye
               left, right eye right)

           sbs2r
               side by side crosseye with half width resolution (right eye
               left, left eye right)

           abl above-below (left eye above, right eye below)

           abr above-below (right eye above, left eye below)

           ab2l
               above-below with half height resolution (left eye above, right
               eye below)

           ab2r
               above-below with half height resolution (right eye above, left
               eye below)

           al  alternating frames (left eye first, right eye second)

           ar  alternating frames (right eye first, left eye second)

           irl interleaved rows (left eye has top row, right eye starts on
               next row)

           irr interleaved rows (right eye has top row, left eye starts on
               next row)

           arbg
               anaglyph red/blue gray (red filter on left eye, blue filter on
               right eye)

           argg
               anaglyph red/green gray (red filter on left eye, green filter
               on right eye)

           arcg
               anaglyph red/cyan gray (red filter on left eye, cyan filter on
               right eye)

           arch
               anaglyph red/cyan half colored (red filter on left eye, cyan
               filter on right eye)

           arcc
               anaglyph red/cyan color (red filter on left eye, cyan filter on
               right eye)

           arcd
               anaglyph red/cyan color optimized with the least squares
               projection of dubois (red filter on left eye, cyan filter on
               right eye)

           agmg
               anaglyph green/magenta gray (green filter on left eye, magenta
               filter on right eye)

           agmh
               anaglyph green/magenta half colored (green filter on left eye,
               magenta filter on right eye)

           agmc
               anaglyph green/magenta colored (green filter on left eye,
               magenta filter on right eye)

           agmd
               anaglyph green/magenta color optimized with the least squares
               projection of dubois (green filter on left eye, magenta filter
               on right eye)

           aybg
               anaglyph yellow/blue gray (yellow filter on left eye, blue
               filter on right eye)

           aybh
               anaglyph yellow/blue half colored (yellow filter on left eye,
               blue filter on right eye)

           aybc
               anaglyph yellow/blue colored (yellow filter on left eye, blue
               filter on right eye)

           aybd
               anaglyph yellow/blue color optimized with the least squares
               projection of dubois (yellow filter on left eye, blue filter on
               right eye)

           ml  mono output (left eye only)

           mr  mono output (right eye only)

           chl checkerboard, left eye first

           chr checkerboard, right eye first

           icl interleaved columns, left eye first

           icr interleaved columns, right eye first

           hdmi
               HDMI frame pack

           Default value is arcd.

       Examples

       o   Convert input video from side by side parallel to anaglyph
           yellow/blue dubois:

                   stereo3d=sbsl:aybd

       o   Convert input video from above below (left eye above, right eye
           below) to side by side crosseye.

                   stereo3d=abl:sbsr

   streamselect, astreamselect
       Select video or audio streams.

       The filter accepts the following options:

       inputs
           Set number of inputs. Default is 2.

       map Set input indexes to remap to outputs.

       Commands

       The "streamselect" and "astreamselect" filter supports the following
       commands:

       map Set input indexes to remap to outputs.

       Examples

       o   Select first 5 seconds 1st stream and rest of time 2nd stream:

                   sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0

       o   Same as above, but for audio:

                   asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0

   sobel
       Apply sobel operator to input video stream.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

       scale
           Set value which will be multiplied with filtered result.

       delta
           Set value which will be added to filtered result.

   spp
       Apply a simple postprocessing filter that compresses and decompresses
       the image at several (or - in the case of quality level 6 - all) shifts
       and average the results.

       The filter accepts the following options:

       quality
           Set quality. This option defines the number of levels for
           averaging. It accepts an integer in the range 0-6. If set to 0, the
           filter will have no effect. A value of 6 means the higher quality.
           For each increment of that value the speed drops by a factor of
           approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If not set, the filter
           will use the QP from the video stream (if available).

       mode
           Set thresholding mode. Available modes are:

           hard
               Set hard thresholding (default).

           soft
               Set soft thresholding (better de-ringing effect, but likely
               blurrier).

       use_bframe_qp
           Enable the use of the QP from the B-Frames if set to 1. Using this
           option may cause flicker since the B-Frames have often larger QP.
           Default is 0 (not enabled).

   subtitles
       Draw subtitles on top of input video using the libass library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libass". This filter also requires a build with libavcodec
       and libavformat to convert the passed subtitles file to ASS (Advanced
       Substation Alpha) subtitles format.

       The filter accepts the following options:

       filename, f
           Set the filename of the subtitle file to read. It must be
           specified.

       original_size
           Specify the size of the original video, the video for which the ASS
           file was composed. For the syntax of this option, check the "Video
           size" section in the ffmpeg-utils manual.  Due to a misdesign in
           ASS aspect ratio arithmetic, this is necessary to correctly scale
           the fonts if the aspect ratio has been changed.

       fontsdir
           Set a directory path containing fonts that can be used by the
           filter.  These fonts will be used in addition to whatever the font
           provider uses.

       alpha
           Process alpha channel, by default alpha channel is untouched.

       charenc
           Set subtitles input character encoding. "subtitles" filter only.
           Only useful if not UTF-8.

       stream_index, si
           Set subtitles stream index. "subtitles" filter only.

       force_style
           Override default style or script info parameters of the subtitles.
           It accepts a string containing ASS style format "KEY=VALUE" couples
           separated by ",".

       If the first key is not specified, it is assumed that the first value
       specifies the filename.

       For example, to render the file sub.srt on top of the input video, use
       the command:

               subtitles=sub.srt

       which is equivalent to:

               subtitles=filename=sub.srt

       To render the default subtitles stream from file video.mkv, use:

               subtitles=video.mkv

       To render the second subtitles stream from that file, use:

               subtitles=video.mkv:si=1

       To make the subtitles stream from sub.srt appear in transparent green
       "DejaVu Serif", use:

               subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'

   super2xsai
       Scale the input by 2x and smooth using the Super2xSaI (Scale and
       Interpolate) pixel art scaling algorithm.

       Useful for enlarging pixel art images without reducing sharpness.

   swaprect
       Swap two rectangular objects in video.

       This filter accepts the following options:

       w   Set object width.

       h   Set object height.

       x1  Set 1st rect x coordinate.

       y1  Set 1st rect y coordinate.

       x2  Set 2nd rect x coordinate.

       y2  Set 2nd rect y coordinate.

           All expressions are evaluated once for each frame.

       The all options are expressions containing the following constants:

       w
       h   The input width and height.

       a   same as w / h

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (w / h) * sar

       n   The number of the input frame, starting from 0.

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
           is unknown.

       pos the position in the file of the input frame, NAN if unknown

   swapuv
       Swap U & V plane.

   telecine
       Apply telecine process to the video.

       This filter accepts the following options:

       first_field
           top, t
               top field first

           bottom, b
               bottom field first The default value is "top".

       pattern
           A string of numbers representing the pulldown pattern you wish to
           apply.  The default value is 23.

               Some typical patterns:

               NTSC output (30i):
               27.5p: 32222
               24p: 23 (classic)
               24p: 2332 (preferred)
               20p: 33
               18p: 334
               16p: 3444

               PAL output (25i):
               27.5p: 12222
               24p: 222222222223 ("Euro pulldown")
               16.67p: 33
               16p: 33333334

   threshold
       Apply threshold effect to video stream.

       This filter needs four video streams to perform thresholding.  First
       stream is stream we are filtering.  Second stream is holding threshold
       values, third stream is holding min values, and last, fourth stream is
       holding max values.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

       For example if first stream pixel's component value is less then
       threshold value of pixel component from 2nd threshold stream, third
       stream value will picked, otherwise fourth stream pixel component value
       will be picked.

       Using color source filter one can perform various types of
       thresholding:

       Examples

       o   Binary threshold, using gray color as threshold:

                   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi

       o   Inverted binary threshold, using gray color as threshold:

                   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi

       o   Truncate binary threshold, using gray color as threshold:

                   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi

       o   Threshold to zero, using gray color as threshold:

                   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi

       o   Inverted threshold to zero, using gray color as threshold:

                   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi

   thumbnail
       Select the most representative frame in a given sequence of consecutive
       frames.

       The filter accepts the following options:

       n   Set the frames batch size to analyze; in a set of n frames, the
           filter will pick one of them, and then handle the next batch of n
           frames until the end. Default is 100.

       Since the filter keeps track of the whole frames sequence, a bigger n
       value will result in a higher memory usage, so a high value is not
       recommended.

       Examples

       o   Extract one picture each 50 frames:

                   thumbnail=50

       o   Complete example of a thumbnail creation with ffmpeg:

                   ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

   tile
       Tile several successive frames together.

       The filter accepts the following options:

       layout
           Set the grid size (i.e. the number of lines and columns). For the
           syntax of this option, check the "Video size" section in the
           ffmpeg-utils manual.

       nb_frames
           Set the maximum number of frames to render in the given area. It
           must be less than or equal to wxh. The default value is 0, meaning
           all the area will be used.

       margin
           Set the outer border margin in pixels.

       padding
           Set the inner border thickness (i.e. the number of pixels between
           frames). For more advanced padding options (such as having
           different values for the edges), refer to the pad video filter.

       color
           Specify the color of the unused area. For the syntax of this
           option, check the "Color" section in the ffmpeg-utils manual. The
           default value of color is "black".

       Examples

       o   Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a
           movie:

                   ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

           The -vsync 0 is necessary to prevent ffmpeg from duplicating each
           output frame to accommodate the originally detected frame rate.

       o   Display 5 pictures in an area of "32" frames, with 7 pixels
           between them, and 2 pixels of initial margin, using mixed flat and
           named options:

                   tile=32:nb_frames=5:padding=7:margin=2

   tinterlace
       Perform various types of temporal field interlacing.

       Frames are counted starting from 1, so the first input frame is
       considered odd.

       The filter accepts the following options:

       mode
           Specify the mode of the interlacing. This option can also be
           specified as a value alone. See below for a list of values for this
           option.

           Available values are:

           merge, 0
               Move odd frames into the upper field, even into the lower
               field, generating a double height frame at half frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444

                       Output:
                       11111                           33333
                       22222                           44444
                       11111                           33333
                       22222                           44444
                       11111                           33333
                       22222                           44444
                       11111                           33333
                       22222                           44444

           drop_even, 1
               Only output odd frames, even frames are dropped, generating a
               frame with unchanged height at half frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444

                       Output:
                       11111                           33333
                       11111                           33333
                       11111                           33333
                       11111                           33333

           drop_odd, 2
               Only output even frames, odd frames are dropped, generating a
               frame with unchanged height at half frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444

                       Output:
                                       22222                           44444
                                       22222                           44444
                                       22222                           44444
                                       22222                           44444

           pad, 3
               Expand each frame to full height, but pad alternate lines with
               black, generating a frame with double height at the same input
               frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444

                       Output:
                       11111           .....           33333           .....
                       .....           22222           .....           44444
                       11111           .....           33333           .....
                       .....           22222           .....           44444
                       11111           .....           33333           .....
                       .....           22222           .....           44444
                       11111           .....           33333           .....
                       .....           22222           .....           44444

           interleave_top, 4
               Interleave the upper field from odd frames with the lower field
               from even frames, generating a frame with unchanged height at
               half frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111<-         22222           33333<-         44444
                       11111           22222<-         33333           44444<-
                       11111<-         22222           33333<-         44444
                       11111           22222<-         33333           44444<-

                       Output:
                       11111                           33333
                       22222                           44444
                       11111                           33333
                       22222                           44444

           interleave_bottom, 5
               Interleave the lower field from odd frames with the upper field
               from even frames, generating a frame with unchanged height at
               half frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222<-         33333           44444<-
                       11111<-         22222           33333<-         44444
                       11111           22222<-         33333           44444<-
                       11111<-         22222           33333<-         44444

                       Output:
                       22222                           44444
                       11111                           33333
                       22222                           44444
                       11111                           33333

           interlacex2, 6
               Double frame rate with unchanged height. Frames are inserted
               each containing the second temporal field from the previous
               input frame and the first temporal field from the next input
               frame. This mode relies on the top_field_first flag. Useful for
               interlaced video displays with no field synchronisation.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                        11111           22222           33333           44444
                       11111           22222           33333           44444
                        11111           22222           33333           44444

                       Output:
                       11111   22222   22222   33333   33333   44444   44444
                        11111   11111   22222   22222   33333   33333   44444
                       11111   22222   22222   33333   33333   44444   44444
                        11111   11111   22222   22222   33333   33333   44444

           mergex2, 7
               Move odd frames into the upper field, even into the lower
               field, generating a double height frame at same frame rate.

                        ------> time
                       Input:
                       Frame 1         Frame 2         Frame 3         Frame 4

                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444
                       11111           22222           33333           44444

                       Output:
                       11111           33333           33333           55555
                       22222           22222           44444           44444
                       11111           33333           33333           55555
                       22222           22222           44444           44444
                       11111           33333           33333           55555
                       22222           22222           44444           44444
                       11111           33333           33333           55555
                       22222           22222           44444           44444

           Numeric values are deprecated but are accepted for backward
           compatibility reasons.

           Default mode is "merge".

       flags
           Specify flags influencing the filter process.

           Available value for flags is:

           low_pass_filter, vlfp
               Enable linear vertical low-pass filtering in the filter.
               Vertical low-pass filtering is required when creating an
               interlaced destination from a progressive source which contains
               high-frequency vertical detail. Filtering will reduce interlace
               'twitter' and Moire patterning.

           complex_filter, cvlfp
               Enable complex vertical low-pass filtering.  This will slightly
               less reduce interlace 'twitter' and Moire patterning but better
               retain detail and subjective sharpness impression.

           Vertical low-pass filtering can only be enabled for mode
           interleave_top and interleave_bottom.

   tonemap
       Tone map colors from different dynamic ranges.

       This filter expects data in single precision floating point, as it
       needs to operate on (and can output) out-of-range values. Another
       filter, such as zscale, is needed to convert the resulting frame to a
       usable format.

       The tonemapping algorithms implemented only work on linear light, so
       input data should be linearized beforehand (and possibly correctly
       tagged).

               ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT

       Options

       The filter accepts the following options.

       tonemap
           Set the tone map algorithm to use.

           Possible values are:

           none
               Do not apply any tone map, only desaturate overbright pixels.

           clip
               Hard-clip any out-of-range values. Use it for perfect color
               accuracy for in-range values, while distorting out-of-range
               values.

           linear
               Stretch the entire reference gamut to a linear multiple of the
               display.

           gamma
               Fit a logarithmic transfer between the tone curves.

           reinhard
               Preserve overall image brightness with a simple curve, using
               nonlinear contrast, which results in flattening details and
               degrading color accuracy.

           hable
               Preserve both dark and bright details better than reinhard, at
               the cost of slightly darkening everything. Use it when detail
               preservation is more important than color and brightness
               accuracy.

           mobius
               Smoothly map out-of-range values, while retaining contrast and
               colors for in-range material as much as possible. Use it when
               color accuracy is more important than detail preservation.

           Default is none.

       param
           Tune the tone mapping algorithm.

           This affects the following algorithms:

           none
               Ignored.

           linear
               Specifies the scale factor to use while stretching.  Default to
               1.0.

           gamma
               Specifies the exponent of the function.  Default to 1.8.

           clip
               Specify an extra linear coefficient to multiply into the signal
               before clipping.  Default to 1.0.

           reinhard
               Specify the local contrast coefficient at the display peak.
               Default to 0.5, which means that in-gamut values will be about
               half as bright as when clipping.

           hable
               Ignored.

           mobius
               Specify the transition point from linear to mobius transform.
               Every value below this point is guaranteed to be mapped 1:1.
               The higher the value, the more accurate the result will be, at
               the cost of losing bright details.  Default to 0.3, which due
               to the steep initial slope still preserves in-range colors
               fairly accurately.

       desat
           Apply desaturation for highlights that exceed this level of
           brightness. The higher the parameter, the more color information
           will be preserved. This setting helps prevent unnaturally blown-out
           colors for super-highlights, by (smoothly) turning into white
           instead. This makes images feel more natural, at the cost of
           reducing information about out-of-range colors.

           The default of 2.0 is somewhat conservative and will mostly just
           apply to skies or directly sunlit surfaces. A setting of 0.0
           disables this option.

           This option works only if the input frame has a supported color
           tag.

       peak
           Override signal/nominal/reference peak with this value. Useful when
           the embedded peak information in display metadata is not reliable
           or when tone mapping from a lower range to a higher range.

   transpose
       Transpose rows with columns in the input video and optionally flip it.

       It accepts the following parameters:

       dir Specify the transposition direction.

           Can assume the following values:

           0, 4, cclock_flip
               Rotate by 90 degrees counterclockwise and vertically flip
               (default), that is:

                       L.R     L.l
                       . . ->  . .
                       l.r     R.r

           1, 5, clock
               Rotate by 90 degrees clockwise, that is:

                       L.R     l.L
                       . . ->  . .
                       l.r     r.R

           2, 6, cclock
               Rotate by 90 degrees counterclockwise, that is:

                       L.R     R.r
                       . . ->  . .
                       l.r     L.l

           3, 7, clock_flip
               Rotate by 90 degrees clockwise and vertically flip, that is:

                       L.R     r.R
                       . . ->  . .
                       l.r     l.L

           For values between 4-7, the transposition is only done if the input
           video geometry is portrait and not landscape. These values are
           deprecated, the "passthrough" option should be used instead.

           Numerical values are deprecated, and should be dropped in favor of
           symbolic constants.

       passthrough
           Do not apply the transposition if the input geometry matches the
           one specified by the specified value. It accepts the following
           values:

           none
               Always apply transposition.

           portrait
               Preserve portrait geometry (when height >= width).

           landscape
               Preserve landscape geometry (when width >= height).

           Default value is "none".

       For example to rotate by 90 degrees clockwise and preserve portrait
       layout:

               transpose=dir=1:passthrough=portrait

       The command above can also be specified as:

               transpose=1:portrait

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

       It accepts the following parameters:

       start
           Specify the time of the start of the kept section, i.e. the frame
           with the timestamp start will be the first frame in the output.

       end Specify the time of the first frame that will be dropped, i.e. the
           frame immediately preceding the one with the timestamp end will be
           the last frame in the output.

       start_pts
           This is the same as start, except this option sets the start
           timestamp in timebase units instead of seconds.

       end_pts
           This is the same as end, except this option sets the end timestamp
           in timebase units instead of seconds.

       duration
           The maximum duration of the output in seconds.

       start_frame
           The number of the first frame that should be passed to the output.

       end_frame
           The number of the first frame that should be dropped.

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

       Note that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the _frame variants simply
       count the frames that pass through the filter. Also note that this
       filter does not modify the timestamps. If you wish for the output
       timestamps to start at zero, insert a setpts filter after the trim
       filter.

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

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

       Examples:

       o   Drop everything except the second minute of input:

                   ffmpeg -i INPUT -vf trim=60:120

       o   Keep only the first second:

                   ffmpeg -i INPUT -vf trim=duration=1

   unpremultiply
       Apply alpha unpremultiply effect to input video stream using first
       plane of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be
           copied.  By default value 0xf, all planes will be processed.

           If the format has 1 or 2 components, then luma is bit 0.  If the
           format has 3 or 4 components: for RGB formats bit 0 is green, bit 1
           is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
           chroma-U and bit 2 is chroma-V.  If present, the alpha channel is
           always the last bit.

       inplace
           Do not require 2nd input for processing, instead use alpha plane
           from input stream.

   unsharp
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
           Set the luma matrix horizontal size. It must be an odd integer
           between 3 and 23. The default value is 5.

       luma_msize_y, ly
           Set the luma matrix vertical size. It must be an odd integer
           between 3 and 23. The default value is 5.

       luma_amount, la
           Set the luma effect strength. It must be a floating point number,
           reasonable values lay between -1.5 and 1.5.

           Negative values will blur the input video, while positive values
           will sharpen it, a value of zero will disable the effect.

           Default value is 1.0.

       chroma_msize_x, cx
           Set the chroma matrix horizontal size. It must be an odd integer
           between 3 and 23. The default value is 5.

       chroma_msize_y, cy
           Set the chroma matrix vertical size. It must be an odd integer
           between 3 and 23. The default value is 5.

       chroma_amount, ca
           Set the chroma effect strength. It must be a floating point number,
           reasonable values lay between -1.5 and 1.5.

           Negative values will blur the input video, while positive values
           will sharpen it, a value of zero will disable the effect.

           Default value is 0.0.

       opencl
           If set to 1, specify using OpenCL capabilities, only available if
           FFmpeg was configured with "--enable-opencl". Default value is 0.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       o   Apply strong luma sharpen effect:

                   unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

       o   Apply a strong blur of both luma and chroma parameters:

                   unsharp=7:7:-2:7:7:-2

   uspp
       Apply ultra slow/simple postprocessing filter that compresses and
       decompresses the image at several (or - in the case of quality level 8
       - all) shifts and average the results.

       The way this differs from the behavior of spp is that uspp actually
       encodes & decodes each case with libavcodec Snow, whereas spp uses a
       simplified intra only 8x8 DCT similar to MJPEG.

       The filter accepts the following options:

       quality
           Set quality. This option defines the number of levels for
           averaging. It accepts an integer in the range 0-8. If set to 0, the
           filter will have no effect. A value of 8 means the higher quality.
           For each increment of that value the speed drops by a factor of
           approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If not set, the filter
           will use the QP from the video stream (if available).

   vaguedenoiser
       Apply a wavelet based denoiser.

       It transforms each frame from the video input into the wavelet domain,
       using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to
       the obtained coefficients. It does an inverse wavelet transform after.
       Due to wavelet properties, it should give a nice smoothed result, and
       reduced noise, without blurring picture features.

       This filter accepts the following options:

       threshold
           The filtering strength. The higher, the more filtered the video
           will be.  Hard thresholding can use a higher threshold than soft
           thresholding before the video looks overfiltered. Default value is
           2.

       method
           The filtering method the filter will use.

           It accepts the following values:

           hard
               All values under the threshold will be zeroed.

           soft
               All values under the threshold will be zeroed. All values above
               will be reduced by the threshold.

           garrote
               Scales or nullifies coefficients - intermediary between (more)
               soft and (less) hard thresholding.

           Default is garrote.

       nsteps
           Number of times, the wavelet will decompose the picture. Picture
           can't be decomposed beyond a particular point (typically, 8 for a
           640x480 frame - as 2^9 = 512 > 480). Valid values are integers
           between 1 and 32. Default value is 6.

       percent
           Partial of full denoising (limited coefficients shrinking), from 0
           to 100. Default value is 85.

       planes
           A list of the planes to process. By default all planes are
           processed.

   vectorscope
       Display 2 color component values in the two dimensional graph (which is
       called a vectorscope).

       This filter accepts the following options:

       mode, m
           Set vectorscope mode.

           It accepts the following values:

           gray
               Gray values are displayed on graph, higher brightness means
               more pixels have same component color value on location in
               graph. This is the default mode.

           color
               Gray values are displayed on graph. Surrounding pixels values
               which are not present in video frame are drawn in gradient of 2
               color components which are set by option "x" and "y". The 3rd
               color component is static.

           color2
               Actual color components values present in video frame are
               displayed on graph.

           color3
               Similar as color2 but higher frequency of same values "x" and
               "y" on graph increases value of another color component, which
               is luminance by default values of "x" and "y".

           color4
               Actual colors present in video frame are displayed on graph. If
               two different colors map to same position on graph then color
               with higher value of component not present in graph is picked.

           color5
               Gray values are displayed on graph. Similar to "color" but with
               3rd color component picked from radial gradient.

       x   Set which color component will be represented on X-axis. Default is
           1.

       y   Set which color component will be represented on Y-axis. Default is
           2.

       intensity, i
           Set intensity, used by modes: gray, color, color3 and color5 for
           increasing brightness of color component which represents frequency
           of (X, Y) location in graph.

       envelope, e
           none
               No envelope, this is default.

           instant
               Instant envelope, even darkest single pixel will be clearly
               highlighted.

           peak
               Hold maximum and minimum values presented in graph over time.
               This way you can still spot out of range values without
               constantly looking at vectorscope.

           peak+instant
               Peak and instant envelope combined together.

       graticule, g
           Set what kind of graticule to draw.

           none
           green
           color
       opacity, o
           Set graticule opacity.

       flags, f
           Set graticule flags.

           white
               Draw graticule for white point.

           black
               Draw graticule for black point.

           name
               Draw color points short names.

       bgopacity, b
           Set background opacity.

       lthreshold, l
           Set low threshold for color component not represented on X or Y
           axis.  Values lower than this value will be ignored. Default is 0.
           Note this value is multiplied with actual max possible value one
           pixel component can have. So for 8-bit input and low threshold
           value of 0.1 actual threshold is 0.1 * 255 = 25.

       hthreshold, h
           Set high threshold for color component not represented on X or Y
           axis.  Values higher than this value will be ignored. Default is 1.
           Note this value is multiplied with actual max possible value one
           pixel component can have. So for 8-bit input and high threshold
           value of 0.9 actual threshold is 0.9 * 255 = 230.

       colorspace, c
           Set what kind of colorspace to use when drawing graticule.

           auto
           601
           709

           Default is auto.

   vidstabdetect
       Analyze video stabilization/deshaking. Perform pass 1 of 2, see
       vidstabtransform for pass 2.

       This filter generates a file with relative translation and rotation
       transform information about subsequent frames, which is then used by
       the vidstabtransform filter.

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

       This filter accepts the following options:

       result
           Set the path to the file used to write the transforms information.
           Default value is transforms.trf.

       shakiness
           Set how shaky the video is and how quick the camera is. It accepts
           an integer in the range 1-10, a value of 1 means little shakiness,
           a value of 10 means strong shakiness. Default value is 5.

       accuracy
           Set the accuracy of the detection process. It must be a value in
           the range 1-15. A value of 1 means low accuracy, a value of 15
           means high accuracy. Default value is 15.

       stepsize
           Set stepsize of the search process. The region around minimum is
           scanned with 1 pixel resolution. Default value is 6.

       mincontrast
           Set minimum contrast. Below this value a local measurement field is
           discarded. Must be a floating point value in the range 0-1. Default
           value is 0.3.

       tripod
           Set reference frame number for tripod mode.

           If enabled, the motion of the frames is compared to a reference
           frame in the filtered stream, identified by the specified number.
           The idea is to compensate all movements in a more-or-less static
           scene and keep the camera view absolutely still.

           If set to 0, it is disabled. The frames are counted starting from
           1.

       show
           Show fields and transforms in the resulting frames. It accepts an
           integer in the range 0-2. Default value is 0, which disables any
           visualization.

       Examples

       o   Use default values:

                   vidstabdetect

       o   Analyze strongly shaky movie and put the results in file
           mytransforms.trf:

                   vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

       o   Visualize the result of internal transformations in the resulting
           video:

                   vidstabdetect=show=1

       o   Analyze a video with medium shakiness using ffmpeg:

                   ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

   vidstabtransform
       Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
       1.

       Read a file with transform information for each frame and
       apply/compensate them. Together with the vidstabdetect filter this can
       be used to deshake videos. See also
       <http://public.hronopik.de/vid.stab>. It is important to also use the
       unsharp filter, see below.

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

       Options

       input
           Set path to the file used to read the transforms. Default value is
           transforms.trf.

       smoothing
           Set the number of frames (value*2 + 1) used for lowpass filtering
           the camera movements. Default value is 10.

           For example a number of 10 means that 21 frames are used (10 in the
           past and 10 in the future) to smoothen the motion in the video. A
           larger value leads to a smoother video, but limits the acceleration
           of the camera (pan/tilt movements). 0 is a special case where a
           static camera is simulated.

       optalgo
           Set the camera path optimization algorithm.

           Accepted values are:

           gauss
               gaussian kernel low-pass filter on camera motion (default)

           avg averaging on transformations

       maxshift
           Set maximal number of pixels to translate frames. Default value is
           -1, meaning no limit.

       maxangle
           Set maximal angle in radians (degree*PI/180) to rotate frames.
           Default value is -1, meaning no limit.

       crop
           Specify how to deal with borders that may be visible due to
           movement compensation.

           Available values are:

           keep
               keep image information from previous frame (default)

           black
               fill the border black

       invert
           Invert transforms if set to 1. Default value is 0.

       relative
           Consider transforms as relative to previous frame if set to 1,
           absolute if set to 0. Default value is 0.

       zoom
           Set percentage to zoom. A positive value will result in a zoom-in
           effect, a negative value in a zoom-out effect. Default value is 0
           (no zoom).

       optzoom
           Set optimal zooming to avoid borders.

           Accepted values are:

           0   disabled

           1   optimal static zoom value is determined (only very strong
               movements will lead to visible borders) (default)

           2   optimal adaptive zoom value is determined (no borders will be
               visible), see zoomspeed

           Note that the value given at zoom is added to the one calculated
           here.

       zoomspeed
           Set percent to zoom maximally each frame (enabled when optzoom is
           set to 2). Range is from 0 to 5, default value is 0.25.

       interpol
           Specify type of interpolation.

           Available values are:

           no  no interpolation

           linear
               linear only horizontal

           bilinear
               linear in both directions (default)

           bicubic
               cubic in both directions (slow)

       tripod
           Enable virtual tripod mode if set to 1, which is equivalent to
           "relative=0:smoothing=0". Default value is 0.

           Use also "tripod" option of vidstabdetect.

       debug
           Increase log verbosity if set to 1. Also the detected global
           motions are written to the temporary file global_motions.trf.
           Default value is 0.

       Examples

       o   Use ffmpeg for a typical stabilization with default values:

                   ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

           Note the use of the unsharp filter which is always recommended.

       o   Zoom in a bit more and load transform data from a given file:

                   vidstabtransform=zoom=5:input="mytransforms.trf"

       o   Smoothen the video even more:

                   vidstabtransform=smoothing=30

   vflip
       Flip the input video vertically.

       For example, to vertically flip a video with ffmpeg:

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

   vignette
       Make or reverse a natural vignetting effect.

       The filter accepts the following options:

       angle, a
           Set lens angle expression as a number of radians.

           The value is clipped in the "[0,PI/2]" range.

           Default value: "PI/5"

       x0
       y0  Set center coordinates expressions. Respectively "w/2" and "h/2" by
           default.

       mode
           Set forward/backward mode.

           Available modes are:

           forward
               The larger the distance from the central point, the darker the
               image becomes.

           backward
               The larger the distance from the central point, the brighter
               the image becomes.  This can be used to reverse a vignette
               effect, though there is no automatic detection to extract the
               lens angle and other settings (yet). It can also be used to
               create a burning effect.

           Default value is forward.

       eval
           Set evaluation mode for the expressions (angle, x0, y0).

           It accepts the following values:

           init
               Evaluate expressions only once during the filter
               initialization.

           frame
               Evaluate expressions for each incoming frame. This is way
               slower than the init mode since it requires all the scalers to
               be re-computed, but it allows advanced dynamic expressions.

           Default value is init.

       dither
           Set dithering to reduce the circular banding effects. Default is 1
           (enabled).

       aspect
           Set vignette aspect. This setting allows one to adjust the shape of
           the vignette.  Setting this value to the SAR of the input will make
           a rectangular vignetting following the dimensions of the video.

           Default is "1/1".

       Expressions

       The alpha, x0 and y0 expressions can contain the following parameters.

       w
       h   input width and height

       n   the number of input frame, starting from 0

       pts the PTS (Presentation TimeStamp) time of the filtered video frame,
           expressed in TB units, NAN if undefined

       r   frame rate of the input video, NAN if the input frame rate is
           unknown

       t   the PTS (Presentation TimeStamp) of the filtered video frame,
           expressed in seconds, NAN if undefined

       tb  time base of the input video

       Examples

       o   Apply simple strong vignetting effect:

                   vignette=PI/4

       o   Make a flickering vignetting:

                   vignette='PI/4+random(1)*PI/50':eval=frame

   vmafmotion
       Obtain the average vmaf motion score of a video.  It is one of the
       component filters of VMAF.

       The obtained average motion score is printed through the logging
       system.

       In the below example the input file ref.mpg is being processed and
       score is computed.

               ffmpeg -i ref.mpg -lavfi vmafmotion -f null -

   vstack
       Stack input videos vertically.

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

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

       The filter accept the following option:

       inputs
           Set number of input streams. Default is 2.

       shortest
           If set to 1, force the output to terminate when the shortest input
           terminates. Default value is 0.

   w3fdif
       Deinterlace the input video ("w3fdif" stands for "Weston 3 Field
       Deinterlacing Filter").

       Based on the process described by Martin Weston for BBC R&D, and
       implemented based on the de-interlace algorithm written by Jim
       Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses
       filter coefficients calculated by BBC R&D.

       There are two sets of filter coefficients, so called "simple": and
       "complex". Which set of filter coefficients is used can be set by
       passing an optional parameter:

       filter
           Set the interlacing filter coefficients. Accepts one of the
           following values:

           simple
               Simple filter coefficient set.

           complex
               More-complex filter coefficient set.

           Default value is complex.

       deint
           Specify which frames to deinterlace. Accept one of the following
           values:

           all Deinterlace all frames,

           interlaced
               Only deinterlace frames marked as interlaced.

           Default value is all.

   waveform
       Video waveform monitor.

       The waveform monitor plots color component intensity. By default
       luminance only. Each column of the waveform corresponds to a column of
       pixels in the source video.

       It accepts the following options:

       mode, m
           Can be either "row", or "column". Default is "column".  In row
           mode, the graph on the left side represents color component value 0
           and the right side represents value = 255. In column mode, the top
           side represents color component value = 0 and bottom side
           represents value = 255.

       intensity, i
           Set intensity. Smaller values are useful to find out how many
           values of the same luminance are distributed across input
           rows/columns.  Default value is 0.04. Allowed range is [0, 1].

       mirror, r
           Set mirroring mode. 0 means unmirrored, 1 means mirrored.  In
           mirrored mode, higher values will be represented on the left side
           for "row" mode and at the top for "column" mode. Default is 1
           (mirrored).

       display, d
           Set display mode.  It accepts the following values:

           overlay
               Presents information identical to that in the "parade", except
               that the graphs representing color components are superimposed
               directly over one another.

               This display mode makes it easier to spot relative differences
               or similarities in overlapping areas of the color components
               that are supposed to be identical, such as neutral whites,
               grays, or blacks.

           stack
               Display separate graph for the color components side by side in
               "row" mode or one below the other in "column" mode.

           parade
               Display separate graph for the color components side by side in
               "column" mode or one below the other in "row" mode.

               Using this display mode makes it easy to spot color casts in
               the highlights and shadows of an image, by comparing the
               contours of the top and the bottom graphs of each waveform.
               Since whites, grays, and blacks are characterized by exactly
               equal amounts of red, green, and blue, neutral areas of the
               picture should display three waveforms of roughly equal
               width/height. If not, the correction is easy to perform by
               making level adjustments the three waveforms.

           Default is "stack".

       components, c
           Set which color components to display. Default is 1, which means
           only luminance or red color component if input is in RGB
           colorspace. If is set for example to 7 it will display all 3 (if)
           available color components.

       envelope, e
           none
               No envelope, this is default.

           instant
               Instant envelope, minimum and maximum values presented in graph
               will be easily visible even with small "step" value.

           peak
               Hold minimum and maximum values presented in graph across time.
               This way you can still spot out of range values without
               constantly looking at waveforms.

           peak+instant
               Peak and instant envelope combined together.

       filter, f
           lowpass
               No filtering, this is default.

           flat
               Luma and chroma combined together.

           aflat
               Similar as above, but shows difference between blue and red
               chroma.

           chroma
               Displays only chroma.

           color
               Displays actual color value on waveform.

           acolor
               Similar as above, but with luma showing frequency of chroma
               values.

       graticule, g
           Set which graticule to display.

           none
               Do not display graticule.

           green
               Display green graticule showing legal broadcast ranges.

       opacity, o
           Set graticule opacity.

       flags, fl
           Set graticule flags.

           numbers
               Draw numbers above lines. By default enabled.

           dots
               Draw dots instead of lines.

       scale, s
           Set scale used for displaying graticule.

           digital
           millivolts
           ire

           Default is digital.

       bgopacity, b
           Set background opacity.

   weave, doubleweave
       The "weave" takes a field-based video input and join each two
       sequential fields into single frame, producing a new double height clip
       with half the frame rate and half the frame count.

       The "doubleweave" works same as "weave" but without halving frame rate
       and frame count.

       It accepts the following option:

       first_field
           Set first field. Available values are:

           top, t
               Set the frame as top-field-first.

           bottom, b
               Set the frame as bottom-field-first.

       Examples

       o   Interlace video using select and separatefields filter:

                   separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave

   xbr
       Apply the xBR high-quality magnification filter which is designed for
       pixel art. It follows a set of edge-detection rules, see
       <http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "2xBR", 3 for "3BR" and 4 for
           "4xBR".  Default is 3.

   yadif
       Deinterlace the input video ("yadif" means "yet another deinterlacing
       filter").

       It accepts the following parameters:

       mode
           The interlacing mode to adopt. It accepts one of the following
           values:

           0, send_frame
               Output one frame for each frame.

           1, send_field
               Output one frame for each field.

           2, send_frame_nospatial
               Like "send_frame", but it skips the spatial interlacing check.

           3, send_field_nospatial
               Like "send_field", but it skips the spatial interlacing check.

           The default value is "send_frame".

       parity
           The picture field parity assumed for the input interlaced video. It
           accepts one of the following values:

           0, tff
               Assume the top field is first.

           1, bff
               Assume the bottom field is first.

           -1, auto
               Enable automatic detection of field parity.

           The default value is "auto".  If the interlacing is unknown or the
           decoder does not export this information, top field first will be
           assumed.

       deint
           Specify which frames to deinterlace. Accept one of the following
           values:

           0, all
               Deinterlace all frames.

           1, interlaced
               Only deinterlace frames marked as interlaced.

           The default value is "all".

   zoompan
       Apply Zoom & Pan effect.

       This filter accepts the following options:

       zoom, z
           Set the zoom expression. Default is 1.

       x
       y   Set the x and y expression. Default is 0.

       d   Set the duration expression in number of frames.  This sets for how
           many number of frames effect will last for single input image.

       s   Set the output image size, default is 'hd720'.

       fps Set the output frame rate, default is '25'.

       Each expression can contain the following constants:

       in_w, iw
           Input width.

       in_h, ih
           Input height.

       out_w, ow
           Output width.

       out_h, oh
           Output height.

       in  Input frame count.

       on  Output frame count.

       x
       y   Last calculated 'x' and 'y' position from 'x' and 'y' expression
           for current input frame.

       px
       py  'x' and 'y' of last output frame of previous input frame or 0 when
           there was not yet such frame (first input frame).

       zoom
           Last calculated zoom from 'z' expression for current input frame.

       pzoom
           Last calculated zoom of last output frame of previous input frame.

       duration
           Number of output frames for current input frame. Calculated from
           'd' expression for each input frame.

       pduration
           number of output frames created for previous input frame

       a   Rational number: input width / input height

       sar sample aspect ratio

       dar display aspect ratio

       Examples

       o   Zoom-in up to 1.5 and pan at same time to some spot near center of
           picture:

                   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

       o   Zoom-in up to 1.5 and pan always at center of picture:

                   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       o   Same as above but without pausing:

                   zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

   zscale
       Scale (resize) the input video, using the z.lib library:
       https://github.com/sekrit-twc/zimg.

       The zscale filter forces the output display aspect ratio to be the same
       as the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the zscale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options.

       width, w
       height, h
           Set the output video dimension expression. Default value is the
           input dimension.

           If the width or w value is 0, the input width is used for the
           output. If the height or h value is 0, the input height is used for
           the output.

           If one and only one of the values is -n with n >= 1, the zscale
           filter will use a value that maintains the aspect ratio of the
           input image, calculated from the other specified dimension. After
           that it will, however, make sure that the calculated dimension is
           divisible by n and adjust the value if necessary.

           If both values are -n with n >= 1, the behavior will be identical
           to both values being set to 0 as previously detailed.

           See below for the list of accepted constants for use in the
           dimension expression.

       size, s
           Set the video size. For the syntax of this option, check the "Video
           size" section in the ffmpeg-utils manual.

       dither, d
           Set the dither type.

           Possible values are:

           none
           ordered
           random
           error_diffusion

           Default is none.

       filter, f
           Set the resize filter type.

           Possible values are:

           point
           bilinear
           bicubic
           spline16
           spline36
           lanczos

           Default is bilinear.

       range, r
           Set the color range.

           Possible values are:

           input
           limited
           full

           Default is same as input.

       primaries, p
           Set the color primaries.

           Possible values are:

           input
           709
           unspecified
           170m
           240m
           2020

           Default is same as input.

       transfer, t
           Set the transfer characteristics.

           Possible values are:

           input
           709
           unspecified
           601
           linear
           2020_10
           2020_12
           smpte2084
           iec61966-2-1
           arib-std-b67

           Default is same as input.

       matrix, m
           Set the colorspace matrix.

           Possible value are:

           input
           709
           unspecified
           470bg
           170m
           2020_ncl
           2020_cl

           Default is same as input.

       rangein, rin
           Set the input color range.

           Possible values are:

           input
           limited
           full

           Default is same as input.

       primariesin, pin
           Set the input color primaries.

           Possible values are:

           input
           709
           unspecified
           170m
           240m
           2020

           Default is same as input.

       transferin, tin
           Set the input transfer characteristics.

           Possible values are:

           input
           709
           unspecified
           601
           linear
           2020_10
           2020_12

           Default is same as input.

       matrixin, min
           Set the input colorspace matrix.

           Possible value are:

           input
           709
           unspecified
           470bg
           170m
           2020_ncl
           2020_cl
       chromal, c
           Set the output chroma location.

           Possible values are:

           input
           left
           center
           topleft
           top
           bottomleft
           bottom
       chromalin, cin
           Set the input chroma location.

           Possible values are:

           input
           left
           center
           topleft
           top
           bottomleft
           bottom
       npl Set the nominal peak luminance.

       The values of the w and h options are expressions containing the
       following constants:

       in_w
       in_h
           The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
           horizontal and vertical input chroma subsample values. For example
           for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
           horizontal and vertical output chroma subsample values. For example
           for the pixel format "yuv422p" hsub is 2 and vsub is 1.


VIDEO SOURCES

       Below is a description of the currently available video sources.

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

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

       It accepts the following parameters:

       video_size
           Specify the size (width and height) of the buffered video frames.
           For the syntax of this option, check the "Video size" section in
           the ffmpeg-utils manual.

       width
           The input video width.

       height
           The input video height.

       pix_fmt
           A string representing the pixel format of the buffered video
           frames.  It may be a number corresponding to a pixel format, or a
           pixel format name.

       time_base
           Specify the timebase assumed by the timestamps of the buffered
           frames.

       frame_rate
           Specify the frame rate expected for the video stream.

       pixel_aspect, sar
           The sample (pixel) aspect ratio of the input video.

       sws_param
           Specify the optional parameters to be used for the scale filter
           which is automatically inserted when an input change is detected in
           the input size or format.

       hw_frames_ctx
           When using a hardware pixel format, this should be a reference to
           an AVHWFramesContext describing input frames.

       For example:

               buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

       will instruct the source to accept video frames with size 320x240 and
       with format "yuv410p", assuming 1/24 as the timestamps timebase and
       square pixels (1:1 sample aspect ratio).  Since the pixel format with
       name "yuv410p" corresponds to the number 6 (check the enum
       AVPixelFormat definition in libavutil/pixfmt.h), this example
       corresponds to:

               buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

       Alternatively, the options can be specified as a flat string, but this
       syntax is deprecated:

       width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]

   cellauto
       Create a pattern generated by an elementary cellular automaton.

       The initial state of the cellular automaton can be defined through the
       filename and pattern options. If such options are not specified an
       initial state is created randomly.

       At each new frame a new row in the video is filled with the result of
       the cellular automaton next generation. The behavior when the whole
       frame is filled is defined by the scroll option.

       This source accepts the following options:

       filename, f
           Read the initial cellular automaton state, i.e. the starting row,
           from the specified file.  In the file, each non-whitespace
           character is considered an alive cell, a newline will terminate the
           row, and further characters in the file will be ignored.

       pattern, p
           Read the initial cellular automaton state, i.e. the starting row,
           from the specified string.

           Each non-whitespace character in the string is considered an alive
           cell, a newline will terminate the row, and further characters in
           the string will be ignored.

       rate, r
           Set the video rate, that is the number of frames generated per
           second.  Default is 25.

       random_fill_ratio, ratio
           Set the random fill ratio for the initial cellular automaton row.
           It is a floating point number value ranging from 0 to 1, defaults
           to 1/PHI.

           This option is ignored when a file or a pattern is specified.

       random_seed, seed
           Set the seed for filling randomly the initial row, must be an
           integer included between 0 and UINT32_MAX. If not specified, or if
           explicitly set to -1, the filter will try to use a good random seed
           on a best effort basis.

       rule
           Set the cellular automaton rule, it is a number ranging from 0 to
           255.  Default value is 110.

       size, s
           Set the size of the output video. For the syntax of this option,
           check the "Video size" section in the ffmpeg-utils manual.

           If filename or pattern is specified, the size is set by default to
           the width of the specified initial state row, and the height is set
           to width * PHI.

           If size is set, it must contain the width of the specified pattern
           string, and the specified pattern will be centered in the larger
           row.

           If a filename or a pattern string is not specified, the size value
           defaults to "320x518" (used for a randomly generated initial
           state).

       scroll
           If set to 1, scroll the output upward when all the rows in the
           output have been already filled. If set to 0, the new generated row
           will be written over the top row just after the bottom row is
           filled.  Defaults to 1.

       start_full, full
           If set to 1, completely fill the output with generated rows before
           outputting the first frame.  This is the default behavior, for
           disabling set the value to 0.

       stitch
           If set to 1, stitch the left and right row edges together.  This is
           the default behavior, for disabling set the value to 0.

       Examples

       o   Read the initial state from pattern, and specify an output of size
           200x400.

                   cellauto=f=pattern:s=200x400

       o   Generate a random initial row with a width of 200 cells, with a
           fill ratio of 2/3:

                   cellauto=ratio=2/3:s=200x200

       o   Create a pattern generated by rule 18 starting by a single alive
           cell centered on an initial row with width 100:

                   cellauto=p=@s=100x400:full=0:rule=18

       o   Specify a more elaborated initial pattern:

                   cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

   coreimagesrc
       Video source generated on GPU using Apple's CoreImage API on OSX.

       This video source is a specialized version of the coreimage video
       filter.  Use a core image generator at the beginning of the applied
       filterchain to generate the content.

       The coreimagesrc video source accepts the following options:

       list_generators
           List all available generators along with all their respective
           options as well as possible minimum and maximum values along with
           the default values.

                   list_generators=true

       size, s
           Specify the size of the sourced video. For the syntax of this
           option, check the "Video size" section in the ffmpeg-utils manual.
           The default value is "320x240".

       rate, r
           Specify the frame rate of the sourced video, as the number of
           frames generated per second. It has to be a string in the format
           frame_rate_num/frame_rate_den, an integer number, a floating point
           number or a valid video frame rate abbreviation. The default value
           is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
           Set the duration of the sourced video. See the Time duration
           section in the ffmpeg-utils(1) manual for the accepted syntax.

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

       Additionally, all options of the coreimage video filter are accepted.
       A complete filterchain can be used for further processing of the
       generated input without CPU-HOST transfer. See coreimage documentation
       and examples for details.

       Examples

       o   Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage,
           given as complete and escaped command-line for Apple's standard
           bash shell:

                   ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

           This example is equivalent to the QRCode example of coreimage
           without the need for a nullsrc video source.

   mandelbrot
       Generate a Mandelbrot set fractal, and progressively zoom towards the
       point specified with start_x and start_y.

       This source accepts the following options:

       end_pts
           Set the terminal pts value. Default value is 400.

       end_scale
           Set the terminal scale value.  Must be a floating point value.
           Default value is 0.3.

       inner
           Set the inner coloring mode, that is the algorithm used to draw the
           Mandelbrot fractal internal region.

           It shall assume one of the following values:

           black
               Set black mode.

           convergence
               Show time until convergence.

           mincol
               Set color based on point closest to the origin of the
               iterations.

           period
               Set period mode.

           Default value is mincol.

       bailout
           Set the bailout value. Default value is 10.0.

       maxiter
           Set the maximum of iterations performed by the rendering algorithm.
           Default value is 7189.

       outer
           Set outer coloring mode.  It shall assume one of following values:

           iteration_count
               Set iteration cound mode.

           normalized_iteration_count
               set normalized iteration count mode.

           Default value is normalized_iteration_count.

       rate, r
           Set frame rate, expressed as number of frames per second. Default
           value is "25".

       size, s
           Set frame size. For the syntax of this option, check the "Video
           size" section in the ffmpeg-utils manual. Default value is
           "640x480".

       start_scale
           Set the initial scale value. Default value is 3.0.

       start_x
           Set the initial x position. Must be a floating point value between
           -100 and 100. Default value is
           -0.743643887037158704752191506114774.

       start_y
           Set the initial y position. Must be a floating point value between
           -100 and 100. Default value is
           -0.131825904205311970493132056385139.

   mptestsrc
       Generate various test patterns, as generated by the MPlayer test
       filter.

       The size of the generated video is fixed, and is 256x256.  This source
       is useful in particular for testing encoding features.

       This source accepts the following options:

       rate, r
           Specify the frame rate of the sourced video, as the number of
           frames generated per second. It has to be a string in the format
           frame_rate_num/frame_rate_den, an integer number, a floating point
           number or a valid video frame rate abbreviation. The default value
           is "25".

       duration, d
           Set the duration of the sourced video. See the Time duration
           section in the ffmpeg-utils(1) manual for the accepted syntax.

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

       test, t
           Set the number or the name of the test to perform. Supported tests
           are:

           dc_luma
           dc_chroma
           freq_luma
           freq_chroma
           amp_luma
           amp_chroma
           cbp
           mv
           ring1
           ring2
           all

           Default value is "all", which will cycle through the list of all
           tests.

       Some examples:

               mptestsrc=t=dc_luma

       will generate a "dc_luma" test pattern.

   frei0r_src
       Provide a frei0r source.

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

       This source accepts the following parameters:

       size
           The size of the video to generate. For the syntax of this option,
           check the "Video size" section in the ffmpeg-utils manual.

       framerate
           The framerate of the generated video. It may be a string of the
           form num/den or a frame rate abbreviation.

       filter_name
           The name to the frei0r source to load. For more information
           regarding frei0r and how to set the parameters, read the frei0r
           section in the video filters documentation.

       filter_params
           A '|'-separated list of parameters to pass to the frei0r source.

       For example, to generate a frei0r partik0l source with size 200x200 and
       frame rate 10 which is overlaid on the overlay filter main input:

               frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

   life
       Generate a life pattern.

       This source is based on a generalization of John Conway's life game.

       The sourced input represents a life grid, each pixel represents a cell
       which can be in one of two possible states, alive or dead. Every cell
       interacts with its eight neighbours, which are the cells that are
       horizontally, vertically, or diagonally adjacent.

       At each interaction the grid evolves according to the adopted rule,
       which specifies the number of neighbor alive cells which will make a
       cell stay alive or born. The rule option allows one to specify the rule
       to adopt.

       This source accepts the following options:

       filename, f
           Set the file from which to read the initial grid state. In the
           file, each non-whitespace character is considered an alive cell,
           and newline is used to delimit the end of each row.

           If this option is not specified, the initial grid is generated
           randomly.

       rate, r
           Set the video rate, that is the number of frames generated per
           second.  Default is 25.

       random_fill_ratio, ratio
           Set the random fill ratio for the initial random grid. It is a
           floating point number value ranging from 0 to 1, defaults to 1/PHI.
           It is ignored when a file is specified.

       random_seed, seed
           Set the seed for filling the initial random grid, must be an
           integer included between 0 and UINT32_MAX. If not specified, or if
           explicitly set to -1, the filter will try to use a good random seed
           on a best effort basis.

       rule
           Set the life rule.

           A rule can be specified with a code of the kind "SNS/BNB", where NS
           and NB are sequences of numbers in the range 0-8, NS specifies the
           number of alive neighbor cells which make a live cell stay alive,
           and NB the number of alive neighbor cells which make a dead cell to
           become alive (i.e. to "born").  "s" and "b" can be used in place of
           "S" and "B", respectively.

           Alternatively a rule can be specified by an 18-bits integer. The 9
           high order bits are used to encode the next cell state if it is
           alive for each number of neighbor alive cells, the low order bits
           specify the rule for "borning" new cells. Higher order bits encode
           for an higher number of neighbor cells.  For example the number
           6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born
           rule of 9, which corresponds to "S23/B03".

           Default value is "S23/B3", which is the original Conway's game of
           life rule, and will keep a cell alive if it has 2 or 3 neighbor
           alive cells, and will born a new cell if there are three alive
           cells around a dead cell.

       size, s
           Set the size of the output video. For the syntax of this option,
           check the "Video size" section in the ffmpeg-utils manual.

           If filename is specified, the size is set by default to the same
           size of the input file. If size is set, it must contain the size
           specified in the input file, and the initial grid defined in that
           file is centered in the larger resulting area.

           If a filename is not specified, the size value defaults to
           "320x240" (used for a randomly generated initial grid).

       stitch
           If set to 1, stitch the left and right grid edges together, and the
           top and bottom edges also. Defaults to 1.

       mold
           Set cell mold speed. If set, a dead cell will go from death_color
           to mold_color with a step of mold. mold can have a value from 0 to
           255.

       life_color
           Set the color of living (or new born) cells.

       death_color
           Set the color of dead cells. If mold is set, this is the first
           color used to represent a dead cell.

       mold_color
           Set mold color, for definitely dead and moldy cells.

           For the syntax of these 3 color options, check the "Color" section
           in the ffmpeg-utils manual.

       Examples

       o   Read a grid from pattern, and center it on a grid of size 300x300
           pixels:

                   life=f=pattern:s=300x300

       o   Generate a random grid of size 200x200, with a fill ratio of 2/3:

                   life=ratio=2/3:s=200x200

       o   Specify a custom rule for evolving a randomly generated grid:

                   life=rule=S14/B34

       o   Full example with slow death effect (mold) using ffplay:

                   ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

   allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars,
       smptehdbars, testsrc, testsrc2, yuvtestsrc
       The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

       The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

       The "color" source provides an uniformly colored input.

       The "haldclutsrc" source provides an identity Hald CLUT. See also
       haldclut filter.

       The "nullsrc" source returns unprocessed video frames. It is mainly
       useful to be employed in analysis / debugging tools, or as the source
       for filters which ignore the input data.

       The "rgbtestsrc" source generates an RGB test pattern useful for
       detecting RGB vs BGR issues. You should see a red, green and blue
       stripe from top to bottom.

       The "smptebars" source generates a color bars pattern, based on the
       SMPTE Engineering Guideline EG 1-1990.

       The "smptehdbars" source generates a color bars pattern, based on the
       SMPTE RP 219-2002.

       The "testsrc" source generates a test video pattern, showing a color
       pattern, a scrolling gradient and a timestamp. This is mainly intended
       for testing purposes.

       The "testsrc2" source is similar to testsrc, but supports more pixel
       formats instead of just "rgb24". This allows using it as an input for
       other tests without requiring a format conversion.

       The "yuvtestsrc" source generates an YUV test pattern. You should see a
       y, cb and cr stripe from top to bottom.

       The sources accept the following parameters:

       alpha
           Specify the alpha (opacity) of the background, only available in
           the "testsrc2" source. The value must be between 0 (fully
           transparent) and 255 (fully opaque, the default).

       color, c
           Specify the color of the source, only available in the "color"
           source. For the syntax of this option, check the "Color" section in
           the ffmpeg-utils manual.

       level
           Specify the level of the Hald CLUT, only available in the
           "haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
           by "N*N*N" pixels to be used as identity matrix for 3D lookup
           tables. Each component is coded on a "1/(N*N)" scale.

       size, s
           Specify the size of the sourced video. For the syntax of this
           option, check the "Video size" section in the ffmpeg-utils manual.
           The default value is "320x240".

           This option is not available with the "haldclutsrc" filter.

       rate, r
           Specify the frame rate of the sourced video, as the number of
           frames generated per second. It has to be a string in the format
           frame_rate_num/frame_rate_den, an integer number, a floating point
           number or a valid video frame rate abbreviation. The default value
           is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
           Set the duration of the sourced video. See the Time duration
           section in the ffmpeg-utils(1) manual for the accepted syntax.

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

       decimals, n
           Set the number of decimals to show in the timestamp, only available
           in the "testsrc" source.

           The displayed timestamp value will correspond to the original
           timestamp value multiplied by the power of 10 of the specified
           value. Default value is 0.

       For example the following:

               testsrc=duration=5.3:size=qcif:rate=10

       will generate a video with a duration of 5.3 seconds, with size 176x144
       and a frame rate of 10 frames per second.

       The following graph description will generate a red source with an
       opacity of 0.2, with size "qcif" and a frame rate of 10 frames per
       second.

               color=c=red@0.2:s=qcif:r=10

       If the input content is to be ignored, "nullsrc" can be used. The
       following command generates noise in the luminance plane by employing
       the "geq" filter:

               nullsrc=s=256x256, geq=random(1)*255:128:128

       Commands

       The "color" source supports the following commands:

       c, color
           Set the color of the created image. Accepts the same syntax of the
           corresponding color option.


VIDEO SINKS

       Below is a description of the currently available video sinks.

   buffersink
       Buffer video frames, and make them available to the end of the filter
       graph.

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

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

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


MULTIMEDIA FILTERS

       Below is a description of the currently available multimedia filters.

   abitscope
       Convert input audio to a video output, displaying the audio bit scope.

       The filter accepts the following options:

       rate, r
           Set frame rate, expressed as number of frames per second. Default
           value is "25".

       size, s
           Specify the video size for the output. For the syntax of this
           option, check the "Video size" section in the ffmpeg-utils manual.
           Default value is "1024x256".

       colors
           Specify list of colors separated by space or by '|' which will be
           used to draw channels. Unrecognized or missing colors will be
           replaced by white color.

   ahistogram
       Convert input audio to a video output, displaying the volume histogram.

       The filter accepts the following options:

       dmode
           Specify how histogram is calculated.

           It accepts the following values:

           single
               Use single histogram for all channels.

           separate
               Use separate histogram for each channel.

           Default is "single".

       rate, r
           Set frame rate, expressed as number of frames per second. Default
           value is "25".

       size, s
           Specify the video size for the output. For the syntax of this
           option, check the "Video size" section in the ffmpeg-utils manual.
           Default value is "hd720".

       scale
           Set display scale.

           It accepts the following values:

           log logarithmic

           sqrt
               square root

           cbrt
               cubic root

           lin linear

           rlog
               reverse logarithmic

           Default is "log".

       ascale
           Set amplitude scale.

           It accepts the following values:

           log logarithmic

           lin linear

           Default is "log".

       acount
           Set how much frames to accumulate in histogram.  Defauls is 1.
           Setting this to -1 accumulates all frames.

       rheight
           Set histogram ratio of window height.

       slide
           Set sonogram sliding.

           It accepts the following values:

           replace
               replace old rows with new ones.

           scroll
               scroll from top to bottom.

           Default is "replace".

   aphasemeter
       Convert input audio to a video output, displaying the audio phase.

       The filter accepts the following options:

       rate, r
           Set the output frame rate. Default value is 25.

       size, s
           Set the video size for the output. For the syntax of this option,
           check the "Video size" section in the ffmpeg-utils manual.  Default
           value is "800x400".

       rc
       gc
       bc  Specify the red, green, blue contrast. Default values are 2, 7 and
           1.  Allowed range is "[0, 255]".

       mpc Set color which will be used for drawing median phase. If color is
           "none" which is default, no median phase value will be drawn.

       video
           Enable video output. Default is enabled.

       The filter also exports the frame metadata "lavfi.aphasemeter.phase"
       which represents mean phase of current audio frame. Value is in range
       "[-1, 1]".  The "-1" means left and right channels are completely out
       of phase and 1 means channels are in phase.

   avectorscope
       Convert input audio to a video output, representing the audio vector
       scope.

       The filter is used to measure the difference between channels of stereo
       audio stream. A monoaural signal, consisting of identical left and
       right signal, results in straight vertical line. Any stereo separation
       is visible as a deviation from this line, creating a Lissajous figure.
       If the straight (or deviation from it) but horizontal line appears this
       indicates that the left and right channels are out of phase.

       The filter accepts the followin