FFMPEG-ALL(1)							 FFMPEG-ALL(1)

NAME
       ffmpeg - ffmpeg video converter

SYNOPSIS
       ffmpeg [global_options] {[input_file_options] -i input_url} ...
       {[output_file_options] output_url} ...

DESCRIPTION
       ffmpeg is a very fast video and audio converter that can also grab from
       a live audio/video source. It can also convert between arbitrary sample
       rates and resize video on the fly with a high quality polyphase filter.

       ffmpeg reads from an arbitrary number of input "files" (which can be
       regular files, pipes, network streams, grabbing devices, etc.),
       specified by the "-i" option, and writes to an arbitrary number of
       output "files", which are specified by a plain output url. Anything
       found on the command line which cannot be interpreted as an option is
       considered to be an output url.

       Each input or output url can, in principle, contain any number of
       streams of different types (video/audio/subtitle/attachment/data). The
       allowed number and/or types of streams may be limited by the container
       format. Selecting which streams from which inputs will go into which
       output is either done automatically or with the "-map" option (see the
       Stream selection chapter).

       To refer to input files in options, you must use their indices
       (0-based). E.g.	the first input file is 0, the second is 1, etc.
       Similarly, streams within a file are referred to by their indices. E.g.
       "2:3" refers to the fourth stream in the third input file. Also see the
       Stream specifiers chapter.

       As a general rule, options are applied to the next specified file.
       Therefore, order is important, and you can have the same option on the
       command line multiple times. Each occurrence is then applied to the
       next input or output file.  Exceptions from this rule are the global
       options (e.g. verbosity level), which should be specified first.

       Do not mix input and output files -- first specify all input files,
       then all output files. Also do not mix options which belong to
       different files. All options apply ONLY to the next input or output
       file and are reset between files.

       ·   To set the video bitrate of the output file to 64 kbit/s:

		   ffmpeg -i input.avi -b:v 64k -bufsize 64k output.avi

       ·   To force the frame rate of the output file to 24 fps:

		   ffmpeg -i input.avi -r 24 output.avi

       ·   To force the frame rate of the input file (valid for raw formats
	   only) to 1 fps and the frame rate of the output file to 24 fps:

		   ffmpeg -r 1 -i input.m2v -r 24 output.avi

       The format option may be needed for raw input files.

DETAILED DESCRIPTION
       The transcoding process in ffmpeg for each output can be described by
       the following diagram:

		_______ 	     ______________
	       |       |	    |		   |
	       | input |  demuxer   | encoded data |   decoder
	       | file  | ---------> | packets	   | -----+
	       |_______|	    |______________|	  |
							  v
						      _________
						     |	       |
						     | decoded |
						     | frames  |
						     |_________|
		________	     ______________	  |
	       |	|	    |		   |	  |
	       | output | <-------- | encoded data | <----+
	       | file	|   muxer   | packets	   |   encoder
	       |________|	    |______________|

       ffmpeg calls the libavformat library (containing demuxers) to read
       input files and get packets containing encoded data from them. When
       there are multiple input files, ffmpeg tries to keep them synchronized
       by tracking lowest timestamp on any active input stream.

       Encoded packets are then passed to the decoder (unless streamcopy is
       selected for the stream, see further for a description). The decoder
       produces uncompressed frames (raw video/PCM audio/...) which can be
       processed further by filtering (see next section). After filtering, the
       frames are passed to the encoder, which encodes them and outputs
       encoded packets. Finally those are passed to the muxer, which writes
       the encoded packets to the output file.

   Filtering
       Before encoding, ffmpeg can process raw audio and video frames using
       filters from the libavfilter library. Several chained filters form a
       filter graph. ffmpeg distinguishes between two types of filtergraphs:
       simple and complex.

       Simple filtergraphs

       Simple filtergraphs are those that have exactly one input and output,
       both of the same type. In the above diagram they can be represented by
       simply inserting an additional step between decoding and encoding:

		_________			 ______________
	       |	 |			|	       |
	       | decoded |			| encoded data |
	       | frames  |\		      _ | packets      |
	       |_________| \		      /||______________|
			    \	__________   /
		 simple     _\||	  | /  encoder
		 filtergraph   | filtered |/
			       | frames   |
			       |__________|

       Simple filtergraphs are configured with the per-stream -filter option
       (with -vf and -af aliases for video and audio respectively).  A simple
       filtergraph for video can look for example like this:

		_______        _____________	    _______	   ________
	       |       |      | 	    |	   |	   |	  |	   |
	       | input | ---> | deinterlace | ---> | scale | ---> | output |
	       |_______|      |_____________|	   |_______|	  |________|

       Note that some filters change frame properties but not frame contents.
       E.g. the "fps" filter in the example above changes number of frames,
       but does not touch the frame contents. Another example is the "setpts"
       filter, which only sets timestamps and otherwise passes the frames
       unchanged.

       Complex filtergraphs

       Complex filtergraphs are those which cannot be described as simply a
       linear processing chain applied to one stream. This is the case, for
       example, when the graph has more than one input and/or output, or when
       output stream type is different from input. They can be represented
       with the following diagram:

		_________
	       |	 |
	       | input 0 |\		       __________
	       |_________| \		      | 	 |
			    \	_________    /| output 0 |
			     \ |	 |  / |__________|
		_________     \| complex | /
	       |	 |     |	 |/
	       | input 1 |---->| filter  |\
	       |_________|     |	 | \   __________
			      /| graph	 |  \ | 	 |
			     / |	 |   \| output 1 |
		_________   /  |_________|    |__________|
	       |	 | /
	       | input 2 |/
	       |_________|

       Complex filtergraphs are configured with the -filter_complex option.
       Note that this option is global, since a complex filtergraph, by its
       nature, cannot be unambiguously associated with a single stream or
       file.

       The -lavfi option is equivalent to -filter_complex.

       A trivial example of a complex filtergraph is the "overlay" filter,
       which has two video inputs and one video output, containing one video
       overlaid on top of the other. Its audio counterpart is the "amix"
       filter.

   Stream copy
       Stream copy is a mode selected by supplying the "copy" parameter to the
       -codec option. It makes ffmpeg omit the decoding and encoding step for
       the specified stream, so it does only demuxing and muxing. It is useful
       for changing the container format or modifying container-level
       metadata. The diagram above will, in this case, simplify to this:

		_______ 	     ______________	       ________
	       |       |	    |		   |	      |        |
	       | input |  demuxer   | encoded data |  muxer   | output |
	       | file  | ---------> | packets	   | -------> | file   |
	       |_______|	    |______________|	      |________|

       Since there is no decoding or encoding, it is very fast and there is no
       quality loss. However, it might not work in some cases because of many
       factors. Applying filters is obviously also impossible, since filters
       work on uncompressed data.

STREAM SELECTION
       By default, ffmpeg includes only one stream of each type (video, audio,
       subtitle) present in the input files and adds them to each output file.
       It picks the "best" of each based upon the following criteria: for
       video, it is the stream with the highest resolution, for audio, it is
       the stream with the most channels, for subtitles, it is the first
       subtitle stream. In the case where several streams of the same type
       rate equally, the stream with the lowest index is chosen.

       You can disable some of those defaults by using the "-vn/-an/-sn/-dn"
       options. For full manual control, use the "-map" option, which disables
       the defaults just described.

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 fmt (input/output)
	   Force input or output file format. The format is normally auto
	   detected for input files and guessed from the file extension for
	   output files, so this option is not needed in most cases.

       -i url (input)
	   input file url

       -y (global)
	   Overwrite output files without asking.

       -n (global)
	   Do not overwrite output files, and exit immediately if a specified
	   output file already exists.

       -stream_loop number (input)
	   Set number of times input stream shall be looped. Loop 0 means no
	   loop, loop -1 means infinite loop.

       -c[:stream_specifier] codec (input/output,per-stream)
       -codec[:stream_specifier] codec (input/output,per-stream)
	   Select an encoder (when used before an output file) or a decoder
	   (when used before an input file) for one or more streams. codec is
	   the name of a decoder/encoder or a special value "copy" (output
	   only) to indicate that the stream is not to be re-encoded.

	   For example

		   ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT

	   encodes all video streams with libx264 and copies all audio
	   streams.

	   For each stream, the last matching "c" option is applied, so

		   ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT

	   will copy all the streams except the second video, which will be
	   encoded with libx264, and the 138th audio, which will be encoded
	   with libvorbis.

       -t duration (input/output)
	   When used as an input option (before "-i"), limit the duration of
	   data read from the input file.

	   When used as an output option (before an output url), stop writing
	   the output after its duration reaches duration.

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

	   -to and -t are mutually exclusive and -t has priority.

       -to position (output)
	   Stop writing the output at position.  position must be a time
	   duration specification, see the Time duration section in the
	   ffmpeg-utils(1) manual.

	   -to and -t are mutually exclusive and -t has priority.

       -fs limit_size (output)
	   Set the file size limit, expressed in bytes. No further chunk of
	   bytes is written after the limit is exceeded. The size of the
	   output file is slightly more than the requested file size.

       -ss position (input/output)
	   When used as an input option (before "-i"), seeks in this input
	   file to position. Note that in most formats it is not possible to
	   seek exactly, so ffmpeg will seek to the closest seek point before
	   position.  When transcoding and -accurate_seek is enabled (the
	   default), this extra segment between the seek point and position
	   will be decoded and discarded. When doing stream copy or when
	   -noaccurate_seek is used, it will be preserved.

	   When used as an output option (before an output url), decodes but
	   discards input until the timestamps reach position.

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

       -sseof position (input/output)
	   Like the "-ss" option but relative to the "end of file". That is
	   negative values are earlier in the file, 0 is at EOF.

       -itsoffset offset (input)
	   Set the input time offset.

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

	   The offset is added to the timestamps of the input files.
	   Specifying a positive offset means that the corresponding streams
	   are delayed by the time duration specified in offset.

       -timestamp date (output)
	   Set the recording timestamp in the container.

	   date must be a date specification, see the Date section in the
	   ffmpeg-utils(1) manual.

       -metadata[:metadata_specifier] key=value (output,per-metadata)
	   Set a metadata key/value pair.

	   An optional metadata_specifier may be given to set metadata on
	   streams, chapters or programs. See "-map_metadata" documentation
	   for details.

	   This option overrides metadata set with "-map_metadata". It is also
	   possible to delete metadata by using an empty value.

	   For example, for setting the title in the output file:

		   ffmpeg -i in.avi -metadata title="my title" out.flv

	   To set the language of the first audio stream:

		   ffmpeg -i INPUT -metadata:s:a:0 language=eng OUTPUT

       -disposition[:stream_specifier] value (output,per-stream)
	   Sets the disposition for a stream.

	   This option overrides the disposition copied from the input stream.
	   It is also possible to delete the disposition by setting it to 0.

	   The following dispositions are recognized:

	   default
	   dub
	   original
	   comment
	   lyrics
	   karaoke
	   forced
	   hearing_impaired
	   visual_impaired
	   clean_effects
	   captions
	   descriptions
	   metadata

	   For example, to make the second audio stream the default stream:

		   ffmpeg -i in.mkv -disposition:a:1 default out.mkv

	   To make the second subtitle stream the default stream and remove
	   the default disposition from the first subtitle stream:

		   ffmpeg -i INPUT -disposition:s:0 0 -disposition:s:1 default OUTPUT

       -program
       [title=title:][program_num=program_num:]st=stream[:st=stream...]
       (output)
	   Creates a program with the specified title, program_num and adds
	   the specified stream(s) to it.

       -target type (output)
	   Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). type
	   may be prefixed with "pal-", "ntsc-" or "film-" to use the
	   corresponding standard. All the format options (bitrate, codecs,
	   buffer sizes) are then set automatically. You can just type:

		   ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg

	   Nevertheless you can specify additional options as long as you know
	   they do not conflict with the standard, as in:

		   ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg

       -dframes number (output)
	   Set the number of data frames to output. This is an obsolete alias
	   for "-frames:d", which you should use instead.

       -frames[:stream_specifier] framecount (output,per-stream)
	   Stop writing to the stream after framecount frames.

       -q[:stream_specifier] q (output,per-stream)
       -qscale[:stream_specifier] q (output,per-stream)
	   Use fixed quality scale (VBR). The meaning of q/qscale is codec-
	   dependent.  If qscale is used without a stream_specifier then it
	   applies only to the video stream, this is to maintain compatibility
	   with previous behavior and as specifying the same codec specific
	   value to 2 different codecs that is audio and video generally is
	   not what is intended when no stream_specifier is used.

       -filter[:stream_specifier] filtergraph (output,per-stream)
	   Create the filtergraph specified by filtergraph and use it to
	   filter the stream.

	   filtergraph is a description of the filtergraph to apply to the
	   stream, and must have a single input and a single output of the
	   same type of the stream. In the filtergraph, the input is
	   associated to the label "in", and the output to the label "out".
	   See the ffmpeg-filters manual for more information about the
	   filtergraph syntax.

	   See the -filter_complex option if you want to create filtergraphs
	   with multiple inputs and/or outputs.

       -filter_script[:stream_specifier] filename (output,per-stream)
	   This option is similar to -filter, the only difference is that its
	   argument is the name of the file from which a filtergraph
	   description is to be read.

       -filter_threads nb_threads (global)
	   Defines how many threads are used to process a filter pipeline.
	   Each pipeline will produce a thread pool with this many threads
	   available for parallel processing.  The default is the number of
	   available CPUs.

       -pre[:stream_specifier] preset_name (output,per-stream)
	   Specify the preset for matching stream(s).

       -stats (global)
	   Print encoding progress/statistics. It is on by default, to
	   explicitly disable it you need to specify "-nostats".

       -progress url (global)
	   Send program-friendly progress information to url.

	   Progress information is written approximately every second and at
	   the end of the encoding process. It is made of "key=value" lines.
	   key consists of only alphanumeric characters. The last key of a
	   sequence of progress information is always "progress".

       -stdin
	   Enable interaction on standard input. On by default unless standard
	   input is used as an input. To explicitly disable interaction you
	   need to specify "-nostdin".

	   Disabling interaction on standard input is useful, for example, if
	   ffmpeg is in the background process group. Roughly the same result
	   can be achieved with "ffmpeg ... < /dev/null" but it requires a
	   shell.

       -debug_ts (global)
	   Print timestamp information. It is off by default. This option is
	   mostly useful for testing and debugging purposes, and the output
	   format may change from one version to another, so it should not be
	   employed by portable scripts.

	   See also the option "-fdebug ts".

       -attach filename (output)
	   Add an attachment to the output file. This is supported by a few
	   formats like Matroska for e.g. fonts used in rendering subtitles.
	   Attachments are implemented as a specific type of stream, so this
	   option will add a new stream to the file. It is then possible to
	   use per-stream options on this stream in the usual way. Attachment
	   streams created with this option will be created after all the
	   other streams (i.e. those created with "-map" or automatic
	   mappings).

	   Note that for Matroska you also have to set the mimetype metadata
	   tag:

		   ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv

	   (assuming that the attachment stream will be third in the output
	   file).

       -dump_attachment[:stream_specifier] filename (input,per-stream)
	   Extract the matching attachment stream into a file named filename.
	   If filename is empty, then the value of the "filename" metadata tag
	   will be used.

	   E.g. to extract the first attachment to a file named 'out.ttf':

		   ffmpeg -dump_attachment:t:0 out.ttf -i INPUT

	   To extract all attachments to files determined by the "filename"
	   tag:

		   ffmpeg -dump_attachment:t "" -i INPUT

	   Technical note -- attachments are implemented as codec extradata,
	   so this option can actually be used to extract extradata from any
	   stream, not just attachments.

       -noautorotate
	   Disable automatically rotating video based on file metadata.

   Video Options
       -vframes number (output)
	   Set the number of video frames to output. This is an obsolete alias
	   for "-frames:v", which you should use instead.

       -r[:stream_specifier] fps (input/output,per-stream)
	   Set frame rate (Hz value, fraction or abbreviation).

	   As an input option, ignore any timestamps stored in the file and
	   instead generate timestamps assuming constant frame rate fps.  This
	   is not the same as the -framerate option used for some input
	   formats like image2 or v4l2 (it used to be the same in older
	   versions of FFmpeg).  If in doubt use -framerate instead of the
	   input option -r.

	   As an output option, duplicate or drop input frames to achieve
	   constant output frame rate fps.

       -s[:stream_specifier] size (input/output,per-stream)
	   Set frame size.

	   As an input option, this is a shortcut for the video_size private
	   option, recognized by some demuxers for which the frame size is
	   either not stored in the file or is configurable -- e.g. raw video
	   or video grabbers.

	   As an output option, this inserts the "scale" video filter to the
	   end of the corresponding filtergraph. Please use the "scale" filter
	   directly to insert it at the beginning or some other place.

	   The format is wxh (default - same as source).

       -aspect[:stream_specifier] aspect (output,per-stream)
	   Set the video display aspect ratio specified by aspect.

	   aspect can be a floating point number string, or a string of the
	   form num:den, where num and den are the numerator and denominator
	   of the aspect ratio. For example "4:3", "16:9", "1.3333", and
	   "1.7777" are valid argument values.

	   If used together with -vcodec copy, it will affect the aspect ratio
	   stored at container level, but not the aspect ratio stored in
	   encoded frames, if it exists.

       -vn (output)
	   Disable video recording.

       -vcodec codec (output)
	   Set the video codec. This is an alias for "-codec:v".

       -pass[:stream_specifier] n (output,per-stream)
	   Select the pass number (1 or 2). It is used to do two-pass video
	   encoding. The statistics of the video are recorded in the first
	   pass into a log file (see also the option -passlogfile), and in the
	   second pass that log file is used to generate the video at the
	   exact requested bitrate.  On pass 1, you may just deactivate audio
	   and set output to null, examples for Windows and Unix:

		   ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL
		   ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null

       -passlogfile[:stream_specifier] prefix (output,per-stream)
	   Set two-pass log file name prefix to prefix, the default file name
	   prefix is ``ffmpeg2pass''. The complete file name will be
	   PREFIX-N.log, where N is a number specific to the output stream

       -vf filtergraph (output)
	   Create the filtergraph specified by filtergraph and use it to
	   filter the stream.

	   This is an alias for "-filter:v", see the -filter option.

   Advanced Video options
       -pix_fmt[:stream_specifier] format (input/output,per-stream)
	   Set pixel format. Use "-pix_fmts" to show all the supported pixel
	   formats.  If the selected pixel format can not be selected, ffmpeg
	   will print a warning and select the best pixel format supported by
	   the encoder.  If pix_fmt is prefixed by a "+", ffmpeg will exit
	   with an error if the requested pixel format can not be selected,
	   and automatic conversions inside filtergraphs are disabled.	If
	   pix_fmt is a single "+", ffmpeg selects the same pixel format as
	   the input (or graph output) and automatic conversions are disabled.

       -sws_flags flags (input/output)
	   Set SwScaler flags.

       -vdt n
	   Discard threshold.

       -rc_override[:stream_specifier] override (output,per-stream)
	   Rate control override for specific intervals, formatted as
	   "int,int,int" list separated with slashes. Two first values are the
	   beginning and end frame numbers, last one is quantizer to use if
	   positive, or quality factor if negative.

       -ilme
	   Force interlacing support in encoder (MPEG-2 and MPEG-4 only).  Use
	   this option if your input file is interlaced and you want to keep
	   the interlaced format for minimum losses.  The alternative is to
	   deinterlace the input stream with -deinterlace, but deinterlacing
	   introduces losses.

       -psnr
	   Calculate PSNR of compressed frames.

       -vstats
	   Dump video coding statistics to vstats_HHMMSS.log.

       -vstats_file file
	   Dump video coding statistics to file.

       -vstats_version file
	   Specifies which version of the vstats format to use. Default is 2.

	   version = 1 :

	   "frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d s_size= %8.0fkB time=
	   %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"

	   version > 1:

	   "out= %2d st= %2d frame= %5d q= %2.1f PSNR= %6.2f f_size= %6d
	   s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s"

       -top[:stream_specifier] n (output,per-stream)
	   top=1/bottom=0/auto=-1 field first

       -dc precision
	   Intra_dc_precision.

       -vtag fourcc/tag (output)
	   Force video tag/fourcc. This is an alias for "-tag:v".

       -qphist (global)
	   Show QP histogram

       -vbsf bitstream_filter
	   Deprecated see -bsf

       -force_key_frames[:stream_specifier] time[,time...] (output,per-stream)
       -force_key_frames[:stream_specifier] expr:expr (output,per-stream)
	   Force key frames at the specified timestamps, more precisely at the
	   first frames after each specified time.

	   If the argument is prefixed with "expr:", the string expr is
	   interpreted like an expression and is evaluated for each frame. A
	   key frame is forced in case the evaluation is non-zero.

	   If one of the times is ""chapters"[delta]", it is expanded into the
	   time of the beginning of all chapters in the file, shifted by
	   delta, expressed as a time in seconds.  This option can be useful
	   to ensure that a seek point is present at a chapter mark or any
	   other designated place in the output file.

	   For example, to insert a key frame at 5 minutes, plus key frames
	   0.1 second before the beginning of every chapter:

		   -force_key_frames 0:05:00,chapters-0.1

	   The expression in expr can contain the following constants:

	   n   the number of current processed frame, starting from 0

	   n_forced
	       the number of forced frames

	   prev_forced_n
	       the number of the previous forced frame, it is "NAN" when no
	       keyframe was forced yet

	   prev_forced_t
	       the time of the previous forced frame, it is "NAN" when no
	       keyframe was forced yet

	   t   the time of the current processed frame

	   For example to force a key frame every 5 seconds, you can specify:

		   -force_key_frames expr:gte(t,n_forced*5)

	   To force a key frame 5 seconds after the time of the last forced
	   one, starting from second 13:

		   -force_key_frames expr:if(isnan(prev_forced_t),gte(t,13),gte(t,prev_forced_t+5))

	   Note that forcing too many keyframes is very harmful for the
	   lookahead algorithms of certain encoders: using fixed-GOP options
	   or similar would be more efficient.

       -copyinkf[:stream_specifier] (output,per-stream)
	   When doing stream copy, copy also non-key frames found at the
	   beginning.

       -init_hw_device type[=name][:device[,key=value...]]
	   Initialise a new hardware device of type type called name, using
	   the given device parameters.  If no name is specified it will
	   receive a default name of the form "type%d".

	   The meaning of device and the following arguments depends on the
	   device type:

	   cuda
	       device is the number of the CUDA device.

	   dxva2
	       device is the number of the Direct3D 9 display adapter.

	   vaapi
	       device is either an X11 display name or a DRM render node.  If
	       not specified, it will attempt to open the default X11 display
	       ($DISPLAY) and then the first DRM render node
	       (/dev/dri/renderD128).

	   vdpau
	       device is an X11 display name.  If not specified, it will
	       attempt to open the default X11 display ($DISPLAY).

	   qsv device selects a value in MFX_IMPL_*. Allowed values are:

	       auto
	       sw
	       hw
	       auto_any
	       hw_any
	       hw2
	       hw3
	       hw4

	       If not specified, auto_any is used.  (Note that it may be
	       easier to achieve the desired result for QSV by creating the
	       platform-appropriate subdevice (dxva2 or vaapi) and then
	       deriving a QSV device from that.)

       -init_hw_device type[=name]@source
	   Initialise a new hardware device of type type called name, deriving
	   it from the existing device with the name source.

       -init_hw_device list
	   List all hardware device types supported in this build of ffmpeg.

       -filter_hw_device name
	   Pass the hardware device called name to all filters in any filter
	   graph.  This can be used to set the device to upload to with the
	   "hwupload" filter, or the device to map to with the "hwmap" filter.
	   Other filters may also make use of this parameter when they require
	   a hardware device.  Note that this is typically only required when
	   the input is not already in hardware frames - when it is, filters
	   will derive the device they require from the context of the frames
	   they receive as input.

	   This is a global setting, so all filters will receive the same
	   device.

       -hwaccel[:stream_specifier] hwaccel (input,per-stream)
	   Use hardware acceleration to decode the matching stream(s). The
	   allowed values of hwaccel are:

	   none
	       Do not use any hardware acceleration (the default).

	   auto
	       Automatically select the hardware acceleration method.

	   vda Use Apple VDA hardware acceleration.

	   vdpau
	       Use VDPAU (Video Decode and Presentation API for Unix) hardware
	       acceleration.

	   dxva2
	       Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

	   vaapi
	       Use VAAPI (Video Acceleration API) hardware acceleration.

	   qsv Use the Intel QuickSync Video acceleration for video
	       transcoding.

	       Unlike most other values, this option does not enable
	       accelerated decoding (that is used automatically whenever a qsv
	       decoder is selected), but accelerated transcoding, without
	       copying the frames into the system memory.

	       For it to work, both the decoder and the encoder must support
	       QSV acceleration and no filters must be used.

	   This option has no effect if the selected hwaccel is not available
	   or not supported by the chosen decoder.

	   Note that most acceleration methods are intended for playback and
	   will not be faster than software decoding on modern CPUs.
	   Additionally, ffmpeg will usually need to copy the decoded frames
	   from the GPU memory into the system memory, resulting in further
	   performance loss. This option is thus mainly useful for testing.

       -hwaccel_device[:stream_specifier] hwaccel_device (input,per-stream)
	   Select a device to use for hardware acceleration.

	   This option only makes sense when the -hwaccel option is also
	   specified.  It can either refer to an existing device created with
	   -init_hw_device by name, or it can create a new device as if
	   -init_hw_device type:hwaccel_device were called immediately before.

       -hwaccels
	   List all hardware acceleration methods supported in this build of
	   ffmpeg.

   Audio Options
       -aframes number (output)
	   Set the number of audio frames to output. This is an obsolete alias
	   for "-frames:a", which you should use instead.

       -ar[:stream_specifier] freq (input/output,per-stream)
	   Set the audio sampling frequency. For output streams it is set by
	   default to the frequency of the corresponding input stream. For
	   input streams this option only makes sense for audio grabbing
	   devices and raw demuxers and is mapped to the corresponding demuxer
	   options.

       -aq q (output)
	   Set the audio quality (codec-specific, VBR). This is an alias for
	   -q:a.

       -ac[:stream_specifier] channels (input/output,per-stream)
	   Set the number of audio channels. For output streams it is set by
	   default to the number of input audio channels. For input streams
	   this option only makes sense for audio grabbing devices and raw
	   demuxers and is mapped to the corresponding demuxer options.

       -an (output)
	   Disable audio recording.

       -acodec codec (input/output)
	   Set the audio codec. This is an alias for "-codec:a".

       -sample_fmt[:stream_specifier] sample_fmt (output,per-stream)
	   Set the audio sample format. Use "-sample_fmts" to get a list of
	   supported sample formats.

       -af filtergraph (output)
	   Create the filtergraph specified by filtergraph and use it to
	   filter the stream.

	   This is an alias for "-filter:a", see the -filter option.

   Advanced Audio options
       -atag fourcc/tag (output)
	   Force audio tag/fourcc. This is an alias for "-tag:a".

       -absf bitstream_filter
	   Deprecated, see -bsf

       -guess_layout_max channels (input,per-stream)
	   If some input channel layout is not known, try to guess only if it
	   corresponds to at most the specified number of channels. For
	   example, 2 tells to ffmpeg to recognize 1 channel as mono and 2
	   channels as stereo but not 6 channels as 5.1. The default is to
	   always try to guess. Use 0 to disable all guessing.

   Subtitle options
       -scodec codec (input/output)
	   Set the subtitle codec. This is an alias for "-codec:s".

       -sn (output)
	   Disable subtitle recording.

       -sbsf bitstream_filter
	   Deprecated, see -bsf

   Advanced Subtitle options
       -fix_sub_duration
	   Fix subtitles durations. For each subtitle, wait for the next
	   packet in the same stream and adjust the duration of the first to
	   avoid overlap. This is necessary with some subtitles codecs,
	   especially DVB subtitles, because the duration in the original
	   packet is only a rough estimate and the end is actually marked by
	   an empty subtitle frame. Failing to use this option when necessary
	   can result in exaggerated durations or muxing failures due to non-
	   monotonic timestamps.

	   Note that this option will delay the output of all data until the
	   next subtitle packet is decoded: it may increase memory consumption
	   and latency a lot.

       -canvas_size size
	   Set the size of the canvas used to render subtitles.

   Advanced options
       -map
       [-]input_file_id[:stream_specifier][?][,sync_file_id[:stream_specifier]]
       | [linklabel] (output)
	   Designate one or more input streams as a source for the output
	   file. Each input stream is identified by the input file index
	   input_file_id and the input stream index input_stream_id within the
	   input file. Both indices start at 0. If specified,
	   sync_file_id:stream_specifier sets which input stream is used as a
	   presentation sync reference.

	   The first "-map" option on the command line specifies the source
	   for output stream 0, the second "-map" option specifies the source
	   for output stream 1, etc.

	   A "-" character before the stream identifier creates a "negative"
	   mapping.  It disables matching streams from already created
	   mappings.

	   A trailing "?" after the stream index will allow the map to be
	   optional: if the map matches no streams the map will be ignored
	   instead of failing. Note the map will still fail if an invalid
	   input file index is used; such as if the map refers to a non-
	   existent input.

	   An alternative [linklabel] form will map outputs from complex
	   filter graphs (see the -filter_complex option) to the output file.
	   linklabel must correspond to a defined output link label in the
	   graph.

	   For example, to map ALL streams from the first input file to output

		   ffmpeg -i INPUT -map 0 output

	   For example, if you have two audio streams in the first input file,
	   these streams are identified by "0:0" and "0:1". You can use "-map"
	   to select which streams to place in an output file. For example:

		   ffmpeg -i INPUT -map 0:1 out.wav

	   will map the input stream in INPUT identified by "0:1" to the
	   (single) output stream in out.wav.

	   For example, to select the stream with index 2 from input file
	   a.mov (specified by the identifier "0:2"), and stream with index 6
	   from input b.mov (specified by the identifier "1:6"), and copy them
	   to the output file out.mov:

		   ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov

	   To select all video and the third audio stream from an input file:

		   ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT

	   To map all the streams except the second audio, use negative
	   mappings

		   ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT

	   To map the video and audio streams from the first input, and using
	   the trailing "?", ignore the audio mapping if no audio streams
	   exist in the first input:

		   ffmpeg -i INPUT -map 0:v -map 0:a? OUTPUT

	   To pick the English audio stream:

		   ffmpeg -i INPUT -map 0:m:language:eng OUTPUT

	   Note that using this option disables the default mappings for this
	   output file.

       -ignore_unknown
	   Ignore input streams with unknown type instead of failing if
	   copying such streams is attempted.

       -copy_unknown
	   Allow input streams with unknown type to be copied instead of
	   failing if copying such streams is attempted.

       -map_channel
       [input_file_id.stream_specifier.channel_id|-1][?][:output_file_id.stream_specifier]
	   Map an audio channel from a given input to an output. If
	   output_file_id.stream_specifier is not set, the audio channel will
	   be mapped on all the audio streams.

	   Using "-1" instead of input_file_id.stream_specifier.channel_id
	   will map a muted channel.

	   A trailing "?" will allow the map_channel to be optional: if the
	   map_channel matches no channel the map_channel will be ignored
	   instead of failing.

	   For example, assuming INPUT is a stereo audio file, you can switch
	   the two audio channels with the following command:

		   ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT

	   If you want to mute the first channel and keep the second:

		   ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT

	   The order of the "-map_channel" option specifies the order of the
	   channels in the output stream. The output channel layout is guessed
	   from the number of channels mapped (mono if one "-map_channel",
	   stereo if two, etc.). Using "-ac" in combination of "-map_channel"
	   makes the channel gain levels to be updated if input and output
	   channel layouts don't match (for instance two "-map_channel"
	   options and "-ac 6").

	   You can also extract each channel of an input to specific outputs;
	   the following command extracts two channels of the INPUT audio
	   stream (file 0, stream 0) to the respective OUTPUT_CH0 and
	   OUTPUT_CH1 outputs:

		   ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1

	   The following example splits the channels of a stereo input into
	   two separate streams, which are put into the same output file:

		   ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg

	   Note that currently each output stream can only contain channels
	   from a single input stream; you can't for example use
	   "-map_channel" to pick multiple input audio channels contained in
	   different streams (from the same or different files) and merge them
	   into a single output stream. It is therefore not currently
	   possible, for example, to turn two separate mono streams into a
	   single stereo stream. However splitting a stereo stream into two
	   single channel mono streams is possible.

	   If you need this feature, a possible workaround is to use the
	   amerge filter. For example, if you need to merge a media (here
	   input.mkv) with 2 mono audio streams into one single stereo channel
	   audio stream (and keep the video stream), you can use the following
	   command:

		   ffmpeg -i input.mkv -filter_complex "[0:1] [0:2] amerge" -c:a pcm_s16le -c:v copy output.mkv

	   To map the first two audio channels from the first input, and using
	   the trailing "?", ignore the audio channel mapping if the first
	   input is mono instead of stereo:

		   ffmpeg -i INPUT -map_channel 0.0.0 -map_channel 0.0.1? OUTPUT

       -map_metadata[:metadata_spec_out] infile[:metadata_spec_in]
       (output,per-metadata)
	   Set metadata information of the next output file from infile. Note
	   that those are file indices (zero-based), not filenames.  Optional
	   metadata_spec_in/out parameters specify, which metadata to copy.  A
	   metadata specifier can have the following forms:

	   g   global metadata, i.e. metadata that applies to the whole file

	   s[:stream_spec]
	       per-stream metadata. stream_spec is a stream specifier as
	       described in the Stream specifiers chapter. In an input
	       metadata specifier, the first matching stream is copied from.
	       In an output metadata specifier, all matching streams are
	       copied to.

	   c:chapter_index
	       per-chapter metadata. chapter_index is the zero-based chapter
	       index.

	   p:program_index
	       per-program metadata. program_index is the zero-based program
	       index.

	   If metadata specifier is omitted, it defaults to global.

	   By default, global metadata is copied from the first input file,
	   per-stream and per-chapter metadata is copied along with
	   streams/chapters. These default mappings are disabled by creating
	   any mapping of the relevant type. A negative file index can be used
	   to create a dummy mapping that just disables automatic copying.

	   For example to copy metadata from the first stream of the input
	   file to global metadata of the output file:

		   ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3

	   To do the reverse, i.e. copy global metadata to all audio streams:

		   ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv

	   Note that simple 0 would work as well in this example, since global
	   metadata is assumed by default.

       -map_chapters input_file_index (output)
	   Copy chapters from input file with index input_file_index to the
	   next output file. If no chapter mapping is specified, then chapters
	   are copied from the first input file with at least one chapter. Use
	   a negative file index to disable any chapter copying.

       -benchmark (global)
	   Show benchmarking information at the end of an encode.  Shows CPU
	   time used and maximum memory consumption.  Maximum memory
	   consumption is not supported on all systems, it will usually
	   display as 0 if not supported.

       -benchmark_all (global)
	   Show benchmarking information during the encode.  Shows CPU time
	   used in various steps (audio/video encode/decode).

       -timelimit duration (global)
	   Exit after ffmpeg has been running for duration seconds.

       -dump (global)
	   Dump each input packet to stderr.

       -hex (global)
	   When dumping packets, also dump the payload.

       -re (input)
	   Read input at native frame rate. Mainly used to simulate a grab
	   device, or live input stream (e.g. when reading from a file).
	   Should not be used with actual grab devices or live input streams
	   (where it can cause packet loss).  By default ffmpeg attempts to
	   read the input(s) as fast as possible.  This option will slow down
	   the reading of the input(s) to the native frame rate of the
	   input(s). It is useful for real-time output (e.g. live streaming).

       -loop_input
	   Loop over the input stream. Currently it works only for image
	   streams. This option is used for automatic FFserver testing.  This
	   option is deprecated, use -loop 1.

       -loop_output number_of_times
	   Repeatedly loop output for formats that support looping such as
	   animated GIF (0 will loop the output infinitely).  This option is
	   deprecated, use -loop.

       -vsync parameter
	   Video sync method.  For compatibility reasons old values can be
	   specified as numbers.  Newly added values will have to be specified
	   as strings always.

	   0, passthrough
	       Each frame is passed with its timestamp from the demuxer to the
	       muxer.

	   1, cfr
	       Frames will be duplicated and dropped to achieve exactly the
	       requested constant frame rate.

	   2, vfr
	       Frames are passed through with their timestamp or dropped so as
	       to prevent 2 frames from having the same timestamp.

	   drop
	       As passthrough but destroys all timestamps, making the muxer
	       generate fresh timestamps based on frame-rate.

	   -1, auto
	       Chooses between 1 and 2 depending on muxer capabilities. This
	       is the default method.

	   Note that the timestamps may be further modified by the muxer,
	   after this.	For example, in the case that the format option
	   avoid_negative_ts is enabled.

	   With -map you can select from which stream the timestamps should be
	   taken. You can leave either video or audio unchanged and sync the
	   remaining stream(s) to the unchanged one.

       -frame_drop_threshold parameter
	   Frame drop threshold, which specifies how much behind video frames
	   can be before they are dropped. In frame rate units, so 1.0 is one
	   frame.  The default is -1.1. One possible usecase is to avoid
	   framedrops in case of noisy timestamps or to increase frame drop
	   precision in case of exact timestamps.

       -async samples_per_second
	   Audio sync method. "Stretches/squeezes" the audio stream to match
	   the timestamps, the parameter is the maximum samples per second by
	   which the audio is changed.	-async 1 is a special case where only
	   the start of the audio stream is corrected without any later
	   correction.

	   Note that the timestamps may be further modified by the muxer,
	   after this.	For example, in the case that the format option
	   avoid_negative_ts is enabled.

	   This option has been deprecated. Use the "aresample" audio filter
	   instead.

       -copyts
	   Do not process input timestamps, but keep their values without
	   trying to sanitize them. In particular, do not remove the initial
	   start time offset value.

	   Note that, depending on the vsync option or on specific muxer
	   processing (e.g. in case the format option avoid_negative_ts is
	   enabled) the output timestamps may mismatch with the input
	   timestamps even when this option is selected.

       -start_at_zero
	   When used with copyts, shift input timestamps so they start at
	   zero.

	   This means that using e.g. "-ss 50" will make output timestamps
	   start at 50 seconds, regardless of what timestamp the input file
	   started at.

       -copytb mode
	   Specify how to set the encoder timebase when stream copying.  mode
	   is an integer numeric value, and can assume one of the following
	   values:

	   1   Use the demuxer timebase.

	       The time base is copied to the output encoder from the
	       corresponding input demuxer. This is sometimes required to
	       avoid non monotonically increasing timestamps when copying
	       video streams with variable frame rate.

	   0   Use the decoder timebase.

	       The time base is copied to the output encoder from the
	       corresponding input decoder.

	   -1  Try to make the choice automatically, in order to generate a
	       sane output.

	   Default value is -1.

       -enc_time_base[:stream_specifier] timebase (output,per-stream)
	   Set the encoder timebase. timebase is a floating point number, and
	   can assume one of the following values:

	   0   Assign a default value according to the media type.

	       For video - use 1/framerate, for audio - use 1/samplerate.

	   -1  Use the input stream timebase when possible.

	       If an input stream is not available, the default timebase will
	       be used.

	   >0  Use the provided number as the timebase.

	       This field can be provided as a ratio of two integers (e.g.
	       1:24, 1:48000) or as a floating point number (e.g. 0.04166,
	       2.0833e-5)

	   Default value is 0.

       -shortest (output)
	   Finish encoding when the shortest input stream ends.

       -dts_delta_threshold
	   Timestamp discontinuity delta threshold.

       -muxdelay seconds (input)
	   Set the maximum demux-decode delay.

       -muxpreload seconds (input)
	   Set the initial demux-decode delay.

       -streamid output-stream-index:new-value (output)
	   Assign a new stream-id value to an output stream. This option
	   should be specified prior to the output filename to which it
	   applies.  For the situation where multiple output files exist, a
	   streamid may be reassigned to a different value.

	   For example, to set the stream 0 PID to 33 and the stream 1 PID to
	   36 for an output mpegts file:

		   ffmpeg -i inurl -streamid 0:33 -streamid 1:36 out.ts

       -bsf[:stream_specifier] bitstream_filters (output,per-stream)
	   Set bitstream filters for matching streams. bitstream_filters is a
	   comma-separated list of bitstream filters. Use the "-bsfs" option
	   to get the list of bitstream filters.

		   ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264

		   ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt

       -tag[:stream_specifier] codec_tag (input/output,per-stream)
	   Force a tag/fourcc for matching streams.

       -timecode hh:mm:ssSEPff
	   Specify Timecode for writing. SEP is ':' for non drop timecode and
	   ';' (or '.') for drop.

		   ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg

       -filter_complex filtergraph (global)
	   Define a complex filtergraph, i.e. one with arbitrary number of
	   inputs and/or outputs. For simple graphs -- those with one input
	   and one output of the same type -- see the -filter options.
	   filtergraph is a description of the filtergraph, as described in
	   the ``Filtergraph syntax'' section of the ffmpeg-filters manual.

	   Input link labels must refer to input streams using the
	   "[file_index:stream_specifier]" syntax (i.e. the same as -map
	   uses). If stream_specifier matches multiple streams, the first one
	   will be used. An unlabeled input will be connected to the first
	   unused input stream of the matching type.

	   Output link labels are referred to with -map. Unlabeled outputs are
	   added to the first output file.

	   Note that with this option it is possible to use only lavfi sources
	   without normal input files.

	   For example, to overlay an image over video

		   ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map
		   '[out]' out.mkv

	   Here "[0:v]" refers to the first video stream in the first input
	   file, which is linked to the first (main) input of the overlay
	   filter. Similarly the first video stream in the second input is
	   linked to the second (overlay) input of overlay.

	   Assuming there is only one video stream in each input file, we can
	   omit input labels, so the above is equivalent to

		   ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map
		   '[out]' out.mkv

	   Furthermore we can omit the output label and the single output from
	   the filter graph will be added to the output file automatically, so
	   we can simply write

		   ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv

	   To generate 5 seconds of pure red video using lavfi "color" source:

		   ffmpeg -filter_complex 'color=c=red' -t 5 out.mkv

       -filter_complex_threads nb_threads (global)
	   Defines how many threads are used to process a filter_complex
	   graph.  Similar to filter_threads but used for "-filter_complex"
	   graphs only.  The default is the number of available CPUs.

       -lavfi filtergraph (global)
	   Define a complex filtergraph, i.e. one with arbitrary number of
	   inputs and/or outputs. Equivalent to -filter_complex.

       -filter_complex_script filename (global)
	   This option is similar to -filter_complex, the only difference is
	   that its argument is the name of the file from which a complex
	   filtergraph description is to be read.

       -accurate_seek (input)
	   This option enables or disables accurate seeking in input files
	   with the -ss option. It is enabled by default, so seeking is
	   accurate when transcoding. Use -noaccurate_seek to disable it,
	   which may be useful e.g. when copying some streams and transcoding
	   the others.

       -seek_timestamp (input)
	   This option enables or disables seeking by timestamp in input files
	   with the -ss option. It is disabled by default. If enabled, the
	   argument to the -ss option is considered an actual timestamp, and
	   is not offset by the start time of the file. This matters only for
	   files which do not start from timestamp 0, such as transport
	   streams.

       -thread_queue_size size (input)
	   This option sets the maximum number of queued packets when reading
	   from the file or device. With low latency / high rate live streams,
	   packets may be discarded if they are not read in a timely manner;
	   raising this value can avoid it.

       -override_ffserver (global)
	   Overrides the input specifications from ffserver. Using this option
	   you can map any input stream to ffserver and control many aspects
	   of the encoding from ffmpeg. Without this option ffmpeg will
	   transmit to ffserver what is requested by ffserver.

	   The option is intended for cases where features are needed that
	   cannot be specified to ffserver but can be to ffmpeg.

       -sdp_file file (global)
	   Print sdp information for an output stream to file.	This allows
	   dumping sdp information when at least one output isn't an rtp
	   stream. (Requires at least one of the output formats to be rtp).

       -discard (input)
	   Allows discarding specific streams or frames of streams at the
	   demuxer.  Not all demuxers support this.

	   none
	       Discard no frame.

	   default
	       Default, which discards no frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   all Discard all frames.

       -abort_on flags (global)
	   Stop and abort on various conditions. The following flags are
	   available:

	   empty_output
	       No packets were passed to the muxer, the output is empty.

       -xerror (global)
	   Stop and exit on error

       -max_muxing_queue_size packets (output,per-stream)
	   When transcoding audio and/or video streams, ffmpeg will not begin
	   writing into the output until it has one packet for each such
	   stream. While waiting for that to happen, packets for other streams
	   are buffered. This option sets the size of this buffer, in packets,
	   for the matching output stream.

	   The default value of this option should be high enough for most
	   uses, so only touch this option if you are sure that you need it.

       As a special exception, you can use a bitmap subtitle stream as input:
       it will be converted into a video with the same size as the largest
       video in the file, or 720x576 if no video is present. Note that this is
       an experimental and temporary solution. It will be removed once
       libavfilter has proper support for subtitles.

       For example, to hardcode subtitles on top of a DVB-T recording stored
       in MPEG-TS format, delaying the subtitles by 1 second:

	       ffmpeg -i input.ts -filter_complex \
		 '[#0x2ef] setpts=PTS+1/TB [sub] ; [#0x2d0] [sub] overlay' \
		 -sn -map '#0x2dc' output.mkv

       (0x2d0, 0x2dc and 0x2ef are the MPEG-TS PIDs of respectively the video,
       audio and subtitles streams; 0:0, 0:3 and 0:7 would have worked too)

   Preset files
       A preset file contains a sequence of option=value pairs, one for each
       line, specifying a sequence of options which would be awkward to
       specify on the command line. Lines starting with the hash ('#')
       character are ignored and are used to provide comments. Check the
       presets directory in the FFmpeg source tree for examples.

       There are two types of preset files: ffpreset and avpreset files.

       ffpreset files

       ffpreset files are specified with the "vpre", "apre", "spre", and
       "fpre" options. The "fpre" option takes the filename of the preset
       instead of a preset name as input and can be used for any kind of
       codec. For the "vpre", "apre", and "spre" options, the options
       specified in a preset file are applied to the currently selected codec
       of the same type as the preset option.

       The argument passed to the "vpre", "apre", and "spre" preset options
       identifies the preset file to use according to the following rules:

       First ffmpeg searches for a file named arg.ffpreset in the directories
       $FFMPEG_DATADIR (if set), and $HOME/.ffmpeg, and in the datadir defined
       at configuration time (usually PREFIX/share/ffmpeg) or in a ffpresets
       folder along the executable on win32, in that order. For example, if
       the argument is "libvpx-1080p", it will search for the file
       libvpx-1080p.ffpreset.

       If no such file is found, then ffmpeg will search for a file named
       codec_name-arg.ffpreset in the above-mentioned directories, where
       codec_name is the name of the codec to which the preset file options
       will be applied. For example, if you select the video codec with
       "-vcodec libvpx" and use "-vpre 1080p", then it will search for the
       file libvpx-1080p.ffpreset.

       avpreset files

       avpreset files are specified with the "pre" option. They work similar
       to ffpreset files, but they only allow encoder- specific options.
       Therefore, an option=value pair specifying an encoder cannot be used.

       When the "pre" option is specified, ffmpeg will look for files with the
       suffix .avpreset in the directories $AVCONV_DATADIR (if set), and
       $HOME/.avconv, and in the datadir defined at configuration time
       (usually PREFIX/share/ffmpeg), in that order.

       First ffmpeg searches for a file named codec_name-arg.avpreset in the
       above-mentioned directories, where codec_name is the name of the codec
       to which the preset file options will be applied. For example, if you
       select the video codec with "-vcodec libvpx" and use "-pre 1080p", then
       it will search for the file libvpx-1080p.avpreset.

       If no such file is found, then ffmpeg will search for a file named
       arg.avpreset in the same directories.

EXAMPLES
   Video and Audio grabbing
       If you specify the input format and device then ffmpeg can grab video
       and audio directly.

	       ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg

       Or with an ALSA audio source (mono input, card id 1) instead of OSS:

	       ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg

       Note that you must activate the right video source and channel before
       launching ffmpeg with any TV viewer such as
       <http://linux.bytesex.org/xawtv/> by Gerd Knorr. You also have to set
       the audio recording levels correctly with a standard mixer.

   X11 grabbing
       Grab the X11 display with ffmpeg via

	       ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0 /tmp/out.mpg

       0.0 is display.screen number of your X11 server, same as the DISPLAY
       environment variable.

	       ffmpeg -f x11grab -video_size cif -framerate 25 -i :0.0+10,20 /tmp/out.mpg

       0.0 is display.screen number of your X11 server, same as the DISPLAY
       environment variable. 10 is the x-offset and 20 the y-offset for the
       grabbing.

   Video and Audio file format conversion
       Any supported file format and protocol can serve as input to ffmpeg:

       Examples:

       ·   You can use YUV files as input:

		   ffmpeg -i /tmp/test%d.Y /tmp/out.mpg

	   It will use the files:

		   /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
		   /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...

	   The Y files use twice the resolution of the U and V files. They are
	   raw files, without header. They can be generated by all decent
	   video decoders. You must specify the size of the image with the -s
	   option if ffmpeg cannot guess it.

       ·   You can input from a raw YUV420P file:

		   ffmpeg -i /tmp/test.yuv /tmp/out.avi

	   test.yuv is a file containing raw YUV planar data. Each frame is
	   composed of the Y plane followed by the U and V planes at half
	   vertical and horizontal resolution.

       ·   You can output to a raw YUV420P file:

		   ffmpeg -i mydivx.avi hugefile.yuv

       ·   You can set several input files and output files:

		   ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg

	   Converts the audio file a.wav and the raw YUV video file a.yuv to
	   MPEG file a.mpg.

       ·   You can also do audio and video conversions at the same time:

		   ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2

	   Converts a.wav to MPEG audio at 22050 Hz sample rate.

       ·   You can encode to several formats at the same time and define a
	   mapping from input stream to output streams:

		   ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2

	   Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits.
	   '-map file:index' specifies which input stream is used for each
	   output stream, in the order of the definition of output streams.

       ·   You can transcode decrypted VOBs:

		   ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi

	   This is a typical DVD ripping example; the input is a VOB file, the
	   output an AVI file with MPEG-4 video and MP3 audio. Note that in
	   this command we use B-frames so the MPEG-4 stream is DivX5
	   compatible, and GOP size is 300 which means one intra frame every
	   10 seconds for 29.97fps input video. Furthermore, the audio stream
	   is MP3-encoded so you need to enable LAME support by passing
	   "--enable-libmp3lame" to configure.	The mapping is particularly
	   useful for DVD transcoding to get the desired audio language.

	   NOTE: To see the supported input formats, use "ffmpeg -demuxers".

       ·   You can extract images from a video, or create a video from many
	   images:

	   For extracting images from a video:

		   ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg

	   This will extract one video frame per second from the video and
	   will output them in files named foo-001.jpeg, foo-002.jpeg, etc.
	   Images will be rescaled to fit the new WxH values.

	   If you want to extract just a limited number of frames, you can use
	   the above command in combination with the "-frames:v" or "-t"
	   option, or in combination with -ss to start extracting from a
	   certain point in time.

	   For creating a video from many images:

		   ffmpeg -f image2 -framerate 12 -i foo-%03d.jpeg -s WxH foo.avi

	   The syntax "foo-%03d.jpeg" specifies to use a decimal number
	   composed of three digits padded with zeroes to express the sequence
	   number. It is the same syntax supported by the C printf function,
	   but only formats accepting a normal integer are suitable.

	   When importing an image sequence, -i also supports expanding shell-
	   like wildcard patterns (globbing) internally, by selecting the
	   image2-specific "-pattern_type glob" option.

	   For example, for creating a video from filenames matching the glob
	   pattern "foo-*.jpeg":

		   ffmpeg -f image2 -pattern_type glob -framerate 12 -i 'foo-*.jpeg' -s WxH foo.avi

       ·   You can put many streams of the same type in the output:

		   ffmpeg -i test1.avi -i test2.avi -map 1:1 -map 1:0 -map 0:1 -map 0:0 -c copy -y test12.nut

	   The resulting output file test12.nut will contain the first four
	   streams from the input files in reverse order.

       ·   To force CBR video output:

		   ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v

       ·   The four options lmin, lmax, mblmin and mblmax use 'lambda' units,
	   but you may use the QP2LAMBDA constant to easily convert from 'q'
	   units:

		   ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext

SYNTAX
       This section documents the syntax and formats employed by the FFmpeg
       libraries and tools.

   Quoting and escaping
       FFmpeg adopts the following quoting and escaping mechanism, unless
       explicitly specified. The following rules are applied:

       ·   ' and \ are special characters (respectively used for quoting and
	   escaping). In addition to them, there might be other special
	   characters depending on the specific syntax where the escaping and
	   quoting are employed.

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

       ·   All characters enclosed between '' are included literally in the
	   parsed string. The quote character ' itself cannot be quoted, so
	   you may need to close the quote and escape it.

       ·   Leading and trailing whitespaces, unless escaped or quoted, are
	   removed from the parsed string.

       Note that you may need to add a second level of escaping when using the
       command line or a script, which depends on the syntax of the adopted
       shell language.

       The function "av_get_token" defined in libavutil/avstring.h can be used
       to parse a token quoted or escaped according to the rules defined
       above.

       The tool tools/ffescape in the FFmpeg source tree can be used to
       automatically quote or escape a string in a script.

       Examples

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

		   Crime d\'Amour

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

		   'Crime d'\''Amour'

       ·   Include leading or trailing whitespaces using quoting:

		   '  this string starts and ends with whitespaces  '

       ·   Escaping and quoting can be mixed together:

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

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

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

   Date
       The accepted syntax is:

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

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

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

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

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

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

       or

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

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

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

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

       23.189
	   23.189 seconds

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

       The following abbreviations are recognized:

       ntsc
	   720x480

       pal 720x576

       qntsc
	   352x240

       qpal
	   352x288

       sntsc
	   640x480

       spal
	   768x576

       film
	   352x240

       ntsc-film
	   352x240

       sqcif
	   128x96

       qcif
	   176x144

       cif 352x288

       4cif
	   704x576

       16cif
	   1408x1152

       qqvga
	   160x120

       qvga
	   320x240

       vga 640x480

       svga
	   800x600

       xga 1024x768

       uxga
	   1600x1200

       qxga
	   2048x1536

       sxga
	   1280x1024

       qsxga
	   2560x2048

       hsxga
	   5120x4096

       wvga
	   852x480

       wxga
	   1366x768

       wsxga
	   1600x1024

       wuxga
	   1920x1200

       woxga
	   2560x1600

       wqsxga
	   3200x2048

       wquxga
	   3840x2400

       whsxga
	   6400x4096

       whuxga
	   7680x4800

       cga 320x200

       ega 640x350

       hd480
	   852x480

       hd720
	   1280x720

       hd1080
	   1920x1080

       2k  2048x1080

       2kflat
	   1998x1080

       2kscope
	   2048x858

       4k  4096x2160

       4kflat
	   3996x2160

       4kscope
	   4096x1716

       nhd 640x360

       hqvga
	   240x160

       wqvga
	   400x240

       fwqvga
	   432x240

       hvga
	   480x320

       qhd 960x540

       2kdci
	   2048x1080

       4kdci
	   4096x2160

       uhd2160
	   3840x2160

       uhd4320
	   7680x4320

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

       The following abbreviations are recognized:

       ntsc
	   30000/1001

       pal 25/1

       qntsc
	   30000/1001

       qpal
	   25/1

       sntsc
	   30000/1001

       spal
	   25/1

       film
	   24/1

       ntsc-film
	   24000/1001

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

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

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

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

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

       The string random will result in a random color.

       The following names of colors are recognized:

       AliceBlue
	   0xF0F8FF

       AntiqueWhite
	   0xFAEBD7

       Aqua
	   0x00FFFF

       Aquamarine
	   0x7FFFD4

       Azure
	   0xF0FFFF

       Beige
	   0xF5F5DC

       Bisque
	   0xFFE4C4

       Black
	   0x000000

       BlanchedAlmond
	   0xFFEBCD

       Blue
	   0x0000FF

       BlueViolet
	   0x8A2BE2

       Brown
	   0xA52A2A

       BurlyWood
	   0xDEB887

       CadetBlue
	   0x5F9EA0

       Chartreuse
	   0x7FFF00

       Chocolate
	   0xD2691E

       Coral
	   0xFF7F50

       CornflowerBlue
	   0x6495ED

       Cornsilk
	   0xFFF8DC

       Crimson
	   0xDC143C

       Cyan
	   0x00FFFF

       DarkBlue
	   0x00008B

       DarkCyan
	   0x008B8B

       DarkGoldenRod
	   0xB8860B

       DarkGray
	   0xA9A9A9

       DarkGreen
	   0x006400

       DarkKhaki
	   0xBDB76B

       DarkMagenta
	   0x8B008B

       DarkOliveGreen
	   0x556B2F

       Darkorange
	   0xFF8C00

       DarkOrchid
	   0x9932CC

       DarkRed
	   0x8B0000

       DarkSalmon
	   0xE9967A

       DarkSeaGreen
	   0x8FBC8F

       DarkSlateBlue
	   0x483D8B

       DarkSlateGray
	   0x2F4F4F

       DarkTurquoise
	   0x00CED1

       DarkViolet
	   0x9400D3

       DeepPink
	   0xFF1493

       DeepSkyBlue
	   0x00BFFF

       DimGray
	   0x696969

       DodgerBlue
	   0x1E90FF

       FireBrick
	   0xB22222

       FloralWhite
	   0xFFFAF0

       ForestGreen
	   0x228B22

       Fuchsia
	   0xFF00FF

       Gainsboro
	   0xDCDCDC

       GhostWhite
	   0xF8F8FF

       Gold
	   0xFFD700

       GoldenRod
	   0xDAA520

       Gray
	   0x808080

       Green
	   0x008000

       GreenYellow
	   0xADFF2F

       HoneyDew
	   0xF0FFF0

       HotPink
	   0xFF69B4

       IndianRed
	   0xCD5C5C

       Indigo
	   0x4B0082

       Ivory
	   0xFFFFF0

       Khaki
	   0xF0E68C

       Lavender
	   0xE6E6FA

       LavenderBlush
	   0xFFF0F5

       LawnGreen
	   0x7CFC00

       LemonChiffon
	   0xFFFACD

       LightBlue
	   0xADD8E6

       LightCoral
	   0xF08080

       LightCyan
	   0xE0FFFF

       LightGoldenRodYellow
	   0xFAFAD2

       LightGreen
	   0x90EE90

       LightGrey
	   0xD3D3D3

       LightPink
	   0xFFB6C1

       LightSalmon
	   0xFFA07A

       LightSeaGreen
	   0x20B2AA

       LightSkyBlue
	   0x87CEFA

       LightSlateGray
	   0x778899

       LightSteelBlue
	   0xB0C4DE

       LightYellow
	   0xFFFFE0

       Lime
	   0x00FF00

       LimeGreen
	   0x32CD32

       Linen
	   0xFAF0E6

       Magenta
	   0xFF00FF

       Maroon
	   0x800000

       MediumAquaMarine
	   0x66CDAA

       MediumBlue
	   0x0000CD

       MediumOrchid
	   0xBA55D3

       MediumPurple
	   0x9370D8

       MediumSeaGreen
	   0x3CB371

       MediumSlateBlue
	   0x7B68EE

       MediumSpringGreen
	   0x00FA9A

       MediumTurquoise
	   0x48D1CC

       MediumVioletRed
	   0xC71585

       MidnightBlue
	   0x191970

       MintCream
	   0xF5FFFA

       MistyRose
	   0xFFE4E1

       Moccasin
	   0xFFE4B5

       NavajoWhite
	   0xFFDEAD

       Navy
	   0x000080

       OldLace
	   0xFDF5E6

       Olive
	   0x808000

       OliveDrab
	   0x6B8E23

       Orange
	   0xFFA500

       OrangeRed
	   0xFF4500

       Orchid
	   0xDA70D6

       PaleGoldenRod
	   0xEEE8AA

       PaleGreen
	   0x98FB98

       PaleTurquoise
	   0xAFEEEE

       PaleVioletRed
	   0xD87093

       PapayaWhip
	   0xFFEFD5

       PeachPuff
	   0xFFDAB9

       Peru
	   0xCD853F

       Pink
	   0xFFC0CB

       Plum
	   0xDDA0DD

       PowderBlue
	   0xB0E0E6

       Purple
	   0x800080

       Red 0xFF0000

       RosyBrown
	   0xBC8F8F

       RoyalBlue
	   0x4169E1

       SaddleBrown
	   0x8B4513

       Salmon
	   0xFA8072

       SandyBrown
	   0xF4A460

       SeaGreen
	   0x2E8B57

       SeaShell
	   0xFFF5EE

       Sienna
	   0xA0522D

       Silver
	   0xC0C0C0

       SkyBlue
	   0x87CEEB

       SlateBlue
	   0x6A5ACD

       SlateGray
	   0x708090

       Snow
	   0xFFFAFA

       SpringGreen
	   0x00FF7F

       SteelBlue
	   0x4682B4

       Tan 0xD2B48C

       Teal
	   0x008080

       Thistle
	   0xD8BFD8

       Tomato
	   0xFF6347

       Turquoise
	   0x40E0D0

       Violet
	   0xEE82EE

       Wheat
	   0xF5DEB3

       White
	   0xFFFFFF

       WhiteSmoke
	   0xF5F5F5

       Yellow
	   0xFFFF00

       YellowGreen
	   0x9ACD32

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

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

       FL  front left

       FR  front right

       FC  front center

       LFE low frequency

       BL  back left

       BR  back right

       FLC front left-of-center

       FRC front right-of-center

       BC  back center

       SL  side left

       SR  side right

       TC  top center

       TFL top front left

       TFC top front center

       TFR top front right

       TBL top back left

       TBC top back center

       TBR top back right

       DL  downmix left

       DR  downmix right

       WL  wide left

       WR  wide right

       SDL surround direct left

       SDR surround direct right

       LFE2
	   low frequency 2

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

       mono
	   FC

       stereo
	   FL+FR

       2.1 FL+FR+LFE

       3.0 FL+FR+FC

       3.0(back)
	   FL+FR+BC

       4.0 FL+FR+FC+BC

       quad
	   FL+FR+BL+BR

       quad(side)
	   FL+FR+SL+SR

       3.1 FL+FR+FC+LFE

       5.0 FL+FR+FC+BL+BR

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

       4.1 FL+FR+FC+LFE+BC

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       downmix
	   DL+DR

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

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

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

       ·   a number of channels, in decimal, followed by 'c', yielding the
	   default channel layout for that number of channels (see the
	   function "av_get_default_channel_layout"). Note that not all
	   channel counts have a default layout.

       ·   a number of channels, in decimal, followed by 'C', yielding an
	   unknown channel layout with the specified number of channels. Note
	   that not all channel layout specification strings support unknown
	   channel layouts.

       ·   a channel layout mask, in hexadecimal starting with "0x" (see the
	   "AV_CH_*" macros in libavutil/channel_layout.h.

       Before libavutil version 53 the trailing character "c" to specify a
       number of channels was optional, but now it is required, while a
       channel layout mask can also be specified as a decimal number (if and
       only if not followed by "c" or "C").

       See also the function "av_get_channel_layout" defined in
       libavutil/channel_layout.h.

EXPRESSION EVALUATION
       When evaluating an arithmetic expression, FFmpeg uses an internal
       formula evaluator, implemented through the libavutil/eval.h interface.

       An expression may contain unary, binary operators, constants, and
       functions.

       Two expressions expr1 and expr2 can be combined to form another
       expression "expr1;expr2".  expr1 and expr2 are evaluated in turn, and
       the new expression evaluates to the value of expr2.

       The following binary operators are available: "+", "-", "*", "/", "^".

       The following unary operators are available: "+", "-".

       The following functions are available:

       abs(x)
	   Compute absolute value of x.

       acos(x)
	   Compute arccosine of x.

       asin(x)
	   Compute arcsine of x.

       atan(x)
	   Compute arctangent of x.

       atan2(x, y)
	   Compute principal value of the arc tangent of y/x.

       between(x, min, max)
	   Return 1 if x is greater than or equal to min and lesser than or
	   equal to max, 0 otherwise.

       bitand(x, y)
       bitor(x, y)
	   Compute bitwise and/or operation on x and y.

	   The results of the evaluation of x and y are converted to integers
	   before executing the bitwise operation.

	   Note that both the conversion to integer and the conversion back to
	   floating point can lose precision. Beware of unexpected results for
	   large numbers (usually 2^53 and larger).

       ceil(expr)
	   Round the value of expression expr upwards to the nearest integer.
	   For example, "ceil(1.5)" is "2.0".

       clip(x, min, max)
	   Return the value of x clipped between min and max.

       cos(x)
	   Compute cosine of x.

       cosh(x)
	   Compute hyperbolic cosine of x.

       eq(x, y)
	   Return 1 if x and y are equivalent, 0 otherwise.

       exp(x)
	   Compute exponential of x (with base "e", the Euler's number).

       floor(expr)
	   Round the value of expression expr downwards to the nearest
	   integer. For example, "floor(-1.5)" is "-2.0".

       gauss(x)
	   Compute Gauss function of x, corresponding to "exp(-x*x/2) /
	   sqrt(2*PI)".

       gcd(x, y)
	   Return the greatest common divisor of x and y. If both x and y are
	   0 or either or both are less than zero then behavior is undefined.

       gt(x, y)
	   Return 1 if x is greater than y, 0 otherwise.

       gte(x, y)
	   Return 1 if x is greater than or equal to y, 0 otherwise.

       hypot(x, y)
	   This function is similar to the C function with the same name; it
	   returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right
	   triangle with sides of length x and y, or the distance of the point
	   (x, y) from the origin.

       if(x, y)
	   Evaluate x, and if the result is non-zero return the result of the
	   evaluation of y, return 0 otherwise.

       if(x, y, z)
	   Evaluate x, and if the result is non-zero return the evaluation
	   result of y, otherwise the evaluation result of z.

       ifnot(x, y)
	   Evaluate x, and if the result is zero return the result of the
	   evaluation of y, return 0 otherwise.

       ifnot(x, y, z)
	   Evaluate x, and if the result is zero return the evaluation result
	   of y, otherwise the evaluation result of z.

       isinf(x)
	   Return 1.0 if x is +/-INFINITY, 0.0 otherwise.

       isnan(x)
	   Return 1.0 if x is NAN, 0.0 otherwise.

       ld(var)
	   Load the value of the internal variable with number var, which was
	   previously stored with st(var, expr).  The function returns the
	   loaded value.

       lerp(x, y, z)
	   Return linear interpolation between x and y by amount of z.

       log(x)
	   Compute natural logarithm of x.

       lt(x, y)
	   Return 1 if x is lesser than y, 0 otherwise.

       lte(x, y)
	   Return 1 if x is lesser than or equal to y, 0 otherwise.

       max(x, y)
	   Return the maximum between x and y.

       min(x, y)
	   Return the minimum between x and y.

       mod(x, y)
	   Compute the remainder of division of x by y.

       not(expr)
	   Return 1.0 if expr is zero, 0.0 otherwise.

       pow(x, y)
	   Compute the power of x elevated y, it is equivalent to "(x)^(y)".

       print(t)
       print(t, l)
	   Print the value of expression t with loglevel l. If l is not
	   specified then a default log level is used.	Returns the value of
	   the expression printed.

	   Prints t with loglevel l

       random(x)
	   Return a pseudo random value between 0.0 and 1.0. x is the index of
	   the internal variable which will be used to save the seed/state.

       root(expr, max)
	   Find an input value for which the function represented by expr with
	   argument ld(0) is 0 in the interval 0..max.

	   The expression in expr must denote a continuous function or the
	   result is undefined.

	   ld(0) is used to represent the function input value, which means
	   that the given expression will be evaluated multiple times with
	   various input values that the expression can access through ld(0).
	   When the expression evaluates to 0 then the corresponding input
	   value will be returned.

       round(expr)
	   Round the value of expression expr to the nearest integer. For
	   example, "round(1.5)" is "2.0".

       sin(x)
	   Compute sine of x.

       sinh(x)
	   Compute hyperbolic sine of x.

       sqrt(expr)
	   Compute the square root of expr. This is equivalent to "(expr)^.5".

       squish(x)
	   Compute expression "1/(1 + exp(4*x))".

       st(var, expr)
	   Store the value of the expression expr in an internal variable. var
	   specifies the number of the variable where to store the value, and
	   it is a value ranging from 0 to 9. The function returns the value
	   stored in the internal variable.  Note, Variables are currently not
	   shared between expressions.

       tan(x)
	   Compute tangent of x.

       tanh(x)
	   Compute hyperbolic tangent of x.

       taylor(expr, x)
       taylor(expr, x, id)
	   Evaluate a Taylor series at x, given an expression representing the
	   "ld(id)"-th derivative of a function at 0.

	   When the series does not converge the result is undefined.

	   ld(id) is used to represent the derivative order in expr, which
	   means that the given expression will be evaluated multiple times
	   with various input values that the expression can access through
	   "ld(id)". If id is not specified then 0 is assumed.

	   Note, when you have the derivatives at y instead of 0,
	   "taylor(expr, x-y)" can be used.

       time(0)
	   Return the current (wallclock) time in seconds.

       trunc(expr)
	   Round the value of expression expr towards zero to the nearest
	   integer. For example, "trunc(-1.5)" is "-1.0".

       while(cond, expr)
	   Evaluate expression expr while the expression cond is non-zero, and
	   returns the value of the last expr evaluation, or NAN if cond was
	   always false.

       The following constants are available:

       PI  area of the unit disc, approximately 3.14

       E   exp(1) (Euler's number), approximately 2.718

       PHI golden ratio (1+sqrt(5))/2, approximately 1.618

       Assuming that an expression is considered "true" if it has a non-zero
       value, note that:

       "*" works like AND

       "+" works like OR

       For example the construct:

	       if (A AND B) then C

       is equivalent to:

	       if(A*B, C)

       In your C code, you can extend the list of unary and binary functions,
       and define recognized constants, so that they are available for your
       expressions.

       The evaluator also recognizes the International System unit prefixes.
       If 'i' is appended after the prefix, binary prefixes are used, which
       are based on powers of 1024 instead of powers of 1000.  The 'B' postfix
       multiplies the value by 8, and can be appended after a unit prefix or
       used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as
       number postfix.

       The list of available International System prefixes follows, with
       indication of the corresponding powers of 10 and of 2.

       y   10^-24 / 2^-80

       z   10^-21 / 2^-70

       a   10^-18 / 2^-60

       f   10^-15 / 2^-50

       p   10^-12 / 2^-40

       n   10^-9 / 2^-30

       u   10^-6 / 2^-20

       m   10^-3 / 2^-10

       c   10^-2

       d   10^-1

       h   10^2

       k   10^3 / 2^10

       K   10^3 / 2^10

       M   10^6 / 2^20

       G   10^9 / 2^30

       T   10^12 / 2^40

       P   10^15 / 2^40

       E   10^18 / 2^50

       Z   10^21 / 2^60

       Y   10^24 / 2^70

OPENCL OPTIONS
       When FFmpeg is configured with "--enable-opencl", it is possible to set
       the options for the global OpenCL context.

       The list of supported options follows:

       build_options
	   Set build options used to compile the registered kernels.

	   See reference "OpenCL Specification Version: 1.2 chapter 5.6.4".

       platform_idx
	   Select the index of the platform to run OpenCL code.

	   The specified index must be one of the indexes in the device list
	   which can be obtained with "ffmpeg -opencl_bench" or
	   "av_opencl_get_device_list()".

       device_idx
	   Select the index of the device used to run OpenCL code.

	   The specified index must be one of the indexes in the device list
	   which can be obtained with "ffmpeg -opencl_bench" or
	   "av_opencl_get_device_list()".

CODEC OPTIONS
       libavcodec provides some generic global options, which can be set on
       all the encoders and decoders. In addition each codec may support so-
       called private options, which are specific for a given codec.

       Sometimes, a global option may only affect a specific kind of codec,
       and may be nonsensical or ignored by another, so you need to be aware
       of the meaning of the specified options. Also some options are meant
       only for decoding or encoding.

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

       The list of supported options follow:

       b integer (encoding,audio,video)
	   Set bitrate in bits/s. Default value is 200K.

       ab integer (encoding,audio)
	   Set audio bitrate (in bits/s). Default value is 128K.

       bt integer (encoding,video)
	   Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate
	   tolerance specifies how far ratecontrol is willing to deviate from
	   the target average bitrate value. This is not related to min/max
	   bitrate. Lowering tolerance too much has an adverse effect on
	   quality.

       flags flags (decoding/encoding,audio,video,subtitles)
	   Set generic flags.

	   Possible values:

	   mv4 Use four motion vector by macroblock (mpeg4).

	   qpel
	       Use 1/4 pel motion compensation.

	   loop
	       Use loop filter.

	   qscale
	       Use fixed qscale.

	   gmc Use gmc.

	   mv0 Always try a mb with mv=<0,0>.

	   input_preserved
	   pass1
	       Use internal 2pass ratecontrol in first pass mode.

	   pass2
	       Use internal 2pass ratecontrol in second pass mode.

	   gray
	       Only decode/encode grayscale.

	   emu_edge
	       Do not draw edges.

	   psnr
	       Set error[?] variables during encoding.

	   truncated
	   naq Normalize adaptive quantization.

	   ildct
	       Use interlaced DCT.

	   low_delay
	       Force low delay.

	   global_header
	       Place global headers in extradata instead of every keyframe.

	   bitexact
	       Only write platform-, build- and time-independent data. (except
	       (I)DCT).  This ensures that file and data checksums are
	       reproducible and match between platforms. Its primary use is
	       for regression testing.

	   aic Apply H263 advanced intra coding / mpeg4 ac prediction.

	   cbp Deprecated, use mpegvideo private options instead.

	   qprd
	       Deprecated, use mpegvideo private options instead.

	   ilme
	       Apply interlaced motion estimation.

	   cgop
	       Use closed gop.

       me_method integer (encoding,video)
	   Set motion estimation method.

	   Possible values:

	   zero
	       zero motion estimation (fastest)

	   full
	       full motion estimation (slowest)

	   epzs
	       EPZS motion estimation (default)

	   esa esa motion estimation (alias for full)

	   tesa
	       tesa motion estimation

	   dia dia motion estimation (alias for epzs)

	   log log motion estimation

	   phods
	       phods motion estimation

	   x1  X1 motion estimation

	   hex hex motion estimation

	   umh umh motion estimation

	   iter
	       iter motion estimation

       extradata_size integer
	   Set extradata size.

       time_base rational number
	   Set codec time base.

	   It is the fundamental unit of time (in seconds) in terms of which
	   frame timestamps are represented. For fixed-fps content, timebase
	   should be "1 / frame_rate" and timestamp increments should be
	   identically 1.

       g integer (encoding,video)
	   Set the group of picture (GOP) size. Default value is 12.

       ar integer (decoding/encoding,audio)
	   Set audio sampling rate (in Hz).

       ac integer (decoding/encoding,audio)
	   Set number of audio channels.

       cutoff integer (encoding,audio)
	   Set cutoff bandwidth. (Supported only by selected encoders, see
	   their respective documentation sections.)

       frame_size integer (encoding,audio)
	   Set audio frame size.

	   Each submitted frame except the last must contain exactly
	   frame_size samples per channel. May be 0 when the codec has
	   CODEC_CAP_VARIABLE_FRAME_SIZE set, in that case the frame size is
	   not restricted. It is set by some decoders to indicate constant
	   frame size.

       frame_number integer
	   Set the frame number.

       delay integer
       qcomp float (encoding,video)
	   Set video quantizer scale compression (VBR). It is used as a
	   constant in the ratecontrol equation. Recommended range for default
	   rc_eq: 0.0-1.0.

       qblur float (encoding,video)
	   Set video quantizer scale blur (VBR).

       qmin integer (encoding,video)
	   Set min video quantizer scale (VBR). Must be included between -1
	   and 69, default value is 2.

       qmax integer (encoding,video)
	   Set max video quantizer scale (VBR). Must be included between -1
	   and 1024, default value is 31.

       qdiff integer (encoding,video)
	   Set max difference between the quantizer scale (VBR).

       bf integer (encoding,video)
	   Set max number of B frames between non-B-frames.

	   Must be an integer between -1 and 16. 0 means that B-frames are
	   disabled. If a value of -1 is used, it will choose an automatic
	   value depending on the encoder.

	   Default value is 0.

       b_qfactor float (encoding,video)
	   Set qp factor between P and B frames.

       rc_strategy integer (encoding,video)
	   Set ratecontrol method.

       b_strategy integer (encoding,video)
	   Set strategy to choose between I/P/B-frames.

       ps integer (encoding,video)
	   Set RTP payload size in bytes.

       mv_bits integer
       header_bits integer
       i_tex_bits integer
       p_tex_bits integer
       i_count integer
       p_count integer
       skip_count integer
       misc_bits integer
       frame_bits integer
       codec_tag integer
       bug flags (decoding,video)
	   Workaround not auto detected encoder bugs.

	   Possible values:

	   autodetect
	   old_msmpeg4
	       some old lavc generated msmpeg4v3 files (no autodetection)

	   xvid_ilace
	       Xvid interlacing bug (autodetected if fourcc==XVIX)

	   ump4
	       (autodetected if fourcc==UMP4)

	   no_padding
	       padding bug (autodetected)

	   amv
	   ac_vlc
	       illegal vlc bug (autodetected per fourcc)

	   qpel_chroma
	   std_qpel
	       old standard qpel (autodetected per fourcc/version)

	   qpel_chroma2
	   direct_blocksize
	       direct-qpel-blocksize bug (autodetected per fourcc/version)

	   edge
	       edge padding bug (autodetected per fourcc/version)

	   hpel_chroma
	   dc_clip
	   ms  Workaround various bugs in microsoft broken decoders.

	   trunc
	       trancated frames

       lelim integer (encoding,video)
	   Set single coefficient elimination threshold for luminance
	   (negative values also consider DC coefficient).

       celim integer (encoding,video)
	   Set single coefficient elimination threshold for chrominance
	   (negative values also consider dc coefficient)

       strict integer (decoding/encoding,audio,video)
	   Specify how strictly to follow the standards.

	   Possible values:

	   very
	       strictly conform to an older more strict version of the spec or
	       reference software

	   strict
	       strictly conform to all the things in the spec no matter what
	       consequences

	   normal
	   unofficial
	       allow unofficial extensions

	   experimental
	       allow non standardized experimental things, experimental
	       (unfinished/work in progress/not well tested) decoders and
	       encoders.  Note: experimental decoders can pose a security
	       risk, do not use this for decoding untrusted input.

       b_qoffset float (encoding,video)
	   Set QP offset between P and B frames.

       err_detect flags (decoding,audio,video)
	   Set error detection flags.

	   Possible values:

	   crccheck
	       verify embedded CRCs

	   bitstream
	       detect bitstream specification deviations

	   buffer
	       detect improper bitstream length

	   explode
	       abort decoding on minor error detection

	   ignore_err
	       ignore decoding errors, and continue decoding.  This is useful
	       if you want to analyze the content of a video and thus want
	       everything to be decoded no matter what. This option will not
	       result in a video that is pleasing to watch in case of errors.

	   careful
	       consider things that violate the spec and have not been seen in
	       the wild as errors

	   compliant
	       consider all spec non compliancies as errors

	   aggressive
	       consider things that a sane encoder should not do as an error

       has_b_frames integer
       block_align integer
       mpeg_quant integer (encoding,video)
	   Use MPEG quantizers instead of H.263.

       qsquish float (encoding,video)
	   How to keep quantizer between qmin and qmax (0 = clip, 1 = use
	   differentiable function).

       rc_qmod_amp float (encoding,video)
	   Set experimental quantizer modulation.

       rc_qmod_freq integer (encoding,video)
	   Set experimental quantizer modulation.

       rc_override_count integer
       rc_eq string (encoding,video)
	   Set rate control equation. When computing the expression, besides
	   the standard functions defined in the section 'Expression
	   Evaluation', the following functions are available: bits2qp(bits),
	   qp2bits(qp). Also the following constants are available: iTex pTex
	   tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex
	   avgPITex avgPPTex avgBPTex avgTex.

       maxrate integer (encoding,audio,video)
	   Set max bitrate tolerance (in bits/s). Requires bufsize to be set.

       minrate integer (encoding,audio,video)
	   Set min bitrate tolerance (in bits/s). Most useful in setting up a
	   CBR encode. It is of little use elsewise.

       bufsize integer (encoding,audio,video)
	   Set ratecontrol buffer size (in bits).

       rc_buf_aggressivity float (encoding,video)
	   Currently useless.

       i_qfactor float (encoding,video)
	   Set QP factor between P and I frames.

       i_qoffset float (encoding,video)
	   Set QP offset between P and I frames.

       rc_init_cplx float (encoding,video)
	   Set initial complexity for 1-pass encoding.

       dct integer (encoding,video)
	   Set DCT algorithm.

	   Possible values:

	   auto
	       autoselect a good one (default)

	   fastint
	       fast integer

	   int accurate integer

	   mmx
	   altivec
	   faan
	       floating point AAN DCT

       lumi_mask float (encoding,video)
	   Compress bright areas stronger than medium ones.

       tcplx_mask float (encoding,video)
	   Set temporal complexity masking.

       scplx_mask float (encoding,video)
	   Set spatial complexity masking.

       p_mask float (encoding,video)
	   Set inter masking.

       dark_mask float (encoding,video)
	   Compress dark areas stronger than medium ones.

       idct integer (decoding/encoding,video)
	   Select IDCT implementation.

	   Possible values:

	   auto
	   int
	   simple
	   simplemmx
	   simpleauto
	       Automatically pick a IDCT compatible with the simple one

	   arm
	   altivec
	   sh4
	   simplearm
	   simplearmv5te
	   simplearmv6
	   simpleneon
	   simplealpha
	   ipp
	   xvidmmx
	   faani
	       floating point AAN IDCT

       slice_count integer
       ec flags (decoding,video)
	   Set error concealment strategy.

	   Possible values:

	   guess_mvs
	       iterative motion vector (MV) search (slow)

	   deblock
	       use strong deblock filter for damaged MBs

	   favor_inter
	       favor predicting from the previous frame instead of the current

       bits_per_coded_sample integer
       pred integer (encoding,video)
	   Set prediction method.

	   Possible values:

	   left
	   plane
	   median
       aspect rational number (encoding,video)
	   Set sample aspect ratio.

       sar rational number (encoding,video)
	   Set sample aspect ratio. Alias to aspect.

       debug flags (decoding/encoding,audio,video,subtitles)
	   Print specific debug info.

	   Possible values:

	   pict
	       picture info

	   rc  rate control

	   bitstream
	   mb_type
	       macroblock (MB) type

	   qp  per-block quantization parameter (QP)

	   mv  motion vector

	   dct_coeff
	   green_metadata
	       display complexity metadata for the upcoming frame, GoP or for
	       a given duration.

	   skip
	   startcode
	   pts
	   er  error recognition

	   mmco
	       memory management control operations (H.264)

	   bugs
	   vis_qp
	       visualize quantization parameter (QP), lower QP are tinted
	       greener

	   vis_mb_type
	       visualize block types

	   buffers
	       picture buffer allocations

	   thread_ops
	       threading operations

	   nomc
	       skip motion compensation

       vismv integer (decoding,video)
	   Visualize motion vectors (MVs).

	   This option is deprecated, see the codecview filter instead.

	   Possible values:

	   pf  forward predicted MVs of P-frames

	   bf  forward predicted MVs of B-frames

	   bb  backward predicted MVs of B-frames

       cmp integer (encoding,video)
	   Set full pel me compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       subcmp integer (encoding,video)
	   Set sub pel me compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       mbcmp integer (encoding,video)
	   Set macroblock compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       ildctcmp integer (encoding,video)
	   Set interlaced dct compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       dia_size integer (encoding,video)
	   Set diamond type & size for motion estimation.

       last_pred integer (encoding,video)
	   Set amount of motion predictors from the previous frame.

       preme integer (encoding,video)
	   Set pre motion estimation.

       precmp integer (encoding,video)
	   Set pre motion estimation compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       pre_dia_size integer (encoding,video)
	   Set diamond type & size for motion estimation pre-pass.

       subq integer (encoding,video)
	   Set sub pel motion estimation quality.

       dtg_active_format integer
       me_range integer (encoding,video)
	   Set limit motion vectors range (1023 for DivX player).

       ibias integer (encoding,video)
	   Set intra quant bias.

       pbias integer (encoding,video)
	   Set inter quant bias.

       color_table_id integer
       global_quality integer (encoding,audio,video)
       coder integer (encoding,video)
	   Possible values:

	   vlc variable length coder / huffman coder

	   ac  arithmetic coder

	   raw raw (no encoding)

	   rle run-length coder

	   deflate
	       deflate-based coder

       context integer (encoding,video)
	   Set context model.

       slice_flags integer
       xvmc_acceleration integer
       mbd integer (encoding,video)
	   Set macroblock decision algorithm (high quality mode).

	   Possible values:

	   simple
	       use mbcmp (default)

	   bits
	       use fewest bits

	   rd  use best rate distortion

       stream_codec_tag integer
       sc_threshold integer (encoding,video)
	   Set scene change threshold.

       lmin integer (encoding,video)
	   Set min lagrange factor (VBR).

       lmax integer (encoding,video)
	   Set max lagrange factor (VBR).

       nr integer (encoding,video)
	   Set noise reduction.

       rc_init_occupancy integer (encoding,video)
	   Set number of bits which should be loaded into the rc buffer before
	   decoding starts.

       flags2 flags (decoding/encoding,audio,video)
	   Possible values:

	   fast
	       Allow non spec compliant speedup tricks.

	   sgop
	       Deprecated, use mpegvideo private options instead.

	   noout
	       Skip bitstream encoding.

	   ignorecrop
	       Ignore cropping information from sps.

	   local_header
	       Place global headers at every keyframe instead of in extradata.

	   chunks
	       Frame data might be split into multiple chunks.

	   showall
	       Show all frames before the first keyframe.

	   skiprd
	       Deprecated, use mpegvideo private options instead.

	   export_mvs
	       Export motion vectors into frame side-data (see
	       "AV_FRAME_DATA_MOTION_VECTORS") for codecs that support it. See
	       also doc/examples/export_mvs.c.

       error integer (encoding,video)
       qns integer (encoding,video)
	   Deprecated, use mpegvideo private options instead.

       threads integer (decoding/encoding,video)
	   Set the number of threads to be used, in case the selected codec
	   implementation supports multi-threading.

	   Possible values:

	   auto, 0
	       automatically select the number of threads to set

	   Default value is auto.

       me_threshold integer (encoding,video)
	   Set motion estimation threshold.

       mb_threshold integer (encoding,video)
	   Set macroblock threshold.

       dc integer (encoding,video)
	   Set intra_dc_precision.

       nssew integer (encoding,video)
	   Set nsse weight.

       skip_top integer (decoding,video)
	   Set number of macroblock rows at the top which are skipped.

       skip_bottom integer (decoding,video)
	   Set number of macroblock rows at the bottom which are skipped.

       profile integer (encoding,audio,video)
	   Possible values:

	   unknown
	   aac_main
	   aac_low
	   aac_ssr
	   aac_ltp
	   aac_he
	   aac_he_v2
	   aac_ld
	   aac_eld
	   mpeg2_aac_low
	   mpeg2_aac_he
	   mpeg4_sp
	   mpeg4_core
	   mpeg4_main
	   mpeg4_asp
	   dts
	   dts_es
	   dts_96_24
	   dts_hd_hra
	   dts_hd_ma
       level integer (encoding,audio,video)
	   Possible values:

	   unknown
       lowres integer (decoding,audio,video)
	   Decode at 1= 1/2, 2=1/4, 3=1/8 resolutions.

       skip_threshold integer (encoding,video)
	   Set frame skip threshold.

       skip_factor integer (encoding,video)
	   Set frame skip factor.

       skip_exp integer (encoding,video)
	   Set frame skip exponent.  Negative values behave identical to the
	   corresponding positive ones, except that the score is normalized.
	   Positive values exist primarily for compatibility reasons and are
	   not so useful.

       skipcmp integer (encoding,video)
	   Set frame skip compare function.

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

	   psnr
	       sum of squared quantization errors (avoid, low quality)

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

	   dctmax
	   chroma
       border_mask float (encoding,video)
	   Increase the quantizer for macroblocks close to borders.

       mblmin integer (encoding,video)
	   Set min macroblock lagrange factor (VBR).

       mblmax integer (encoding,video)
	   Set max macroblock lagrange factor (VBR).

       mepc integer (encoding,video)
	   Set motion estimation bitrate penalty compensation (1.0 = 256).

       skip_loop_filter integer (decoding,video)
       skip_idct	integer (decoding,video)
       skip_frame	integer (decoding,video)
	   Make decoder discard processing depending on the frame type
	   selected by the option value.

	   skip_loop_filter skips frame loop filtering, skip_idct skips frame
	   IDCT/dequantization, skip_frame skips decoding.

	   Possible values:

	   none
	       Discard no frame.

	   default
	       Discard useless frames like 0-sized frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   all Discard all frames.

	   Default value is default.

       bidir_refine integer (encoding,video)
	   Refine the two motion vectors used in bidirectional macroblocks.

       brd_scale integer (encoding,video)
	   Downscale frames for dynamic B-frame decision.

       keyint_min integer (encoding,video)
	   Set minimum interval between IDR-frames.

       refs integer (encoding,video)
	   Set reference frames to consider for motion compensation.

       chromaoffset integer (encoding,video)
	   Set chroma qp offset from luma.

       trellis integer (encoding,audio,video)
	   Set rate-distortion optimal quantization.

       sc_factor integer (encoding,video)
	   Set value multiplied by qscale for each frame and added to
	   scene_change_score.

       mv0_threshold integer (encoding,video)
       b_sensitivity integer (encoding,video)
	   Adjust sensitivity of b_frame_strategy 1.

       compression_level integer (encoding,audio,video)
       min_prediction_order integer (encoding,audio)
       max_prediction_order integer (encoding,audio)
       timecode_frame_start integer (encoding,video)
	   Set GOP timecode frame start number, in non drop frame format.

       request_channels integer (decoding,audio)
	   Set desired number of audio channels.

       bits_per_raw_sample integer
       channel_layout integer (decoding/encoding,audio)
	   Possible values:

       request_channel_layout integer (decoding,audio)
	   Possible values:

       rc_max_vbv_use float (encoding,video)
       rc_min_vbv_use float (encoding,video)
       ticks_per_frame integer (decoding/encoding,audio,video)
       color_primaries integer (decoding/encoding,video)
	   Possible values:

	   bt709
	       BT.709

	   bt470m
	       BT.470 M

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   film
	       Film

	   bt2020
	       BT.2020

	   smpte428
	   smpte428_1
	       SMPTE ST 428-1

	   smpte431
	       SMPTE 431-2

	   smpte432
	       SMPTE 432-1

	   jedec-p22
	       JEDEC P22

       color_trc integer (decoding/encoding,video)
	   Possible values:

	   bt709
	       BT.709

	   gamma22
	       BT.470 M

	   gamma28
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   linear
	       Linear

	   log
	   log100
	       Log

	   log_sqrt
	   log316
	       Log square root

	   iec61966_2_4
	   iec61966-2-4
	       IEC 61966-2-4

	   bt1361
	   bt1361e
	       BT.1361

	   iec61966_2_1
	   iec61966-2-1
	       IEC 61966-2-1

	   bt2020_10
	   bt2020_10bit
	       BT.2020 - 10 bit

	   bt2020_12
	   bt2020_12bit
	       BT.2020 - 12 bit

	   smpte2084
	       SMPTE ST 2084

	   smpte428
	   smpte428_1
	       SMPTE ST 428-1

	   arib-std-b67
	       ARIB STD-B67

       colorspace integer (decoding/encoding,video)
	   Possible values:

	   rgb RGB

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   ycocg
	       YCOCG

	   bt2020nc
	   bt2020_ncl
	       BT.2020 NCL

	   bt2020c
	   bt2020_cl
	       BT.2020 CL

	   smpte2085
	       SMPTE 2085

       color_range integer (decoding/encoding,video)
	   If used as input parameter, it serves as a hint to the decoder,
	   which color_range the input has.  Possible values:

	   tv
	   mpeg
	       MPEG (219*2^(n-8))

	   pc
	   jpeg
	       JPEG (2^n-1)

       chroma_sample_location integer (decoding/encoding,video)
	   Possible values:

	   left
	   center
	   topleft
	   top
	   bottomleft
	   bottom
       log_level_offset integer
	   Set the log level offset.

       slices integer (encoding,video)
	   Number of slices, used in parallelized encoding.

       thread_type flags (decoding/encoding,video)
	   Select which multithreading methods to use.

	   Use of frame will increase decoding delay by one frame per thread,
	   so clients which cannot provide future frames should not use it.

	   Possible values:

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

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

	   frame
	       Decode more than one frame at once.

	   Default value is slice+frame.

       audio_service_type integer (encoding,audio)
	   Set audio service type.

	   Possible values:

	   ma  Main Audio Service

	   ef  Effects

	   vi  Visually Impaired

	   hi  Hearing Impaired

	   di  Dialogue

	   co  Commentary

	   em  Emergency

	   vo  Voice Over

	   ka  Karaoke

       request_sample_fmt sample_fmt (decoding,audio)
	   Set sample format audio decoders should prefer. Default value is
	   "none".

       pkt_timebase rational number
       sub_charenc encoding (decoding,subtitles)
	   Set the input subtitles character encoding.

       field_order  field_order (video)
	   Set/override the field order of the video.  Possible values:

	   progressive
	       Progressive video

	   tt  Interlaced video, top field coded and displayed first

	   bb  Interlaced video, bottom field coded and displayed first

	   tb  Interlaced video, top coded first, bottom displayed first

	   bt  Interlaced video, bottom coded first, top displayed first

       skip_alpha bool (decoding,video)
	   Set to 1 to disable processing alpha (transparency). This works
	   like the gray flag in the flags option which skips chroma
	   information instead of alpha. Default is 0.

       codec_whitelist list (input)
	   "," separated list of allowed decoders. By default all are allowed.

       dump_separator string (input)
	   Separator used to separate the fields printed on the command line
	   about the Stream parameters.  For example to separate the fields
	   with newlines and indention:

		   ffprobe -dump_separator "
					     "	-i ~/videos/matrixbench_mpeg2.mpg

       max_pixels integer (decoding/encoding,video)
	   Maximum number of pixels per image. This value can be used to avoid
	   out of memory failures due to large images.

       apply_cropping bool (decoding,video)
	   Enable cropping if cropping parameters are multiples of the
	   required alignment for the left and top parameters. If the
	   alignment is not met the cropping will be partially applied to
	   maintain alignment.	Default is 1 (enabled).  Note: The required
	   alignment depends on if "AV_CODEC_FLAG_UNALIGNED" is set and the
	   CPU. "AV_CODEC_FLAG_UNALIGNED" cannot be changed from the command
	   line. Also hardware decoders will not apply left/top Cropping.

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

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

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

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

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

   hevc
       HEVC / H.265 decoder.

       Note: the skip_loop_filter option has effect only at level "all".

   rawvideo
       Raw video decoder.

       This decoder decodes rawvideo streams.

       Options

       top top_field_first
	   Specify the assumed field type of the input video.

	   -1  the video is assumed to be progressive (default)

	   0   bottom-field-first is assumed

	   1   top-field-first is assumed

AUDIO DECODERS
       A description of some of the currently available audio decoders
       follows.

   ac3
       AC-3 audio decoder.

       This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as
       well as the undocumented RealAudio 3 (a.k.a. dnet).

       AC-3 Decoder Options

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

	   drc_scale == 0
	       DRC disabled. Produces full range audio.

	   0 < drc_scale <= 1
	       DRC enabled.  Applies a fraction of the stream DRC value.
	       Audio reproduction is between full range and full compression.

	   drc_scale > 1
	       DRC enabled. Applies drc_scale asymmetrically.  Loud sounds are
	       fully compressed.  Soft sounds are enhanced.

   flac
       FLAC audio decoder.

       This decoder aims to implement the complete FLAC specification from
       Xiph.

       FLAC Decoder options

       -use_buggy_lpc
	   The lavc FLAC encoder used to produce buggy streams with high lpc
	   values (like the default value). This option makes it possible to
	   decode such streams correctly by using lavc's old buggy lpc logic
	   for decoding.

   ffwavesynth
       Internal wave synthesizer.

       This decoder generates wave patterns according to predefined sequences.
       Its use is purely internal and the format of the data it accepts is not
       publicly documented.

   libcelt
       libcelt decoder wrapper.

       libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio
       codec.  Requires the presence of the libcelt headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-libcelt".

   libgsm
       libgsm decoder wrapper.

       libgsm allows libavcodec to decode the GSM full rate audio codec.
       Requires the presence of the libgsm headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libgsm".

       This decoder supports both the ordinary GSM and the Microsoft variant.

   libilbc
       libilbc decoder wrapper.

       libilbc allows libavcodec to decode the Internet Low Bitrate Codec
       (iLBC) audio codec. Requires the presence of the libilbc headers and
       library during configuration. You need to explicitly configure the
       build with "--enable-libilbc".

       Options

       The following option is supported by the libilbc wrapper.

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

   libopencore-amrnb
       libopencore-amrnb decoder wrapper.

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

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

   libopencore-amrwb
       libopencore-amrwb decoder wrapper.

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

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

   libopus
       libopus decoder wrapper.

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

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

SUBTITLES DECODERS
   dvbsub
       Options

       compute_clut
	   -1  Compute clut if no matching CLUT is in the stream.

	   0   Never compute CLUT

	   1   Always compute CLUT and override the one provided in the
	       stream.

       dvb_substream
	   Selects the dvb substream, or all substreams if -1 which is
	   default.

   dvdsub
       This codec decodes the bitmap subtitles used in DVDs; the same
       subtitles can also be found in VobSub file pairs and in some Matroska
       files.

       Options

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

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

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

       forced_subs_only
	   Only decode subtitle entries marked as forced. Some titles have
	   forced and non-forced subtitles in the same track. Setting this
	   flag to 1 will only keep the forced subtitles. Default value is 0.

   libzvbi-teletext
       Libzvbi allows libavcodec to decode DVB teletext pages and DVB teletext
       subtitles. Requires the presence of the libzvbi headers and library
       during configuration. You need to explicitly configure the build with
       "--enable-libzvbi".

       Options

       txt_page
	   List of teletext page numbers to decode. You may use the special *
	   string to match all pages. Pages that do not match the specified
	   list are dropped.  Default value is *.

       txt_chop_top
	   Discards the top teletext line. Default value is 1.

       txt_format
	   Specifies the format of the decoded subtitles. The teletext decoder
	   is capable of decoding the teletext pages to bitmaps or to simple
	   text, you should use "bitmap" for teletext pages, because certain
	   graphics and colors cannot be expressed in simple text. You might
	   use "text" for teletext based subtitles if your application can
	   handle simple text based subtitles. Default value is bitmap.

       txt_left
	   X offset of generated bitmaps, default is 0.

       txt_top
	   Y offset of generated bitmaps, default is 0.

       txt_chop_spaces
	   Chops leading and trailing spaces and removes empty lines from the
	   generated text. This option is useful for teletext based subtitles
	   where empty spaces may be present at the start or at the end of the
	   lines or empty lines may be present between the subtitle lines
	   because of double-sized teletext characters.  Default value is 1.

       txt_duration
	   Sets the display duration of the decoded teletext pages or
	   subtitles in milliseconds. Default value is 30000 which is 30
	   seconds.

       txt_transparent
	   Force transparent background of the generated teletext bitmaps.
	   Default value is 0 which means an opaque background.

       txt_opacity
	   Sets the opacity (0-255) of the teletext background. If
	   txt_transparent is not set, it only affects characters between a
	   start box and an end box, typically subtitles. Default value is 0
	   if txt_transparent is set, 255 otherwise.

ENCODERS
       Encoders are configured elements in FFmpeg which allow the encoding of
       multimedia streams.

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

       You can disable all the encoders with the configure option
       "--disable-encoders" and selectively enable / disable single encoders
       with the options "--enable-encoder=ENCODER" /
       "--disable-encoder=ENCODER".

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

AUDIO ENCODERS
       A description of some of the currently available audio encoders
       follows.

   aac
       Advanced Audio Coding (AAC) encoder.

       This encoder is the default AAC encoder, natively implemented into
       FFmpeg. Its quality is on par or better than libfdk_aac at the default
       bitrate of 128kbps.  This encoder also implements more options,
       profiles and samplerates than other encoders (with only the AAC-HE
       profile pending to be implemented) so this encoder has become the
       default and is the recommended choice.

       Options

       b   Set bit rate in bits/s. Setting this automatically activates
	   constant bit rate (CBR) mode. If this option is unspecified it is
	   set to 128kbps.

       q   Set quality for variable bit rate (VBR) mode. This option is valid
	   only using the ffmpeg command-line tool. For library interface
	   users, use global_quality.

       cutoff
	   Set cutoff frequency. If unspecified will allow the encoder to
	   dynamically adjust the cutoff to improve clarity on low bitrates.

       aac_coder
	   Set AAC encoder coding method. Possible values:

	   twoloop
	       Two loop searching (TLS) method.

	       This method first sets quantizers depending on band thresholds
	       and then tries to find an optimal combination by adding or
	       subtracting a specific value from all quantizers and adjusting
	       some individual quantizer a little.  Will tune itself based on
	       whether aac_is, aac_ms and aac_pns are enabled.	This is the
	       default choice for a coder.

	   anmr
	       Average noise to mask ratio (ANMR) trellis-based solution.

	       This is an experimental coder which currently produces a lower
	       quality, is more unstable and is slower than the default
	       twoloop coder but has potential.  Currently has no support for
	       the aac_is or aac_pns options.  Not currently recommended.

	   fast
	       Constant quantizer method.

	       This method sets a constant quantizer for all bands. This is
	       the fastest of all the methods and has no rate control or
	       support for aac_is or aac_pns.  Not recommended.

       aac_ms
	   Sets mid/side coding mode. The default value of "auto" will
	   automatically use M/S with bands which will benefit from such
	   coding. Can be forced for all bands using the value "enable", which
	   is mainly useful for debugging or disabled using "disable".

       aac_is
	   Sets intensity stereo coding tool usage. By default, it's enabled
	   and will automatically toggle IS for similar pairs of stereo bands
	   if it's beneficial.	Can be disabled for debugging by setting the
	   value to "disable".

       aac_pns
	   Uses perceptual noise substitution to replace low entropy high
	   frequency bands with imperceptible white noise during the decoding
	   process. By default, it's enabled, but can be disabled for
	   debugging purposes by using "disable".

       aac_tns
	   Enables the use of a multitap FIR filter which spans through the
	   high frequency bands to hide quantization noise during the encoding
	   process and is reverted by the decoder. As well as decreasing
	   unpleasant artifacts in the high range this also reduces the
	   entropy in the high bands and allows for more bits to be used by
	   the mid-low bands. By default it's enabled but can be disabled for
	   debugging by setting the option to "disable".

       aac_ltp
	   Enables the use of the long term prediction extension which
	   increases coding efficiency in very low bandwidth situations such
	   as encoding of voice or solo piano music by extending constant
	   harmonic peaks in bands throughout frames. This option is implied
	   by profile:a aac_low and is incompatible with aac_pred. Use in
	   conjunction with -ar to decrease the samplerate.

       aac_pred
	   Enables the use of a more traditional style of prediction where the
	   spectral coefficients transmitted are replaced by the difference of
	   the current coefficients minus the previous "predicted"
	   coefficients. In theory and sometimes in practice this can improve
	   quality for low to mid bitrate audio.  This option implies the
	   aac_main profile and is incompatible with aac_ltp.

       profile
	   Sets the encoding profile, possible values:

	   aac_low
	       The default, AAC "Low-complexity" profile. Is the most
	       compatible and produces decent quality.

	   mpeg2_aac_low
	       Equivalent to "-profile:a aac_low -aac_pns 0". PNS was
	       introduced with the MPEG4 specifications.

	   aac_ltp
	       Long term prediction profile, is enabled by and will enable the
	       aac_ltp option. Introduced in MPEG4.

	   aac_main
	       Main-type prediction profile, is enabled by and will enable the
	       aac_pred option. Introduced in MPEG2.

	   If this option is unspecified it is set to aac_low.

   ac3 and ac3_fixed
       AC-3 audio encoders.

       These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as
       well as the undocumented RealAudio 3 (a.k.a. dnet).

       The ac3 encoder uses floating-point math, while the ac3_fixed encoder
       only uses fixed-point integer math. This does not mean that one is
       always faster, just that one or the other may be better suited to a
       particular system. The floating-point encoder will generally produce
       better quality audio for a given bitrate. The ac3_fixed encoder is not
       the default codec for any of the output formats, so it must be
       specified explicitly using the option "-acodec ac3_fixed" in order to
       use it.

       AC-3 Metadata

       The AC-3 metadata options are used to set parameters that describe the
       audio, but in most cases do not affect the audio encoding itself. Some
       of the options do directly affect or influence the decoding and
       playback of the resulting bitstream, while others are just for
       informational purposes. A few of the options will add bits to the
       output stream that could otherwise be used for audio data, and will
       thus affect the quality of the output. Those will be indicated
       accordingly with a note in the option list below.

       These parameters are described in detail in several publicly-available
       documents.

       *<<http://www.atsc.org/cms/standards/a_52-2010.pdf>>
       *<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
       *<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
       *<<http://www.dolby.com/uploadedFiles/zz-_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>

       Metadata Control Options

       -per_frame_metadata boolean
	   Allow Per-Frame Metadata. Specifies if the encoder should check for
	   changing metadata for each frame.

	   0   The metadata values set at initialization will be used for
	       every frame in the stream. (default)

	   1   Metadata values can be changed before encoding each frame.

       Downmix Levels

       -center_mixlev level
	   Center Mix Level. The amount of gain the decoder should apply to
	   the center channel when downmixing to stereo. This field will only
	   be written to the bitstream if a center channel is present. The
	   value is specified as a scale factor. There are 3 valid values:

	   0.707
	       Apply -3dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6dB gain

       -surround_mixlev level
	   Surround Mix Level. The amount of gain the decoder should apply to
	   the surround channel(s) when downmixing to stereo. This field will
	   only be written to the bitstream if one or more surround channels
	   are present. The value is specified as a scale factor.  There are 3
	   valid values:

	   0.707
	       Apply -3dB gain

	   0.500
	       Apply -6dB gain (default)

	   0.000
	       Silence Surround Channel(s)

       Audio Production Information

       Audio Production Information is optional information describing the
       mixing environment.  Either none or both of the fields are written to
       the bitstream.

       -mixing_level number
	   Mixing Level. Specifies peak sound pressure level (SPL) in the
	   production environment when the mix was mastered. Valid values are
	   80 to 111, or -1 for unknown or not indicated. The default value is
	   -1, but that value cannot be used if the Audio Production
	   Information is written to the bitstream. Therefore, if the
	   "room_type" option is not the default value, the "mixing_level"
	   option must not be -1.

       -room_type type
	   Room Type. Describes the equalization used during the final mixing
	   session at the studio or on the dubbing stage. A large room is a
	   dubbing stage with the industry standard X-curve equalization; a
	   small room has flat equalization.  This field will not be written
	   to the bitstream if both the "mixing_level" option and the
	   "room_type" option have the default values.

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   large
	       Large Room

	   2
	   small
	       Small Room

       Other Metadata Options

       -copyright boolean
	   Copyright Indicator. Specifies whether a copyright exists for this
	   audio.

	   0
	   off No Copyright Exists (default)

	   1
	   on  Copyright Exists

       -dialnorm value
	   Dialogue Normalization. Indicates how far the average dialogue
	   level of the program is below digital 100% full scale (0 dBFS).
	   This parameter determines a level shift during audio reproduction
	   that sets the average volume of the dialogue to a preset level. The
	   goal is to match volume level between program sources. A value of
	   -31dB will result in no volume level change, relative to the source
	   volume, during audio reproduction. Valid values are whole numbers
	   in the range -31 to -1, with -31 being the default.

       -dsur_mode mode
	   Dolby Surround Mode. Specifies whether the stereo signal uses Dolby
	   Surround (Pro Logic). This field will only be written to the
	   bitstream if the audio stream is stereo. Using this option does NOT
	   mean the encoder will actually apply Dolby Surround processing.

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   off Not Dolby Surround Encoded

	   2
	   on  Dolby Surround Encoded

       -original boolean
	   Original Bit Stream Indicator. Specifies whether this audio is from
	   the original source and not a copy.

	   0
	   off Not Original Source

	   1
	   on  Original Source (default)

       Extended Bitstream Information

       The extended bitstream options are part of the Alternate Bit Stream
       Syntax as specified in Annex D of the A/52:2010 standard. It is grouped
       into 2 parts.  If any one parameter in a group is specified, all values
       in that group will be written to the bitstream.	Default values are
       used for those that are written but have not been specified.  If the
       mixing levels are written, the decoder will use these values instead of
       the ones specified in the "center_mixlev" and "surround_mixlev" options
       if it supports the Alternate Bit Stream Syntax.

       Extended Bitstream Information - Part 1

       -dmix_mode mode
	   Preferred Stereo Downmix Mode. Allows the user to select either
	   Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred
	   stereo downmix mode.

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   ltrt
	       Lt/Rt Downmix Preferred

	   2
	   loro
	       Lo/Ro Downmix Preferred

       -ltrt_cmixlev level
	   Lt/Rt Center Mix Level. The amount of gain the decoder should apply
	   to the center channel when downmixing to stereo in Lt/Rt mode.

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

       -ltrt_surmixlev level
	   Lt/Rt Surround Mix Level. The amount of gain the decoder should
	   apply to the surround channel(s) when downmixing to stereo in Lt/Rt
	   mode.

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround Channel(s)

       -loro_cmixlev level
	   Lo/Ro Center Mix Level. The amount of gain the decoder should apply
	   to the center channel when downmixing to stereo in Lo/Ro mode.

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

       -loro_surmixlev level
	   Lo/Ro Surround Mix Level. The amount of gain the decoder should
	   apply to the surround channel(s) when downmixing to stereo in Lo/Ro
	   mode.

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround Channel(s)

       Extended Bitstream Information - Part 2

       -dsurex_mode mode
	   Dolby Surround EX Mode. Indicates whether the stream uses Dolby
	   Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean
	   the encoder will actually apply Dolby Surround EX processing.

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Surround EX Off

	   2
	   off Dolby Surround EX On

       -dheadphone_mode mode
	   Dolby Headphone Mode. Indicates whether the stream uses Dolby
	   Headphone encoding (multi-channel matrixed to 2.0 for use with
	   headphones). Using this option does NOT mean the encoder will
	   actually apply Dolby Headphone processing.

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Headphone Off

	   2
	   off Dolby Headphone On

       -ad_conv_type type
	   A/D Converter Type. Indicates whether the audio has passed through
	   HDCD A/D conversion.

	   0
	   standard
	       Standard A/D Converter (default)

	   1
	   hdcd
	       HDCD A/D Converter

       Other AC-3 Encoding Options

       -stereo_rematrixing boolean
	   Stereo Rematrixing. Enables/Disables use of rematrixing for stereo
	   input. This is an optional AC-3 feature that increases quality by
	   selectively encoding the left/right channels as mid/side. This
	   option is enabled by default, and it is highly recommended that it
	   be left as enabled except for testing purposes.

       cutoff frequency
	   Set lowpass cutoff frequency. If unspecified, the encoder selects a
	   default determined by various other encoding parameters.

       Floating-Point-Only AC-3 Encoding Options

       These options are only valid for the floating-point encoder and do not
       exist for the fixed-point encoder due to the corresponding features not
       being implemented in fixed-point.

       -channel_coupling boolean
	   Enables/Disables use of channel coupling, which is an optional AC-3
	   feature that increases quality by combining high frequency
	   information from multiple channels into a single channel. The per-
	   channel high frequency information is sent with less accuracy in
	   both the frequency and time domains. This allows more bits to be
	   used for lower frequencies while preserving enough information to
	   reconstruct the high frequencies. This option is enabled by default
	   for the floating-point encoder and should generally be left as
	   enabled except for testing purposes or to increase encoding speed.

	   -1
	   auto
	       Selected by Encoder (default)

	   0
	   off Disable Channel Coupling

	   1
	   on  Enable Channel Coupling

       -cpl_start_band number
	   Coupling Start Band. Sets the channel coupling start band, from 1
	   to 15. If a value higher than the bandwidth is used, it will be
	   reduced to 1 less than the coupling end band. If auto is used, the
	   start band will be determined by the encoder based on the bit rate,
	   sample rate, and channel layout. This option has no effect if
	   channel coupling is disabled.

	   -1
	   auto
	       Selected by Encoder (default)

   flac
       FLAC (Free Lossless Audio Codec) Encoder

       Options

       The following options are supported by FFmpeg's flac encoder.

       compression_level
	   Sets the compression level, which chooses defaults for many other
	   options if they are not set explicitly. Valid values are from 0 to
	   12, 5 is the default.

       frame_size
	   Sets the size of the frames in samples per channel.

       lpc_coeff_precision
	   Sets the LPC coefficient precision, valid values are from 1 to 15,
	   15 is the default.

       lpc_type
	   Sets the first stage LPC algorithm

	   none
	       LPC is not used

	   fixed
	       fixed LPC coefficients

	   levinson
	   cholesky
       lpc_passes
	   Number of passes to use for Cholesky factorization during LPC
	   analysis

       min_partition_order
	   The minimum partition order

       max_partition_order
	   The maximum partition order

       prediction_order_method
	   estimation
	   2level
	   4level
	   8level
	   search
	       Bruteforce search

	   log
       ch_mode
	   Channel mode

	   auto
	       The mode is chosen automatically for each frame

	   indep
	       Channels are independently coded

	   left_side
	   right_side
	   mid_side
       exact_rice_parameters
	   Chooses if rice parameters are calculated exactly or approximately.
	   if set to 1 then they are chosen exactly, which slows the code down
	   slightly and improves compression slightly.

       multi_dim_quant
	   Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC
	   algorithm is applied after the first stage to finetune the
	   coefficients. This is quite slow and slightly improves compression.

   opus
       Opus encoder.

       This is a native FFmpeg encoder for the Opus format. Currently its in
       development and only implements the CELT part of the codec. Its quality
       is usually worse and at best is equal to the libopus encoder.

       Options

       b   Set bit rate in bits/s. If unspecified it uses the number of
	   channels and the layout to make a good guess.

       opus_delay
	   Sets the maximum delay in milliseconds. Lower delays than 20ms will
	   very quickly decrease quality.

   libfdk_aac
       libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

       The libfdk-aac library is based on the Fraunhofer FDK AAC code from the
       Android project.

       Requires the presence of the libfdk-aac headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libfdk-aac". The library is also incompatible with GPL, so if
       you allow the use of GPL, you should configure with "--enable-gpl
       --enable-nonfree --enable-libfdk-aac".

       This encoder is considered to produce output on par or worse at 128kbps
       to the the native FFmpeg AAC encoder but can often produce better
       sounding audio at identical or lower bitrates and has support for the
       AAC-HE profiles.

       VBR encoding, enabled through the vbr or flags +qscale options, is
       experimental and only works with some combinations of parameters.

       Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3
       or higher.

       For more information see the fdk-aac project at
       <http://sourceforge.net/p/opencore-amr/fdk-aac/>.

       Options

       The following options are mapped on the shared FFmpeg codec options.

       b   Set bit rate in bits/s. If the bitrate is not explicitly specified,
	   it is automatically set to a suitable value depending on the
	   selected profile.

	   In case VBR mode is enabled the option is ignored.

       ar  Set audio sampling rate (in Hz).

       channels
	   Set the number of audio channels.

       flags +qscale
	   Enable fixed quality, VBR (Variable Bit Rate) mode.	Note that VBR
	   is implicitly enabled when the vbr value is positive.

       cutoff
	   Set cutoff frequency. If not specified (or explicitly set to 0) it
	   will use a value automatically computed by the library. Default
	   value is 0.

       profile
	   Set audio profile.

	   The following profiles are recognized:

	   aac_low
	       Low Complexity AAC (LC)

	   aac_he
	       High Efficiency AAC (HE-AAC)

	   aac_he_v2
	       High Efficiency AAC version 2 (HE-AACv2)

	   aac_ld
	       Low Delay AAC (LD)

	   aac_eld
	       Enhanced Low Delay AAC (ELD)

	   If not specified it is set to aac_low.

       The following are private options of the libfdk_aac encoder.

       afterburner
	   Enable afterburner feature if set to 1, disabled if set to 0. This
	   improves the quality but also the required processing power.

	   Default value is 1.

       eld_sbr
	   Enable SBR (Spectral Band Replication) for ELD if set to 1,
	   disabled if set to 0.

	   Default value is 0.

       signaling
	   Set SBR/PS signaling style.

	   It can assume one of the following values:

	   default
	       choose signaling implicitly (explicit hierarchical by default,
	       implicit if global header is disabled)

	   implicit
	       implicit backwards compatible signaling

	   explicit_sbr
	       explicit SBR, implicit PS signaling

	   explicit_hierarchical
	       explicit hierarchical signaling

	   Default value is default.

       latm
	   Output LATM/LOAS encapsulated data if set to 1, disabled if set to
	   0.

	   Default value is 0.

       header_period
	   Set StreamMuxConfig and PCE repetition period (in frames) for
	   sending in-band configuration buffers within LATM/LOAS transport
	   layer.

	   Must be a 16-bits non-negative integer.

	   Default value is 0.

       vbr Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty
	   good) and 5 is highest quality. A value of 0 will disable VBR, and
	   CBR (Constant Bit Rate) is enabled.

	   Currently only the aac_low profile supports VBR encoding.

	   VBR modes 1-5 correspond to roughly the following average bit
	   rates:

	   1   32 kbps/channel

	   2   40 kbps/channel

	   3   48-56 kbps/channel

	   4   64 kbps/channel

	   5   about 80-96 kbps/channel

	   Default value is 0.

       Examples

       ·   Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4)
	   container:

		   ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a

       ·   Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the
	   High-Efficiency AAC profile:

		   ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a

   libmp3lame
       LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.

       Requires the presence of the libmp3lame headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libmp3lame".

       See libshine for a fixed-point MP3 encoder, although with a lower
       quality.

       Options

       The following options are supported by the libmp3lame wrapper. The
       lame-equivalent of the options are listed in parentheses.

       b (-b)
	   Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is
	   expressed in kilobits/s.

       q (-V)
	   Set constant quality setting for VBR. This option is valid only
	   using the ffmpeg command-line tool. For library interface users,
	   use global_quality.

       compression_level (-q)
	   Set algorithm quality. Valid arguments are integers in the 0-9
	   range, with 0 meaning highest quality but slowest, and 9 meaning
	   fastest while producing the worst quality.

       cutoff (--lowpass)
	   Set lowpass cutoff frequency. If unspecified, the encoder
	   dynamically adjusts the cutoff.

       reservoir
	   Enable use of bit reservoir when set to 1. Default value is 1. LAME
	   has this enabled by default, but can be overridden by use --nores
	   option.

       joint_stereo (-m j)
	   Enable the encoder to use (on a frame by frame basis) either L/R
	   stereo or mid/side stereo. Default value is 1.

       abr (--abr)
	   Enable the encoder to use ABR when set to 1. The lame --abr sets
	   the target bitrate, while this options only tells FFmpeg to use ABR
	   still relies on b to set bitrate.

   libopencore-amrnb
       OpenCORE Adaptive Multi-Rate Narrowband encoder.

       Requires the presence of the libopencore-amrnb headers and library
       during configuration. You need to explicitly configure the build with
       "--enable-libopencore-amrnb --enable-version3".

       This is a mono-only encoder. Officially it only supports 8000Hz sample
       rate, but you can override it by setting strict to unofficial or lower.

       Options

       b   Set bitrate in bits per second. Only the following bitrates are
	   supported, otherwise libavcodec will round to the nearest valid
	   bitrate.

	   4750
	   5150
	   5900
	   6700
	   7400
	   7950
	   10200
	   12200
       dtx Allow discontinuous transmission (generate comfort noise) when set
	   to 1. The default value is 0 (disabled).

   libopus
       libopus Opus Interactive Audio Codec encoder wrapper.

       Requires the presence of the libopus headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libopus".

       Option Mapping

       Most libopus options are modelled after the opusenc utility from opus-
       tools. The following is an option mapping chart describing options
       supported by the libopus wrapper, and their opusenc-equivalent in
       parentheses.

       b (bitrate)
	   Set the bit rate in bits/s.	FFmpeg's b option is expressed in
	   bits/s, while opusenc's bitrate in kilobits/s.

       vbr (vbr, hard-cbr, and cvbr)
	   Set VBR mode. The FFmpeg vbr option has the following valid
	   arguments, with the opusenc equivalent options in parentheses:

	   off (hard-cbr)
	       Use constant bit rate encoding.

	   on (vbr)
	       Use variable bit rate encoding (the default).

	   constrained (cvbr)
	       Use constrained variable bit rate encoding.

       compression_level (comp)
	   Set encoding algorithm complexity. Valid options are integers in
	   the 0-10 range. 0 gives the fastest encodes but lower quality,
	   while 10 gives the highest quality but slowest encoding. The
	   default is 10.

       frame_duration (framesize)
	   Set maximum frame size, or duration of a frame in milliseconds. The
	   argument must be exactly the following: 2.5, 5, 10, 20, 40, 60.
	   Smaller frame sizes achieve lower latency but less quality at a
	   given bitrate.  Sizes greater than 20ms are only interesting at
	   fairly low bitrates.  The default is 20ms.

       packet_loss (expect-loss)
	   Set expected packet loss percentage. The default is 0.

       application (N.A.)
	   Set intended application type. Valid options are listed below:

	   voip
	       Favor improved speech intelligibility.

	   audio
	       Favor faithfulness to the input (the default).

	   lowdelay
	       Restrict to only the lowest delay modes.

       cutoff (N.A.)
	   Set cutoff bandwidth in Hz. The argument must be exactly one of the
	   following: 4000, 6000, 8000, 12000, or 20000, corresponding to
	   narrowband, mediumband, wideband, super wideband, and fullband
	   respectively. The default is 0 (cutoff disabled).

       mapping_family (mapping_family)
	   Set channel mapping family to be used by the encoder. The default
	   value of -1 uses mapping family 0 for mono and stereo inputs, and
	   mapping family 1 otherwise. The default also disables the surround
	   masking and LFE bandwidth optimzations in libopus, and requires
	   that the input contains 8 channels or fewer.

	   Other values include 0 for mono and stereo, 1 for surround sound
	   with masking and LFE bandwidth optimizations, and 255 for
	   independent streams with an unspecified channel layout.

   libshine
       Shine Fixed-Point MP3 encoder wrapper.

       Shine is a fixed-point MP3 encoder. It has a far better performance on
       platforms without an FPU, e.g. armel CPUs, and some phones and tablets.
       However, as it is more targeted on performance than quality, it is not
       on par with LAME and other production-grade encoders quality-wise.
       Also, according to the project's homepage, this encoder may not be free
       of bugs as the code was written a long time ago and the project was
       dead for at least 5 years.

       This encoder only supports stereo and mono input. This is also CBR-
       only.

       The original project (last updated in early 2007) is at
       <http://sourceforge.net/projects/libshine-fxp/>. We only support the
       updated fork by the Savonet/Liquidsoap project at
       <https://github.com/savonet/shine>.

       Requires the presence of the libshine headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libshine".

       See also libmp3lame.

       Options

       The following options are supported by the libshine wrapper. The
       shineenc-equivalent of the options are listed in parentheses.

       b (-b)
	   Set bitrate expressed in bits/s for CBR. shineenc -b option is
	   expressed in kilobits/s.

   libtwolame
       TwoLAME MP2 encoder wrapper.

       Requires the presence of the libtwolame headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libtwolame".

       Options

       The following options are supported by the libtwolame wrapper. The
       twolame-equivalent options follow the FFmpeg ones and are in
       parentheses.

       b (-b)
	   Set bitrate expressed in bits/s for CBR. twolame b option is
	   expressed in kilobits/s. Default value is 128k.

       q (-V)
	   Set quality for experimental VBR support. Maximum value range is
	   from -50 to 50, useful range is from -10 to 10. The higher the
	   value, the better the quality. This option is valid only using the
	   ffmpeg command-line tool. For library interface users, use
	   global_quality.

       mode (--mode)
	   Set the mode of the resulting audio. Possible values:

	   auto
	       Choose mode automatically based on the input. This is the
	       default.

	   stereo
	       Stereo

	   joint_stereo
	       Joint stereo

	   dual_channel
	       Dual channel

	   mono
	       Mono

       psymodel (--psyc-mode)
	   Set psychoacoustic model to use in encoding. The argument must be
	   an integer between -1 and 4, inclusive. The higher the value, the
	   better the quality. The default value is 3.

       energy_levels (--energy)
	   Enable energy levels extensions when set to 1. The default value is
	   0 (disabled).

       error_protection (--protect)
	   Enable CRC error protection when set to 1. The default value is 0
	   (disabled).

       copyright (--copyright)
	   Set MPEG audio copyright flag when set to 1. The default value is 0
	   (disabled).

       original (--original)
	   Set MPEG audio original flag when set to 1. The default value is 0
	   (disabled).

   libvo-amrwbenc
       VisualOn Adaptive Multi-Rate Wideband encoder.

       Requires the presence of the libvo-amrwbenc headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libvo-amrwbenc --enable-version3".

       This is a mono-only encoder. Officially it only supports 16000Hz sample
       rate, but you can override it by setting strict to unofficial or lower.

       Options

       b   Set bitrate in bits/s. Only the following bitrates are supported,
	   otherwise libavcodec will round to the nearest valid bitrate.

	   6600
	   8850
	   12650
	   14250
	   15850
	   18250
	   19850
	   23050
	   23850
       dtx Allow discontinuous transmission (generate comfort noise) when set
	   to 1. The default value is 0 (disabled).

   libvorbis
       libvorbis encoder wrapper.

       Requires the presence of the libvorbisenc headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libvorbis".

       Options

       The following options are supported by the libvorbis wrapper. The
       oggenc-equivalent of the options are listed in parentheses.

       To get a more accurate and extensive documentation of the libvorbis
       options, consult the libvorbisenc's and oggenc's documentations.  See
       <http://xiph.org/vorbis/>, <http://wiki.xiph.org/Vorbis-tools>, and
       oggenc(1).

       b (-b)
	   Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in
	   kilobits/s.

       q (-q)
	   Set constant quality setting for VBR. The value should be a float
	   number in the range of -1.0 to 10.0. The higher the value, the
	   better the quality. The default value is 3.0.

	   This option is valid only using the ffmpeg command-line tool.  For
	   library interface users, use global_quality.

       cutoff (--advanced-encode-option lowpass_frequency=N)
	   Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's
	   related option is expressed in kHz. The default value is 0 (cutoff
	   disabled).

       minrate (-m)
	   Set minimum bitrate expressed in bits/s. oggenc -m is expressed in
	   kilobits/s.

       maxrate (-M)
	   Set maximum bitrate expressed in bits/s. oggenc -M is expressed in
	   kilobits/s. This only has effect on ABR mode.

       iblock (--advanced-encode-option impulse_noisetune=N)
	   Set noise floor bias for impulse blocks. The value is a float
	   number from -15.0 to 0.0. A negative bias instructs the encoder to
	   pay special attention to the crispness of transients in the encoded
	   audio. The tradeoff for better transient response is a higher
	   bitrate.

   libwavpack
       A wrapper providing WavPack encoding through libwavpack.

       Only lossless mode using 32-bit integer samples is supported currently.

       Requires the presence of the libwavpack headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libwavpack".

       Note that a libavcodec-native encoder for the WavPack codec exists so
       users can encode audios with this codec without using this encoder. See
       wavpackenc.

       Options

       wavpack command line utility's corresponding options are listed in
       parentheses, if any.

       frame_size (--blocksize)
	   Default is 32768.

       compression_level
	   Set speed vs. compression tradeoff. Acceptable arguments are listed
	   below:

	   0 (-f)
	       Fast mode.

	   1   Normal (default) settings.

	   2 (-h)
	       High quality.

	   3 (-hh)
	       Very high quality.

	   4-8 (-hh -xEXTRAPROC)
	       Same as 3, but with extra processing enabled.

	       4 is the same as -x2 and 8 is the same as -x6.

   mjpeg
       Motion JPEG encoder.

       Options

       huffman
	   Set the huffman encoding strategy. Possible values:

	   default
	       Use the default huffman tables. This is the default strategy.

	   optimal
	       Compute and use optimal huffman tables.

   wavpack
       WavPack lossless audio encoder.

       This is a libavcodec-native WavPack encoder. There is also an encoder
       based on libwavpack, but there is virtually no reason to use that
       encoder.

       See also libwavpack.

       Options

       The equivalent options for wavpack command line utility are listed in
       parentheses.

       Shared options

       The following shared options are effective for this encoder. Only
       special notes about this particular encoder will be documented here.
       For the general meaning of the options, see the Codec Options chapter.

       frame_size (--blocksize)
	   For this encoder, the range for this option is between 128 and
	   131072. Default is automatically decided based on sample rate and
	   number of channel.

	   For the complete formula of calculating default, see
	   libavcodec/wavpackenc.c.

       compression_level (-f, -h, -hh, and -x)
	   This option's syntax is consistent with libwavpack's.

       Private options

       joint_stereo (-j)
	   Set whether to enable joint stereo. Valid values are:

	   on (1)
	       Force mid/side audio encoding.

	   off (0)
	       Force left/right audio encoding.

	   auto
	       Let the encoder decide automatically.

       optimize_mono
	   Set whether to enable optimization for mono. This option is only
	   effective for non-mono streams. Available values:

	   on  enabled

	   off disabled

VIDEO ENCODERS
       A description of some of the currently available video encoders
       follows.

   Hap
       Vidvox Hap video encoder.

       Options

       format integer
	   Specifies the Hap format to encode.

	   hap
	   hap_alpha
	   hap_q

	   Default value is hap.

       chunks integer
	   Specifies the number of chunks to split frames into, between 1 and
	   64. This permits multithreaded decoding of large frames,
	   potentially at the cost of data-rate. The encoder may modify this
	   value to divide frames evenly.

	   Default value is 1.

       compressor integer
	   Specifies the second-stage compressor to use. If set to none,
	   chunks will be limited to 1, as chunked uncompressed frames offer
	   no benefit.

	   none
	   snappy

	   Default value is snappy.

   jpeg2000
       The native jpeg 2000 encoder is lossy by default, the "-q:v" option can
       be used to set the encoding quality. Lossless encoding can be selected
       with "-pred 1".

       Options

       format
	   Can be set to either "j2k" or "jp2" (the default) that makes it
	   possible to store non-rgb pix_fmts.

   libkvazaar
       Kvazaar H.265/HEVC encoder.

       Requires the presence of the libkvazaar headers and library during
       configuration. You need to explicitly configure the build with
       --enable-libkvazaar.

       Options

       b   Set target video bitrate in bit/s and enable rate control.

       kvazaar-params
	   Set kvazaar parameters as a list of name=value pairs separated by
	   commas (,). See kvazaar documentation for a list of options.

   libopenh264
       Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libopenh264 headers and
       library during configuration. You need to explicitly configure the
       build with "--enable-libopenh264". The library is detected using pkg-
       config.

       For more information about the library see <http://www.openh264.org>.

       Options

       The following FFmpeg global options affect the configurations of the
       libopenh264 encoder.

       b   Set the bitrate (as a number of bits per second).

       g   Set the GOP size.

       maxrate
	   Set the max bitrate (as a number of bits per second).

       flags +global_header
	   Set global header in the bitstream.

       slices
	   Set the number of slices, used in parallelized encoding. Default
	   value is 0. This is only used when slice_mode is set to fixed.

       slice_mode
	   Set slice mode. Can assume one of the following possible values:

	   fixed
	       a fixed number of slices

	   rowmb
	       one slice per row of macroblocks

	   auto
	       automatic number of slices according to number of threads

	   dyn dynamic slicing

	   Default value is auto.

       loopfilter
	   Enable loop filter, if set to 1 (automatically enabled). To disable
	   set a value of 0.

       profile
	   Set profile restrictions. If set to the value of main enable CABAC
	   (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

       max_nal_size
	   Set maximum NAL size in bytes.

       allow_skip_frames
	   Allow skipping frames to hit the target bitrate if set to 1.

   libtheora
       libtheora Theora encoder wrapper.

       Requires the presence of the libtheora headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-libtheora".

       For more information about the libtheora project see
       <http://www.theora.org/>.

       Options

       The following global options are mapped to internal libtheora options
       which affect the quality and the bitrate of the encoded stream.

       b   Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.
	   In case VBR (Variable Bit Rate) mode is enabled this option is
	   ignored.

       flags
	   Used to enable constant quality mode (VBR) encoding through the
	   qscale flag, and to enable the "pass1" and "pass2" modes.

       g   Set the GOP size.

       global_quality
	   Set the global quality as an integer in lambda units.

	   Only relevant when VBR mode is enabled with "flags +qscale". The
	   value is converted to QP units by dividing it by "FF_QP2LAMBDA",
	   clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
	   value in the native libtheora range [0-63]. A higher value
	   corresponds to a higher quality.

       q   Enable VBR mode when set to a non-negative value, and set constant
	   quality value as a double floating point value in QP units.

	   The value is clipped in the [0-10] range, and then multiplied by
	   6.3 to get a value in the native libtheora range [0-63].

	   This option is valid only using the ffmpeg command-line tool. For
	   library interface users, use global_quality.

       Examples

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

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

       ·   Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

		   ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

   libvpx
       VP8/VP9 format supported through libvpx.

       Requires the presence of the libvpx headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-libvpx".

       Options

       The following options are supported by the libvpx wrapper. The
       vpxenc-equivalent options or values are listed in parentheses for easy
       migration.

       To reduce the duplication of documentation, only the private options
       and some others requiring special attention are documented here. For
       the documentation of the undocumented generic options, see the Codec
       Options chapter.

       To get more documentation of the libvpx options, invoke the command
       ffmpeg -h encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc
       --help. Further information is available in the libvpx API
       documentation.

       b (target-bitrate)
	   Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
	   bits/s, while vpxenc's target-bitrate is in kilobits/s.

       g (kf-max-dist)
       keyint_min (kf-min-dist)
       qmin (min-q)
       qmax (max-q)
       bufsize (buf-sz, buf-optimal-sz)
	   Set ratecontrol buffer size (in bits). Note vpxenc's options are
	   specified in milliseconds, the libvpx wrapper converts this value
	   as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz =
	   bufsize * 1000 / bitrate * 5 / 6".

       rc_init_occupancy (buf-initial-sz)
	   Set number of bits which should be loaded into the rc buffer before
	   decoding starts. Note vpxenc's option is specified in milliseconds,
	   the libvpx wrapper converts this value as follows:
	   "rc_init_occupancy * 1000 / bitrate".

       undershoot-pct
	   Set datarate undershoot (min) percentage of the target bitrate.

       overshoot-pct
	   Set datarate overshoot (max) percentage of the target bitrate.

       skip_threshold (drop-frame)
       qcomp (bias-pct)
       maxrate (maxsection-pct)
	   Set GOP max bitrate in bits/s. Note vpxenc's option is specified as
	   a percentage of the target bitrate, the libvpx wrapper converts
	   this value as follows: "(maxrate * 100 / bitrate)".

       minrate (minsection-pct)
	   Set GOP min bitrate in bits/s. Note vpxenc's option is specified as
	   a percentage of the target bitrate, the libvpx wrapper converts
	   this value as follows: "(minrate * 100 / bitrate)".

       minrate, maxrate, b end-usage=cbr
	   "(minrate == maxrate == bitrate)".

       crf (end-usage=cq, cq-level)
       tune (tune)
	   psnr (psnr)
	   ssim (ssim)
       quality, deadline (deadline)
	   best
	       Use best quality deadline. Poorly named and quite slow, this
	       option should be avoided as it may give worse quality output
	       than good.

	   good
	       Use good quality deadline. This is a good trade-off between
	       speed and quality when used with the cpu-used option.

	   realtime
	       Use realtime quality deadline.

       speed, cpu-used (cpu-used)
	   Set quality/speed ratio modifier. Higher values speed up the encode
	   at the cost of quality.

       nr (noise-sensitivity)
       static-thresh
	   Set a change threshold on blocks below which they will be skipped
	   by the encoder.

       slices (token-parts)
	   Note that FFmpeg's slices option gives the total number of
	   partitions, while vpxenc's token-parts is given as
	   "log2(partitions)".

       max-intra-rate
	   Set maximum I-frame bitrate as a percentage of the target bitrate.
	   A value of 0 means unlimited.

       force_key_frames
	   "VPX_EFLAG_FORCE_KF"

       Alternate reference frame related
	   auto-alt-ref
	       Enable use of alternate reference frames (2-pass only).

	   arnr-max-frames
	       Set altref noise reduction max frame count.

	   arnr-type
	       Set altref noise reduction filter type: backward, forward,
	       centered.

	   arnr-strength
	       Set altref noise reduction filter strength.

	   rc-lookahead, lag-in-frames (lag-in-frames)
	       Set number of frames to look ahead for frametype and
	       ratecontrol.

       error-resilient
	   Enable error resiliency features.

       VP9-specific options
	   lossless
	       Enable lossless mode.

	   tile-columns
	       Set number of tile columns to use. Note this is given as
	       "log2(tile_columns)". For example, 8 tile columns would be
	       requested by setting the tile-columns option to 3.

	   tile-rows
	       Set number of tile rows to use. Note this is given as
	       "log2(tile_rows)".  For example, 4 tile rows would be requested
	       by setting the tile-rows option to 2.

	   frame-parallel
	       Enable frame parallel decodability features.

	   aq-mode
	       Set adaptive quantization mode (0: off (default), 1: variance
	       2: complexity, 3: cyclic refresh, 4: equator360).

	   colorspace color-space
	       Set input color space. The VP9 bitstream supports signaling the
	       following colorspaces:

	       rgb sRGB
	       bt709 bt709
	       unspecified unknown
	       bt470bg bt601
	       smpte170m smpte170
	       smpte240m smpte240
	       bt2020_ncl bt2020
	   row-mt boolean
	       Enable row based multi-threading.

       For more information about libvpx see: <http://www.webmproject.org/>

   libwebp
       libwebp WebP Image encoder wrapper

       libwebp is Google's official encoder for WebP images. It can encode in
       either lossy or lossless mode. Lossy images are essentially a wrapper
       around a VP8 frame. Lossless images are a separate codec developed by
       Google.

       Pixel Format

       Currently, libwebp only supports YUV420 for lossy and RGB for lossless
       due to limitations of the format and libwebp. Alpha is supported for
       either mode.  Because of API limitations, if RGB is passed in when
       encoding lossy or YUV is passed in for encoding lossless, the pixel
       format will automatically be converted using functions from libwebp.
       This is not ideal and is done only for convenience.

       Options

       -lossless boolean
	   Enables/Disables use of lossless mode. Default is 0.

       -compression_level integer
	   For lossy, this is a quality/speed tradeoff. Higher values give
	   better quality for a given size at the cost of increased encoding
	   time. For lossless, this is a size/speed tradeoff. Higher values
	   give smaller size at the cost of increased encoding time. More
	   specifically, it controls the number of extra algorithms and
	   compression tools used, and varies the combination of these tools.
	   This maps to the method option in libwebp. The valid range is 0 to
	   6.  Default is 4.

       -qscale float
	   For lossy encoding, this controls image quality, 0 to 100. For
	   lossless encoding, this controls the effort and time spent at
	   compressing more. The default value is 75. Note that for usage via
	   libavcodec, this option is called global_quality and must be
	   multiplied by FF_QP2LAMBDA.

       -preset type
	   Configuration preset. This does some automatic settings based on
	   the general type of the image.

	   none
	       Do not use a preset.

	   default
	       Use the encoder default.

	   picture
	       Digital picture, like portrait, inner shot

	   photo
	       Outdoor photograph, with natural lighting

	   drawing
	       Hand or line drawing, with high-contrast details

	   icon
	       Small-sized colorful images

	   text
	       Text-like

   libx264, libx264rgb
       x264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libx264 headers and library
       during configuration. You need to explicitly configure the build with
       "--enable-libx264".

       libx264 supports an impressive number of features, including 8x8 and
       4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC
       entropy coding, interlacing (MBAFF), lossless mode, psy optimizations
       for detail retention (adaptive quantization, psy-RD, psy-trellis).

       Many libx264 encoder options are mapped to FFmpeg global codec options,
       while unique encoder options are provided through private options.
       Additionally the x264opts and x264-params private options allows one to
       pass a list of key=value tuples as accepted by the libx264
       "x264_param_parse" function.

       The x264 project website is at
       <http://www.videolan.org/developers/x264.html>.

       The libx264rgb encoder is the same as libx264, except it accepts packed
       RGB pixel formats as input instead of YUV.

       Supported Pixel Formats

       x264 supports 8- to 10-bit color spaces. The exact bit depth is
       controlled at x264's configure time. FFmpeg only supports one bit depth
       in one particular build. In other words, it is not possible to build
       one FFmpeg with multiple versions of x264 with different bit depths.

       Options

       The following options are supported by the libx264 wrapper. The
       x264-equivalent options or values are listed in parentheses for easy
       migration.

       To reduce the duplication of documentation, only the private options
       and some others requiring special attention are documented here. For
       the documentation of the undocumented generic options, see the Codec
       Options chapter.

       To get a more accurate and extensive documentation of the libx264
       options, invoke the command x264 --fullhelp or consult the libx264
       documentation.

       b (bitrate)
	   Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
	   bits/s, while x264's bitrate is in kilobits/s.

       bf (bframes)
       g (keyint)
       qmin (qpmin)
	   Minimum quantizer scale.

       qmax (qpmax)
	   Maximum quantizer scale.

       qdiff (qpstep)
	   Maximum difference between quantizer scales.

       qblur (qblur)
	   Quantizer curve blur

       qcomp (qcomp)
	   Quantizer curve compression factor

       refs (ref)
	   Number of reference frames each P-frame can use. The range is from
	   0-16.

       sc_threshold (scenecut)
	   Sets the threshold for the scene change detection.

       trellis (trellis)
	   Performs Trellis quantization to increase efficiency. Enabled by
	   default.

       nr  (nr)
       me_range (merange)
	   Maximum range of the motion search in pixels.

       me_method (me)
	   Set motion estimation method. Possible values in the decreasing
	   order of speed:

	   dia (dia)
	   epzs (dia)
	       Diamond search with radius 1 (fastest). epzs is an alias for
	       dia.

	   hex (hex)
	       Hexagonal search with radius 2.

	   umh (umh)
	       Uneven multi-hexagon search.

	   esa (esa)
	       Exhaustive search.

	   tesa (tesa)
	       Hadamard exhaustive search (slowest).

       forced-idr
	   Normally, when forcing a I-frame type, the encoder can select any
	   type of I-frame. This option forces it to choose an IDR-frame.

       subq (subme)
	   Sub-pixel motion estimation method.

       b_strategy (b-adapt)
	   Adaptive B-frame placement decision algorithm. Use only on first-
	   pass.

       keyint_min (min-keyint)
	   Minimum GOP size.

       coder
	   Set entropy encoder. Possible values:

	   ac  Enable CABAC.

	   vlc Enable CAVLC and disable CABAC. It generates the same effect as
	       x264's --no-cabac option.

       cmp Set full pixel motion estimation comparison algorithm. Possible
	   values:

	   chroma
	       Enable chroma in motion estimation.

	   sad Ignore chroma in motion estimation. It generates the same
	       effect as x264's --no-chroma-me option.

       threads (threads)
	   Number of encoding threads.

       thread_type
	   Set multithreading technique. Possible values:

	   slice
	       Slice-based multithreading. It generates the same effect as
	       x264's --sliced-threads option.

	   frame
	       Frame-based multithreading.

       flags
	   Set encoding flags. It can be used to disable closed GOP and enable
	   open GOP by setting it to "-cgop". The result is similar to the
	   behavior of x264's --open-gop option.

       rc_init_occupancy (vbv-init)
       preset (preset)
	   Set the encoding preset.

       tune (tune)
	   Set tuning of the encoding params.

       profile (profile)
	   Set profile restrictions.

       fastfirstpass
	   Enable fast settings when encoding first pass, when set to 1. When
	   set to 0, it has the same effect of x264's --slow-firstpass option.

       crf (crf)
	   Set the quality for constant quality mode.

       crf_max (crf-max)
	   In CRF mode, prevents VBV from lowering quality beyond this point.

       qp (qp)
	   Set constant quantization rate control method parameter.

       aq-mode (aq-mode)
	   Set AQ method. Possible values:

	   none (0)
	       Disabled.

	   variance (1)
	       Variance AQ (complexity mask).

	   autovariance (2)
	       Auto-variance AQ (experimental).

       aq-strength (aq-strength)
	   Set AQ strength, reduce blocking and blurring in flat and textured
	   areas.

       psy Use psychovisual optimizations when set to 1. When set to 0, it has
	   the same effect as x264's --no-psy option.

       psy-rd  (psy-rd)
	   Set strength of psychovisual optimization, in psy-rd:psy-trellis
	   format.

       rc-lookahead (rc-lookahead)
	   Set number of frames to look ahead for frametype and ratecontrol.

       weightb
	   Enable weighted prediction for B-frames when set to 1. When set to
	   0, it has the same effect as x264's --no-weightb option.

       weightp (weightp)
	   Set weighted prediction method for P-frames. Possible values:

	   none (0)
	       Disabled

	   simple (1)
	       Enable only weighted refs

	   smart (2)
	       Enable both weighted refs and duplicates

       ssim (ssim)
	   Enable calculation and printing SSIM stats after the encoding.

       intra-refresh (intra-refresh)
	   Enable the use of Periodic Intra Refresh instead of IDR frames when
	   set to 1.

       avcintra-class (class)
	   Configure the encoder to generate AVC-Intra.  Valid values are
	   50,100 and 200

       bluray-compat (bluray-compat)
	   Configure the encoder to be compatible with the bluray standard.
	   It is a shorthand for setting "bluray-compat=1 force-cfr=1".

       b-bias (b-bias)
	   Set the influence on how often B-frames are used.

       b-pyramid (b-pyramid)
	   Set method for keeping of some B-frames as references. Possible
	   values:

	   none (none)
	       Disabled.

	   strict (strict)
	       Strictly hierarchical pyramid.

	   normal (normal)
	       Non-strict (not Blu-ray compatible).

       mixed-refs
	   Enable the use of one reference per partition, as opposed to one
	   reference per macroblock when set to 1. When set to 0, it has the
	   same effect as x264's --no-mixed-refs option.

       8x8dct
	   Enable adaptive spatial transform (high profile 8x8 transform) when
	   set to 1. When set to 0, it has the same effect as x264's
	   --no-8x8dct option.

       fast-pskip
	   Enable early SKIP detection on P-frames when set to 1. When set to
	   0, it has the same effect as x264's --no-fast-pskip option.

       aud (aud)
	   Enable use of access unit delimiters when set to 1.

       mbtree
	   Enable use macroblock tree ratecontrol when set to 1. When set to
	   0, it has the same effect as x264's --no-mbtree option.

       deblock (deblock)
	   Set loop filter parameters, in alpha:beta form.

       cplxblur (cplxblur)
	   Set fluctuations reduction in QP (before curve compression).

       partitions (partitions)
	   Set partitions to consider as a comma-separated list of. Possible
	   values in the list:

	   p8x8
	       8x8 P-frame partition.

	   p4x4
	       4x4 P-frame partition.

	   b8x8
	       4x4 B-frame partition.

	   i8x8
	       8x8 I-frame partition.

	   i4x4
	       4x4 I-frame partition.  (Enabling p4x4 requires p8x8 to be
	       enabled. Enabling i8x8 requires adaptive spatial transform
	       (8x8dct option) to be enabled.)

	   none (none)
	       Do not consider any partitions.

	   all (all)
	       Consider every partition.

       direct-pred (direct)
	   Set direct MV prediction mode. Possible values:

	   none (none)
	       Disable MV prediction.

	   spatial (spatial)
	       Enable spatial predicting.

	   temporal (temporal)
	       Enable temporal predicting.

	   auto (auto)
	       Automatically decided.

       slice-max-size (slice-max-size)
	   Set the limit of the size of each slice in bytes. If not specified
	   but RTP payload size (ps) is specified, that is used.

       stats (stats)
	   Set the file name for multi-pass stats.

       nal-hrd (nal-hrd)
	   Set signal HRD information (requires vbv-bufsize to be set).
	   Possible values:

	   none (none)
	       Disable HRD information signaling.

	   vbr (vbr)
	       Variable bit rate.

	   cbr (cbr)
	       Constant bit rate (not allowed in MP4 container).

       x264opts (N.A.)
	   Set any x264 option, see x264 --fullhelp for a list.

	   Argument is a list of key=value couples separated by ":". In filter
	   and psy-rd options that use ":" as a separator themselves, use ","
	   instead. They accept it as well since long ago but this is kept
	   undocumented for some reason.

	   For example to specify libx264 encoding options with ffmpeg:

		   ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

       a53cc boolean
	   Import closed captions (which must be ATSC compatible format) into
	   output.  Only the mpeg2 and h264 decoders provide these. Default is
	   1 (on).

       x264-params (N.A.)
	   Override the x264 configuration using a :-separated list of
	   key=value parameters.

	   This option is functionally the same as the x264opts, but is
	   duplicated for compatibility with the Libav fork.

	   For example to specify libx264 encoding options with ffmpeg:

		   ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
		   cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
		   no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT

       Encoding ffpresets for common usages are provided so they can be used
       with the general presets system (e.g. passing the pre option).

   libx265
       x265 H.265/HEVC encoder wrapper.

       This encoder requires the presence of the libx265 headers and library
       during configuration. You need to explicitly configure the build with
       --enable-libx265.

       Options

       preset
	   Set the x265 preset.

       tune
	   Set the x265 tune parameter.

       forced-idr
	   Normally, when forcing a I-frame type, the encoder can select any
	   type of I-frame. This option forces it to choose an IDR-frame.

       x265-params
	   Set x265 options using a list of key=value couples separated by
	   ":". See x265 --help for a list of options.

	   For example to specify libx265 encoding options with -x265-params:

		   ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

   libxvid
       Xvid MPEG-4 Part 2 encoder wrapper.

       This encoder requires the presence of the libxvidcore headers and
       library during configuration. You need to explicitly configure the
       build with "--enable-libxvid --enable-gpl".

       The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users
       can encode to this format without this library.

       Options

       The following options are supported by the libxvid wrapper. Some of the
       following options are listed but are not documented, and correspond to
       shared codec options. See the Codec Options chapter for their
       documentation. The other shared options which are not listed have no
       effect for the libxvid encoder.

       b
       g
       qmin
       qmax
       mpeg_quant
       threads
       bf
       b_qfactor
       b_qoffset
       flags
	   Set specific encoding flags. Possible values:

	   mv4 Use four motion vector by macroblock.

	   aic Enable high quality AC prediction.

	   gray
	       Only encode grayscale.

	   gmc Enable the use of global motion compensation (GMC).

	   qpel
	       Enable quarter-pixel motion compensation.

	   cgop
	       Enable closed GOP.

	   global_header
	       Place global headers in extradata instead of every keyframe.

       trellis
       me_method
	   Set motion estimation method. Possible values in decreasing order
	   of speed and increasing order of quality:

	   zero
	       Use no motion estimation (default).

	   phods
	   x1
	   log Enable advanced diamond zonal search for 16x16 blocks and half-
	       pixel refinement for 16x16 blocks. x1 and log are aliases for
	       phods.

	   epzs
	       Enable all of the things described above, plus advanced diamond
	       zonal search for 8x8 blocks, half-pixel refinement for 8x8
	       blocks, and motion estimation on chroma planes.

	   full
	       Enable all of the things described above, plus extended 16x16
	       and 8x8 blocks search.

       mbd Set macroblock decision algorithm. Possible values in the
	   increasing order of quality:

	   simple
	       Use macroblock comparing function algorithm (default).

	   bits
	       Enable rate distortion-based half pixel and quarter pixel
	       refinement for 16x16 blocks.

	   rd  Enable all of the things described above, plus rate distortion-
	       based half pixel and quarter pixel refinement for 8x8 blocks,
	       and rate distortion-based search using square pattern.

       lumi_aq
	   Enable lumi masking adaptive quantization when set to 1. Default is
	   0 (disabled).

       variance_aq
	   Enable variance adaptive quantization when set to 1. Default is 0
	   (disabled).

	   When combined with lumi_aq, the resulting quality will not be
	   better than any of the two specified individually. In other words,
	   the resulting quality will be the worse one of the two effects.

       ssim
	   Set structural similarity (SSIM) displaying method. Possible
	   values:

	   off Disable displaying of SSIM information.

	   avg Output average SSIM at the end of encoding to stdout. The
	       format of showing the average SSIM is:

		       Average SSIM: %f

	       For users who are not familiar with C, %f means a float number,
	       or a decimal (e.g. 0.939232).

	   frame
	       Output both per-frame SSIM data during encoding and average
	       SSIM at the end of encoding to stdout. The format of per-frame
	       information is:

			      SSIM: avg: %1.3f min: %1.3f max: %1.3f

	       For users who are not familiar with C, %1.3f means a float
	       number rounded to 3 digits after the dot (e.g. 0.932).

       ssim_acc
	   Set SSIM accuracy. Valid options are integers within the range of
	   0-4, while 0 gives the most accurate result and 4 computes the
	   fastest.

   mpeg2
       MPEG-2 video encoder.

       Options

       seq_disp_ext integer
	   Specifies if the encoder should write a sequence_display_extension
	   to the output.

	   -1
	   auto
	       Decide automatically to write it or not (this is the default)
	       by checking if the data to be written is different from the
	       default or unspecified values.

	   0
	   never
	       Never write it.

	   1
	   always
	       Always write it.

   png
       PNG image encoder.

       Private options

       dpi integer
	   Set physical density of pixels, in dots per inch, unset by default

       dpm integer
	   Set physical density of pixels, in dots per meter, unset by default

   ProRes
       Apple ProRes encoder.

       FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
       The used encoder can be chosen with the "-vcodec" option.

       Private Options for prores-ks

       profile integer
	   Select the ProRes profile to encode

	   proxy
	   lt
	   standard
	   hq
	   4444
	   4444xq
       quant_mat integer
	   Select quantization matrix.

	   auto
	   default
	   proxy
	   lt
	   standard
	   hq

	   If set to auto, the matrix matching the profile will be picked.  If
	   not set, the matrix providing the highest quality, default, will be
	   picked.

       bits_per_mb integer
	   How many bits to allot for coding one macroblock. Different
	   profiles use between 200 and 2400 bits per macroblock, the maximum
	   is 8000.

       mbs_per_slice integer
	   Number of macroblocks in each slice (1-8); the default value (8)
	   should be good in almost all situations.

       vendor string
	   Override the 4-byte vendor ID.  A custom vendor ID like apl0 would
	   claim the stream was produced by the Apple encoder.

       alpha_bits integer
	   Specify number of bits for alpha component.	Possible values are 0,
	   8 and 16.  Use 0 to disable alpha plane coding.

       Speed considerations

       In the default mode of operation the encoder has to honor frame
       constraints (i.e. not produce frames with size bigger than requested)
       while still making output picture as good as possible.  A frame
       containing a lot of small details is harder to compress and the encoder
       would spend more time searching for appropriate quantizers for each
       slice.

       Setting a higher bits_per_mb limit will improve the speed.

       For the fastest encoding speed set the qscale parameter (4 is the
       recommended value) and do not set a size constraint.

   QSV encoders
       The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)

       The ratecontrol method is selected as follows:

       ·   When global_quality is specified, a quality-based mode is used.
	   Specifically this means either

	   -   CQP - constant quantizer scale, when the qscale codec flag is
	       also set (the -qscale ffmpeg option).

	   -   LA_ICQ - intelligent constant quality with lookahead, when the
	       look_ahead option is also set.

	   -   ICQ -- intelligent constant quality otherwise.

       ·   Otherwise, a bitrate-based mode is used. For all of those, you
	   should specify at least the desired average bitrate with the b
	   option.

	   -   LA - VBR with lookahead, when the look_ahead option is
	       specified.

	   -   VCM - video conferencing mode, when the vcm option is set.

	   -   CBR - constant bitrate, when maxrate is specified and equal to
	       the average bitrate.

	   -   VBR - variable bitrate, when maxrate is specified, but is
	       higher than the average bitrate.

	   -   AVBR - average VBR mode, when maxrate is not specified. This
	       mode is further configured by the avbr_accuracy and
	       avbr_convergence options.

       Note that depending on your system, a different mode than the one you
       specified may be selected by the encoder. Set the verbosity level to
       verbose or higher to see the actual settings used by the QSV runtime.

       Additional libavcodec global options are mapped to MSDK options as
       follows:

       ·   g/gop_size -> GopPicSize

       ·   bf/max_b_frames+1 -> GopRefDist

       ·   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       ·   slices -> NumSlice

       ·   refs -> NumRefFrame

       ·   b_strategy/b_frame_strategy -> BRefType

       ·   cgop/CLOSED_GOP codec flag -> GopOptFlag

       ·   For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset
	   set the difference between QPP and QPI, and QPP and QPB
	   respectively.

       ·   Setting the coder option to the value vlc will make the H.264
	   encoder use CAVLC instead of CABAC.

   snow
       Options

       iterative_dia_size
	   dia size for the iterative motion estimation

   VAAPI encoders
       Wrappers for hardware encoders accessible via VAAPI.

       These encoders only accept input in VAAPI hardware surfaces.  If you
       have input in software frames, use the hwupload filter to upload them
       to the GPU.

       The following standard libavcodec options are used:

       ·   g / gop_size

       ·   bf / max_b_frames

       ·   profile

       ·   level

       ·   b / bit_rate

       ·   maxrate / rc_max_rate

       ·   bufsize / rc_buffer_size

       ·   rc_init_occupancy / rc_initial_buffer_occupancy

       ·   compression_level

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

       ·   q / global_quality

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

       ·   qmin (only: qmax is not supported)

       ·   i_qfactor / i_quant_factor

       ·   i_qoffset / i_quant_offset

       ·   b_qfactor / b_quant_factor

       ·   b_qoffset / b_quant_offset

       h264_vaapi
	   profile sets the value of profile_idc and the
	   constraint_set*_flags.  level sets the value of level_idc.

	   low_power
	       Use low-power encoding mode.

	   coder
	       Set entropy encoder (default is cabac).	Possible values:

	       ac
	       cabac
		   Use CABAC.

	       vlc
	       cavlc
		   Use CAVLC.

       hevc_vaapi
	   profile and level set the values of general_profile_idc and
	   general_level_idc respectively.

       mjpeg_vaapi
	   Always encodes using the standard quantisation and huffman tables -
	   global_quality scales the standard quantisation table (range
	   1-100).

       mpeg2_vaapi
	   profile and level set the value of profile_and_level_indication.

	   No rate control is supported.

       vp8_vaapi
	   B-frames are not supported.

	   global_quality sets the q_idx used for non-key frames (range
	   0-127).

	   loop_filter_level
	   loop_filter_sharpness
	       Manually set the loop filter parameters.

       vp9_vaapi
	   global_quality sets the q_idx used for P-frames (range 0-255).

	   loop_filter_level
	   loop_filter_sharpness
	       Manually set the loop filter parameters.

	   B-frames are supported, but the output stream is always in encode
	   order rather than display order.  If B-frames are enabled, it may
	   be necessary to use the vp9_raw_reorder bitstream filter to modify
	   the output stream to display frames in the correct order.

	   Only normal frames are produced - the vp9_superframe bitstream
	   filter may be required to produce a stream usable with all
	   decoders.

   vc2
       SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed
       at professional broadcasting but since it supports yuv420, yuv422 and
       yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it
       suitable for other tasks which require low overhead and low compression
       (like screen recording).

       Options

       b   Sets target video bitrate. Usually that's around 1:6 of the
	   uncompressed video bitrate (e.g. for 1920x1080 50fps yuv422p10
	   that's around 400Mbps). Higher values (close to the uncompressed
	   bitrate) turn on lossless compression mode.

       field_order
	   Enables field coding when set (e.g. to tt - top field first) for
	   interlaced inputs. Should increase compression with interlaced
	   content as it splits the fields and encodes each separately.

       wavelet_depth
	   Sets the total amount of wavelet transforms to apply, between 1 and
	   5 (default).  Lower values reduce compression and quality. Less
	   capable decoders may not be able to handle values of wavelet_depth
	   over 3.

       wavelet_type
	   Sets the transform type. Currently only 5_3 (LeGall) and 9_7
	   (Deslauriers-Dubuc) are implemented, with 9_7 being the one with
	   better compression and thus is the default.

       slice_width
       slice_height
	   Sets the slice size for each slice. Larger values result in better
	   compression.  For compatibility with other more limited decoders
	   use slice_width of 32 and slice_height of 8.

       tolerance
	   Sets the undershoot tolerance of the rate control system in
	   percent. This is to prevent an expensive search from being run.

       qm  Sets the quantization matrix preset to use by default or when
	   wavelet_depth is set to 5

	   -   default Uses the default quantization matrix from the
	       specifications, extended with values for the fifth level. This
	       provides a good balance between keeping detail and omitting
	       artifacts.

	   -   flat Use a completely zeroed out quantization matrix. This
	       increases PSNR but might reduce perception. Use in bogus
	       benchmarks.

	   -   color Reduces detail but attempts to preserve color at
	       extremely low bitrates.

SUBTITLES ENCODERS
   dvdsub
       This codec encodes the bitmap subtitle format that is used in DVDs.
       Typically they are stored in VOBSUB file pairs (*.idx + *.sub), and
       they can also be used in Matroska files.

       Options

       even_rows_fix
	   When set to 1, enable a work-around that makes the number of pixel
	   rows even in all subtitles.	This fixes a problem with some players
	   that cut off the bottom row if the number is odd.  The work-around
	   just adds a fully transparent row if needed.  The overhead is low,
	   typically one byte per subtitle on average.

	   By default, this work-around is disabled.

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

       You can disable all the bitstream filters using the configure option
       "--disable-bsfs", and selectively enable any bitstream filter using the
       option "--enable-bsf=BSF", or you can disable a particular bitstream
       filter using the option "--disable-bsf=BSF".

       The option "-bsfs" of the ff* tools will display the list of all the
       supported bitstream filters included in your build.

       The ff* tools have a -bsf option applied per stream, taking a comma-
       separated list of filters, whose parameters follow the filter name
       after a '='.

	       ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

       Below is a description of the currently available bitstream filters,
       with their parameters, if any.

   aac_adtstoasc
       Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration
       bitstream.

       This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS
       header and removes the ADTS header.

       This filter is required for example when copying an AAC stream from a
       raw ADTS AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
       to MOV/MP4 files and related formats such as 3GP or M4A. Please note
       that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.

   chomp
       Remove zero padding at the end of a packet.

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

   dump_extra
       Add extradata to the beginning of the filtered packets.

       The additional argument specifies which packets should be filtered.  It
       accepts the values:

       a   add extradata to all key packets, but only if local_header is set
	   in the flags2 codec context field

       k   add extradata to all key packets

       e   add extradata to all packets

       If not specified it is assumed k.

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

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

   extract_extradata
       Extract the in-band extradata.

       Certain codecs allow the long-term headers (e.g. MPEG-2 sequence
       headers, or H.264/HEVC (VPS/)SPS/PPS) to be transmitted either "in-
       band" (i.e. as a part of the bitstream containing the coded frames) or
       "out of band" (e.g. on the container level). This latter form is called
       "extradata" in FFmpeg terminology.

       This bitstream filter detects the in-band headers and makes them
       available as extradata.

       remove
	   When this option is enabled, the long-term headers are removed from
	   the bitstream after extraction.

   h264_mp4toannexb
       Convert an H.264 bitstream from length prefixed mode to start code
       prefixed mode (as defined in the Annex B of the ITU-T H.264
       specification).

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

       For example to remux an MP4 file containing an H.264 stream to mpegts
       format with ffmpeg, you can use the command:

	       ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

       Please note that this filter is auto-inserted for MPEG-TS (muxer
       "mpegts") and raw H.264 (muxer "h264") output formats.

   hevc_mp4toannexb
       Convert an HEVC/H.265 bitstream from length prefixed mode to start code
       prefixed mode (as defined in the Annex B of the ITU-T H.265
       specification).

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

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

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

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

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

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

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

   mjpeg2jpeg
       Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

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

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

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

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

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

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

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

   mov2textsub
       Extract a representable text file from MOV subtitles, stripping the
       metadata header from each subtitle packet.

       See also the text2movsub filter.

   mp3decomp
       Decompress non-standard compressed MP3 audio headers.

   mpeg4_unpack_bframes
       Unpack DivX-style packed B-frames.

       DivX-style packed B-frames are not valid MPEG-4 and were only a
       workaround for the broken Video for Windows subsystem.  They use more
       space, can cause minor AV sync issues, require more CPU power to decode
       (unless the player has some decoded picture queue to compensate the
       2,0,2,0 frame per packet style) and cause trouble if copied into a
       standard container like mp4 or mpeg-ps/ts, because MPEG-4 decoders may
       not be able to decode them, since they are not valid MPEG-4.

       For example to fix an AVI file containing an MPEG-4 stream with DivX-
       style packed B-frames using ffmpeg, you can use the command:

	       ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

   noise
       Damages the contents of packets or simply drops them without damaging
       the container. Can be used for fuzzing or testing error
       resilience/concealment.

       Parameters:

       amount
	   A numeral string, whose value is related to how often output bytes
	   will be modified. Therefore, values below or equal to 0 are
	   forbidden, and the lower the more frequent bytes will be modified,
	   with 1 meaning every byte is modified.

       dropamount
	   A numeral string, whose value is related to how often packets will
	   be dropped.	Therefore, values below or equal to 0 are forbidden,
	   and the lower the more frequent packets will be dropped, with 1
	   meaning every packet is dropped.

       The following example applies the modification to every byte but does
       not drop any packets.

	       ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

   null
       This bitstream filter passes the packets through unchanged.

   remove_extra
       Remove extradata from packets.

       It accepts the following parameter:

       freq
	   Set which frame types to remove extradata from.

	   k   Remove extradata from non-keyframes only.

	   keyframe
	       Remove extradata from keyframes only.

	   e, all
	       Remove extradata from all frames.

   text2movsub
       Convert text subtitles to MOV subtitles (as used by the "mov_text"
       codec) with metadata headers.

       See also the mov2textsub filter.

   vp9_superframe
       Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This
       fixes merging of split/segmented VP9 streams where the alt-ref frame
       was split from its visible counterpart.

   vp9_superframe_split
       Split VP9 superframes into single frames.

   vp9_raw_reorder
       Given a VP9 stream with correct timestamps but possibly out of order,
       insert additional show-existing-frame packets to correct the ordering.

FORMAT OPTIONS
       The libavformat library provides some generic global options, which can
       be set on all the muxers and demuxers. In addition each muxer or
       demuxer may support so-called private options, which are specific for
       that component.

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

       The list of supported options follows:

       avioflags flags (input/output)
	   Possible values:

	   direct
	       Reduce buffering.

       probesize integer (input)
	   Set probing size in bytes, i.e. the size of the data to analyze to
	   get stream information. A higher value will enable detecting more
	   information in case it is dispersed into the stream, but will
	   increase latency. Must be an integer not lesser than 32. It is
	   5000000 by default.

       packetsize integer (output)
	   Set packet size.

       fflags flags (input/output)
	   Set format flags.

	   Possible values:

	   ignidx
	       Ignore index.

	   fastseek
	       Enable fast, but inaccurate seeks for some formats.

	   genpts
	       Generate PTS.

	   nofillin
	       Do not fill in missing values that can be exactly calculated.

	   noparse
	       Disable AVParsers, this needs "+nofillin" too.

	   igndts
	       Ignore DTS.

	   discardcorrupt
	       Discard corrupted frames.

	   sortdts
	       Try to interleave output packets by DTS.

	   keepside
	       Do not merge side data.

	   latm
	       Enable RTP MP4A-LATM payload.

	   nobuffer
	       Reduce the latency introduced by optional buffering

	   bitexact
	       Only write platform-, build- and time-independent data.	This
	       ensures that file and data checksums are reproducible and match
	       between platforms. Its primary use is for regression testing.

	   shortest
	       Stop muxing at the end of the shortest stream.  It may be
	       needed to increase max_interleave_delta to avoid flushing the
	       longer streams before EOF.

       seek2any integer (input)
	   Allow seeking to non-keyframes on demuxer level when supported if
	   set to 1.  Default is 0.

       analyzeduration integer (input)
	   Specify how many microseconds are analyzed to probe the input. A
	   higher value will enable detecting more accurate information, but
	   will increase latency. It defaults to 5,000,000 microseconds = 5
	   seconds.

       cryptokey hexadecimal string (input)
	   Set decryption key.

       indexmem integer (input)
	   Set max memory used for timestamp index (per stream).

       rtbufsize integer (input)
	   Set max memory used for buffering real-time frames.

       fdebug flags (input/output)
	   Print specific debug info.

	   Possible values:

	   ts
       max_delay integer (input/output)
	   Set maximum muxing or demuxing delay in microseconds.

       fpsprobesize integer (input)
	   Set number of frames used to probe fps.

       audio_preload integer (output)
	   Set microseconds by which audio packets should be interleaved
	   earlier.

       chunk_duration integer (output)
	   Set microseconds for each chunk.

       chunk_size integer (output)
	   Set size in bytes for each chunk.

       err_detect, f_err_detect flags (input)
	   Set error detection flags. "f_err_detect" is deprecated and should
	   be used only via the ffmpeg tool.

	   Possible values:

	   crccheck
	       Verify embedded CRCs.

	   bitstream
	       Detect bitstream specification deviations.

	   buffer
	       Detect improper bitstream length.

	   explode
	       Abort decoding on minor error detection.

	   careful
	       Consider things that violate the spec and have not been seen in
	       the wild as errors.

	   compliant
	       Consider all spec non compliancies as errors.

	   aggressive
	       Consider things that a sane encoder should not do as an error.

       max_interleave_delta integer (output)
	   Set maximum buffering duration for interleaving. The duration is
	   expressed in microseconds, and defaults to 1000000 (1 second).

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

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

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

       use_wallclock_as_timestamps integer (input)
	   Use wallclock as timestamps if set to 1. Default is 0.

       avoid_negative_ts integer (output)
	   Possible values:

	   make_non_negative
	       Shift timestamps to make them non-negative.  Also note that
	       this affects only leading negative timestamps, and not non-
	       monotonic negative timestamps.

	   make_zero
	       Shift timestamps so that the first timestamp is 0.

	   auto (default)
	       Enables shifting when required by the target format.

	   disabled
	       Disables shifting of timestamp.

	   When shifting is enabled, all output timestamps are shifted by the
	   same amount. Audio, video, and subtitles desynching and relative
	   timestamp differences are preserved compared to how they would have
	   been without shifting.

       skip_initial_bytes integer (input)
	   Set number of bytes to skip before reading header and frames if set
	   to 1.  Default is 0.

       correct_ts_overflow integer (input)
	   Correct single timestamp overflows if set to 1. Default is 1.

       flush_packets integer (output)
	   Flush the underlying I/O stream after each packet. Default is -1
	   (auto), which means that the underlying protocol will decide, 1
	   enables it, and has the effect of reducing the latency, 0 disables
	   it and may increase IO throughput in some cases.

       output_ts_offset offset (output)
	   Set the output time offset.

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

	   The offset is added by the muxer to the output timestamps.

	   Specifying a positive offset means that the corresponding streams
	   are delayed bt the time duration specified in offset. Default value
	   is 0 (meaning that no offset is applied).

       format_whitelist list (input)
	   "," separated list of allowed demuxers. By default all are allowed.

       dump_separator string (input)
	   Separator used to separate the fields printed on the command line
	   about the Stream parameters.  For example to separate the fields
	   with newlines and indention:

		   ffprobe -dump_separator "
					     "	-i ~/videos/matrixbench_mpeg2.mpg

       max_streams integer (input)
	   Specifies the maximum number of streams. This can be used to reject
	   files that would require too many resources due to a large number
	   of streams.

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

       Possible forms of stream specifiers are:

       stream_index
	   Matches the stream with this index.

       stream_type[:stream_index]
	   stream_type is one of following: 'v' for video, 'a' for audio, 's'
	   for subtitle, 'd' for data, and 't' for attachments. If
	   stream_index is given, then it matches the stream number
	   stream_index of this type. Otherwise, it matches all streams of
	   this type.

       p:program_id[:stream_index]
	   If stream_index is given, then it matches the stream with number
	   stream_index in the program with the id program_id. Otherwise, it
	   matches all streams in the program.

       #stream_id
	   Matches the stream by a format-specific ID.

       The exact semantics of stream specifiers is defined by the
       "avformat_match_stream_specifier()" function declared in the
       libavformat/avformat.h header.

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

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

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

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

       The description of some of the currently available demuxers follows.

   aa
       Audible Format 2, 3, and 4 demuxer.

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

   applehttp
       Apple HTTP Live Streaming demuxer.

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

   apng
       Animated Portable Network Graphics demuxer.

       This demuxer is used to demux APNG files.  All headers, but the PNG
       signature, up to (but not including) the first fcTL chunk are
       transmitted as extradata.  Frames are then split as being all the
       chunks between two fcTL ones, or between the last fcTL and IEND chunks.

       -ignore_loop bool
	   Ignore the loop variable in the file if set.

       -max_fps int
	   Maximum framerate in frames per second (0 for no limit).

       -default_fps int
	   Default framerate in frames per second when none is specified in
	   the file (0 meaning as fast as possible).

   asf
       Advanced Systems Format demuxer.

       This demuxer is used to demux ASF files and MMS network streams.

       -no_resync_search bool
	   Do not try to resynchronize by looking for a certain optional start
	   code.

   concat
       Virtual concatenation script demuxer.

       This demuxer reads a list of files and other directives from a text
       file and demuxes them one after the other, as if all their packets had
       been muxed together.

       The timestamps in the files are adjusted so that the first file starts
       at 0 and each next file starts where the previous one finishes. Note
       that it is done globally and may cause gaps if all streams do not have
       exactly the same length.

       All files must have the same streams (same codecs, same time base,
       etc.).

       The duration of each file is used to adjust the timestamps of the next
       file: if the duration is incorrect (because it was computed using the
       bit-rate or because the file is truncated, for example), it can cause
       artifacts. The "duration" directive can be used to override the
       duration stored in each file.

       Syntax

       The script is a text file in extended-ASCII, with one directive per
       line.  Empty lines, leading spaces and lines starting with '#' are
       ignored. The following directive is recognized:

       "file path"
	   Path to a file to read; special characters and spaces must be
	   escaped with backslash or single quotes.

	   All subsequent file-related directives apply to that file.

       "ffconcat version 1.0"
	   Identify the script type and version. It also sets the safe option
	   to 1 if it was -1.

	   To make FFmpeg recognize the format automatically, this directive
	   must appear exactly as is (no extra space or byte-order-mark) on
	   the very first line of the script.

       "duration dur"
	   Duration of the file. This information can be specified from the
	   file; specifying it here may be more efficient or help if the
	   information from the file is not available or accurate.

	   If the duration is set for all files, then it is possible to seek
	   in the whole concatenated video.

       "inpoint timestamp"
	   In point of the file. When the demuxer opens the file it instantly
	   seeks to the specified timestamp. Seeking is done so that all
	   streams can be presented successfully at In point.

	   This directive works best with intra frame codecs, because for non-
	   intra frame ones you will usually get extra packets before the
	   actual In point and the decoded content will most likely contain
	   frames before In point too.

	   For each file, packets before the file In point will have
	   timestamps less than the calculated start timestamp of the file
	   (negative in case of the first file), and the duration of the files
	   (if not specified by the "duration" directive) will be reduced
	   based on their specified In point.

	   Because of potential packets before the specified In point, packet
	   timestamps may overlap between two concatenated files.

       "outpoint timestamp"
	   Out point of the file. When the demuxer reaches the specified
	   decoding timestamp in any of the streams, it handles it as an end
	   of file condition and skips the current and all the remaining
	   packets from all streams.

	   Out point is exclusive, which means that the demuxer will not
	   output packets with a decoding timestamp greater or equal to Out
	   point.

	   This directive works best with intra frame codecs and formats where
	   all streams are tightly interleaved. For non-intra frame codecs you
	   will usually get additional packets with presentation timestamp
	   after Out point therefore the decoded content will most likely
	   contain frames after Out point too. If your streams are not tightly
	   interleaved you may not get all the packets from all streams before
	   Out point and you may only will be able to decode the earliest
	   stream until Out point.

	   The duration of the files (if not specified by the "duration"
	   directive) will be reduced based on their specified Out point.

       "file_packet_metadata key=value"
	   Metadata of the packets of the file. The specified metadata will be
	   set for each file packet. You can specify this directive multiple
	   times to add multiple metadata entries.

       "stream"
	   Introduce a stream in the virtual file.  All subsequent stream-
	   related directives apply to the last introduced stream.  Some
	   streams properties must be set in order to allow identifying the
	   matching streams in the subfiles.  If no streams are defined in the
	   script, the streams from the first file are copied.

       "exact_stream_id id"
	   Set the id of the stream.  If this directive is given, the string
	   with the corresponding id in the subfiles will be used.  This is
	   especially useful for MPEG-PS (VOB) files, where the order of the
	   streams is not reliable.

       Options

       This demuxer accepts the following option:

       safe
	   If set to 1, reject unsafe file paths. A file path is considered
	   safe if it does not contain a protocol specification and is
	   relative and all components only contain characters from the
	   portable character set (letters, digits, period, underscore and
	   hyphen) and have no period at the beginning of a component.

	   If set to 0, any file name is accepted.

	   The default is 1.

	   -1 is equivalent to 1 if the format was automatically probed and 0
	   otherwise.

       auto_convert
	   If set to 1, try to perform automatic conversions on packet data to
	   make the streams concatenable.  The default is 1.

	   Currently, the only conversion is adding the h264_mp4toannexb
	   bitstream filter to H.264 streams in MP4 format. This is necessary
	   in particular if there are resolution changes.

       segment_time_metadata
	   If set to 1, every packet will contain the lavf.concat.start_time
	   and the lavf.concat.duration packet metadata values which are the
	   start_time and the duration of the respective file segments in the
	   concatenated output expressed in microseconds. The duration
	   metadata is only set if it is known based on the concat file.  The
	   default is 0.

       Examples

       ·   Use absolute filenames and include some comments:

		   # my first filename
		   file /mnt/share/file-1.wav
		   # my second filename including whitespace
		   file '/mnt/share/file 2.wav'
		   # my third filename including whitespace plus single quote
		   file '/mnt/share/file 3'\''.wav'

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

		   ffconcat version 1.0

		   file file-1.wav
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   flv, live_flv
       Adobe Flash Video Format demuxer.

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

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

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

   gif
       Animated GIF demuxer.

       It accepts the following options:

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

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

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

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

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

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

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

   hls
       HLS demuxer

       It accepts the following options:

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

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

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

   image2
       Image file demuxer.

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

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

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

       This demuxer accepts the following options:

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

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

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

	   pattern_type accepts one of the following values.

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

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

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

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

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

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

		       ffmpeg -i img.jpeg img.png

	   glob
	       Select a glob wildcard pattern type.

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

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

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

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

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

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

	   Default value is glob_sequence.

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

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

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

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

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

       Examples

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

		   ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

       ·   As above, but start by reading from a file with index 100 in the
	   sequence:

		   ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

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

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

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

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

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

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

   libopenmpt
       libopenmpt based module demuxer

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

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

       It accepts the following options:

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

	   The default value is to let libopenmpt choose.

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

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

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

       This demuxer accepts the following options:

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

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

   mpegts
       MPEG-2 transport stream demuxer.

       This demuxer accepts the following options:

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

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

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

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

   mpjpeg
       MJPEG encapsulated in multi-part MIME demuxer.

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

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

   rawvideo
       Raw video demuxer.

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

       This demuxer accepts the following options:

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

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

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

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

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

   sbg
       SBaGen script demuxer.

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

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

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

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

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

       This demuxer accepts the following option:

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

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

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

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

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

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

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

       A description of some of the currently available muxers follows.

   aiff
       Audio Interchange File Format muxer.

       Options

       It accepts the following options:

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

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

   asf
       Advanced Systems Format muxer.

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

       Options

       It accepts the following options:

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

   avi
       Audio Video Interleaved muxer.

       Options

       It accepts the following options:

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

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

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

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

   chromaprint
       Chromaprint fingerprinter

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

       Options

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

       algorithm
	   Algorithm index to fingerprint with.

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

	   raw Binary raw fingerprint

	   compressed
	       Binary compressed fingerprint

	   base64
	       Base64 compressed fingerprint

   crc
       CRC (Cyclic Redundancy Check) testing format.

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

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

       See also the framecrc muxer.

       Examples

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

	       ffmpeg -i INPUT -f crc out.crc

       You can print the CRC to stdout with the command:

	       ffmpeg -i INPUT -f crc -

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

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

   flv
       Adobe Flash Video Format muxer.

       This muxer accepts the following options:

       flvflags flags
	   Possible values:

	   aac_seq_header_detect
	       Place AAC sequence header based on audio stream data.

	   no_sequence_end
	       Disable sequence end tag.

	   no_metadata
	       Disable metadata tag.

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

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

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

       For more information see:

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

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

       It creates a MPD manifest file and segment files for each stream.

       The segment filename might contain pre-defined identifiers used with
       SegmentTemplate as defined in section 5.3.9.4.4 of the standard.
       Available identifiers are "$RepresentationID$", "$Number$",
       "$Bandwidth$" and "$Time$".

	       ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
	       -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
	       -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
	       -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
	       -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
	       -f dash /path/to/out.mpd

       -min_seg_duration microseconds
	   Set the segment length in microseconds.

       -window_size size
	   Set the maximum number of segments kept in the manifest.

       -extra_window_size size
	   Set the maximum number of segments kept outside of the manifest
	   before removing from disk.

       -remove_at_exit remove
	   Enable (1) or disable (0) removal of all segments when finished.

       -use_template template
	   Enable (1) or disable (0) use of SegmentTemplate instead of
	   SegmentList.

       -use_timeline timeline
	   Enable (1) or disable (0) use of SegmentTimeline in
	   SegmentTemplate.

       -single_file single_file
	   Enable (1) or disable (0) storing all segments in one file,
	   accessed using byte ranges.

       -single_file_name file_name
	   DASH-templated name to be used for baseURL. Implies single_file set
	   to "1".

       -init_seg_name init_name
	   DASH-templated name to used for the initialization segment. Default
	   is "init-stream$RepresentationID$.m4s"

       -media_seg_name segment_name
	   DASH-templated name to used for the media segments. Default is
	   "chunk-stream$RepresentationID$-$Number%05d$.m4s"

       -utc_timing_url utc_url
	   URL of the page that will return the UTC timestamp in ISO format.
	   Example: "https://time.akamai.com/?iso"

       -adaptation_sets adaptation_sets
	   Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c
	   id=y,streams=d,e" with x and y being the IDs of the adaptation sets
	   and a,b,c,d and e are the indices of the mapped streams.

	   To map all video (or audio) streams to an AdaptationSet, "v" (or
	   "a") can be used as stream identifier instead of IDs.

	   When no assignment is defined, this defaults to an AdaptationSet
	   for each stream.

   framecrc
       Per-packet CRC (Cyclic Redundancy Check) testing format.

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

       The output of the muxer consists of a line for each audio and video
       packet of the form:

	       <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>

       CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of
       the packet.

       Examples

       For example to compute the CRC of the audio and video frames in INPUT,
       converted to raw audio and video packets, and store it in the file
       out.crc:

	       ffmpeg -i INPUT -f framecrc out.crc

       To print the information to stdout, use the command:

	       ffmpeg -i INPUT -f framecrc -

       With ffmpeg, you can select the output format to which the audio and
       video frames are encoded before computing the CRC for each packet by
       specifying the audio and video codec. For example, to compute the CRC
       of each decoded input audio frame converted to PCM unsigned 8-bit and
       of each decoded input video frame converted to MPEG-2 video, use the
       command:

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

       See also the crc muxer.

   framehash
       Per-packet hash testing format.

       This muxer computes and prints a cryptographic hash for each audio and
       video packet. This can be used for packet-by-packet equality checks
       without having to individually do a binary comparison on each.

       By default audio frames are converted to signed 16-bit raw audio and
       video frames to raw video before computing the hash, but the output of
       explicit conversions to other codecs can also be used. It uses the
       SHA-256 cryptographic hash function by default, but supports several
       other algorithms.

       The output of the muxer consists of a line for each audio and video
       packet of the form:

	       <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>

       hash is a hexadecimal number representing the computed hash for the
       packet.

       hash algorithm
	   Use the cryptographic hash function specified by the string
	   algorithm.  Supported values include "MD5", "murmur3", "RIPEMD128",
	   "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
	   (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
	   and "adler32".

       Examples

       To compute the SHA-256 hash of the audio and video frames in INPUT,
       converted to raw audio and video packets, and store it in the file
       out.sha256:

	       ffmpeg -i INPUT -f framehash out.sha256

       To print the information to stdout, using the MD5 hash function, use
       the command:

	       ffmpeg -i INPUT -f framehash -hash md5 -

       See also the hash muxer.

   framemd5
       Per-packet MD5 testing format.

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

       Examples

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

	       ffmpeg -i INPUT -f framemd5 out.md5

       To print the information to stdout, use the command:

	       ffmpeg -i INPUT -f framemd5 -

       See also the framehash and md5 muxers.

   gif
       Animated GIF muxer.

       It accepts the following options:

       loop
	   Set the number of times to loop the output. Use "-1" for no loop, 0
	   for looping indefinitely (default).

       final_delay
	   Force the delay (expressed in centiseconds) after the last frame.
	   Each frame ends with a delay until the next frame. The default is
	   "-1", which is a special value to tell the muxer to re-use the
	   previous delay. In case of a loop, you might want to customize this
	   value to mark a pause for instance.

       For example, to encode a gif looping 10 times, with a 5 seconds delay
       between the loops:

	       ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif

       Note 1: if you wish to extract the frames into separate GIF files, you
       need to force the image2 muxer:

	       ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"

       Note 2: the GIF format has a very large time base: the delay between
       two frames can therefore not be smaller than one centi second.

   hash
       Hash testing format.

       This muxer computes and prints a cryptographic hash of all the input
       audio and video frames. This can be used for equality checks without
       having to do a complete binary comparison.

       By default audio frames are converted to signed 16-bit raw audio and
       video frames to raw video before computing the hash, but the output of
       explicit conversions to other codecs can also be used. Timestamps are
       ignored. It uses the SHA-256 cryptographic hash function by default,
       but supports several other algorithms.

       The output of the muxer consists of a single line of the form:
       algo=hash, where algo is a short string representing the hash function
       used, and hash is a hexadecimal number representing the computed hash.

       hash algorithm
	   Use the cryptographic hash function specified by the string
	   algorithm.  Supported values include "MD5", "murmur3", "RIPEMD128",
	   "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
	   (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
	   and "adler32".

       Examples

       To compute the SHA-256 hash of the input converted to raw audio and
       video, and store it in the file out.sha256:

	       ffmpeg -i INPUT -f hash out.sha256

       To print an MD5 hash to stdout use the command:

	       ffmpeg -i INPUT -f hash -hash md5 -

       See also the framehash muxer.

   hls
       Apple HTTP Live Streaming muxer that segments MPEG-TS according to the
       HTTP Live Streaming (HLS) specification.

       It creates a playlist file, and one or more segment files. The output
       filename specifies the playlist filename.

       By default, the muxer creates a file for each segment produced. These
       files have the same name as the playlist, followed by a sequential
       number and a .ts extension.

       For example, to convert an input file with ffmpeg:

	       ffmpeg -i in.nut out.m3u8

       This example will produce the playlist, out.m3u8, and segment files:
       out0.ts, out1.ts, out2.ts, etc.

       See also the segment muxer, which provides a more generic and flexible
       implementation of a segmenter, and can be used to perform HLS
       segmentation.

       Options

       This muxer supports the following options:

       hls_init_time seconds
	   Set the initial target segment length in seconds. Default value is
	   0.  Segment will be cut on the next key frame after this time has
	   passed on the first m3u8 list.  After the initial playlist is
	   filled ffmpeg will cut segments at duration equal to "hls_time"

       hls_time seconds
	   Set the target segment length in seconds. Default value is 2.
	   Segment will be cut on the next key frame after this time has
	   passed.

       hls_list_size size
	   Set the maximum number of playlist entries. If set to 0 the list
	   file will contain all the segments. Default value is 5.

       hls_ts_options options_list
	   Set output format options using a :-separated list of key=value
	   parameters. Values containing ":" special characters must be
	   escaped.

       hls_wrap wrap
	   This is a deprecated option, you can use "hls_list_size" and
	   "hls_flags delete_segments" instead it

	   This option is useful to avoid to fill the disk with many segment
	   files, and limits the maximum number of segment files written to
	   disk to wrap.

       hls_start_number_source
	   Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE")
	   according to the specified source.  Unless "hls_flags single_file"
	   is set, it also specifies source of starting sequence numbers of
	   segment and subtitle filenames. In any case, if "hls_flags
	   append_list" is set and read playlist sequence number is greater
	   than the specified start sequence number, then that value will be
	   used as start value.

	   It accepts the following values:

	   generic (default)
	       Set the starting sequence numbers according to start_number
	       option value.

	   epoch
	       The start number will be the seconds since epoch (1970-01-01
	       00:00:00)

	   datetime
	       The start number will be based on the current date/time as
	       YYYYmmddHHMMSS. e.g. 20161231235759.

       start_number number
	   Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") from
	   the specified number when hls_start_number_source value is generic.
	   (This is the default case.)	Unless "hls_flags single_file" is set,
	   it also specifies starting sequence numbers of segment and subtitle
	   filenames.  Default value is 0.

       hls_allow_cache allowcache
	   Explicitly set whether the client MAY (1) or MUST NOT (0) cache
	   media segments.

       hls_base_url baseurl
	   Append baseurl to every entry in the playlist.  Useful to generate
	   playlists with absolute paths.

	   Note that the playlist sequence number must be unique for each
	   segment and it is not to be confused with the segment filename
	   sequence number which can be cyclic, for example if the wrap option
	   is specified.

       hls_segment_filename filename
	   Set the segment filename. Unless "hls_flags single_file" is set,
	   filename is used as a string format with the segment number:

		   ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

	   This example will produce the playlist, out.m3u8, and segment
	   files: file000.ts, file001.ts, file002.ts, etc.

	   filename may contain full path or relative path specification, but
	   only the file name part without any path info will be contained in
	   the m3u8 segment list.  Should a relative path be specified, the
	   path of the created segment files will be relative to the current
	   working directory.  When use_localtime_mkdir is set, the whole
	   expanded value of filename will be written into the m3u8 segment
	   list.

       use_localtime
	   Use strftime() on filename to expand the segment filename with
	   localtime.  The segment number is also available in this mode, but
	   to use it, you need to specify second_level_segment_index hls_flag
	   and %%d will be the specifier.

		   ffmpeg -i in.nut -use_localtime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

	   This example will produce the playlist, out.m3u8, and segment
	   files: file-20160215-1455569023.ts, file-20160215-1455569024.ts,
	   etc.  Note: On some systems/environments, the %s specifier is not
	   available. See
	     "strftime()" documentation.

		   ffmpeg -i in.nut -use_localtime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

	   This example will produce the playlist, out.m3u8, and segment
	   files: file-20160215-0001.ts, file-20160215-0002.ts, etc.

       use_localtime_mkdir
	   Used together with -use_localtime, it will create all
	   subdirectories which is expanded in filename.

		   ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

	   This example will create a directory 201560215 (if it does not
	   exist), and then produce the playlist, out.m3u8, and segment files:
	   20160215/file-20160215-1455569023.ts,
	   20160215/file-20160215-1455569024.ts, etc.

		   ffmpeg -i in.nut -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

	   This example will create a directory hierarchy 2016/02/15 (if any
	   of them do not exist), and then produce the playlist, out.m3u8, and
	   segment files: 2016/02/15/file-20160215-1455569023.ts,
	   2016/02/15/file-20160215-1455569024.ts, etc.

       hls_key_info_file key_info_file
	   Use the information in key_info_file for segment encryption. The
	   first line of key_info_file specifies the key URI written to the
	   playlist. The key URL is used to access the encryption key during
	   playback. The second line specifies the path to the key file used
	   to obtain the key during the encryption process. The key file is
	   read as a single packed array of 16 octets in binary format. The
	   optional third line specifies the initialization vector (IV) as a
	   hexadecimal string to be used instead of the segment sequence
	   number (default) for encryption. Changes to key_info_file will
	   result in segment encryption with the new key/IV and an entry in
	   the playlist for the new key URI/IV if "hls_flags periodic_rekey"
	   is enabled.

	   Key info file format:

		   <key URI>
		   <key file path>
		   <IV> (optional)

	   Example key URIs:

		   http://server/file.key
		   /path/to/file.key
		   file.key

	   Example key file paths:

		   file.key
		   /path/to/file.key

	   Example IV:

		   0123456789ABCDEF0123456789ABCDEF

	   Key info file example:

		   http://server/file.key
		   /path/to/file.key
		   0123456789ABCDEF0123456789ABCDEF

	   Example shell script:

		   #!/bin/sh
		   BASE_URL=${1:-'.'}
		   openssl rand 16 > file.key
		   echo $BASE_URL/file.key > file.keyinfo
		   echo file.key >> file.keyinfo
		   echo $(openssl rand -hex 16) >> file.keyinfo
		   ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
		     -hls_key_info_file file.keyinfo out.m3u8

       -hls_enc enc
	   Enable (1) or disable (0) the AES128 encryption.  When enabled
	   every segment generated is encrypted and the encryption key is
	   saved as playlist name.key.

       -hls_enc_key key
	   Hex-coded 16byte key to encrypt the segments, by default it is
	   randomly generated.

       -hls_enc_key_url keyurl
	   If set, keyurl is prepended instead of baseurl to the key filename
	   in the playlist.

       -hls_enc_iv iv
	   Hex-coded 16byte initialization vector for every segment instead of
	   the autogenerated ones.

       hls_segment_type flags
	   Possible values:

	   mpegts
	       If this flag is set, the hls segment files will format to
	       mpegts.	the mpegts files is used in all hls versions.

	   fmp4
	       If this flag is set, the hls segment files will format to
	       fragment mp4 looks like dash.  the fmp4 files is used in hls
	       after version 7.

       hls_fmp4_init_filename filename
	   set filename to the fragment files header file, default filename is
	   init.mp4.

       hls_flags flags
	   Possible values:

	   single_file
	       If this flag is set, the muxer will store all segments in a
	       single MPEG-TS file, and will use byte ranges in the playlist.
	       HLS playlists generated with this way will have the version
	       number 4.  For example:

		       ffmpeg -i in.nut -hls_flags single_file out.m3u8

	       Will produce the playlist, out.m3u8, and a single segment file,
	       out.ts.

	   delete_segments
	       Segment files removed from the playlist are deleted after a
	       period of time equal to the duration of the segment plus the
	       duration of the playlist.

	   append_list
	       Append new segments into the end of old segment list, and
	       remove the "#EXT-X-ENDLIST" from the old segment list.

	   round_durations
	       Round the duration info in the playlist file segment info to
	       integer values, instead of using floating point.

	   discont_start
	       Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the
	       first segment's information.

	   omit_endlist
	       Do not append the "EXT-X-ENDLIST" tag at the end of the
	       playlist.

	   periodic_rekey
	       The file specified by "hls_key_info_file" will be checked
	       periodically and detect updates to the encryption info. Be sure
	       to replace this file atomically, including the file containing
	       the AES encryption key.

	   split_by_time
	       Allow segments to start on frames other than keyframes. This
	       improves behavior on some players when the time between
	       keyframes is inconsistent, but may make things worse on others,
	       and can cause some oddities during seeking. This flag should be
	       used with the "hls_time" option.

	   program_date_time
	       Generate "EXT-X-PROGRAM-DATE-TIME" tags.

	   second_level_segment_index
	       Makes it possible to use segment indexes as %%d in
	       hls_segment_filename expression besides date/time values when
	       use_localtime is on.  To get fixed width numbers with trailing
	       zeroes, %%0xd format is available where x is the required
	       width.

	   second_level_segment_size
	       Makes it possible to use segment sizes (counted in bytes) as
	       %%s in hls_segment_filename expression besides date/time values
	       when use_localtime is on.  To get fixed width numbers with
	       trailing zeroes, %%0xs format is available where x is the
	       required width.

	   second_level_segment_duration
	       Makes it possible to use segment duration (calculated  in
	       microseconds) as %%t in hls_segment_filename expression besides
	       date/time values when use_localtime is on.  To get fixed width
	       numbers with trailing zeroes, %%0xt format is available where x
	       is the required width.

		       ffmpeg -i sample.mpeg \
			  -f hls -hls_time 3 -hls_list_size 5 \
			  -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
			  -use_localtime 1 -use_localtime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8

	       This will produce segments like this:
	       segment_20170102194334_0003_00122200_0000003000000.ts,
	       segment_20170102194334_0004_00120072_0000003000000.ts etc.

	   temp_file
	       Write segment data to filename.tmp and rename to filename only
	       once the segment is complete. A webserver serving up segments
	       can be configured to reject requests to *.tmp to prevent access
	       to in-progress segments before they have been added to the m3u8
	       playlist.

       hls_playlist_type event
	   Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces
	   hls_list_size to 0; the playlist can only be appended to.

       hls_playlist_type vod
	   Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces
	   hls_list_size to 0; the playlist must not change.

       method
	   Use the given HTTP method to create the hls files.

		   ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

	   This example will upload all the mpegts segment files to the HTTP
	   server using the HTTP PUT method, and update the m3u8 files every
	   "refresh" times using the same method.  Note that the HTTP server
	   must support the given method for uploading files.

       http_user_agent
	   Override User-Agent field in HTTP header. Applicable only for HTTP
	   output.

   ico
       ICO file muxer.

       Microsoft's icon file format (ICO) has some strict limitations that
       should be noted:

       ·   Size cannot exceed 256 pixels in any dimension

       ·   Only BMP and PNG images can be stored

       ·   If a BMP image is used, it must be one of the following pixel
	   formats:

		   BMP Bit Depth      FFmpeg Pixel Format
		   1bit 	      pal8
		   4bit 	      pal8
		   8bit 	      pal8
		   16bit	      rgb555le
		   24bit	      bgr24
		   32bit	      bgra

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

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

   image2
       Image file muxer.

       The image file muxer writes video frames to image files.

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

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

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

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

       Examples

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

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

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

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

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

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

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

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

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

       Options

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

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

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

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

   matroska
       Matroska container muxer.

       This muxer implements the matroska and webm container specs.

       Metadata

       The recognized metadata settings in this muxer are:

       title
	   Set title name provided to a single track.

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

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

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

	   The following values are recognized:

	   mono
	       video is not stereo

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       Options

       This muxer supports the following options:

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

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

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

   md5
       MD5 testing format.

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

       Examples

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

	       ffmpeg -i INPUT -f md5 out.md5

       You can print the MD5 to stdout with the command:

	       ffmpeg -i INPUT -f md5 -

       See also the hash and framemd5 muxers.

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

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

       Options

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

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

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

       -frag_duration duration
	   Create fragments that are duration microseconds long.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       Example

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

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

       Audible AAX

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

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

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

       ·   An ID3v2 metadata header at the beginning (enabled by default).
	   Versions 2.3 and 2.4 are supported, the "id3v2_version" private
	   option controls which one is used (3 or 4). Setting "id3v2_version"
	   to 0 disables the ID3v2 header completely.

	   The muxer supports writing attached pictures (APIC frames) to the
	   ID3v2 header.  The pictures are supplied to the muxer in form of a
	   video stream with a single packet. There can be any number of those
	   streams, each will correspond to a single APIC frame.  The stream
	   metadata tags title and comment map to APIC description and picture
	   type respectively. See <http://id3.org/id3v2.4.0-frames> for
	   allowed picture types.

	   Note that the APIC frames must be written at the beginning, so the
	   muxer will buffer the audio frames until it gets all the pictures.
	   It is therefore advised to provide the pictures as soon as possible
	   to avoid excessive buffering.

       ·   A Xing/LAME frame right after the ID3v2 header (if present). It is
	   enabled by default, but will be written only if the output is
	   seekable. The "write_xing" private option can be used to disable
	   it.	The frame contains various information that may be useful to
	   the decoder, like the audio duration or encoder delay.

       ·   A legacy ID3v1 tag at the end of the file (disabled by default). It
	   may be enabled with the "write_id3v1" private option, but as its
	   capabilities are very limited, its usage is not recommended.

       Examples:

       Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

	       ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

       To attach a picture to an mp3 file select both the audio and the
       picture stream with "map":

	       ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
	       -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

       Write a "clean" MP3 without any extra features:

	       ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

   mpegts
       MPEG transport stream muxer.

       This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

       The recognized metadata settings in mpegts muxer are "service_provider"
       and "service_name". If they are not set the default for
       "service_provider" is FFmpeg and the default for "service_name" is
       Service01.

       Options

       The muxer options are:

       mpegts_transport_stream_id integer
	   Set the transport_stream_id. This identifies a transponder in DVB.
	   Default is 0x0001.

       mpegts_original_network_id integer
	   Set the original_network_id. This is unique identifier of a network
	   in DVB. Its main use is in the unique identification of a service
	   through the path Original_Network_ID, Transport_Stream_ID. Default
	   is 0x0001.

       mpegts_service_id integer
	   Set the service_id, also known as program in DVB. Default is
	   0x0001.

       mpegts_service_type integer
	   Set the program service_type. Default is "digital_tv".  Accepts the
	   following options:

	   hex_value
	       Any hexdecimal value between 0x01 to 0xff as defined in ETSI
	       300 468.

	   digital_tv
	       Digital TV service.

	   digital_radio
	       Digital Radio service.

	   teletext
	       Teletext service.

	   advanced_codec_digital_radio
	       Advanced Codec Digital Radio service.

	   mpeg2_digital_hdtv
	       MPEG2 Digital HDTV service.

	   advanced_codec_digital_sdtv
	       Advanced Codec Digital SDTV service.

	   advanced_codec_digital_hdtv
	       Advanced Codec Digital HDTV service.

       mpegts_pmt_start_pid integer
	   Set the first PID for PMT. Default is 0x1000. Max is 0x1f00.

       mpegts_start_pid integer
	   Set the first PID for data packets. Default is 0x0100. Max is
	   0x0f00.

       mpegts_m2ts_mode boolean
	   Enable m2ts mode if set to 1. Default value is "-1" which disables
	   m2ts mode.

       muxrate integer
	   Set a constant muxrate. Default is VBR.

       pes_payload_size integer
	   Set minimum PES packet payload in bytes. Default is 2930.

       mpegts_flags flags
	   Set mpegts flags. Accepts the following options:

	   resend_headers
	       Reemit PAT/PMT before writing the next packet.

	   latm
	       Use LATM packetization for AAC.

	   pat_pmt_at_frames
	       Reemit PAT and PMT at each video frame.

	   system_b
	       Conform to System B (DVB) instead of System A (ATSC).

	   initial_discontinuity
	       Mark the initial packet of each stream as discontinuity.

       resend_headers integer
	   Reemit PAT/PMT before writing the next packet. This option is
	   deprecated: use mpegts_flags instead.

       mpegts_copyts boolean
	   Preserve original timestamps, if value is set to 1. Default value
	   is "-1", which results in shifting timestamps so that they start
	   from 0.

       omit_video_pes_length boolean
	   Omit the PES packet length for video packets. Default is 1 (true).

       pcr_period integer
	   Override the default PCR retransmission time in milliseconds.
	   Ignored if variable muxrate is selected. Default is 20.

       pat_period double
	   Maximum time in seconds between PAT/PMT tables.

       sdt_period double
	   Maximum time in seconds between SDT tables.

       tables_version integer
	   Set PAT, PMT and SDT version (default 0, valid values are from 0 to
	   31, inclusively).  This option allows updating stream structure so
	   that standard consumer may detect the change. To do so, reopen
	   output "AVFormatContext" (in case of API usage) or restart ffmpeg
	   instance, cyclically changing tables_version value:

		   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
		   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
		   ...
		   ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
		   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
		   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
		   ...

       Example

	       ffmpeg -i file.mpg -c copy \
		    -mpegts_original_network_id 0x1122 \
		    -mpegts_transport_stream_id 0x3344 \
		    -mpegts_service_id 0x5566 \
		    -mpegts_pmt_start_pid 0x1500 \
		    -mpegts_start_pid 0x150 \
		    -metadata service_provider="Some provider" \
		    -metadata service_name="Some Channel" \
		    out.ts

   mxf, mxf_d10
       MXF muxer.

       Options

       The muxer options are:

       store_user_comments bool
	   Set if user comments should be stored if available or never.  IRT
	   D-10 does not allow user comments. The default is thus to write
	   them for mxf but not for mxf_d10

   null
       Null muxer.

       This muxer does not generate any output file, it is mainly useful for
       testing or benchmarking purposes.

       For example to benchmark decoding with ffmpeg you can use the command:

	       ffmpeg -benchmark -i INPUT -f null out.null

       Note that the above command does not read or write the out.null file,
       but specifying the output file is required by the ffmpeg syntax.

       Alternatively you can write the command as:

	       ffmpeg -benchmark -i INPUT -f null -

   nut
       -syncpoints flags
	   Change the syncpoint usage in nut:

	   default use the normal low-overhead seeking aids.
	   none do not use the syncpoints at all, reducing the overhead but
	   making the stream non-seekable;
		   Use of this option is not recommended, as the resulting files are very damage
		   sensitive and seeking is not possible. Also in general the overhead from
		   syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
		   all growing data tables, allowing to mux endless streams with limited memory
		   and without these disadvantages.

	   timestamped extend the syncpoint with a wallclock field.

	   The none and timestamped flags are experimental.

       -write_index bool
	   Write index at the end, the default is to write an index.

	       ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor

   ogg
       Ogg container muxer.

       -page_duration duration
	   Preferred page duration, in microseconds. The muxer will attempt to
	   create pages that are approximately duration microseconds long.
	   This allows the user to compromise between seek granularity and
	   container overhead. The default is 1 second. A value of 0 will fill
	   all segments, making pages as large as possible. A value of 1 will
	   effectively use 1 packet-per-page in most situations, giving a
	   small seek granularity at the cost of additional container
	   overhead.

       -serial_offset value
	   Serial value from which to set the streams serial number.  Setting
	   it to different and sufficiently large values ensures that the
	   produced ogg files can be safely chained.

   segment, stream_segment, ssegment
       Basic stream segmenter.

       This muxer outputs streams to a number of separate files of nearly
       fixed duration. Output filename pattern can be set in a fashion similar
       to image2, or by using a "strftime" template if the strftime option is
       enabled.

       "stream_segment" is a variant of the muxer used to write to streaming
       output formats, i.e. which do not require global headers, and is
       recommended for outputting e.g. to MPEG transport stream segments.
       "ssegment" is a shorter alias for "stream_segment".

       Every segment starts with a keyframe of the selected reference stream,
       which is set through the reference_stream option.

       Note that if you want accurate splitting for a video file, you need to
       make the input key frames correspond to the exact splitting times
       expected by the segmenter, or the segment muxer will start the new
       segment with the key frame found next after the specified start time.

       The segment muxer works best with a single constant frame rate video.

       Optionally it can generate a list of the created segments, by setting
       the option segment_list. The list type is specified by the
       segment_list_type option. The entry filenames in the segment list are
       set by default to the basename of the corresponding segment files.

       See also the hls muxer, which provides a more specific implementation
       for HLS segmentation.

       Options

       The segment muxer supports the following options:

       increment_tc 1|0
	   if set to 1, increment timecode between each segment If this is
	   selected, the input need to have a timecode in the first video
	   stream. Default value is 0.

       reference_stream specifier
	   Set the reference stream, as specified by the string specifier.  If
	   specifier is set to "auto", the reference is chosen automatically.
	   Otherwise it must be a stream specifier (see the ``Stream
	   specifiers'' chapter in the ffmpeg manual) which specifies the
	   reference stream. The default value is "auto".

       segment_format format
	   Override the inner container format, by default it is guessed by
	   the filename extension.

       segment_format_options options_list
	   Set output format options using a :-separated list of key=value
	   parameters. Values containing the ":" special character must be
	   escaped.

       segment_list name
	   Generate also a listfile named name. If not specified no listfile
	   is generated.

       segment_list_flags flags
	   Set flags affecting the segment list generation.

	   It currently supports the following flags:

	   cache
	       Allow caching (only affects M3U8 list files).

	   live
	       Allow live-friendly file generation.

       segment_list_size size
	   Update the list file so that it contains at most size segments. If
	   0 the list file will contain all the segments. Default value is 0.

       segment_list_entry_prefix prefix
	   Prepend prefix to each entry. Useful to generate absolute paths.
	   By default no prefix is applied.

       segment_list_type type
	   Select the listing format.

	   The following values are recognized:

	   flat
	       Generate a flat list for the created segments, one segment per
	       line.

	   csv, ext
	       Generate a list for the created segments, one segment per line,
	       each line matching the format (comma-separated values):

		       <segment_filename>,<segment_start_time>,<segment_end_time>

	       segment_filename is the name of the output file generated by
	       the muxer according to the provided pattern. CSV escaping
	       (according to RFC4180) is applied if required.

	       segment_start_time and segment_end_time specify the segment
	       start and end time expressed in seconds.

	       A list file with the suffix ".csv" or ".ext" will auto-select
	       this format.

	       ext is deprecated in favor or csv.

	   ffconcat
	       Generate an ffconcat file for the created segments. The
	       resulting file can be read using the FFmpeg concat demuxer.

	       A list file with the suffix ".ffcat" or ".ffconcat" will auto-
	       select this format.

	   m3u8
	       Generate an extended M3U8 file, version 3, compliant with
	       <http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

	       A list file with the suffix ".m3u8" will auto-select this
	       format.

	   If not specified the type is guessed from the list file name
	   suffix.

       segment_time time
	   Set segment duration to time, the value must be a duration
	   specification. Default value is "2". See also the segment_times
	   option.

	   Note that splitting may not be accurate, unless you force the
	   reference stream key-frames at the given time. See the introductory
	   notice and the examples below.

       segment_atclocktime 1|0
	   If set to "1" split at regular clock time intervals starting from
	   00:00 o'clock. The time value specified in segment_time is used for
	   setting the length of the splitting interval.

	   For example with segment_time set to "900" this makes it possible
	   to create files at 12:00 o'clock, 12:15, 12:30, etc.

	   Default value is "0".

       segment_clocktime_offset duration
	   Delay the segment splitting times with the specified duration when
	   using segment_atclocktime.

	   For example with segment_time set to "900" and
	   segment_clocktime_offset set to "300" this makes it possible to
	   create files at 12:05, 12:20, 12:35, etc.

	   Default value is "0".

       segment_clocktime_wrap_duration duration
	   Force the segmenter to only start a new segment if a packet reaches
	   the muxer within the specified duration after the segmenting clock
	   time. This way you can make the segmenter more resilient to
	   backward local time jumps, such as leap seconds or transition to
	   standard time from daylight savings time.

	   Default is the maximum possible duration which means starting a new
	   segment regardless of the elapsed time since the last clock time.

       segment_time_delta delta
	   Specify the accuracy time when selecting the start time for a
	   segment, expressed as a duration specification. Default value is
	   "0".

	   When delta is specified a key-frame will start a new segment if its
	   PTS satisfies the relation:

		   PTS >= start_time - time_delta

	   This option is useful when splitting video content, which is always
	   split at GOP boundaries, in case a key frame is found just before
	   the specified split time.

	   In particular may be used in combination with the ffmpeg option
	   force_key_frames. The key frame times specified by force_key_frames
	   may not be set accurately because of rounding issues, with the
	   consequence that a key frame time may result set just before the
	   specified time. For constant frame rate videos a value of
	   1/(2*frame_rate) should address the worst case mismatch between the
	   specified time and the time set by force_key_frames.

       segment_times times
	   Specify a list of split points. times contains a list of comma
	   separated duration specifications, in increasing order. See also
	   the segment_time option.

       segment_frames frames
	   Specify a list of split video frame numbers. frames contains a list
	   of comma separated integer numbers, in increasing order.

	   This option specifies to start a new segment whenever a reference
	   stream key frame is found and the sequential number (starting from
	   0) of the frame is greater or equal to the next value in the list.

       segment_wrap limit
	   Wrap around segment index once it reaches limit.

       segment_start_number number
	   Set the sequence number of the first segment. Defaults to 0.

       strftime 1|0
	   Use the "strftime" function to define the name of the new segments
	   to write. If this is selected, the output segment name must contain
	   a "strftime" function template. Default value is 0.

       break_non_keyframes 1|0
	   If enabled, allow segments to start on frames other than keyframes.
	   This improves behavior on some players when the time between
	   keyframes is inconsistent, but may make things worse on others, and
	   can cause some oddities during seeking. Defaults to 0.

       reset_timestamps 1|0
	   Reset timestamps at the beginning of each segment, so that each
	   segment will start with near-zero timestamps. It is meant to ease
	   the playback of the generated segments. May not work with some
	   combinations of muxers/codecs. It is set to 0 by default.

       initial_offset offset
	   Specify timestamp offset to apply to the output packet timestamps.
	   The argument must be a time duration specification, and defaults to
	   0.

       write_empty_segments 1|0
	   If enabled, write an empty segment if there are no packets during
	   the period a segment would usually span. Otherwise, the segment
	   will be filled with the next packet written. Defaults to 0.

       Examples

       ·   Remux the content of file in.mkv to a list of segments out-000.nut,
	   out-001.nut, etc., and write the list of generated segments to
	   out.list:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut

       ·   Segment input and set output format options for the output
	   segments:

		   ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4

       ·   Segment the input file according to the split points specified by
	   the segment_times option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut

       ·   Use the ffmpeg force_key_frames option to force key frames in the
	   input at the specified location, together with the segment option
	   segment_time_delta to account for possible roundings operated when
	   setting key frame times.

		   ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
		   -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut

	   In order to force key frames on the input file, transcoding is
	   required.

       ·   Segment the input file by splitting the input file according to the
	   frame numbers sequence specified with the segment_frames option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut

       ·   Convert the in.mkv to TS segments using the "libx264" and "aac"
	   encoders:

		   ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts

       ·   Segment the input file, and create an M3U8 live playlist (can be
	   used as live HLS source):

		   ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
		   -segment_list_flags +live -segment_time 10 out%03d.mkv

   smoothstreaming
       Smooth Streaming muxer generates a set of files (Manifest, chunks)
       suitable for serving with conventional web server.

       window_size
	   Specify the number of fragments kept in the manifest. Default 0
	   (keep all).

       extra_window_size
	   Specify the number of fragments kept outside of the manifest before
	   removing from disk. Default 5.

       lookahead_count
	   Specify the number of lookahead fragments. Default 2.

       min_frag_duration
	   Specify the minimum fragment duration (in microseconds). Default
	   5000000.

       remove_at_exit
	   Specify whether to remove all fragments when finished. Default 0
	   (do not remove).

   fifo
       The fifo pseudo-muxer allows the separation of encoding and muxing by
       using first-in-first-out queue and running the actual muxer in a
       separate thread. This is especially useful in combination with the tee
       muxer and can be used to send data to several destinations with
       different reliability/writing speed/latency.

       API users should be aware that callback functions (interrupt_callback,
       io_open and io_close) used within its AVFormatContext must be thread-
       safe.

       The behavior of the fifo muxer if the queue fills up or if the output
       fails is selectable,

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

       ·   encoding can be blocked during temporary failure, or continue
	   transparently dropping packets in case fifo queue fills up.

       fifo_format
	   Specify the format name. Useful if it cannot be guessed from the
	   output name suffix.

       queue_size
	   Specify size of the queue (number of packets). Default value is 60.

       format_opts
	   Specify format options for the underlying muxer. Muxer options can
	   be specified as a list of key=value pairs separated by ':'.

       drop_pkts_on_overflow bool
	   If set to 1 (true), in case the fifo queue fills up, packets will
	   be dropped rather than blocking the encoder. This makes it possible
	   to continue streaming without delaying the input, at the cost of
	   omitting part of the stream. By default this option is set to 0
	   (false), so in such cases the encoder will be blocked until the
	   muxer processes some of the packets and none of them is lost.

       attempt_recovery bool
	   If failure occurs, attempt to recover the output. This is
	   especially useful when used with network output, since it makes it
	   possible to restart streaming transparently.  By default this
	   option is set to 0 (false).

       max_recovery_attempts
	   Sets maximum number of successive unsuccessful recovery attempts
	   after which the output fails permanently. By default this option is
	   set to 0 (unlimited).

       recovery_wait_time duration
	   Waiting time before the next recovery attempt after previous
	   unsuccessful recovery attempt. Default value is 5 seconds.

       recovery_wait_streamtime bool
	   If set to 0 (false), the real time is used when waiting for the
	   recovery attempt (i.e. the recovery will be attempted after at
	   least recovery_wait_time seconds).  If set to 1 (true), the time of
	   the processed stream is taken into account instead (i.e. the
	   recovery will be attempted after at least recovery_wait_time
	   seconds of the stream is omitted).  By default, this option is set
	   to 0 (false).

       recover_any_error bool
	   If set to 1 (true), recovery will be attempted regardless of type
	   of the error causing the failure. By default this option is set to
	   0 (false) and in case of certain (usually permanent) errors the
	   recovery is not attempted even when attempt_recovery is set to 1.

       restart_with_keyframe bool
	   Specify whether to wait for the keyframe after recovering from
	   queue overflow or failure. This option is set to 0 (false) by
	   default.

       Examples

       ·   Stream something to rtmp server, continue processing the stream at
	   real-time rate even in case of temporary failure (network outage)
	   and attempt to recover streaming every second indefinitely.

		   ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
		     -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name

   tee
       The tee muxer can be used to write the same data to several files or
       any other kind of muxer. It can be used, for example, to both stream a
       video to the network and save it to disk at the same time.

       It is different from specifying several outputs to the ffmpeg command-
       line tool because the audio and video data will be encoded only once
       with the tee muxer; encoding can be a very expensive process. It is not
       useful when using the libavformat API directly because it is then
       possible to feed the same packets to several muxers directly.

       use_fifo bool
	   If set to 1, slave outputs will be processed in separate thread
	   using fifo muxer. This allows to compensate for different
	   speed/latency/reliability of outputs and setup transparent
	   recovery. By default this feature is turned off.

       fifo_options
	   Options to pass to fifo pseudo-muxer instances. See fifo.

       The slave outputs are specified in the file name given to the muxer,
       separated by '|'. If any of the slave name contains the '|' separator,
       leading or trailing spaces or any special character, it must be escaped
       (see the "Quoting and escaping" section in the ffmpeg-utils(1) manual).

       Muxer options can be specified for each slave by prepending them as a
       list of key=value pairs separated by ':', between square brackets. If
       the options values contain a special character or the ':' separator,
       they must be escaped; note that this is a second level escaping.

       The following special options are also recognized:

       f   Specify the format name. Useful if it cannot be guessed from the
	   output name suffix.

       bsfs[/spec]
	   Specify a list of bitstream filters to apply to the specified
	   output.

       use_fifo bool
	   This allows to override tee muxer use_fifo option for individual
	   slave muxer.

       fifo_options
	   This allows to override tee muxer fifo_options for individual slave
	   muxer.  See fifo.

	   It is possible to specify to which streams a given bitstream filter
	   applies, by appending a stream specifier to the option separated by
	   "/". spec must be a stream specifier (see Format stream
	   specifiers).  If the stream specifier is not specified, the
	   bitstream filters will be applied to all streams in the output.

	   Several bitstream filters can be specified, separated by ",".

       select
	   Select the streams that should be mapped to the slave output,
	   specified by a stream specifier. If not specified, this defaults to
	   all the input streams. You may use multiple stream specifiers
	   separated by commas (",") e.g.: "a:0,v"

       onfail
	   Specify behaviour on output failure. This can be set to either
	   "abort" (which is default) or "ignore". "abort" will cause whole
	   process to fail in case of failure on this slave output. "ignore"
	   will ignore failure on this output, so other outputs will continue
	   without being affected.

       Examples

       ·   Encode something and both archive it in a WebM file and stream it
	   as MPEG-TS over UDP (the streams need to be explicitly mapped):

		   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		     "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       ·   As above, but continue streaming even if output to local file fails
	   (for example local drive fills up):

		   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		     "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       ·   Use ffmpeg to encode the input, and send the output to three
	   different destinations. The "dump_extra" bitstream filter is used
	   to add extradata information to all the output video keyframes
	   packets, as requested by the MPEG-TS format. The select option is
	   applied to out.aac in order to make it contain only audio packets.

		   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
			  -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

       ·   As below, but select only stream "a:1" for the audio output. Note
	   that a second level escaping must be performed, as ":" is a special
	   character used to separate options.

		   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
			  -f tee "[bsfs/v=dump_extra]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

       Note: some codecs may need different options depending on the output
       format; the auto-detection of this can not work with the tee muxer. The
       main example is the global_header flag.

   webm_dash_manifest
       WebM DASH Manifest muxer.

       This muxer implements the WebM DASH Manifest specification to generate
       the DASH manifest XML. It also supports manifest generation for DASH
       live streams.

       For more information see:

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

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

       Options

       This muxer supports the following options:

       adaptation_sets
	   This option has the following syntax: "id=x,streams=a,b,c
	   id=y,streams=d,e" where x and y are the unique identifiers of the
	   adaptation sets and a,b,c,d and e are the indices of the
	   corresponding audio and video streams. Any number of adaptation
	   sets can be added using this option.

       live
	   Set this to 1 to create a live stream DASH Manifest. Default: 0.

       chunk_start_index
	   Start index of the first chunk. This will go in the startNumber
	   attribute of the SegmentTemplate element in the manifest. Default:
	   0.

       chunk_duration_ms
	   Duration of each chunk in milliseconds. This will go in the
	   duration attribute of the SegmentTemplate element in the manifest.
	   Default: 1000.

       utc_timing_url
	   URL of the page that will return the UTC timestamp in ISO format.
	   This will go in the value attribute of the UTCTiming element in the
	   manifest.  Default: None.

       time_shift_buffer_depth
	   Smallest time (in seconds) shifting buffer for which any
	   Representation is guaranteed to be available. This will go in the
	   timeShiftBufferDepth attribute of the MPD element. Default: 60.

       minimum_update_period
	   Minimum update period (in seconds) of the manifest. This will go in
	   the minimumUpdatePeriod attribute of the MPD element. Default: 0.

       Example

	       ffmpeg -f webm_dash_manifest -i video1.webm \
		      -f webm_dash_manifest -i video2.webm \
		      -f webm_dash_manifest -i audio1.webm \
		      -f webm_dash_manifest -i audio2.webm \
		      -map 0 -map 1 -map 2 -map 3 \
		      -c copy \
		      -f webm_dash_manifest \
		      -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
		      manifest.xml

   webm_chunk
       WebM Live Chunk Muxer.

       This muxer writes out WebM headers and chunks as separate files which
       can be consumed by clients that support WebM Live streams via DASH.

       Options

       This muxer supports the following options:

       chunk_start_index
	   Index of the first chunk (defaults to 0).

       header
	   Filename of the header where the initialization data will be
	   written.

       audio_chunk_duration
	   Duration of each audio chunk in milliseconds (defaults to 5000).

       Example

	       ffmpeg -f v4l2 -i /dev/video0 \
		      -f alsa -i hw:0 \
		      -map 0:0 \
		      -c:v libvpx-vp9 \
		      -s 640x360 -keyint_min 30 -g 30 \
		      -f webm_chunk \
		      -header webm_live_video_360.hdr \
		      -chunk_start_index 1 \
		      webm_live_video_360_%d.chk \
		      -map 1:0 \
		      -c:a libvorbis \
		      -b:a 128k \
		      -f webm_chunk \
		      -header webm_live_audio_128.hdr \
		      -chunk_start_index 1 \
		      -audio_chunk_duration 1000 \
		      webm_live_audio_128_%d.chk

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

       The file format is as follows:

       1.  A file consists of a header and a number of metadata tags divided
	   into sections, each on its own line.

       2.  The header is a ;FFMETADATA string, followed by a version number
	   (now 1).

       3.  Metadata tags are of the form key=value

       4.  Immediately after header follows global metadata

       5.  After global metadata there may be sections with
	   per-stream/per-chapter metadata.

       6.  A section starts with the section name in uppercase (i.e. STREAM or
	   CHAPTER) in brackets ([, ]) and ends with next section or end of
	   file.

       7.  At the beginning of a chapter section there may be an optional
	   timebase to be used for start/end values. It must be in form
	   TIMEBASE=num/den, where num and den are integers. If the timebase
	   is missing then start/end times are assumed to be in milliseconds.

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

       8.  Empty lines and lines starting with ; or # are ignored.

       9.  Metadata keys or values containing special characters (=, ;, #, \
	   and a newline) must be escaped with a backslash \.

       10. Note that whitespace in metadata (e.g. foo = bar) is considered to
	   be a part of the tag (in the example above key is foo , value is
	    bar).

       A ffmetadata file might look like this:

	       ;FFMETADATA1
	       title=bike\\shed
	       ;this is a comment
	       artist=FFmpeg troll team

	       [CHAPTER]
	       TIMEBASE=1/1000
	       START=0
	       #chapter ends at 0:01:00
	       END=60000
	       title=chapter \#1
	       [STREAM]
	       title=multi\
	       line

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

       Extracting an ffmetadata file with ffmpeg goes as follows:

	       ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

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

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

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

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

       The list of supported options follows:

       protocol_whitelist list (input)
	   Set a ","-separated list of allowed protocols. "ALL" matches all
	   protocols. Protocols prefixed by "-" are disabled.  All protocols
	   are allowed by default but protocols used by an another protocol
	   (nested protocols) are restricted to a per protocol subset.

PROTOCOLS
       Protocols are configured elements in FFmpeg that enable access to
       resources that require specific protocols.

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

       You can disable all the protocols using the configure option
       "--disable-protocols", and selectively enable a protocol using the
       option "--enable-protocol=PROTOCOL", or you can disable a particular
       protocol using the option "--disable-protocol=PROTOCOL".

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

       All protocols accept the following options:

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

       A description of the currently available protocols follows.

   async
       Asynchronous data filling wrapper for input stream.

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

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

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
	   BluRay angle

       chapter
	   Start chapter (1...N)

       playlist
	   Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

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

	       bluray:/mnt/bluray

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

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

   cache
       Caching wrapper for input stream.

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

	       cache:<URL>

   concat
       Physical concatenation protocol.

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

       A URL accepted by this protocol has the syntax:

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

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

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

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

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

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set the AES decryption key binary block from given hexadecimal
	   representation.

       iv  Set the AES decryption initialization vector binary block from
	   given hexadecimal representation.

       Accepted URL formats:

	       crypto:<URL>
	       crypto+<URL>

   data
       Data in-line in the URI. See
       <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given inline with ffmpeg:

	       ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

	       file:<filename>

       where filename is the path of the file to read.

       An URL that does not have a protocol prefix will be assumed to be a
       file URL. Depending on the build, an URL that looks like a Windows path
       with the drive letter at the beginning will also be assumed to be a
       file URL (usually not the case in builds for unix-like systems).

       For example to read from a file input.mpeg with ffmpeg use the command:

	       ffmpeg -i file:input.mpeg output.mpeg

       This protocol accepts the following options:

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

       blocksize
	   Set I/O operation maximum block size, in bytes. Default value is
	   "INT_MAX", which results in not limiting the requested block size.
	   Setting this value reasonably low improves user termination request
	   reaction time, which is valuable for files on slow medium.

   ftp
       FTP (File Transfer Protocol).

       Read from or write to remote resources using FTP protocol.

       Following syntax is required.

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

       This protocol accepts the following options.

       timeout
	   Set timeout in microseconds of socket I/O operations used by the
	   underlying low level operation. By default it is set to -1, which
	   means that the timeout is not specified.

       ftp-anonymous-password
	   Password used when login as anonymous user. Typically an e-mail
	   address should be used.

       ftp-write-seekable
	   Control seekability of connection during encoding. If set to 1 the
	   resource is supposed to be seekable, if set to 0 it is assumed not
	   to be seekable. Default value is 0.

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

       This protocol accepts the following options:

       follow
	   If set to 1, the protocol will retry reading at the end of the
	   file, allowing reading files that still are being written. In order
	   for this to terminate, you either need to use the rw_timeout
	   option, or use the interrupt callback (for API users).

   gopher
       Gopher protocol.

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

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

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

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options:

       seekable
	   Control seekability of connection. If set to 1 the resource is
	   supposed to be seekable, if set to 0 it is assumed not to be
	   seekable, if set to -1 it will try to autodetect if it is seekable.
	   Default value is -1.

       chunked_post
	   If set to 1 use chunked Transfer-Encoding for posts, default is 1.

       content_type
	   Set a specific content type for the POST messages or for listen
	   mode.

       http_proxy
	   set HTTP proxy to tunnel through e.g. http://example.com:1234

       headers
	   Set custom HTTP headers, can override built in default headers. The
	   value must be a string encoding the headers.

       multiple_requests
	   Use persistent connections if set to 1, default is 0.

       post_data
	   Set custom HTTP post data.

       user_agent
	   Override the User-Agent header. If not specified the protocol will
	   use a string describing the libavformat build. ("Lavf/<version>")

       user-agent
	   This is a deprecated option, you can use user_agent instead it.

       timeout
	   Set timeout in microseconds of socket I/O operations used by the
	   underlying low level operation. By default it is set to -1, which
	   means that the timeout is not specified.

       reconnect_at_eof
	   If set then eof is treated like an error and causes reconnection,
	   this is useful for live / endless streams.

       reconnect_streamed
	   If set then even streamed/non seekable streams will be reconnected
	   on errors.

       reconnect_delay_max
	   Sets the maximum delay in seconds after which to give up
	   reconnecting

       mime_type
	   Export the MIME type.

       icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
	   the server supports this, the metadata has to be retrieved by the
	   application by reading the icy_metadata_headers and
	   icy_metadata_packet options.  The default is 1.

       icy_metadata_headers
	   If the server supports ICY metadata, this contains the ICY-specific
	   HTTP reply headers, separated by newline characters.

       icy_metadata_packet
	   If the server supports ICY metadata, and icy was set to 1, this
	   contains the last non-empty metadata packet sent by the server. It
	   should be polled in regular intervals by applications interested in
	   mid-stream metadata updates.

       cookies
	   Set the cookies to be sent in future requests. The format of each
	   cookie is the same as the value of a Set-Cookie HTTP response
	   field. Multiple cookies can be delimited by a newline character.

       offset
	   Set initial byte offset.

       end_offset
	   Try to limit the request to bytes preceding this offset.

       method
	   When used as a client option it sets the HTTP method for the
	   request.

	   When used as a server option it sets the HTTP method that is going
	   to be expected from the client(s).  If the expected and the
	   received HTTP method do not match the client will be given a Bad
	   Request response.  When unset the HTTP method is not checked for
	   now. This will be replaced by autodetection in the future.

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

		   # Server side (sending):
		   ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

		   # Client side (receiving):
		   ffmpeg -i http://<server>:<port> -c copy somefile.ogg

		   # Client can also be done with wget:
		   wget http://<server>:<port> -O somefile.ogg

		   # Server side (receiving):
		   ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

		   # Client side (sending):
		   ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

		   # Client can also be done with wget:
		   wget --post-file=somefile.ogg http://<server>:<port>

       HTTP Cookies

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

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

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

   Icecast
       Icecast protocol (stream to Icecast servers)

       This protocol accepts the following options:

       ice_genre
	   Set the stream genre.

       ice_name
	   Set the stream name.

       ice_description
	   Set the stream description.

       ice_url
	   Set the stream website URL.

       ice_public
	   Set if the stream should be public.	The default is 0 (not public).

       user_agent
	   Override the User-Agent header. If not specified a string of the
	   form "Lavf/<version>" will be used.

       password
	   Set the Icecast mountpoint password.

       content_type
	   Set the stream content type. This must be set if it is different
	   from audio/mpeg.

       legacy_icecast
	   This enables support for Icecast versions < 2.4.0, that do not
	   support the HTTP PUT method but the SOURCE method.

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

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

	       mmsh://<server>[:<port>][/<app>][/<playpath>]

   md5
       MD5 output protocol.

       Computes the MD5 hash of the data to be written, and on close writes
       this to the designated output or stdout if none is specified. It can be
       used to test muxers without writing an actual file.

       Some examples follow.

	       # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
	       ffmpeg -i input.flv -f avi -y md5:output.avi.md5

	       # Write the MD5 hash of the encoded AVI file to stdout.
	       ffmpeg -i input.flv -f avi -y md5:

       Note that some formats (typically MOV) require the output protocol to
       be seekable, so they will fail with the MD5 output protocol.

   pipe
       UNIX pipe access protocol.

       Read and write from UNIX pipes.

       The accepted syntax is:

	       pipe:[<number>]

       number is the number corresponding to the file descriptor of the pipe
       (e.g. 0 for stdin, 1 for stdout, 2 for stderr).	If number is not
       specified, by default the stdout file descriptor will be used for
       writing, stdin for reading.

       For example to read from stdin with ffmpeg:

	       cat test.wav | ffmpeg -i pipe:0
	       # ...this is the same as...
	       cat test.wav | ffmpeg -i pipe:

       For writing to stdout with ffmpeg:

	       ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
	       # ...this is the same as...
	       ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       This protocol accepts the following options:

       blocksize
	   Set I/O operation maximum block size, in bytes. Default value is
	   "INT_MAX", which results in not limiting the requested block size.
	   Setting this value reasonably low improves user termination request
	   reaction time, which is valuable if data transmission is slow.

       Note that some formats (typically MOV), require the output protocol to
       be seekable, so they will fail with the pipe output protocol.

   prompeg
       Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

       The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
       mechanism for MPEG-2 Transport Streams sent over RTP.

       This protocol must be used in conjunction with the "rtp_mpegts" muxer
       and the "rtp" protocol.

       The required syntax is:

	       -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

       The destination UDP ports are "port + 2" for the column FEC stream and
       "port + 4" for the row FEC stream.

       This protocol accepts the following options:

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

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

       Example usage:

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

   rtmp
       Real-Time Messaging Protocol.

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

       The required syntax is:

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

       The accepted parameters are:

       username
	   An optional username (mostly for publishing).

       password
	   An optional password (mostly for publishing).

       server
	   The address of the RTMP server.

       port
	   The number of the TCP port to use (by default is 1935).

       app It is the name of the application to access. It usually corresponds
	   to the path where the application is installed on the RTMP server
	   (e.g. /ondemand/, /flash/live/, etc.). You can override the value
	   parsed from the URI through the "rtmp_app" option, too.

       playpath
	   It is the path or name of the resource to play with reference to
	   the application specified in app, may be prefixed by "mp4:". You
	   can override the value parsed from the URI through the
	   "rtmp_playpath" option, too.

       listen
	   Act as a server, listening for an incoming connection.

       timeout
	   Maximum time to wait for the incoming connection. Implies listen.

       Additionally, the following parameters can be set via command line
       options (or in code via "AVOption"s):

       rtmp_app
	   Name of application to connect on the RTMP server. This option
	   overrides the parameter specified in the URI.

       rtmp_buffer
	   Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
	   Extra arbitrary AMF connection parameters, parsed from a string,
	   e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each
	   value is prefixed by a single character denoting the type, B for
	   Boolean, N for number, S for string, O for object, or Z for null,
	   followed by a colon. For Booleans the data must be either 0 or 1
	   for FALSE or TRUE, respectively.  Likewise for Objects the data
	   must be 0 or 1 to end or begin an object, respectively. Data items
	   in subobjects may be named, by prefixing the type with 'N' and
	   specifying the name before the value (i.e. "NB:myFlag:1"). This
	   option may be used multiple times to construct arbitrary AMF
	   sequences.

       rtmp_flashver
	   Version of the Flash plugin used to run the SWF player. The default
	   is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
	   (compatible; <libavformat version>).)

       rtmp_flush_interval
	   Number of packets flushed in the same request (RTMPT only). The
	   default is 10.

       rtmp_live
	   Specify that the media is a live stream. No resuming or seeking in
	   live streams is possible. The default value is "any", which means
	   the subscriber first tries to play the live stream specified in the
	   playpath. If a live stream of that name is not found, it plays the
	   recorded stream. The other possible values are "live" and
	   "recorded".

       rtmp_pageurl
	   URL of the web page in which the media was embedded. By default no
	   value will be sent.

       rtmp_playpath
	   Stream identifier to play or to publish. This option overrides the
	   parameter specified in the URI.

       rtmp_subscribe
	   Name of live stream to subscribe to. By default no value will be
	   sent.  It is only sent if the option is specified or if rtmp_live
	   is set to live.

       rtmp_swfhash
	   SHA256 hash of the decompressed SWF file (32 bytes).

       rtmp_swfsize
	   Size of the decompressed SWF file, required for SWFVerification.

       rtmp_swfurl
	   URL of the SWF player for the media. By default no value will be
	   sent.

       rtmp_swfverify
	   URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
	   URL of the target stream. Defaults to proto://host[:port]/app.

       For example to read with ffplay a multimedia resource named "sample"
       from the application "vod" from an RTMP server "myserver":

	       ffplay rtmp://myserver/vod/sample

       To publish to a password protected server, passing the playpath and app
       names separately:

	       ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

   rtmpe
       Encrypted Real-Time Messaging Protocol.

       The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
       streaming multimedia content within standard cryptographic primitives,
       consisting of Diffie-Hellman key exchange and HMACSHA256, generating a
       pair of RC4 keys.

   rtmps
       Real-Time Messaging Protocol over a secure SSL connection.

       The Real-Time Messaging Protocol (RTMPS) is used for streaming
       multimedia content across an encrypted connection.

   rtmpt
       Real-Time Messaging Protocol tunneled through HTTP.

       The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used
       for streaming multimedia content within HTTP requests to traverse
       firewalls.

   rtmpte
       Encrypted Real-Time Messaging Protocol tunneled through HTTP.

       The Encrypted Real-Time Messaging Protocol tunneled through HTTP
       (RTMPTE) is used for streaming multimedia content within HTTP requests
       to traverse firewalls.

   rtmpts
       Real-Time Messaging Protocol tunneled through HTTPS.

       The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is
       used for streaming multimedia content within HTTPS requests to traverse
       firewalls.

   libsmbclient
       libsmbclient permits one to manipulate CIFS/SMB network resources.

       Following syntax is required.

	       smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

       This protocol accepts the following options.

       timeout
	   Set timeout in milliseconds of socket I/O operations used by the
	   underlying low level operation. By default it is set to -1, which
	   means that the timeout is not specified.

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

       workgroup
	   Set the workgroup used for making connections. By default workgroup
	   is not specified.

       For more information see: <http://www.samba.org/>.

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax is required.

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

       This protocol accepts the following options.

       timeout
	   Set timeout of socket I/O operations used by the underlying low
	   level operation. By default it is set to -1, which means that the
	   timeout is not specified.

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

       private_key
	   Specify the path of the file containing private key to use during
	   authorization.  By default libssh searches for keys in the ~/.ssh/
	   directory.

       Example: Play a file stored on remote server.

	       ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

   librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
       Real-Time Messaging Protocol and its variants supported through
       librtmp.

       Requires the presence of the librtmp headers and library during
       configuration. You need to explicitly configure the build with
       "--enable-librtmp". If enabled this will replace the native RTMP
       protocol.

       This protocol provides most client functions and a few server functions
       needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
       (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these
       encrypted types (RTMPTE, RTMPTS).

       The required syntax is:

	       <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

       where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe",
       "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and
       server, port, app and playpath have the same meaning as specified for
       the RTMP native protocol.  options contains a list of space-separated
       options of the form key=val.

       See the librtmp manual page (man 3 librtmp) for more information.

       For example, to stream a file in real-time to an RTMP server using
       ffmpeg:

	       ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

       To play the same stream using ffplay:

	       ffplay "rtmp://myserver/live/mystream live=1"

   rtp
       Real-time Transport Protocol.

       The required syntax for an RTP URL is:
       rtp://hostname[:port][?option=val...]

       port specifies the RTP port to use.

       The following URL options are supported:

       ttl=n
	   Set the TTL (Time-To-Live) value (for multicast only).

       rtcpport=n
	   Set the remote RTCP port to n.

       localrtpport=n
	   Set the local RTP port to n.

       localrtcpport=n'
	   Set the local RTCP port to n.

       pkt_size=n
	   Set max packet size (in bytes) to n.

       connect=0|1
	   Do a "connect()" on the UDP socket (if set to 1) or not (if set to
	   0).

       sources=ip[,ip]
	   List allowed source IP addresses.

       block=ip[,ip]
	   List disallowed (blocked) source IP addresses.

       write_to_source=0|1
	   Send packets to the source address of the latest received packet
	   (if set to 1) or to a default remote address (if set to 0).

       localport=n
	   Set the local RTP port to n.

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

       Important notes:

       1.  If rtcpport is not set the RTCP port will be set to the RTP port
	   value plus 1.

       2.  If localrtpport (the local RTP port) is not set any available port
	   will be used for the local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it will be set to
	   the local RTP port value plus 1.

   rtsp
       Real-Time Streaming Protocol.

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

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

       The required syntax for a RTSP url is:

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

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

       The following options are supported.

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

       rtsp_transport
	   Set RTSP transport protocols.

	   It accepts the following values:

	   udp Use UDP as lower transport protocol.

	   tcp Use TCP (interleaving within the RTSP control channel) as lower
	       transport protocol.

	   udp_multicast
	       Use UDP multicast as lower transport protocol.

	   http
	       Use HTTP tunneling as lower transport protocol, which is useful
	       for passing proxies.

	   Multiple lower transport protocols may be specified, in that case
	   they are tried one at a time (if the setup of one fails, the next
	   one is tried).  For the muxer, only the tcp and udp options are
	   supported.

       rtsp_flags
	   Set RTSP flags.

	   The following values are accepted:

	   filter_src
	       Accept packets only from negotiated peer address and port.

	   listen
	       Act as a server, listening for an incoming connection.

	   prefer_tcp
	       Try TCP for RTP transport first, if TCP is available as RTSP
	       RTP transport.

	   Default value is none.

       allowed_media_types
	   Set media types to accept from the server.

	   The following flags are accepted:

	   video
	   audio
	   data

	   By default it accepts all media types.

       min_port
	   Set minimum local UDP port. Default value is 5000.

       max_port
	   Set maximum local UDP port. Default value is 65000.

       timeout
	   Set maximum timeout (in seconds) to wait for incoming connections.

	   A value of -1 means infinite (default). This option implies the
	   rtsp_flags set to listen.

       reorder_queue_size
	   Set number of packets to buffer for handling of reordered packets.

       stimeout
	   Set socket TCP I/O timeout in microseconds.

       user-agent
	   Override User-Agent header. If not specified, it defaults to the
	   libavformat identifier string.

       When receiving data over UDP, the demuxer tries to reorder received
       packets (since they may arrive out of order, or packets may get lost
       totally). This can be disabled by setting the maximum demuxing delay to
       zero (via the "max_delay" field of AVFormatContext).

       When watching multi-bitrate Real-RTSP streams with ffplay, the streams
       to display can be chosen with "-vst" n and "-ast" n for video and audio
       respectively, and can be switched on the fly by pressing "v" and "a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       ·   Watch a stream over UDP, with a max reordering delay of 0.5
	   seconds:

		   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

       ·   Watch a stream tunneled over HTTP:

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

       ·   Send a stream in realtime to a RTSP server, for others to watch:

		   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       ·   Receive a stream in realtime:

		   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a
       protocol handler in libavformat, it is a muxer and demuxer.  It is used
       for signalling of RTP streams, by announcing the SDP for the streams
       regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

	       sap://<destination>[:<port>][?<options>]

       The RTP packets are sent to destination on port port, or to port 5004
       if no port is specified.  options is a "&"-separated list. The
       following options are supported:

       announce_addr=address
	   Specify the destination IP address for sending the announcements
	   to.	If omitted, the announcements are sent to the commonly used
	   SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
	   or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
	   Specify the port to send the announcements on, defaults to 9875 if
	   not specified.

       ttl=ttl
	   Specify the time to live value for the announcements and RTP
	   packets, defaults to 255.

       same_port=0|1
	   If set to 1, send all RTP streams on the same port pair. If zero
	   (the default), all streams are sent on unique ports, with each
	   stream on a port 2 numbers higher than the previous.  VLC/Live555
	   requires this to be set to 1, to be able to receive the stream.
	   The RTP stack in libavformat for receiving requires all streams to
	   be sent on unique ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

	       ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

	       sap://[<address>][:<port>]

       address is the multicast address to listen for announcements on, if
       omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
       port that is listened on, 9875 if omitted.

       The demuxers listens for announcements on the given address and port.
       Once an announcement is received, it tries to receive that particular
       stream.

       Example command lines follow.

       To play back the first stream announced on the normal SAP multicast
       address:

	       ffplay sap://

       To play back the first stream announced on one the default IPv6 SAP
       multicast address:

	       ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

	       sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

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

       max_streams
	   Set the maximum number of streams. By default no limit is set.

   srtp
       Secure Real-time Transport Protocol.

       The accepted options are:

       srtp_in_suite
       srtp_out_suite
	   Select input and output encoding suites.

	   Supported values:

	   AES_CM_128_HMAC_SHA1_80
	   SRTP_AES128_CM_HMAC_SHA1_80
	   AES_CM_128_HMAC_SHA1_32
	   SRTP_AES128_CM_HMAC_SHA1_32
       srtp_in_params
       srtp_out_params
	   Set input and output encoding parameters, which are expressed by a
	   base64-encoded representation of a binary block. The first 16 bytes
	   of this binary block are used as master key, the following 14 bytes
	   are used as master salt.

   subfile
       Virtually extract a segment of a file or another stream.  The
       underlying stream must be seekable.

       Accepted options:

       start
	   Start offset of the extracted segment, in bytes.

       end End offset of the extracted segment, in bytes.

       Examples:

       Extract a chapter from a DVD VOB file (start and end sectors obtained
       externally and multiplied by 2048):

	       subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

       Play an AVI file directly from a TAR archive:

	       subfile,,start,183241728,end,366490624,,:archive.tar

   tee
       Writes the output to multiple protocols. The individual outputs are
       separated by |

	       tee:file://path/to/local/this.avi|file://path/to/local/that.avi

   tcp
       Transmission Control Protocol.

       The required syntax for a TCP url is:

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

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

       The list of supported options follows.

       listen=1|0
	   Listen for an incoming connection. Default value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This option is only relevant in read mode: if no data arrived in
	   more than this time interval, raise error.

       listen_timeout=milliseconds
	   Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
	   Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
	   Set send buffer size, expressed bytes.

       The following example shows how to setup a listening TCP connection
       with ffmpeg, which is then accessed with ffplay:

	       ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
	       ffplay tcp://<hostname>:<port>

   tls
       Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

       The required syntax for a TLS/SSL url is:

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

       The following parameters can be set via command line options (or in
       code via "AVOption"s):

       ca_file, cafile=filename
	   A file containing certificate authority (CA) root certificates to
	   treat as trusted. If the linked TLS library contains a default this
	   might not need to be specified for verification to work, but not
	   all libraries and setups have defaults built in.  The file must be
	   in OpenSSL PEM format.

       tls_verify=1|0
	   If enabled, try to verify the peer that we are communicating with.
	   Note, if using OpenSSL, this currently only makes sure that the
	   peer certificate is signed by one of the root certificates in the
	   CA database, but it does not validate that the certificate actually
	   matches the host name we are trying to connect to. (With GnuTLS,
	   the host name is validated as well.)

	   This is disabled by default since it requires a CA database to be
	   provided by the caller in many cases.

       cert_file, cert=filename
	   A file containing a certificate to use in the handshake with the
	   peer.  (When operating as server, in listen mode, this is more
	   often required by the peer, while client certificates only are
	   mandated in certain setups.)

       key_file, key=filename
	   A file containing the private key for the certificate.

       listen=1|0
	   If enabled, listen for connections on the provided port, and assume
	   the server role in the handshake instead of the client role.

       Example command lines:

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

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

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

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

   udp
       User Datagram Protocol.

       The required syntax for an UDP URL is:

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

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

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

       The list of supported options follows.

       buffer_size=size
	   Set the UDP maximum socket buffer size in bytes. This is used to
	   set either the receive or send buffer size, depending on what the
	   socket is used for.	Default is 64KB.  See also fifo_size.

       bitrate=bitrate
	   If set to nonzero, the output will have the specified constant
	   bitrate if the input has enough packets to sustain it.

       burst_bits=bits
	   When using bitrate this specifies the maximum number of bits in
	   packet bursts.

       localport=port
	   Override the local UDP port to bind with.

       localaddr=addr
	   Choose the local IP address. This is useful e.g. if sending
	   multicast and the host has multiple interfaces, where the user can
	   choose which interface to send on by specifying the IP address of
	   that interface.

       pkt_size=size
	   Set the size in bytes of UDP packets.

       reuse=1|0
	   Explicitly allow or disallow reusing UDP sockets.

       ttl=ttl
	   Set the time to live value (for multicast only).

       connect=1|0
	   Initialize the UDP socket with "connect()". In this case, the
	   destination address can't be changed with ff_udp_set_remote_url
	   later.  If the destination address isn't known at the start, this
	   option can be specified in ff_udp_set_remote_url, too.  This allows
	   finding out the source address for the packets with getsockname,
	   and makes writes return with AVERROR(ECONNREFUSED) if "destination
	   unreachable" is received.  For receiving, this gives the benefit of
	   only receiving packets from the specified peer address/port.

       sources=address[,address]
	   Only receive packets sent to the multicast group from one of the
	   specified sender IP addresses.

       block=address[,address]
	   Ignore packets sent to the multicast group from the specified
	   sender IP addresses.

       fifo_size=units
	   Set the UDP receiving circular buffer size, expressed as a number
	   of packets with size of 188 bytes. If not specified defaults to
	   7*4096.

       overrun_nonfatal=1|0
	   Survive in case of UDP receiving circular buffer overrun. Default
	   value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This option is only relevant in read mode: if no data arrived in
	   more than this time interval, raise error.

       broadcast=1|0
	   Explicitly allow or disallow UDP broadcasting.

	   Note that broadcasting may not work properly on networks having a
	   broadcast storm protection.

       Examples

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

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

       ·   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
	   packets, using a large input buffer:

		   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

       ·   Use ffmpeg to receive over UDP from a remote endpoint:

		   ffmpeg -i udp://[<multicast-address>]:<port> ...

   unix
       Unix local socket

       The required syntax for a Unix socket URL is:

	       unix://<filepath>

       The following parameters can be set via command line options (or in
       code via "AVOption"s):

       timeout
	   Timeout in ms.

       listen
	   Create the Unix socket in listening mode.

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

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

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

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

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

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

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

       A description of the currently available input devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) input device.

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

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

       An ALSA identifier has the syntax:

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

       where the DEV and SUBDEV components are optional.

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

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

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

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

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

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   avfoundation
       AVFoundation input device.

       AVFoundation is the currently recommended framework by Apple for
       streamgrabbing on OSX >= 10.7 as well as on iOS.

       The input filename has to be given in the following syntax:

	       -i "[[VIDEO]:[AUDIO]]"

       The first entry selects the video input while the latter selects the
       audio input.  The stream has to be specified by the device name or the
       device index as shown by the device list.  Alternatively, the video
       and/or audio input device can be chosen by index using the

	   B<-video_device_index E<lt>INDEXE<gt>>

       and/or

	   B<-audio_device_index E<lt>INDEXE<gt>>

       , overriding any device name or index given in the input filename.

       All available devices can be enumerated by using -list_devices true,
       listing all device names and corresponding indices.

       There are two device name aliases:

       "default"
	   Select the AVFoundation default device of the corresponding type.

       "none"
	   Do not record the corresponding media type.	This is equivalent to
	   specifying an empty device name or index.

       Options

       AVFoundation supports the following options:

       -list_devices <TRUE|FALSE>
	   If set to true, a list of all available input devices is given
	   showing all device names and indices.

       -video_device_index <INDEX>
	   Specify the video device by its index. Overrides anything given in
	   the input filename.

       -audio_device_index <INDEX>
	   Specify the audio device by its index. Overrides anything given in
	   the input filename.

       -pixel_format <FORMAT>
	   Request the video device to use a specific pixel format.  If the
	   specified format is not supported, a list of available formats is
	   given and the first one in this list is used instead. Available
	   pixel formats are: "monob, rgb555be, rgb555le, rgb565be, rgb565le,
	   rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
	    bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16,
	   yuv422p10, yuv444p10,
	    yuv420p, nv12, yuyv422, gray"

       -framerate
	   Set the grabbing frame rate. Default is "ntsc", corresponding to a
	   frame rate of "30000/1001".

       -video_size
	   Set the video frame size.

       -capture_cursor
	   Capture the mouse pointer. Default is 0.

       -capture_mouse_clicks
	   Capture the screen mouse clicks. Default is 0.

       Examples

       ·   Print the list of AVFoundation supported devices and exit:

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

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

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

       ·   Record video from video device 2 and audio from audio device 1 into
	   out.avi:

		   $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

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

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

   bktr
       BSD video input device.

       Options

       framerate
	   Set the frame rate.

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

       standard
	   Available values are:

	   pal
	   ntsc
	   secam
	   paln
	   palm
	   ntscj

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

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

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

       Options

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       Examples

       ·   List input devices:

		   ffmpeg -f decklink -list_devices 1 -i dummy

       ·   List supported formats:

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

       ·   Capture video clip at 1080i50:

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

       ·   Capture video clip at 1080i50 10 bit:

		   ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

       ·   Capture video clip at 1080i50 with 16 audio channels:

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

   kmsgrab
       KMS video input device.

       Captures the KMS scanout framebuffer associated with a specified CRTC
       or plane as a DRM object that can be passed to other hardware
       functions.

       Requires either DRM master or CAP_SYS_ADMIN to run.

       If you don't understand what all of that means, you probably don't want
       this.  Look at x11grab instead.

       Options

       device
	   DRM device to capture on.  Defaults to /dev/dri/card0.

       format
	   Pixel format of the framebuffer.  Defaults to bgr0.

       format_modifier
	   Format modifier to signal on output frames.	This is necessary to
	   import correctly into some APIs, but can't be autodetected.	See
	   the libdrm documentation for possible values.

       crtc_id
	   KMS CRTC ID to define the capture source.  The first active plane
	   on the given CRTC will be used.

       plane_id
	   KMS plane ID to define the capture source.  Defaults to the first
	   active plane found if neither crtc_id nor plane_id are specified.

       framerate
	   Framerate to capture at.  This is not synchronised to any page
	   flipping or framebuffer changes - it just defines the interval at
	   which the framebuffer is sampled.  Sampling faster than the
	   framebuffer update rate will generate independent frames with the
	   same content.  Defaults to 30.

       Examples

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

		   ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4

       ·   Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert
	   to NV12 and encode as H.264.

		   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

   libndi_newtek
       The libndi_newtek input device provides capture capabilities for using
       NDI (Network Device Interface, standard created by NewTek).

       Input filename is a NDI source name that could be found by sending
       -find_sources 1 to command line - it has no specific syntax but human-
       readable formatted.

       To enable this input device, you need the NDI SDK and you need to
       configure with the appropriate "--extra-cflags" and "--extra-ldflags".

       Options

       find_sources
	   If set to true, print a list of found/available NDI sources and
	   exit.  Defaults to false.

       wait_sources
	   Override time to wait until the number of online sources have
	   changed.  Defaults to 0.5.

       allow_video_fields
	   When this flag is false, all video that you receive will be
	   progressive.  Defaults to true.

       Examples

       ·   List input devices:

		   ffmpeg -f libndi_newtek -find_sources 1 -i dummy

       ·   Restream to NDI:

		   ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2

   dshow
       Windows DirectShow input device.

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

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

       The input name should be in the format:

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

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

       Options

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

       video_size
	   Set the video size in the captured video.

       framerate
	   Set the frame rate in the captured video.

       sample_rate
	   Set the sample rate (in Hz) of the captured audio.

       sample_size
	   Set the sample size (in bits) of the captured audio.

       channels
	   Set the number of channels in the captured audio.

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

       list_options
	   If set to true, print a list of selected device's options and exit.

       video_device_number
	   Set video device number for devices with the same name (starts at
	   0, defaults to 0).

       audio_device_number
	   Set audio device number for devices with the same name (starts at
	   0, defaults to 0).

       pixel_format
	   Select pixel format to be used by DirectShow. This may only be set
	   when the video codec is not set or set to rawvideo.

       audio_buffer_size
	   Set audio device buffer size in milliseconds (which can directly
	   impact latency, depending on the device).  Defaults to using the
	   audio device's default buffer size (typically some multiple of
	   500ms).  Setting this value too low can degrade performance.  See
	   also
	   <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>

       video_pin_name
	   Select video capture pin to use by name or alternative name.

       audio_pin_name
	   Select audio capture pin to use by name or alternative name.

       crossbar_video_input_pin_number
	   Select video input pin number for crossbar device. This will be
	   routed to the crossbar device's Video Decoder output pin.  Note
	   that changing this value can affect future invocations (sets a new
	   default) until system reboot occurs.

       crossbar_audio_input_pin_number
	   Select audio input pin number for crossbar device. This will be
	   routed to the crossbar device's Audio Decoder output pin.  Note
	   that changing this value can affect future invocations (sets a new
	   default) until system reboot occurs.

       show_video_device_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to change video filter properties and
	   configurations manually.  Note that for crossbar devices, adjusting
	   values in this dialog may be needed at times to toggle between PAL
	   (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing,
	   etc.  Changing these values can enable different scan rates/frame
	   rates and avoiding green bars at the bottom, flickering scan lines,
	   etc.  Note that with some devices, changing these properties can
	   also affect future invocations (sets new defaults) until system
	   reboot occurs.

       show_audio_device_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to change audio filter properties and
	   configurations manually.

       show_video_crossbar_connection_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to manually modify crossbar pin
	   routings, when it opens a video device.

       show_audio_crossbar_connection_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to manually modify crossbar pin
	   routings, when it opens an audio device.

       show_analog_tv_tuner_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to manually modify TV channels and
	   frequencies.

       show_analog_tv_tuner_audio_dialog
	   If set to true, before capture starts, popup a display dialog to
	   the end user, allowing them to manually modify TV audio (like mono
	   vs. stereo, Language A,B or C).

       audio_device_load
	   Load an audio capture filter device from file instead of searching
	   it by name. It may load additional parameters too, if the filter
	   supports the serialization of its properties to.  To use this an
	   audio capture source has to be specified, but it can be anything
	   even fake one.

       audio_device_save
	   Save the currently used audio capture filter device and its
	   parameters (if the filter supports it) to a file.  If a file with
	   the same name exists it will be overwritten.

       video_device_load
	   Load a video capture filter device from file instead of searching
	   it by name. It may load additional parameters too, if the filter
	   supports the serialization of its properties to.  To use this a
	   video capture source has to be specified, but it can be anything
	   even fake one.

       video_device_save
	   Save the currently used video capture filter device and its
	   parameters (if the filter supports it) to a file.  If a file with
	   the same name exists it will be overwritten.

       Examples

       ·   Print the list of DirectShow supported devices and exit:

		   $ ffmpeg -list_devices true -f dshow -i dummy

       ·   Open video device Camera:

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

       ·   Open second video device with name Camera:

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

       ·   Open video device Camera and audio device Microphone:

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

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

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

       ·   Specify pin names to capture by name or alternative name, specify
	   alternative device name:

		   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

       ·   Configure a crossbar device, specifying crossbar pins, allow user
	   to adjust video capture properties at startup:

		   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
			-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

   fbdev
       Linux framebuffer input device.

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

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

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

       To record from the framebuffer device /dev/fb0 with ffmpeg:

	       ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

       You can take a single screenshot image with the command:

	       ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

       Options

       framerate
	   Set the frame rate. Default is 25.

   gdigrab
       Win32 GDI-based screen capture device.

       This device allows you to capture a region of the display on Windows.

       There are two options for the input filename:

	       desktop

       or

	       title=<window_title>

       The first option will capture the entire desktop, or a fixed region of
       the desktop. The second option will instead capture the contents of a
       single window, regardless of its position on the screen.

       For example, to grab the entire desktop using ffmpeg:

	       ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

       Grab a 640x480 region at position "10,20":

	       ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

       Grab the contents of the window named "Calculator"

	       ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. Use the value 0 to not
	   draw the pointer. Default value is 1.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

       show_region
	   Show grabbed region on screen.

	   If show_region is specified with 1, then the grabbing region will
	   be indicated on screen. With this option, it is easy to know what
	   is being grabbed if only a portion of the screen is grabbed.

	   Note that show_region is incompatible with grabbing the contents of
	   a single window.

	   For example:

		   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

       video_size
	   Set the video frame size. The default is to capture the full screen
	   if desktop is selected, or the full window size if
	   title=window_title is selected.

       offset_x
	   When capturing a region with video_size, set the distance from the
	   left edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary monitor on Windows. If you have a monitor positioned to the
	   left of your primary monitor, you will need to use a negative
	   offset_x value to move the region to that monitor.

       offset_y
	   When capturing a region with video_size, set the distance from the
	   top edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary monitor on Windows. If you have a monitor positioned above
	   your primary monitor, you will need to use a negative offset_y
	   value to move the region to that monitor.

   iec61883
       FireWire DV/HDV input device using libiec61883.

       To enable this input device, you need libiec61883, libraw1394 and
       libavc1394 installed on your system. Use the configure option
       "--enable-libiec61883" to compile with the device enabled.

       The iec61883 capture device supports capturing from a video device
       connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
       FireWire stack (juju). This is the default DV/HDV input method in Linux
       Kernel 2.6.37 and later, since the old FireWire stack was removed.

       Specify the FireWire port to be used as input file, or "auto" to choose
       the first port connected.

       Options

       dvtype
	   Override autodetection of DV/HDV. This should only be used if auto
	   detection does not work, or if usage of a different device type
	   should be prohibited. Treating a DV device as HDV (or vice versa)
	   will not work and result in undefined behavior.  The values auto,
	   dv and hdv are supported.

       dvbuffer
	   Set maximum size of buffer for incoming data, in frames. For DV,
	   this is an exact value. For HDV, it is not frame exact, since HDV
	   does not have a fixed frame size.

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

       Examples

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

		   ffplay -f iec61883 -i auto

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

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

   jack
       JACK input device.

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

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

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

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

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

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

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

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

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

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

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

       Options

       channels
	   Set the number of channels. Default is 2.

   lavfi
       Libavfilter input virtual device.

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

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

       Options

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

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

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

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

       dumpgraph
	   Dump graph to stderr.

       Examples

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

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

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

		   ffplay -f lavfi color=c=pink

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

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

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

		   ffplay -f lavfi "amovie=test.wav"

       ·   Read an audio stream and a video stream and play it back with
	   ffplay:

		   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

       ·   Dump decoded frames to images and closed captions to a file
	   (experimental):

		   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

   libcdio
       Audio-CD input device based on libcdio.

       To enable this input device during configuration you need libcdio
       installed on your system. It requires the configure option
       "--enable-libcdio".

       This device allows playing and grabbing from an Audio-CD.

       For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you
       may run the command:

	       ffmpeg -f libcdio -i /dev/sr0 cd.wav

       Options

       speed
	   Set drive reading speed. Default value is 0.

	   The speed is specified CD-ROM speed units. The speed is set through
	   the libcdio "cdio_cddap_speed_set" function. On many CD-ROM drives,
	   specifying a value too large will result in using the fastest
	   speed.

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

	   disable
	   verify
	   overlap
	   neverskip
	   full

	   Default value is disable.

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

   libdc1394
       IIDC1394 input device, based on libdc1394 and libraw1394.

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

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

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

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

       An incomplete list of OpenAL implementations follows:

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

       OpenAL Soft
	   Portable, open source (LGPL) software implementation. Includes
	   backends for the most common sound APIs on the Windows, Linux,
	   Solaris, and BSD operating systems.	See
	   <http://kcat.strangesoft.net/openal.html>.

       Apple
	   OpenAL is part of Core Audio, the official Mac OS X Audio
	   interface.  See
	   <http://developer.apple.com/technologies/mac/audio-and-video.html>

       This device allows one to capture from an audio input device handled
       through OpenAL.

       You need to specify the name of the device to capture in the provided
       filename. If the empty string is provided, the device will
       automatically select the default device. You can get the list of the
       supported devices by using the option list_devices.

       Options

       channels
	   Set the number of channels in the captured audio. Only the values 1
	   (monaural) and 2 (stereo) are currently supported.  Defaults to 2.

       sample_size
	   Set the sample size (in bits) of the captured audio. Only the
	   values 8 and 16 are currently supported. Defaults to 16.

       sample_rate
	   Set the sample rate (in Hz) of the captured audio.  Defaults to
	   44.1k.

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

       Examples

       Print the list of OpenAL supported devices and exit:

	       $ ffmpeg -list_devices true -f openal -i dummy out.ogg

       Capture from the OpenAL device DR-BT101 via PulseAudio:

	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

       Capture from the default device (note the empty string '' as filename):

	       $ ffmpeg -f openal -i '' out.ogg

       Capture from two devices simultaneously, writing to two different
       files, within the same ffmpeg command:

	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

       Note: not all OpenAL implementations support multiple simultaneous
       capture - try the latest OpenAL Soft if the above does not work.

   oss
       Open Sound System input device.

       The filename to provide to the input device is the device node
       representing the OSS input device, and is usually set to /dev/dsp.

       For example to grab from /dev/dsp using ffmpeg use the command:

	       ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

       For more information about OSS see:
       <http://manuals.opensound.com/usersguide/dsp.html>

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   pulse
       PulseAudio input device.

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

       The filename to provide to the input device is a source device or the
       string "default"

       To list the PulseAudio source devices and their properties you can
       invoke the command pactl list sources.

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

       Options

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

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

       stream_name
	   Specify the stream name PulseAudio will use when showing active
	   streams, by default it is "record".

       sample_rate
	   Specify the samplerate in Hz, by default 48kHz is used.

       channels
	   Specify the channels in use, by default 2 (stereo) is set.

       frame_size
	   Specify the number of bytes per frame, by default it is set to
	   1024.

       fragment_size
	   Specify the minimal buffering fragment in PulseAudio, it will
	   affect the audio latency. By default it is unset.

       wallclock
	   Set the initial PTS using the current time. Default is 1.

       Examples

       Record a stream from default device:

	       ffmpeg -f pulse -i default /tmp/pulse.wav

   sndio
       sndio input device.

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

       The filename to provide to the input device is the device node
       representing the sndio input device, and is usually set to /dev/audio0.

       For example to grab from /dev/audio0 using ffmpeg use the command:

	       ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   video4linux2, v4l2
       Video4Linux2 input video device.

       "v4l2" can be used as alias for "video4linux2".

       If FFmpeg is built with v4l-utils support (by using the
       "--enable-libv4l2" configure option), it is possible to use it with the
       "-use_libv4l2" input device option.

       The name of the device to grab is a file device node, usually Linux
       systems tend to automatically create such nodes when the device (e.g.
       an USB webcam) is plugged into the system, and has a name of the kind
       /dev/videoN, where N is a number associated to the device.

       Video4Linux2 devices usually support a limited set of widthxheight
       sizes and frame rates. You can check which are supported using
       -list_formats all for Video4Linux2 devices.  Some devices, like TV
       cards, support one or more standards. It is possible to list all the
       supported standards using -list_standards all.

       The time base for the timestamps is 1 microsecond. Depending on the
       kernel version and configuration, the timestamps may be derived from
       the real time clock (origin at the Unix Epoch) or the monotonic clock
       (origin usually at boot time, unaffected by NTP or manual changes to
       the clock). The -timestamps abs or -ts abs option can be used to force
       conversion into the real time clock.

       Some usage examples of the video4linux2 device with ffmpeg and ffplay:

       ·   List supported formats for a video4linux2 device:

		   ffplay -f video4linux2 -list_formats all /dev/video0

       ·   Grab and show the input of a video4linux2 device:

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

       ·   Grab and record the input of a video4linux2 device, leave the frame
	   rate and size as previously set:

		   ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

       For more information about Video4Linux, check <http://linuxtv.org/>.

       Options

       standard
	   Set the standard. Must be the name of a supported standard. To get
	   a list of the supported standards, use the list_standards option.

       channel
	   Set the input channel number. Default to -1, which means using the
	   previously selected channel.

       video_size
	   Set the video frame size. The argument must be a string in the form
	   WIDTHxHEIGHT or a valid size abbreviation.

       pixel_format
	   Select the pixel format (only valid for raw video input).

       input_format
	   Set the preferred pixel format (for raw video) or a codec name.
	   This option allows one to select the input format, when several are
	   available.

       framerate
	   Set the preferred video frame rate.

       list_formats
	   List available formats (supported pixel formats, codecs, and frame
	   sizes) and exit.

	   Available values are:

	   all Show all available (compressed and non-compressed) formats.

	   raw Show only raw video (non-compressed) formats.

	   compressed
	       Show only compressed formats.

       list_standards
	   List supported standards and exit.

	   Available values are:

	   all Show all supported standards.

       timestamps, ts
	   Set type of timestamps for grabbed frames.

	   Available values are:

	   default
	       Use timestamps from the kernel.

	   abs Use absolute timestamps (wall clock).

	   mono2abs
	       Force conversion from monotonic to absolute timestamps.

	   Default value is "default".

       use_libv4l2
	   Use libv4l2 (v4l-utils) conversion functions. Default is 0.

   vfwcap
       VfW (Video for Windows) capture input device.

       The filename passed as input is the capture driver number, ranging from
       0 to 9. You may use "list" as filename to print a list of drivers. Any
       other filename will be interpreted as device number 0.

       Options

       video_size
	   Set the video frame size.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

   x11grab
       X11 video input device.

       To enable this input device during configuration you need libxcb
       installed on your system. It will be automatically detected during
       configuration.

       This device allows one to capture a region of an X11 display.

       The filename passed as input has the syntax:

	       [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

       hostname:display_number.screen_number specifies the X11 display name of
       the screen to grab from. hostname can be omitted, and defaults to
       "localhost". The environment variable DISPLAY contains the default
       display name.

       x_offset and y_offset specify the offsets of the grabbed area with
       respect to the top-left border of the X11 screen. They default to 0.

       Check the X11 documentation (e.g. man X) for more detailed information.

       Use the xdpyinfo program for getting basic information about the
       properties of your X11 display (e.g. grep for "name" or "dimensions").

       For example to grab from :0.0 using ffmpeg:

	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg

       Grab at position "10,20":

	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. A value of 0 specifies
	   not to draw the pointer. Default value is 1.

       follow_mouse
	   Make the grabbed area follow the mouse. The argument can be
	   "centered" or a number of pixels PIXELS.

	   When it is specified with "centered", the grabbing region follows
	   the mouse pointer and keeps the pointer at the center of region;
	   otherwise, the region follows only when the mouse pointer reaches
	   within PIXELS (greater than zero) to the edge of region.

	   For example:

		   ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

	   To follow only when the mouse pointer reaches within 100 pixels to
	   edge:

		   ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

       show_region
	   Show grabbed region on screen.

	   If show_region is specified with 1, then the grabbing region will
	   be indicated on screen. With this option, it is easy to know what
	   is being grabbed if only a portion of the screen is grabbed.

       region_border
	   Set the region border thickness if -show_region 1 is used.  Range
	   is 1 to 128 and default is 3 (XCB-based x11grab only).

	   For example:

		   ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

	   With follow_mouse:

		   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

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

       grab_x
       grab_y
	   Set the grabbing region coordinates. They are expressed as offset
	   from the top left corner of the X11 window and correspond to the
	   x_offset and y_offset parameters in the device name. The default
	   value for both options is 0.

OUTPUT DEVICES
       Output devices are configured elements in FFmpeg that can write
       multimedia data to an output device attached to your system.

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

       You can disable all the output devices using the configure option
       "--disable-outdevs", and selectively enable an output device using the
       option "--enable-outdev=OUTDEV", or you can disable a particular input
       device using the option "--disable-outdev=OUTDEV".

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

       A description of the currently available output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) output device.

       Examples

       ·   Play a file on default ALSA device:

		   ffmpeg -i INPUT -f alsa default

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

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

   caca
       CACA output device.

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

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

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

       Options

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

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

       driver
	   Set display driver.

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

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

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

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

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

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

       Examples

       ·   The following command shows the ffmpeg output is an CACA window,
	   forcing its size to 80x25:

		   ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

       ·   Show the list of available drivers and exit:

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

       ·   Show the list of available dither colors and exit:

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

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

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

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

       Options

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

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

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

       Examples

       ·   List output devices:

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

       ·   List supported formats:

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

       ·   Play video clip:

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

       ·   Play video clip with non-standard framerate or video size:

		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

   libndi_newtek
       The libndi_newtek output device provides playback capabilities for
       using NDI (Network Device Interface, standard created by NewTek).

       Output filename is a NDI name.

       To enable this output device, you need the NDI SDK and you need to
       configure with the appropriate "--extra-cflags" and "--extra-ldflags".

       NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0,
       rgba and rgb0.

       Options

       reference_level
	   The audio reference level in dB. This specifies how many dB above
	   the reference level (+4dBU) is the full range of 16 bit audio.
	   Defaults to 0.

       clock_video
	   These specify whether video "clock" themselves.  Defaults to false.

       clock_audio
	   These specify whether audio "clock" themselves.  Defaults to false.

       Examples

       ·   Play video clip:

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

   fbdev
       Linux framebuffer output device.

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

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

       Options

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

       Examples

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

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

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

   opengl
       OpenGL output device.

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

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

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

       Options

       background
	   Set background color. Black is a default.

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

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

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

       Examples

       Play a file on SDL window using OpenGL rendering:

	       ffmpeg  -i INPUT -f opengl "window title"

   oss
       OSS (Open Sound System) output device.

   pulse
       PulseAudio output device.

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

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

       Options

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

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

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

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

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

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

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

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

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

       Examples

       Play a file on default device on default server:

	       ffmpeg  -i INPUT -f pulse "stream name"

   sdl
       SDL (Simple DirectMedia Layer) output device.

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

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

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

       Options

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

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

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

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

       Interactive commands

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

       q, ESC
	   Quit the device immediately.

       Examples

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

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

   sndio
       sndio audio output device.

   xv
       XV (XVideo) output device.

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

       Options

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

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

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

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

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

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

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

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

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

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

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

       Examples

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

		   ffmpeg -i INPUT OUTPUT -f xv display

       ·   Decode and display the input video to multiple X11 windows:

		   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
       The audio resampler supports the following named options.

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

       ich, in_channel_count
	   Set the number of input channels. Default value is 0. Setting this
	   value is not mandatory if the corresponding channel layout
	   in_channel_layout is set.

       och, out_channel_count
	   Set the number of output channels. Default value is 0. Setting this
	   value is not mandatory if the corresponding channel layout
	   out_channel_layout is set.

       uch, used_channel_count
	   Set the number of used input channels. Default value is 0. This
	   option is only used for special remapping.

       isr, in_sample_rate
	   Set the input sample rate. Default value is 0.

       osr, out_sample_rate
	   Set the output sample rate. Default value is 0.

       isf, in_sample_fmt
	   Specify the input sample format. It is set by default to "none".

       osf, out_sample_fmt
	   Specify the output sample format. It is set by default to "none".

       tsf, internal_sample_fmt
	   Set the internal sample format. Default value is "none".  This will
	   automatically be chosen when it is not explicitly set.

       icl, in_channel_layout
       ocl, out_channel_layout
	   Set the input/output channel layout.

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

       clev, center_mix_level
	   Set the center mix level. It is a value expressed in deciBel, and
	   must be in the interval [-32,32].

       slev, surround_mix_level
	   Set the surround mix level. It is a value expressed in deciBel, and
	   must be in the interval [-32,32].

       lfe_mix_level
	   Set LFE mix into non LFE level. It is used when there is a LFE
	   input but no LFE output. It is a value expressed in deciBel, and
	   must be in the interval [-32,32].

       rmvol, rematrix_volume
	   Set rematrix volume. Default value is 1.0.

       rematrix_maxval
	   Set maximum output value for rematrixing.  This can be used to
	   prevent clipping vs. preventing volume reduction.  A value of 1.0
	   prevents clipping.

       flags, swr_flags
	   Set flags used by the converter. Default value is 0.

	   It supports the following individual flags:

	   res force resampling, this flag forces resampling to be used even
	       when the input and output sample rates match.

       dither_scale
	   Set the dither scale. Default value is 1.

       dither_method
	   Set dither method. Default value is 0.

	   Supported values:

	   rectangular
	       select rectangular dither

	   triangular
	       select triangular dither

	   triangular_hp
	       select triangular dither with high pass

	   lipshitz
	       select Lipshitz noise shaping dither.

	   shibata
	       select Shibata noise shaping dither.

	   low_shibata
	       select low Shibata noise shaping dither.

	   high_shibata
	       select high Shibata noise shaping dither.

	   f_weighted
	       select f-weighted noise shaping dither

	   modified_e_weighted
	       select modified-e-weighted noise shaping dither

	   improved_e_weighted
	       select improved-e-weighted noise shaping dither

       resampler
	   Set resampling engine. Default value is swr.

	   Supported values:

	   swr select the native SW Resampler; filter options precision and
	       cheby are not applicable in this case.

	   soxr
	       select the SoX Resampler (where available); compensation, and
	       filter options filter_size, phase_shift, exact_rational,
	       filter_type & kaiser_beta, are not applicable in this case.

       filter_size
	   For swr only, set resampling filter size, default value is 32.

       phase_shift
	   For swr only, set resampling phase shift, default value is 10, and
	   must be in the interval [0,30].

       linear_interp
	   Use linear interpolation when enabled (the default). Disable it if
	   you want to preserve speed instead of quality when exact_rational
	   fails.

       exact_rational
	   For swr only, when enabled, try to use exact phase_count based on
	   input and output sample rate. However, if it is larger than "1 <<
	   phase_shift", the phase_count will be "1 << phase_shift" as
	   fallback. Default is enabled.

       cutoff
	   Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must
	   be a float value between 0 and 1.  Default value is 0.97 with swr,
	   and 0.91 with soxr (which, with a sample-rate of 44100, preserves
	   the entire audio band to 20kHz).

       precision
	   For soxr only, the precision in bits to which the resampled signal
	   will be calculated.	The default value of 20 (which, with suitable
	   dithering, is appropriate for a destination bit-depth of 16) gives
	   SoX's 'High Quality'; a value of 28 gives SoX's 'Very High
	   Quality'.

       cheby
	   For soxr only, selects passband rolloff none (Chebyshev) & higher-
	   precision approximation for 'irrational' ratios. Default value is
	   0.

       async
	   For swr only, simple 1 parameter audio sync to timestamps using
	   stretching, squeezing, filling and trimming. Setting this to 1 will
	   enable filling and trimming, larger values represent the maximum
	   amount in samples that the data may be stretched or squeezed for
	   each second.  Default value is 0, thus no compensation is applied
	   to make the samples match the audio timestamps.

       first_pts
	   For swr only, assume the first pts should be this value. The time
	   unit is 1 / sample rate.  This allows for padding/trimming at the
	   start of stream. By default, no assumption is made about the first
	   frame's expected pts, so no padding or trimming is done. For
	   example, this could be set to 0 to pad the beginning with silence
	   if an audio stream starts after the video stream or to trim any
	   samples with a negative pts due to encoder delay.

       min_comp
	   For swr only, set the minimum difference between timestamps and
	   audio data (in seconds) to trigger stretching/squeezing/filling or
	   trimming of the data to make it match the timestamps. The default
	   is that stretching/squeezing/filling and trimming is disabled
	   (min_comp = "FLT_MAX").

       min_hard_comp
	   For swr only, set the minimum difference between timestamps and
	   audio data (in seconds) to trigger adding/dropping samples to make
	   it match the timestamps.  This option effectively is a threshold to
	   select between hard (trim/fill) and soft (squeeze/stretch)
	   compensation. Note that all compensation is by default disabled
	   through min_comp.  The default is 0.1.

       comp_duration
	   For swr only, set duration (in seconds) over which data is
	   stretched/squeezed to make it match the timestamps. Must be a non-
	   negative double float value, default value is 1.0.

       max_soft_comp
	   For swr only, set maximum factor by which data is
	   stretched/squeezed to make it match the timestamps. Must be a non-
	   negative double float value, default value is 0.

       matrix_encoding
	   Select matrixed stereo encoding.

	   It accepts the following values:

	   none
	       select none

	   dolby
	       select Dolby

	   dplii
	       select Dolby Pro Logic II

	   Default value is "none".

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

	   It accepts the following values:

	   cubic
	       select cubic

	   blackman_nuttall
	       select Blackman Nuttall windowed sinc

	   kaiser
	       select Kaiser windowed sinc

       kaiser_beta
	   For swr only, set Kaiser window beta value. Must be a double float
	   value in the interval [2,16], default value is 9.

       output_sample_bits
	   For swr only, set number of used output sample bits for dithering.
	   Must be an integer in the interval [0,64], default value is 0,
	   which means it's not used.

SCALER OPTIONS
       The video scaler supports the following named options.

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

       sws_flags
	   Set the scaler flags. This is also used to set the scaling
	   algorithm. Only a single algorithm should be selected. Default
	   value is bicubic.

	   It accepts the following values:

	   fast_bilinear
	       Select fast bilinear scaling algorithm.

	   bilinear
	       Select bilinear scaling algorithm.

	   bicubic
	       Select bicubic scaling algorithm.

	   experimental
	       Select experimental scaling algorithm.

	   neighbor
	       Select nearest neighbor rescaling algorithm.

	   area
	       Select averaging area rescaling algorithm.

	   bicublin
	       Select bicubic scaling algorithm for the luma component,
	       bilinear for chroma components.

	   gauss
	       Select Gaussian rescaling algorithm.

	   sinc
	       Select sinc rescaling algorithm.

	   lanczos
	       Select Lanczos rescaling algorithm.

	   spline
	       Select natural bicubic spline rescaling algorithm.

	   print_info
	       Enable printing/debug logging.

	   accurate_rnd
	       Enable accurate rounding.

	   full_chroma_int
	       Enable full chroma interpolation.

	   full_chroma_inp
	       Select full chroma input.

	   bitexact
	       Enable bitexact output.

       srcw
	   Set source width.

       srch
	   Set source height.

       dstw
	   Set destination width.

       dsth
	   Set destination height.

       src_format
	   Set source pixel format (must be expressed as an integer).

       dst_format
	   Set destination pixel format (must be expressed as an integer).

       src_range
	   Select source range.

       dst_range
	   Select destination range.

       param0, param1
	   Set scaling algorithm parameters. The specified values are specific
	   of some scaling algorithms and ignored by others. The specified
	   values are floating point number values.

       sws_dither
	   Set the dithering algorithm. Accepts one of the following values.
	   Default value is auto.

	   auto
	       automatic choice

	   none
	       no dithering

	   bayer
	       bayer dither

	   ed  error diffusion dither

	   a_dither
	       arithmetic dither, based using addition

	   x_dither
	       arithmetic dither, based using xor (more random/less apparent
	       patterning that a_dither).

       alphablend
	   Set the alpha blending to use when the input has alpha but the
	   output does not.  Default value is none.

	   uniform_color
	       Blend onto a uniform background color

	   checkerboard
	       Blend onto a checkerboard

	   none
	       No blending

FILTERING INTRODUCTION
       Filtering in FFmpeg is enabled through the libavfilter library.

       In libavfilter, a filter can have multiple inputs and multiple outputs.
       To illustrate the sorts of things that are possible, we consider the
       following filtergraph.

			       [main]
	       input --> split ---------------------> overlay --> output
			   |				 ^
			   |[tmp]		   [flip]|
			   +-----> crop --> vflip -------+

       This filtergraph splits the input stream in two streams, then sends one
       stream through the crop filter and the vflip filter, before merging it
       back with the other stream by overlaying it on top. You can use the
       following command to achieve this:

	       ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

       The result will be that the top half of the video is mirrored onto the
       bottom half of the output video.

       Filters in the same linear chain are separated by commas, and distinct
       linear chains of filters are separated by semicolons. In our example,
       crop,vflip are in one linear chain, split and overlay are separately in
       another. The points where the linear chains join are labelled by names
       enclosed in square brackets. In the example, the split filter generates
       two outputs that are associated to the labels [main] and [tmp].

       The stream sent to the second output of split, labelled as [tmp], is
       processed through the crop filter, which crops away the lower half part
       of the video, and then vertically flipped. The overlay filter takes in
       input the first unchanged output of the split filter (which was
       labelled as [main]), and overlay on its lower half the output generated
       by the crop,vflip filterchain.

       Some filters take in input a list of parameters: they are specified
       after the filter name and an equal sign, and are separated from each
       other by a colon.

       There exist so-called source filters that do not have an audio/video
       input, and sink filters that will not have audio/video output.

GRAPH
       The graph2dot program included in the FFmpeg tools directory can be
       used to parse a filtergraph description and issue a corresponding
       textual representation in the dot language.

       Invoke the command:

	       graph2dot -h

       to see how to use graph2dot.

       You can then pass the dot description to the dot program (from the
       graphviz suite of programs) and obtain a graphical representation of
       the filtergraph.

       For example the sequence of commands:

	       echo <GRAPH_DESCRIPTION> | \
	       tools/graph2dot -o graph.tmp && \
	       dot -Tpng graph.tmp -o graph.png && \
	       display graph.png

       can be used to create and display an image representing the graph
       described by the GRAPH_DESCRIPTION string. Note that this string must
       be a complete self-contained graph, with its inputs and outputs
       explicitly defined.  For example if your command line is of the form:

	       ffmpeg -i infile -vf scale=640:360 outfile

       your GRAPH_DESCRIPTION string will need to be of the form:

	       nullsrc,scale=640:360,nullsink

       you may also need to set the nullsrc parameters and add a format filter
       in order to simulate a specific input file.

FILTERGRAPH DESCRIPTION
       A filtergraph is a directed graph of connected filters. It can contain
       cycles, and there can be multiple links between a pair of filters. Each
       link has one input pad on one side connecting it to one filter from
       which it takes its input, and one output pad on the other side
       connecting it to one filter accepting its output.

       Each filter in a filtergraph is an instance of a filter class
       registered in the application, which defines the features and the
       number of input and output pads of the filter.

       A filter with no input pads is called a "source", and a filter with no
       output pads is called a "sink".

   Filtergraph syntax
       A filtergraph has a textual representation, which is recognized by the
       -filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in
       ffplay, and by the "avfilter_graph_parse_ptr()" function defined in
       libavfilter/avfilter.h.

       A filterchain consists of a sequence of connected filters, each one
       connected to the previous one in the sequence. A filterchain is
       represented by a list of ","-separated filter descriptions.

       A filtergraph consists of a sequence of filterchains. A sequence of
       filterchains is represented by a list of ";"-separated filterchain
       descriptions.

       A filter is represented by a string of the form:
       [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]

       filter_name is the name of the filter class of which the described
       filter is an instance of, and has to be the name of one of the filter
       classes registered in the program optionally followed by "@id".	The
       name of the filter class is optionally followed by a string
       "=arguments".

       arguments is a string which contains the parameters used to initialize
       the filter instance. It may have one of two forms:

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

       ·   A ':'-separated list of value. In this case, the keys are assumed
	   to be the option names in the order they are declared. E.g. the
	   "fade" filter declares three options in this order -- type,
	   start_frame and nb_frames. Then the parameter list in:0:30 means
	   that the value in is assigned to the option type, 0 to start_frame
	   and 30 to nb_frames.

       ·   A ':'-separated list of mixed direct value and long key=value
	   pairs. The direct value must precede the key=value pairs, and
	   follow the same constraints order of the previous point. The
	   following key=value pairs can be set in any preferred order.

       If the option value itself is a list of items (e.g. the "format" filter
       takes a list of pixel formats), the items in the list are usually
       separated by |.

       The list of arguments can be quoted using the character ' as initial
       and ending mark, and the character \ for escaping the characters within
       the quoted text; otherwise the argument string is considered terminated
       when the next special character (belonging to the set []=;,) is
       encountered.

       The name and arguments of the filter are optionally preceded and
       followed by a list of link labels.  A link label allows one to name a
       link and associate it to a filter output or input pad. The preceding
       labels in_link_1 ... in_link_N, are associated to the filter input
       pads, the following labels out_link_1 ... out_link_M, are associated to
       the output pads.

       When two link labels with the same name are found in the filtergraph, a
       link between the corresponding input and output pad is created.

       If an output pad is not labelled, it is linked by default to the first
       unlabelled input pad of the next filter in the filterchain.  For
       example in the filterchain

	       nullsrc, split[L1], [L2]overlay, nullsink

       the split filter instance has two output pads, and the overlay filter
       instance two input pads. The first output pad of split is labelled
       "L1", the first input pad of overlay is labelled "L2", and the second
       output pad of split is linked to the second input pad of overlay, which
       are both unlabelled.

       In a filter description, if the input label of the first filter is not
       specified, "in" is assumed; if the output label of the last filter is
       not specified, "out" is assumed.

       In a complete filterchain all the unlabelled filter input and output
       pads must be connected. A filtergraph is considered valid if all the
       filter input and output pads of all the filterchains are connected.

       Libavfilter will automatically insert scale filters where format
       conversion is required. It is possible to specify swscale flags for
       those automatically inserted scalers by prepending "sws_flags=flags;"
       to the filtergraph description.

       Here is a BNF description of the filtergraph syntax:

	       <NAME>		  ::= sequence of alphanumeric characters and '_'
	       <FILTER_NAME>	  ::= <NAME>["@"<NAME>]
	       <LINKLABEL>	  ::= "[" <NAME> "]"
	       <LINKLABELS>	  ::= <LINKLABEL> [<LINKLABELS>]
	       <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
	       <FILTER> 	  ::= [<LINKLABELS>] <FILTER_NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
	       <FILTERCHAIN>	  ::= <FILTER> [,<FILTERCHAIN>]
	       <FILTERGRAPH>	  ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

   Notes on filtergraph escaping
       Filtergraph description composition entails several levels of escaping.
       See the "Quoting and escaping" section in the ffmpeg-utils(1) manual
       for more information about the employed escaping procedure.

       A first level escaping affects the content of each filter option value,
       which may contain the special character ":" used to separate values, or
       one of the escaping characters "\'".

       A second level escaping affects the whole filter description, which may
       contain the escaping characters "\'" or the special characters "[],;"
       used by the filtergraph description.

       Finally, when you specify a filtergraph on a shell commandline, you
       need to perform a third level escaping for the shell special characters
       contained within it.

       For example, consider the following string to be embedded in the
       drawtext filter description text value:

	       this is a 'string': may contain one, or more, special characters

       This string contains the "'" special escaping character, and the ":"
       special character, so it needs to be escaped in this way:

	       text=this is a \'string\'\: may contain one, or more, special characters

       A second level of escaping is required when embedding the filter
       description in a filtergraph description, in order to escape all the
       filtergraph special characters. Thus the example above becomes:

	       drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

       (note that in addition to the "\'" escaping special characters, also
       "," needs to be escaped).

       Finally an additional level of escaping is needed when writing the
       filtergraph description in a shell command, which depends on the
       escaping rules of the adopted shell. For example, assuming that "\" is
       special and needs to be escaped with another "\", the previous string
       will finally result in:

	       -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING
       Some filters support a generic enable option. For the filters
       supporting timeline editing, this option can be set to an expression
       which is evaluated before sending a frame to the filter. If the
       evaluation is non-zero, the filter will be enabled, otherwise the frame
       will be sent unchanged to the next filter in the filtergraph.

       The expression accepts the following values:

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

       n   sequential number of the input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       w
       h   width and height of the input frame if video

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

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

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

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

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

OPTIONS FOR FILTERS WITH SEVERAL INPUTS
       Some filters with several inputs support a common set of options.
       These options can only be set by name, not with the short notation.

       eof_action
	   The action to take when EOF is encountered on the secondary input;
	   it accepts one of the following values:

	   repeat
	       Repeat the last frame (the default).

	   endall
	       End both streams.

	   pass
	       Pass the main input through.

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

       repeatlast
	   If set to 1, force the filter to extend the last frame of secondary
	   streams until the end of the primary stream. A value of 0 disables
	   this behavior.  Default value is 1.

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

       Below is a description of the currently available audio filters.

   acompressor
       A compressor is mainly used to reduce the dynamic range of a signal.
       Especially modern music is mostly compressed at a high ratio to improve
       the overall loudness. It's done to get the highest attention of a
       listener, "fatten" the sound and bring more "power" to the track.  If a
       signal is compressed too much it may sound dull or "dead" afterwards or
       it may start to "pump" (which could be a powerful effect but can also
       destroy a track completely).  The right compression is the key to reach
       a professional sound and is the high art of mixing and mastering.
       Because of its complex settings it may take a long time to get the
       right feeling for this kind of effect.

       Compression is done by detecting the volume above a chosen level
       "threshold" and dividing it by the factor set with "ratio".  So if you
       set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1
       will result in a signal at -9dB. Because an exact manipulation of the
       signal would cause distortion of the waveform the reduction can be
       levelled over the time. This is done by setting "Attack" and "Release".
       "attack" determines how long the signal has to rise above the threshold
       before any reduction will occur and "release" sets the time the signal
       has to fall below the threshold to reduce the reduction again. Shorter
       signals than the chosen attack time will be left untouched.  The
       overall reduction of the signal can be made up afterwards with the
       "makeup" setting. So compressing the peaks of a signal about 6dB and
       raising the makeup to this level results in a signal twice as loud than
       the source. To gain a softer entry in the compression the "knee"
       flattens the hard edge at the threshold in the range of the chosen
       decibels.

       The filter accepts the following options:

       level_in
	   Set input gain. Default is 1. Range is between 0.015625 and 64.

       threshold
	   If a signal of stream rises above this level it will affect the
	   gain reduction.  By default it is 0.125. Range is between
	   0.00097563 and 1.

       ratio
	   Set a ratio by which the signal is reduced. 1:2 means that if the
	   level rose 4dB above the threshold, it will be only 2dB above after
	   the reduction.  Default is 2. Range is between 1 and 20.

       attack
	   Amount of milliseconds the signal has to rise above the threshold
	   before gain reduction starts. Default is 20. Range is between 0.01
	   and 2000.

       release
	   Amount of milliseconds the signal has to fall below the threshold
	   before reduction is decreased again. Default is 250. Range is
	   between 0.01 and 9000.

       makeup
	   Set the amount by how much signal will be amplified after
	   processing.	Default is 1. Range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter gain reduction
	   more softly.  Default is 2.82843. Range is between 1 and 8.

       link
	   Choose if the "average" level between all channels of input stream
	   or the louder("maximum") channel of input stream affects the
	   reduction. Default is "average".

       detection
	   Should the exact signal be taken in case of "peak" or an RMS one in
	   case of "rms". Default is "rms" which is mostly smoother.

       mix How much to use compressed signal in output. Default is 1.  Range
	   is between 0 and 1.

   acopy
       Copy the input audio source unchanged to the output. This is mainly
       useful for testing purposes.

   acrossfade
       Apply cross fade from one input audio stream to another input audio
       stream.	The cross fade is applied for specified duration near the end
       of first stream.

       The filter accepts the following options:

       nb_samples, ns
	   Specify the number of samples for which the cross fade effect has
	   to last.  At the end of the cross fade effect the first input audio
	   will be completely silent. Default is 44100.

       duration, d
	   Specify the duration of the cross fade effect. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  By default the duration is determined by nb_samples.  If
	   set this option is used instead of nb_samples.

       overlap, o
	   Should first stream end overlap with second stream start. Default
	   is enabled.

       curve1
	   Set curve for cross fade transition for first stream.

       curve2
	   Set curve for cross fade transition for second stream.

	   For description of available curve types see afade filter
	   description.

       Examples

       ·   Cross fade from one input to another:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

       ·   Cross fade from one input to another but without overlapping:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

   acrusher
       Reduce audio bit resolution.

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

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

       The filter accepts the following options:

       level_in
	   Set level in.

       level_out
	   Set level out.

       bits
	   Set bit reduction.

       mix Set mixing amount.

       mode
	   Can be linear: "lin" or logarithmic: "log".

       dc  Set DC.

       aa  Set anti-aliasing.

       samples
	   Set sample reduction.

       lfo Enable LFO. By default disabled.

       lforange
	   Set LFO range.

       lforate
	   Set LFO rate.

   adelay
       Delay one or more audio channels.

       Samples in delayed channel are filled with silence.

       The filter accepts the following option:

       delays
	   Set list of delays in milliseconds for each channel separated by
	   '|'.  Unused delays will be silently ignored. If number of given
	   delays is smaller than number of channels all remaining channels
	   will not be delayed.  If you want to delay exact number of samples,
	   append 'S' to number.

       Examples

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

		   adelay=1500|0|500

       ·   Delay second channel by 500 samples, the third channel by 700
	   samples and leave the first channel (and any other channels that
	   may be present) unchanged.

		   adelay=0|500S|700S

   aecho
       Apply echoing to the input audio.

       Echoes are reflected sound and can occur naturally amongst mountains
       (and sometimes large buildings) when talking or shouting; digital echo
       effects emulate this behaviour and are often used to help fill out the
       sound of a single instrument or vocal. The time difference between the
       original signal and the reflection is the "delay", and the loudness of
       the reflected signal is the "decay".  Multiple echoes can have
       different delays and decays.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain of reflected signal. Default is 0.6.

       out_gain
	   Set output gain of reflected signal. Default is 0.3.

       delays
	   Set list of time intervals in milliseconds between original signal
	   and reflections separated by '|'. Allowed range for each "delay" is
	   "(0 - 90000.0]".  Default is 1000.

       decays
	   Set list of loudness of reflected signals separated by '|'.
	   Allowed range for each "decay" is "(0 - 1.0]".  Default is 0.5.

       Examples

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

		   aecho=0.8:0.88:60:0.4

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

		   aecho=0.8:0.88:6:0.4

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

		   aecho=0.8:0.9:1000:0.3

       ·   Same as above but with one more mountain:

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

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

       The filter accepts the following options:

       level_in
	   Set input gain.

       level_out
	   Set output gain.

       mode
	   Set filter mode. For restoring material use "reproduction" mode,
	   otherwise use "production" mode. Default is "reproduction" mode.

       type
	   Set filter type. Selects medium. Can be one of the following:

	   col select Columbia.

	   emi select EMI.

	   bsi select BSI (78RPM).

	   riaa
	       select RIAA.

	   cd  select Compact Disc (CD).

	   50fm
	       select 50Xs (FM).

	   75fm
	       select 75Xs (FM).

	   50kf
	       select 50Xs (FM-KF).

	   75kf
	       select 75Xs (FM-KF).

   aeval
       Modify an audio signal according to the specified expressions.

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

       It accepts the following parameters:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   If the number of input channels is greater than the number of
	   expressions, the last specified expression is used for the
	   remaining output channels.

       channel_layout, c
	   Set output channel layout. If not specified, the channel layout is
	   specified by the number of expressions. If set to same, it will use
	   by default the same input channel layout.

       Each expression in exprs can contain the following constants and
       functions:

       ch  channel number of the current expression

       n   number of the evaluated sample, starting from 0

       s   sample rate

       t   time of the evaluated sample expressed in seconds

       nb_in_channels
       nb_out_channels
	   input and output number of channels

       val(CH)
	   the value of input channel with number CH

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

       Examples

       ·   Half volume:

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

       ·   Invert phase of the second channel:

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

   afade
       Apply fade-in/out effect to input audio.

       A description of the accepted parameters follows.

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

       start_sample, ss
	   Specify the number of the start sample for starting to apply the
	   fade effect. Default is 0.

       nb_samples, ns
	   Specify the number of samples for which the fade effect has to
	   last. At the end of the fade-in effect the output audio will have
	   the same volume as the input audio, at the end of the fade-out
	   transition the output audio will be silence. Default is 44100.

       start_time, st
	   Specify the start time of the fade effect. Default is 0.  The value
	   must be specified as a time duration; see the Time duration section
	   in the ffmpeg-utils(1) manual for the accepted syntax.  If set this
	   option is used instead of start_sample.

       duration, d
	   Specify the duration of the fade effect. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.  At
	   the end of the fade-in effect the output audio will have the same
	   volume as the input audio, at the end of the fade-out transition
	   the output audio will be silence.  By default the duration is
	   determined by nb_samples.  If set this option is used instead of
	   nb_samples.

       curve
	   Set curve for fade transition.

	   It accepts the following values:

	   tri select triangular, linear slope (default)

	   qsin
	       select quarter of sine wave

	   hsin
	       select half of sine wave

	   esin
	       select exponential sine wave

	   log select logarithmic

	   ipar
	       select inverted parabola

	   qua select quadratic

	   cub select cubic

	   squ select square root

	   cbr select cubic root

	   par select parabola

	   exp select exponential

	   iqsin
	       select inverted quarter of sine wave

	   ihsin
	       select inverted half of sine wave

	   dese
	       select double-exponential seat

	   desi
	       select double-exponential sigmoid

       Examples

       ·   Fade in first 15 seconds of audio:

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

       ·   Fade out last 25 seconds of a 900 seconds audio:

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

   afftfilt
       Apply arbitrary expressions to samples in frequency domain.

       real
	   Set frequency domain real expression for each separate channel
	   separated by '|'. Default is "1".  If the number of input channels
	   is greater than the number of expressions, the last specified
	   expression is used for the remaining output channels.

       imag
	   Set frequency domain imaginary expression for each separate channel
	   separated by '|'. If not set, real option is used.

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

	   sr  sample rate

	   b   current frequency bin number

	   nb  number of available bins

	   ch  channel number of the current expression

	   chs number of channels

	   pts current frame pts

       win_size
	   Set window size.

	   It accepts the following values:

	   w16
	   w32
	   w64
	   w128
	   w256
	   w512
	   w1024
	   w2048
	   w4096
	   w8192
	   w16384
	   w32768
	   w65536

	   Default is "w4096"

       win_func
	   Set window function. Default is "hann".

       overlap
	   Set window overlap. If set to 1, the recommended overlap for
	   selected window function will be picked. Default is 0.75.

       Examples

       ·   Leave almost only low frequencies in audio:

		   afftfilt="1-clip((b/nb)*b,0,1)"

   afir
       Apply an arbitrary Frequency Impulse Response filter.

       This filter is designed for applying long FIR filters, up to 30 seconds
       long.

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

       This filter uses second stream as FIR coefficients.  If second stream
       holds single channel, it will be used for all input channels in first
       stream, otherwise number of channels in second stream must be same as
       number of channels in first stream.

       It accepts the following parameters:

       dry Set dry gain. This sets input gain.

       wet Set wet gain. This sets final output gain.

       length
	   Set Impulse Response filter length. Default is 1, which means whole
	   IR is processed.

       again
	   Enable applying gain measured from power of IR.

       Examples

       ·   Apply reverb to stream using mono IR file as second input, complete
	   command using ffmpeg:

		   ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

   aformat
       Set output format constraints for the input audio. The framework will
       negotiate the most appropriate format to minimize conversions.

       It accepts the following parameters:

       sample_fmts
	   A '|'-separated list of requested sample formats.

       sample_rates
	   A '|'-separated list of requested sample rates.

       channel_layouts
	   A '|'-separated list of requested channel layouts.

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

       If a parameter is omitted, all values are allowed.

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

	       aformat=sample_fmts=u8|s16:channel_layouts=stereo

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

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

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

       level_in
	   Set input level before filtering.  Default is 1. Allowed range is
	   from 0.015625 to 64.

       range
	   Set the level of gain reduction when the signal is below the
	   threshold.  Default is 0.06125. Allowed range is from 0 to 1.

       threshold
	   If a signal rises above this level the gain reduction is released.
	   Default is 0.125. Allowed range is from 0 to 1.

       ratio
	   Set a ratio by which the signal is reduced.	Default is 2. Allowed
	   range is from 1 to 9000.

       attack
	   Amount of milliseconds the signal has to rise above the threshold
	   before gain reduction stops.  Default is 20 milliseconds. Allowed
	   range is from 0.01 to 9000.

       release
	   Amount of milliseconds the signal has to fall below the threshold
	   before the reduction is increased again. Default is 250
	   milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
	   Set amount of amplification of signal after processing.  Default is
	   1. Allowed range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter gain reduction
	   more softly.  Default is 2.828427125. Allowed range is from 1 to 8.

       detection
	   Choose if exact signal should be taken for detection or an RMS like
	   one.  Default is "rms". Can be "peak" or "rms".

       link
	   Choose if the average level between all channels or the louder
	   channel affects the reduction.  Default is "average". Can be
	   "average" or "maximum".

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

       The filter accepts the following options:

       level_in
	   Set input gain. Default is 1.

       level_out
	   Set output gain. Default is 1.

       limit
	   Don't let signals above this level pass the limiter. Default is 1.

       attack
	   The limiter will reach its attenuation level in this amount of time
	   in milliseconds. Default is 5 milliseconds.

       release
	   Come back from limiting to attenuation 1.0 in this amount of
	   milliseconds.  Default is 50 milliseconds.

       asc When gain reduction is always needed ASC takes care of releasing to
	   an average reduction level rather than reaching a reduction of 0 in
	   the release time.

       asc_level
	   Select how much the release time is affected by ASC, 0 means nearly
	   no changes in release time while 1 produces higher release times.

       level
	   Auto level output signal. Default is enabled.  This normalizes
	   audio back to 0dB if enabled.

       Depending on picked setting it is recommended to upsample input 2x or
       4x times with aresample before applying this filter.

   allpass
       Apply a two-pole all-pass filter with central frequency (in Hz)
       frequency, and filter-width width.  An all-pass filter changes the
       audio's frequency to phase relationship without changing its frequency
       to amplitude relationship.

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

   aloop
       Loop audio samples.

       The filter accepts the following options:

       loop
	   Set the number of loops.

       size
	   Set maximal number of samples.

       start
	   Set first sample of loop.

   amerge
       Merge two or more audio streams into a single multi-channel stream.

       The filter accepts the following options:

       inputs
	   Set the number of inputs. Default is 2.

       If the channel layouts of the inputs are disjoint, and therefore
       compatible, the channel layout of the output will be set accordingly
       and the channels will be reordered as necessary. If the channel layouts
       of the inputs are not disjoint, the output will have all the channels
       of the first input then all the channels of the second input, in that
       order, and the channel layout of the output will be the default value
       corresponding to the total number of channels.

       For example, if the first input is in 2.1 (FL+FR+LF) and the second
       input is FC+BL+BR, then the output will be in 5.1, with the channels in
       the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of
       the first input, b1 is the first channel of the second input).

       On the other hand, if both input are in stereo, the output channels
       will be in the default order: a1, a2, b1, b2, and the channel layout
       will be arbitrarily set to 4.0, which may or may not be the expected
       value.

       All inputs must have the same sample rate, and format.

       If inputs do not have the same duration, the output will stop with the
       shortest.

       Examples

       ·   Merge two mono files into a stereo stream:

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

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

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

   amix
       Mixes multiple audio inputs into a single output.

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

       For example

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

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

       It accepts the following parameters:

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

       duration
	   How to determine the end-of-stream.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

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

   anequalizer
       High-order parametric multiband equalizer for each channel.

       It accepts the following parameters:

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

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

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

	   w   Set band width in hertz.

	   g   Set band gain in dB.

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

	       0   Butterworth, this is default.

	       1   Chebyshev type 1.

	       2   Chebyshev type 2.

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

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

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

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

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

       Examples

       ·   Lower gain by 10 of central frequency 200Hz and width 100 Hz for
	   first 2 channels using Chebyshev type 1 filter:

		   anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

       Commands

       This filter supports the following commands:

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

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

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

   anull
       Pass the audio source unchanged to the output.

   apad
       Pad the end of an audio stream with silence.

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

       A description of the accepted options follows.

       packet_size
	   Set silence packet size. Default value is 4096.

       pad_len
	   Set the number of samples of silence to add to the end. After the
	   value is reached, the stream is terminated. This option is mutually
	   exclusive with whole_len.

       whole_len
	   Set the minimum total number of samples in the output audio stream.
	   If the value is longer than the input audio length, silence is
	   added to the end, until the value is reached. This option is
	   mutually exclusive with pad_len.

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

       Examples

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

		   apad=pad_len=1024

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

		   apad=whole_len=10000

       ·   Use ffmpeg to pad the audio input with silence, so that the video
	   stream will always result the shortest and will be converted until
	   the end in the output file when using the shortest option:

		   ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

   aphaser
       Add a phasing effect to the input audio.

       A phaser filter creates series of peaks and troughs in the frequency
       spectrum.  The position of the peaks and troughs are modulated so that
       they vary over time, creating a sweeping effect.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.74

       delay
	   Set delay in milliseconds. Default is 3.0.

       decay
	   Set decay. Default is 0.4.

       speed
	   Set modulation speed in Hz. Default is 0.5.

       type
	   Set modulation type. Default is triangular.

	   It accepts the following values:

	   triangular, t
	   sinusoidal, s

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

       The filter accepts the following options:

       level_in
	   Set input gain. By default it is 1. Range is [0.015625 - 64].

       level_out
	   Set output gain. By default it is 1. Range is [0.015625 - 64].

       mode
	   Set waveform shape the LFO will use. Can be one of: sine, triangle,
	   square, sawup or sawdown. Default is sine.

       amount
	   Set modulation. Define how much of original signal is affected by
	   the LFO.

       offset_l
	   Set left channel offset. Default is 0. Allowed range is [0 - 1].

       offset_r
	   Set right channel offset. Default is 0.5. Allowed range is [0 - 1].

       width
	   Set pulse width. Default is 1. Allowed range is [0 - 2].

       timing
	   Set possible timing mode. Can be one of: bpm, ms or hz. Default is
	   hz.

       bpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if
	   timing is set to bpm.

       ms  Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if
	   timing is set to ms.

       hz  Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100].
	   Only used if timing is set to hz.

   aresample
       Resample the input audio to the specified parameters, using the
       libswresample library. If none are specified then the filter will
       automatically convert between its input and output.

       This filter is also able to stretch/squeeze the audio data to make it
       match the timestamps or to inject silence / cut out audio to make it
       match the timestamps, do a combination of both or do neither.

       The filter accepts the syntax [sample_rate:]resampler_options, where
       sample_rate expresses a sample rate and resampler_options is a list of
       key=value pairs, separated by ":". See the the "Resampler Options"
       section in the ffmpeg-resampler(1) manual for the complete list of
       supported options.

       Examples

       ·   Resample the input audio to 44100Hz:

		   aresample=44100

       ·   Stretch/squeeze samples to the given timestamps, with a maximum of
	   1000 samples per second compensation:

		   aresample=async=1000

   areverse
       Reverse an audio clip.

       Warning: This filter requires memory to buffer the entire clip, so
       trimming is suggested.

       Examples

       ·   Take the first 5 seconds of a clip, and reverse it.

		   atrim=end=5,areverse

   asetnsamples
       Set the number of samples per each output audio frame.

       The last output packet may contain a different number of samples, as
       the filter will flush all the remaining samples when the input audio
       signals its end.

       The filter accepts the following options:

       nb_out_samples, n
	   Set the number of frames per each output audio frame. The number is
	   intended as the number of samples per each channel.	Default value
	   is 1024.

       pad, p
	   If set to 1, the filter will pad the last audio frame with zeroes,
	   so that the last frame will contain the same number of samples as
	   the previous ones. Default value is 1.

       For example, to set the number of per-frame samples to 1234 and disable
       padding for the last frame, use:

	       asetnsamples=n=1234:p=0

   asetrate
       Set the sample rate without altering the PCM data.  This will result in
       a change of speed and pitch.

       The filter accepts the following options:

       sample_rate, r
	   Set the output sample rate. Default is 44100 Hz.

   ashowinfo
       Show a line containing various information for each input audio frame.
       The input audio is not modified.

       The shown line contains a sequence of key/value pairs of the form
       key:value.

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The presentation timestamp of the input frame, in time base units;
	   the time base depends on the filter input pad, and is usually
	   1/sample_rate.

       pts_time
	   The presentation timestamp of the input frame in seconds.

       pos position of the frame in the input stream, -1 if this information
	   in unavailable and/or meaningless (for example in case of synthetic
	   audio)

       fmt The sample format.

       chlayout
	   The channel layout.

       rate
	   The sample rate for the audio frame.

       nb_samples
	   The number of samples (per channel) in the frame.

       checksum
	   The Adler-32 checksum (printed in hexadecimal) of the audio data.
	   For planar audio, the data is treated as if all the planes were
	   concatenated.

       plane_checksums
	   A list of Adler-32 checksums for each data plane.

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

       It accepts the following option:

       length
	   Short window length in seconds, used for peak and trough RMS
	   measurement.  Default is 0.05 (50 milliseconds). Allowed range is
	   "[0.1 - 10]".

       metadata
	   Set metadata injection. All the metadata keys are prefixed with
	   "lavfi.astats.X", where "X" is channel number starting from 1 or
	   string "Overall". Default is disabled.

	   Available keys for each channel are: DC_offset Min_level Max_level
	   Min_difference Max_difference Mean_difference RMS_difference
	   Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count
	   Bit_depth Dynamic_range

	   and for Overall: DC_offset Min_level Max_level Min_difference
	   Max_difference Mean_difference RMS_difference Peak_level RMS_level
	   RMS_peak RMS_trough Flat_factor Peak_count Bit_depth
	   Number_of_samples

	   For example full key look like this "lavfi.astats.1.DC_offset" or
	   this "lavfi.astats.Overall.Peak_count".

	   For description what each key means read below.

       reset
	   Set number of frame after which stats are going to be recalculated.
	   Default is disabled.

       A description of each shown parameter follows:

       DC offset
	   Mean amplitude displacement from zero.

       Min level
	   Minimal sample level.

       Max level
	   Maximal sample level.

       Min difference
	   Minimal difference between two consecutive samples.

       Max difference
	   Maximal difference between two consecutive samples.

       Mean difference
	   Mean difference between two consecutive samples.  The average of
	   each difference between two consecutive samples.

       RMS difference
	   Root Mean Square difference between two consecutive samples.

       Peak level dB
       RMS level dB
	   Standard peak and RMS level measured in dBFS.

       RMS peak dB
       RMS trough dB
	   Peak and trough values for RMS level measured over a short window.

       Crest factor
	   Standard ratio of peak to RMS level (note: not in dB).

       Flat factor
	   Flatness (i.e. consecutive samples with the same value) of the
	   signal at its peak levels (i.e. either Min level or Max level).

       Peak count
	   Number of occasions (not the number of samples) that the signal
	   attained either Min level or Max level.

       Bit depth
	   Overall bit depth of audio. Number of bits used for each sample.

       Dynamic range
	   Measured dynamic range of audio in dB.

   atempo
       Adjust audio tempo.

       The filter accepts exactly one parameter, the audio tempo. If not
       specified then the filter will assume nominal 1.0 tempo. Tempo must be
       in the [0.5, 2.0] range.

       Examples

       ·   Slow down audio to 80% tempo:

		   atempo=0.8

       ·   To speed up audio to 125% tempo:

		   atempo=1.25

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

       It accepts the following parameters:

       start
	   Timestamp (in seconds) of the start of the section to keep. I.e.
	   the audio sample with the timestamp start will be the first sample
	   in the output.

       end Specify time of the first audio sample that will be dropped, i.e.
	   the audio sample immediately preceding the one with the timestamp
	   end will be the last sample in the output.

       start_pts
	   Same as start, except this option sets the start timestamp in
	   samples instead of seconds.

       end_pts
	   Same as end, except this option sets the end timestamp in samples
	   instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_sample
	   The number of the first sample that should be output.

       end_sample
	   The number of the first sample that should be dropped.

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

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

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

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

       Examples:

       ·   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -af atrim=60:120

       ·   Keep only the first 1000 samples:

		   ffmpeg -i INPUT -af atrim=end_sample=1000

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

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency. Default is 3000.

       csg Constant skirt gain if set to 1. Defaults to 0.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

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

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency. Default is 3000.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

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

       The filter accepts the following options:

       gain, g
	   Give the gain at 0 Hz. Its useful range is about -20 (for a large
	   cut) to +20 (for a large boost).  Beware of clipping when using a
	   positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 100 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Determine how steep is the filter's shelf transition.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

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

   bs2b
       Bauer stereo to binaural transformation, which improves headphone
       listening of stereo audio records.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libbs2b".

       It accepts the following parameters:

       profile
	   Pre-defined crossfeed level.

	   default
	       Default level (fcut=700, feed=50).

	   cmoy
	       Chu Moy circuit (fcut=700, feed=60).

	   jmeier
	       Jan Meier circuit (fcut=650, feed=95).

       fcut
	   Cut frequency (in Hz).

       feed
	   Feed level (in Hz).

   channelmap
       Remap input channels to new locations.

       It accepts the following parameters:

       map Map channels from input to output. The argument is a '|'-separated
	   list of mappings, each in the "in_channel-out_channel" or
	   in_channel form. in_channel can be either the name of the input
	   channel (e.g. FL for front left) or its index in the input channel
	   layout.  out_channel is the name of the output channel or its index
	   in the output channel layout. If out_channel is not given then it
	   is implicitly an index, starting with zero and increasing by one
	   for each mapping.

       channel_layout
	   The channel layout of the output stream.

       If no mapping is present, the filter will implicitly map input channels
       to output channels, preserving indices.

       For example, assuming a 5.1+downmix input MOV file,

	       ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

       will create an output WAV file tagged as stereo from the downmix
       channels of the input.

       To fix a 5.1 WAV improperly encoded in AAC's native channel order

	       ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

   channelsplit
       Split each channel from an input audio stream into a separate output
       stream.

       It accepts the following parameters:

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

       For example, assuming a stereo input MP3 file,

	       ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

       will create an output Matroska file with two audio streams, one
       containing only the left channel and the other the right channel.

       Split a 5.1 WAV file into per-channel files:

	       ffmpeg -i in.wav -filter_complex
	       'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
	       -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
	       front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
	       side_right.wav

   chorus
       Add a chorus effect to the audio.

       Can make a single vocal sound like a chorus, but can also be applied to
       instrumentation.

       Chorus resembles an echo effect with a short delay, but whereas with
       echo the delay is constant, with chorus, it is varied using using
       sinusoidal or triangular modulation.  The modulation depth defines the
       range the modulated delay is played before or after the delay. Hence
       the delayed sound will sound slower or faster, that is the delayed
       sound tuned around the original one, like in a chorus where some vocals
       are slightly off key.

       It accepts the following parameters:

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.4.

       delays
	   Set delays. A typical delay is around 40ms to 60ms.

       decays
	   Set decays.

       speeds
	   Set speeds.

       depths
	   Set depths.

       Examples

       ·   A single delay:

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

       ·   Two delays:

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

       ·   Fuller sounding chorus with three delays:

		   chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

   compand
       Compress or expand the audio's dynamic range.

       It accepts the following parameters:

       attacks
       decays
	   A list of times in seconds for each channel over which the
	   instantaneous level of the input signal is averaged to determine
	   its volume. attacks refers to increase of volume and decays refers
	   to decrease of volume. For most situations, the attack time
	   (response to the audio getting louder) should be shorter than the
	   decay time, because the human ear is more sensitive to sudden loud
	   audio than sudden soft audio. A typical value for attack is 0.3
	   seconds and a typical value for decay is 0.8 seconds.  If specified
	   number of attacks & decays is lower than number of channels, the
	   last set attack/decay will be used for all remaining channels.

       points
	   A list of points for the transfer function, specified in dB
	   relative to the maximum possible signal amplitude. Each key points
	   list must be defined using the following syntax:
	   "x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

	   The input values must be in strictly increasing order but the
	   transfer function does not have to be monotonically rising. The
	   point "0/0" is assumed but may be overridden (by "0/out-dBn").
	   Typical values for the transfer function are "-70/-70|-60/-20|1/0".

       soft-knee
	   Set the curve radius in dB for all joints. It defaults to 0.01.

       gain
	   Set the additional gain in dB to be applied at all points on the
	   transfer function. This allows for easy adjustment of the overall
	   gain.  It defaults to 0.

       volume
	   Set an initial volume, in dB, to be assumed for each channel when
	   filtering starts. This permits the user to supply a nominal level
	   initially, so that, for example, a very large gain is not applied
	   to initial signal levels before the companding has begun to
	   operate. A typical value for audio which is initially quiet is -90
	   dB. It defaults to 0.

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

       Examples

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

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

	   Another example for audio with whisper and explosion parts:

		   compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

       ·   A noise gate for when the noise is at a lower level than the
	   signal:

		   compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

       ·   Here is another noise gate, this time for when the noise is at a
	   higher level than the signal (making it, in some ways, similar to
	   squelch):

		   compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

       ·   2:1 compression starting at -6dB:

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

       ·   2:1 compression starting at -9dB:

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

       ·   2:1 compression starting at -12dB:

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

       ·   2:1 compression starting at -18dB:

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

       ·   3:1 compression starting at -15dB:

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

       ·   Compressor/Gate:

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

       ·   Expander:

		   compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3

       ·   Hard limiter at -6dB:

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

       ·   Hard limiter at -12dB:

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

       ·   Hard noise gate at -35 dB:

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

       ·   Soft limiter:

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

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

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

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

       It accepts the following parameters:

       mm  Set millimeters distance. This is compensation distance for fine
	   tuning.  Default is 0.

       cm  Set cm distance. This is compensation distance for tightening
	   distance setup.  Default is 0.

       m   Set meters distance. This is compensation distance for hard
	   distance setup.  Default is 0.

       dry Set dry amount. Amount of unprocessed (dry) signal.	Default is 0.

       wet Set wet amount. Amount of processed (wet) signal.  Default is 1.

       temp
	   Set temperature degree in Celsius. This is the temperature of the
	   environment.  Default is 20.

   crossfeed
       Apply headphone crossfeed filter.

       Crossfeed is the process of blending the left and right channels of
       stereo audio recording.	It is mainly used to reduce extreme stereo
       separation of low frequencies.

       The intent is to produce more speaker like sound to the listener.

       The filter accepts the following options:

       strength
	   Set strength of crossfeed. Default is 0.2. Allowed range is from 0
	   to 1.  This sets gain of low shelf filter for side part of stereo
	   image.  Default is -6dB. Max allowed is -30db when strength is set
	   to 1.

       range
	   Set soundstage wideness. Default is 0.5. Allowed range is from 0 to
	   1.  This sets cut off frequency of low shelf filter. Default is cut
	   off near 1550 Hz. With range set to 1 cut off frequency is set to
	   2100 Hz.

       level_in
	   Set input gain. Default is 0.9.

       level_out
	   Set output gain. Default is 1.

   crystalizer
       Simple algorithm to expand audio dynamic range.

       The filter accepts the following options:

       i   Sets the intensity of effect (default: 2.0). Must be in range
	   between 0.0 (unchanged sound) to 10.0 (maximum effect).

       c   Enable clipping. By default is enabled.

   dcshift
       Apply a DC shift to the audio.

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

       shift
	   Set the DC shift, allowed range is [-1, 1]. It indicates the amount
	   to shift the audio.

       limitergain
	   Optional. It should have a value much less than 1 (e.g. 0.05 or
	   0.02) and is used to prevent clipping.

   dynaudnorm
       Dynamic Audio Normalizer.

       This filter applies a certain amount of gain to the input audio in
       order to bring its peak magnitude to a target level (e.g. 0 dBFS).
       However, in contrast to more "simple" normalization algorithms, the
       Dynamic Audio Normalizer *dynamically* re-adjusts the gain factor to
       the input audio.  This allows for applying extra gain to the "quiet"
       sections of the audio while avoiding distortions or clipping the "loud"
       sections. In other words: The Dynamic Audio Normalizer will "even out"
       the volume of quiet and loud sections, in the sense that the volume of
       each section is brought to the same target level. Note, however, that
       the Dynamic Audio Normalizer achieves this goal *without* applying
       "dynamic range compressing". It will retain 100% of the dynamic range
       *within* each section of the audio file.

       f   Set the frame length in milliseconds. In range from 10 to 8000
	   milliseconds.  Default is 500 milliseconds.	The Dynamic Audio
	   Normalizer processes the input audio in small chunks, referred to
	   as frames. This is required, because a peak magnitude has no
	   meaning for just a single sample value. Instead, we need to
	   determine the peak magnitude for a contiguous sequence of sample
	   values. While a "standard" normalizer would simply use the peak
	   magnitude of the complete file, the Dynamic Audio Normalizer
	   determines the peak magnitude individually for each frame. The
	   length of a frame is specified in milliseconds. By default, the
	   Dynamic Audio Normalizer uses a frame length of 500 milliseconds,
	   which has been found to give good results with most files.  Note
	   that the exact frame length, in number of samples, will be
	   determined automatically, based on the sampling rate of the
	   individual input audio file.

       g   Set the Gaussian filter window size. In range from 3 to 301, must
	   be odd number. Default is 31.  Probably the most important
	   parameter of the Dynamic Audio Normalizer is the "window size" of
	   the Gaussian smoothing filter. The filter's window size is
	   specified in frames, centered around the current frame. For the
	   sake of simplicity, this must be an odd number. Consequently, the
	   default value of 31 takes into account the current frame, as well
	   as the 15 preceding frames and the 15 subsequent frames. Using a
	   larger window results in a stronger smoothing effect and thus in
	   less gain variation, i.e. slower gain adaptation. Conversely, using
	   a smaller window results in a weaker smoothing effect and thus in
	   more gain variation, i.e. faster gain adaptation.  In other words,
	   the more you increase this value, the more the Dynamic Audio
	   Normalizer will behave like a "traditional" normalization filter.
	   On the contrary, the more you decrease this value, the more the
	   Dynamic Audio Normalizer will behave like a dynamic range
	   compressor.

       p   Set the target peak value. This specifies the highest permissible
	   magnitude level for the normalized audio input. This filter will
	   try to approach the target peak magnitude as closely as possible,
	   but at the same time it also makes sure that the normalized signal
	   will never exceed the peak magnitude.  A frame's maximum local gain
	   factor is imposed directly by the target peak magnitude. The
	   default value is 0.95 and thus leaves a headroom of 5%*.  It is not
	   recommended to go above this value.

       m   Set the maximum gain factor. In range from 1.0 to 100.0. Default is
	   10.0.  The Dynamic Audio Normalizer determines the maximum possible
	   (local) gain factor for each input frame, i.e. the maximum gain
	   factor that does not result in clipping or distortion. The maximum
	   gain factor is determined by the frame's highest magnitude sample.
	   However, the Dynamic Audio Normalizer additionally bounds the
	   frame's maximum gain factor by a predetermined (global) maximum
	   gain factor. This is done in order to avoid excessive gain factors
	   in "silent" or almost silent frames. By default, the maximum gain
	   factor is 10.0, For most inputs the default value should be
	   sufficient and it usually is not recommended to increase this
	   value. Though, for input with an extremely low overall volume
	   level, it may be necessary to allow even higher gain factors. Note,
	   however, that the Dynamic Audio Normalizer does not simply apply a
	   "hard" threshold (i.e. cut off values above the threshold).
	   Instead, a "sigmoid" threshold function will be applied. This way,
	   the gain factors will smoothly approach the threshold value, but
	   never exceed that value.

       r   Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 -
	   disabled.  By default, the Dynamic Audio Normalizer performs "peak"
	   normalization.  This means that the maximum local gain factor for
	   each frame is defined (only) by the frame's highest magnitude
	   sample. This way, the samples can be amplified as much as possible
	   without exceeding the maximum signal level, i.e. without clipping.
	   Optionally, however, the Dynamic Audio Normalizer can also take
	   into account the frame's root mean square, abbreviated RMS. In
	   electrical engineering, the RMS is commonly used to determine the
	   power of a time-varying signal. It is therefore considered that the
	   RMS is a better approximation of the "perceived loudness" than just
	   looking at the signal's peak magnitude. Consequently, by adjusting
	   all frames to a constant RMS value, a uniform "perceived loudness"
	   can be established. If a target RMS value has been specified, a
	   frame's local gain factor is defined as the factor that would
	   result in exactly that RMS value.  Note, however, that the maximum
	   local gain factor is still restricted by the frame's highest
	   magnitude sample, in order to prevent clipping.

       n   Enable channels coupling. By default is enabled.  By default, the
	   Dynamic Audio Normalizer will amplify all channels by the same
	   amount. This means the same gain factor will be applied to all
	   channels, i.e.  the maximum possible gain factor is determined by
	   the "loudest" channel.  However, in some recordings, it may happen
	   that the volume of the different channels is uneven, e.g. one
	   channel may be "quieter" than the other one(s).  In this case, this
	   option can be used to disable the channel coupling. This way, the
	   gain factor will be determined independently for each channel,
	   depending only on the individual channel's highest magnitude
	   sample. This allows for harmonizing the volume of the different
	   channels.

       c   Enable DC bias correction. By default is disabled.  An audio signal
	   (in the time domain) is a sequence of sample values.  In the
	   Dynamic Audio Normalizer these sample values are represented in the
	   -1.0 to 1.0 range, regardless of the original input format.
	   Normally, the audio signal, or "waveform", should be centered
	   around the zero point.  That means if we calculate the mean value
	   of all samples in a file, or in a single frame, then the result
	   should be 0.0 or at least very close to that value. If, however,
	   there is a significant deviation of the mean value from 0.0, in
	   either positive or negative direction, this is referred to as a DC
	   bias or DC offset. Since a DC bias is clearly undesirable, the
	   Dynamic Audio Normalizer provides optional DC bias correction.
	   With DC bias correction enabled, the Dynamic Audio Normalizer will
	   determine the mean value, or "DC correction" offset, of each input
	   frame and subtract that value from all of the frame's sample values
	   which ensures those samples are centered around 0.0 again. Also, in
	   order to avoid "gaps" at the frame boundaries, the DC correction
	   offset values will be interpolated smoothly between neighbouring
	   frames.

       b   Enable alternative boundary mode. By default is disabled.  The
	   Dynamic Audio Normalizer takes into account a certain neighbourhood
	   around each frame. This includes the preceding frames as well as
	   the subsequent frames. However, for the "boundary" frames, located
	   at the very beginning and at the very end of the audio file, not
	   all neighbouring frames are available. In particular, for the first
	   few frames in the audio file, the preceding frames are not known.
	   And, similarly, for the last few frames in the audio file, the
	   subsequent frames are not known. Thus, the question arises which
	   gain factors should be assumed for the missing frames in the
	   "boundary" region. The Dynamic Audio Normalizer implements two
	   modes to deal with this situation. The default boundary mode
	   assumes a gain factor of exactly 1.0 for the missing frames,
	   resulting in a smooth "fade in" and "fade out" at the beginning and
	   at the end of the input, respectively.

       s   Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
	   By default, the Dynamic Audio Normalizer does not apply
	   "traditional" compression. This means that signal peaks will not be
	   pruned and thus the full dynamic range will be retained within each
	   local neighbourhood. However, in some cases it may be desirable to
	   combine the Dynamic Audio Normalizer's normalization algorithm with
	   a more "traditional" compression.  For this purpose, the Dynamic
	   Audio Normalizer provides an optional compression (thresholding)
	   function. If (and only if) the compression feature is enabled, all
	   input frames will be processed by a soft knee thresholding function
	   prior to the actual normalization process. Put simply, the
	   thresholding function is going to prune all samples whose magnitude
	   exceeds a certain threshold value.  However, the Dynamic Audio
	   Normalizer does not simply apply a fixed threshold value. Instead,
	   the threshold value will be adjusted for each individual frame.  In
	   general, smaller parameters result in stronger compression, and
	   vice versa.	Values below 3.0 are not recommended, because audible
	   distortion may appear.

   earwax
       Make audio easier to listen to on headphones.

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

       Ported from SoX.

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

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

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency in Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

       gain, g
	   Set the required gain or attenuation in dB.	Beware of clipping
	   when using a positive gain.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       Examples

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

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

       ·   Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz
	   with Q 2:

		   equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

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

       The filter accepts the following options:

       m   Sets the difference coefficient (default: 2.5). 0.0 means mono
	   sound (average of both channels), with 1.0 sound will be unchanged,
	   with -1.0 left and right channels will be swapped.

       c   Enable clipping. By default is enabled.

   firequalizer
       Apply FIR Equalization using arbitrary frequency response.

       The filter accepts the following option:

       gain
	   Set gain curve equation (in dB). The expression can contain
	   variables:

	   f   the evaluated frequency

	   sr  sample rate

	   ch  channel number, set to 0 when multichannels evaluation is
	       disabled

	   chid
	       channel id, see libavutil/channel_layout.h, set to the first
	       channel id when multichannels evaluation is disabled

	   chs number of channels

	   chlayout
	       channel_layout, see libavutil/channel_layout.h

	   and functions:

	   gain_interpolate(f)
	       interpolate gain on frequency f based on gain_entry

	   cubic_interpolate(f)
	       same as gain_interpolate, but smoother

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

       gain_entry
	   Set gain entry for gain_interpolate function. The expression can
	   contain functions:

	   entry(f, g)
	       store gain entry at frequency f with value g

	   This option is also available as command.

       delay
	   Set filter delay in seconds. Higher value means more accurate.
	   Default is 0.01.

       accuracy
	   Set filter accuracy in Hz. Lower value means more accurate.
	   Default is 5.

       wfunc
	   Set window function. Acceptable values are:

	   rectangular
	       rectangular window, useful when gain curve is already smooth

	   hann
	       hann window (default)

	   hamming
	       hamming window

	   blackman
	       blackman window

	   nuttall3
	       3-terms continuous 1st derivative nuttall window

	   mnuttall3
	       minimum 3-terms discontinuous nuttall window

	   nuttall
	       4-terms continuous 1st derivative nuttall window

	   bnuttall
	       minimum 4-terms discontinuous nuttall (blackman-nuttall) window

	   bharris
	       blackman-harris window

	   tukey
	       tukey window

       fixed
	   If enabled, use fixed number of audio samples. This improves speed
	   when filtering with large delay. Default is disabled.

       multi
	   Enable multichannels evaluation on gain. Default is disabled.

       zero_phase
	   Enable zero phase mode by subtracting timestamp to compensate
	   delay.  Default is disabled.

       scale
	   Set scale used by gain. Acceptable values are:

	   linlin
	       linear frequency, linear gain

	   linlog
	       linear frequency, logarithmic (in dB) gain (default)

	   loglin
	       logarithmic (in octave scale where 20 Hz is 0) frequency,
	       linear gain

	   loglog
	       logarithmic frequency, logarithmic gain

       dumpfile
	   Set file for dumping, suitable for gnuplot.

       dumpscale
	   Set scale for dumpfile. Acceptable values are same with scale
	   option.  Default is linlog.

       fft2
	   Enable 2-channel convolution using complex FFT. This improves speed
	   significantly.  Default is disabled.

       min_phase
	   Enable minimum phase impulse response. Default is disabled.

       Examples

       ·   lowpass at 1000 Hz:

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

       ·   lowpass at 1000 Hz with gain_entry:

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

       ·   custom equalization:

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

       ·   higher delay with zero phase to compensate delay:

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

       ·   lowpass on left channel, highpass on right channel:

		   firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
		   :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on

   flanger
       Apply a flanging effect to the audio.

       The filter accepts the following options:

       delay
	   Set base delay in milliseconds. Range from 0 to 30. Default value
	   is 0.

       depth
	   Set added sweep delay in milliseconds. Range from 0 to 10. Default
	   value is 2.

       regen
	   Set percentage regeneration (delayed signal feedback). Range from
	   -95 to 95.  Default value is 0.

       width
	   Set percentage of delayed signal mixed with original. Range from 0
	   to 100.  Default value is 71.

       speed
	   Set sweeps per second (Hz). Range from 0.1 to 10. Default value is
	   0.5.

       shape
	   Set swept wave shape, can be triangular or sinusoidal.  Default
	   value is sinusoidal.

       phase
	   Set swept wave percentage-shift for multi channel. Range from 0 to
	   100.  Default value is 25.

       interp
	   Set delay-line interpolation, linear or quadratic.  Default is
	   linear.

   haas
       Apply Haas effect to audio.

       Note that this makes most sense to apply on mono signals.  With this
       filter applied to mono signals it give some directionality and
       stretches its stereo image.

       The filter accepts the following options:

       level_in
	   Set input level. By default is 1, or 0dB

       level_out
	   Set output level. By default is 1, or 0dB.

       side_gain
	   Set gain applied to side part of signal. By default is 1.

       middle_source
	   Set kind of middle source. Can be one of the following:

	   left
	       Pick left channel.

	   right
	       Pick right channel.

	   mid Pick middle part signal of stereo image.

	   side
	       Pick side part signal of stereo image.

       middle_phase
	   Change middle phase. By default is disabled.

       left_delay
	   Set left channel delay. By default is 2.05 milliseconds.

       left_balance
	   Set left channel balance. By default is -1.

       left_gain
	   Set left channel gain. By default is 1.

       left_phase
	   Change left phase. By default is disabled.

       right_delay
	   Set right channel delay. By defaults is 2.12 milliseconds.

       right_balance
	   Set right channel balance. By default is 1.

       right_gain
	   Set right channel gain. By default is 1.

       right_phase
	   Change right phase. By default is enabled.

   hdcd
       Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM
       stream with embedded HDCD codes is expanded into a 20-bit PCM stream.

       The filter supports the Peak Extend and Low-level Gain Adjustment
       features of HDCD, and detects the Transient Filter flag.

	       ffmpeg -i HDCD16.flac -af hdcd OUT24.flac

       When using the filter with wav, note the default encoding for wav is
       16-bit, so the resulting 20-bit stream will be truncated back to
       16-bit. Use something like -acodec pcm_s24le after the filter to get
       24-bit PCM output.

	       ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
	       ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav

       The filter accepts the following options:

       disable_autoconvert
	   Disable any automatic format conversion or resampling in the filter
	   graph.

       process_stereo
	   Process the stereo channels together. If target_gain does not match
	   between channels, consider it invalid and use the last valid
	   target_gain.

       cdt_ms
	   Set the code detect timer period in ms.

       force_pe
	   Always extend peaks above -3dBFS even if PE isn't signaled.

       analyze_mode
	   Replace audio with a solid tone and adjust the amplitude to signal
	   some specific aspect of the decoding process. The output file can
	   be loaded in an audio editor alongside the original to aid
	   analysis.

	   "analyze_mode=pe:force_pe=true" can be used to see all samples
	   above the PE level.

	   Modes are:

	   0, off
	       Disabled

	   1, lle
	       Gain adjustment level at each sample

	   2, pe
	       Samples where peak extend occurs

	   3, cdt
	       Samples where the code detect timer is active

	   4, tgm
	       Samples where the target gain does not match between channels

   headphone
       Apply head-related transfer functions (HRTFs) to create virtual
       loudspeakers around the user for binaural listening via headphones.
       The HRIRs are provided via additional streams, for each channel one
       stereo input stream is needed.

       The filter accepts the following options:

       map Set mapping of input streams for convolution.  The argument is a
	   '|'-separated list of channel names in order as they are given as
	   additional stream inputs for filter.  This also specify number of
	   input streams. Number of input streams must be not less than number
	   of channels in first stream plus one.

       gain
	   Set gain applied to audio. Value is in dB. Default is 0.

       type
	   Set processing type. Can be time or freq. time is processing audio
	   in time domain which is slow.  freq is processing audio in
	   frequency domain which is fast.  Default is freq.

       lfe Set custom gain for LFE channels. Value is in dB. Default is 0.

       Examples

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

		   ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
		   output.wav

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

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz. Default is 3000.

       poles, p
	   Set number of poles. Default is 2.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.  Applies
	   only to double-pole filter.	The default is 0.707q and gives a
	   Butterworth response.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

   join
       Join multiple input streams into one multi-channel stream.

       It accepts the following parameters:

       inputs
	   The number of input streams. It defaults to 2.

       channel_layout
	   The desired output channel layout. It defaults to stereo.

       map Map channels from inputs to output. The argument is a '|'-separated
	   list of mappings, each in the "input_idx.in_channel-out_channel"
	   form. input_idx is the 0-based index of the input stream.
	   in_channel can be either the name of the input channel (e.g. FL for
	   front left) or its index in the specified input stream. out_channel
	   is the name of the output channel.

       The filter will attempt to guess the mappings when they are not
       specified explicitly. It does so by first trying to find an unused
       matching input channel and if that fails it picks the first unused
       input channel.

       Join 3 inputs (with properly set channel layouts):

	       ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

       Build a 5.1 output from 6 single-channel streams:

	       ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
	       'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
	       out

   ladspa
       Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

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

       file, f
	   Specifies the name of LADSPA plugin library to load. If the
	   environment variable LADSPA_PATH is defined, the LADSPA plugin is
	   searched in each one of the directories specified by the colon
	   separated list in LADSPA_PATH, otherwise in the standard LADSPA
	   paths, which are in this order: HOME/.ladspa/lib/,
	   /usr/local/lib/ladspa/, /usr/lib/ladspa/.

       plugin, p
	   Specifies the plugin within the library. Some libraries contain
	   only one plugin, but others contain many of them. If this is not
	   set filter will list all available plugins within the specified
	   library.

       controls, c
	   Set the '|' separated list of controls which are zero or more
	   floating point values that determine the behavior of the loaded
	   plugin (for example delay, threshold or gain).  Controls need to be
	   defined using the following syntax:
	   c0=value0|c1=value1|c2=value2|..., where valuei is the value set on
	   the i-th control.  Alternatively they can be also defined using the
	   following syntax: value0|value1|value2|..., where valuei is the
	   value set on the i-th control.  If controls is set to "help", all
	   available controls and their valid ranges are printed.

       sample_rate, s
	   Specify the sample rate, default to 44100. Only used if plugin have
	   zero inputs.

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default is 1024. Only used if plugin have zero inputs.

       duration, d
	   Set the minimum duration of the sourced audio. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater than the
	   specified duration, as the generated audio is always cut at the end
	   of a complete frame.  If not specified, or the expressed duration
	   is negative, the audio is supposed to be generated forever.	Only
	   used if plugin have zero inputs.

       Examples

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

		   ladspa=file=amp

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

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

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

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

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

		   ladspa=file=tap_reverb:tap_reverb

       ·   Generate white noise, with 0.2 amplitude:

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

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

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

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

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

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

		   ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

       ·   Attenuate low frequencies using Multiband EQ from Steve Harris "SWH
	   Plugins" collection:

		   ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0

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

		   ladspa=caps:Narrower

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

		   ladspa=caps:White:.2

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

		   ladspa=caps:Fractal:c=c1=1

       ·   Dynamic volume normalization using "VLevel" plugin:

		   ladspa=vlevel-ladspa:vlevel_mono

       Commands

       This filter supports the following commands:

       cN  Modify the N-th control value.

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

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

       The filter accepts the following options:

       I, i
	   Set integrated loudness target.  Range is -70.0 - -5.0. Default
	   value is -24.0.

       LRA, lra
	   Set loudness range target.  Range is 1.0 - 20.0. Default value is
	   7.0.

       TP, tp
	   Set maximum true peak.  Range is -9.0 - +0.0. Default value is
	   -2.0.

       measured_I, measured_i
	   Measured IL of input file.  Range is -99.0 - +0.0.

       measured_LRA, measured_lra
	   Measured LRA of input file.	Range is  0.0 - 99.0.

       measured_TP, measured_tp
	   Measured true peak of input file.  Range is	-99.0 - +99.0.

       measured_thresh
	   Measured threshold of input file.  Range is -99.0 - +0.0.

       offset
	   Set offset gain. Gain is applied before the true-peak limiter.
	   Range is  -99.0 - +99.0. Default is +0.0.

       linear
	   Normalize linearly if possible.  measured_I, measured_LRA,
	   measured_TP, and measured_thresh must also to be specified in order
	   to use this mode.  Options are true or false. Default is true.

       dual_mono
	   Treat mono input files as "dual-mono". If a mono file is intended
	   for playback on a stereo system, its EBU R128 measurement will be
	   perceptually incorrect.  If set to "true", this option will
	   compensate for this effect.	Multi-channel input files are not
	   affected by this option.  Options are true or false. Default is
	   false.

       print_format
	   Set print format for stats. Options are summary, json, or none.
	   Default value is none.

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

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz. Default is 500.

       poles, p
	   Set number of poles. Default is 2.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.  Applies
	   only to double-pole filter.	The default is 0.707q and gives a
	   Butterworth response.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       Examples

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

		   lowpass=c=LFE

   pan
       Mix channels with specific gain levels. The filter accepts the output
       channel layout followed by a set of channels definitions.

       This filter is also designed to efficiently remap the channels of an
       audio stream.

       The filter accepts parameters of the form: "l|outdef|outdef|..."

       l   output channel layout or number of channels

       outdef
	   output channel specification, of the form:
	   "out_name=[gain*]in_name[(+-)[gain*]in_name...]"

       out_name
	   output channel to define, either a channel name (FL, FR, etc.) or a
	   channel number (c0, c1, etc.)

       gain
	   multiplicative coefficient for the channel, 1 leaving the volume
	   unchanged

       in_name
	   input channel to use, see out_name for details; it is not possible
	   to mix named and numbered input channels

       If the `=' in a channel specification is replaced by `<', then the
       gains for that specification will be renormalized so that the total is
       1, thus avoiding clipping noise.

       Mixing examples

       For example, if you want to down-mix from stereo to mono, but with a
       bigger factor for the left channel:

	       pan=1c|c0=0.9*c0+0.1*c1

       A customized down-mix to stereo that works automatically for 3-, 4-, 5-
       and 7-channels surround:

	       pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

       Note that ffmpeg integrates a default down-mix (and up-mix) system that
       should be preferred (see "-ac" option) unless you have very specific
       needs.

       Remapping examples

       The channel remapping will be effective if, and only if:

       *<gain coefficients are zeroes or ones,>
       *<only one input per channel output,>

       If all these conditions are satisfied, the filter will notify the user
       ("Pure channel mapping detected"), and use an optimized and lossless
       method to do the remapping.

       For example, if you have a 5.1 source and want a stereo audio stream by
       dropping the extra channels:

	       pan="stereo| c0=FL | c1=FR"

       Given the same source, you can also switch front left and front right
       channels and keep the input channel layout:

	       pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

       If the input is a stereo audio stream, you can mute the front left
       channel (and still keep the stereo channel layout) with:

	       pan="stereo|c1=c1"

       Still with a stereo audio stream input, you can copy the right channel
       in both front left and right:

	       pan="stereo| c0=FR | c1=FR"

   replaygain
       ReplayGain scanner filter. This filter takes an audio stream as an
       input and outputs it unchanged.	At end of filtering it displays
       "track_gain" and "track_peak".

   resample
       Convert the audio sample format, sample rate and channel layout. It is
       not meant to be used directly.

   rubberband
       Apply time-stretching and pitch-shifting with librubberband.

       The filter accepts the following options:

       tempo
	   Set tempo scale factor.

       pitch
	   Set pitch scale factor.

       transients
	   Set transients detector.  Possible values are:

	   crisp
	   mixed
	   smooth
       detector
	   Set detector.  Possible values are:

	   compound
	   percussive
	   soft
       phase
	   Set phase.  Possible values are:

	   laminar
	   independent
       window
	   Set processing window size.	Possible values are:

	   standard
	   short
	   long
       smoothing
	   Set smoothing.  Possible values are:

	   off
	   on
       formant
	   Enable formant preservation when shift pitching.  Possible values
	   are:

	   shifted
	   preserved
       pitchq
	   Set pitch quality.  Possible values are:

	   quality
	   speed
	   consistency
       channels
	   Set channels.  Possible values are:

	   apart
	   together

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

       The filter accepts the following options:

       level_in
	   Set input gain. Default is 1. Range is between 0.015625 and 64.

       threshold
	   If a signal of second stream raises above this level it will affect
	   the gain reduction of first stream.	By default is 0.125. Range is
	   between 0.00097563 and 1.

       ratio
	   Set a ratio about which the signal is reduced. 1:2 means that if
	   the level raised 4dB above the threshold, it will be only 2dB above
	   after the reduction.  Default is 2. Range is between 1 and 20.

       attack
	   Amount of milliseconds the signal has to rise above the threshold
	   before gain reduction starts. Default is 20. Range is between 0.01
	   and 2000.

       release
	   Amount of milliseconds the signal has to fall below the threshold
	   before reduction is decreased again. Default is 250. Range is
	   between 0.01 and 9000.

       makeup
	   Set the amount by how much signal will be amplified after
	   processing.	Default is 1. Range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter gain reduction
	   more softly.  Default is 2.82843. Range is between 1 and 8.

       link
	   Choose if the "average" level between all channels of side-chain
	   stream or the louder("maximum") channel of side-chain stream
	   affects the reduction. Default is "average".

       detection
	   Should the exact signal be taken in case of "peak" or an RMS one in
	   case of "rms". Default is "rms" which is mainly smoother.

       level_sc
	   Set sidechain gain. Default is 1. Range is between 0.015625 and 64.

       mix How much to use compressed signal in output. Default is 1.  Range
	   is between 0 and 1.

       Examples

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

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

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

       The filter accepts the following options:

       level_in
	   Set input level before filtering.  Default is 1. Allowed range is
	   from 0.015625 to 64.

       range
	   Set the level of gain reduction when the signal is below the
	   threshold.  Default is 0.06125. Allowed range is from 0 to 1.

       threshold
	   If a signal rises above this level the gain reduction is released.
	   Default is 0.125. Allowed range is from 0 to 1.

       ratio
	   Set a ratio about which the signal is reduced.  Default is 2.
	   Allowed range is from 1 to 9000.

       attack
	   Amount of milliseconds the signal has to rise above the threshold
	   before gain reduction stops.  Default is 20 milliseconds. Allowed
	   range is from 0.01 to 9000.

       release
	   Amount of milliseconds the signal has to fall below the threshold
	   before the reduction is increased again. Default is 250
	   milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
	   Set amount of amplification of signal after processing.  Default is
	   1. Allowed range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter gain reduction
	   more softly.  Default is 2.828427125. Allowed range is from 1 to 8.

       detection
	   Choose if exact signal should be taken for detection or an RMS like
	   one.  Default is rms. Can be peak or rms.

       link
	   Choose if the average level between all channels or the louder
	   channel affects the reduction.  Default is average. Can be average
	   or maximum.

       level_sc
	   Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

   silencedetect
       Detect silence in an audio stream.

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

       The printed times and duration are expressed in seconds.

       The filter accepts the following options:

       duration, d
	   Set silence duration until notification (default is 2 seconds).

       noise, n
	   Set noise tolerance. Can be specified in dB (in case "dB" is
	   appended to the specified value) or amplitude ratio. Default is
	   -60dB, or 0.001.

       Examples

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

		   silencedetect=n=-50dB:d=5

       ·   Complete example with ffmpeg to detect silence with 0.0001 noise
	   tolerance in silence.mp3:

		   ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

   silenceremove
       Remove silence from the beginning, middle or end of the audio.

       The filter accepts the following options:

       start_periods
	   This value is used to indicate if audio should be trimmed at
	   beginning of the audio. A value of zero indicates no silence should
	   be trimmed from the beginning. When specifying a non-zero value, it
	   trims audio up until it finds non-silence. Normally, when trimming
	   silence from beginning of audio the start_periods will be 1 but it
	   can be increased to higher values to trim all audio up to specific
	   count of non-silence periods.  Default value is 0.

       start_duration
	   Specify the amount of time that non-silence must be detected before
	   it stops trimming audio. By increasing the duration, bursts of
	   noises can be treated as silence and trimmed off. Default value is
	   0.

       start_threshold
	   This indicates what sample value should be treated as silence. For
	   digital audio, a value of 0 may be fine but for audio recorded from
	   analog, you may wish to increase the value to account for
	   background noise.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude ratio. Default value is 0.

       stop_periods
	   Set the count for trimming silence from the end of audio.  To
	   remove silence from the middle of a file, specify a stop_periods
	   that is negative. This value is then treated as a positive value
	   and is used to indicate the effect should restart processing as
	   specified by start_periods, making it suitable for removing periods
	   of silence in the middle of the audio.  Default value is 0.

       stop_duration
	   Specify a duration of silence that must exist before audio is not
	   copied any more. By specifying a higher duration, silence that is
	   wanted can be left in the audio.  Default value is 0.

       stop_threshold
	   This is the same as start_threshold but for trimming silence from
	   the end of audio.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude ratio. Default value is 0.

       leave_silence
	   This indicates that stop_duration length of audio should be left
	   intact at the beginning of each period of silence.  For example, if
	   you want to remove long pauses between words but do not want to
	   remove the pauses completely. Default value is 0.

       detection
	   Set how is silence detected. Can be "rms" or "peak". Second is
	   faster and works better with digital silence which is exactly 0.
	   Default value is "rms".

       window
	   Set ratio used to calculate size of window for detecting silence.
	   Default value is 0.02. Allowed range is from 0 to 10.

       Examples

       ·   The following example shows how this filter can be used to start a
	   recording that does not contain the delay at the start which
	   usually occurs between pressing the record button and the start of
	   the performance:

		   silenceremove=1:5:0.02

       ·   Trim all silence encountered from beginning to end where there is
	   more than 1 second of silence in audio:

		   silenceremove=0:0:0:-1:1:-90dB

   sofalizer
       SOFAlizer uses head-related transfer functions (HRTFs) to create
       virtual loudspeakers around the user for binaural listening via
       headphones (audio formats up to 9 channels supported).  The HRTFs are
       stored in SOFA files (see <http://www.sofacoustics.org/> for a
       database).  SOFAlizer is developed at the Acoustics Research Institute
       (ARI) of the Austrian Academy of Sciences.

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

       The filter accepts the following options:

       sofa
	   Set the SOFA file used for rendering.

       gain
	   Set gain applied to audio. Value is in dB. Default is 0.

       rotation
	   Set rotation of virtual loudspeakers in deg. Default is 0.

       elevation
	   Set elevation of virtual speakers in deg. Default is 0.

       radius
	   Set distance in meters between loudspeakers and the listener with
	   near-field HRTFs. Default is 1.

       type
	   Set processing type. Can be time or freq. time is processing audio
	   in time domain which is slow.  freq is processing audio in
	   frequency domain which is fast.  Default is freq.

       speakers
	   Set custom positions of virtual loudspeakers. Syntax for this
	   option is: <CH> <AZIM> <ELEV>[|<CH> <AZIM> <ELEV>|...].  Each
	   virtual loudspeaker is described with short channel name following
	   with azimuth and elevation in degrees.  Each virtual loudspeaker
	   description is separated by '|'.  For example to override front
	   left and front right channel positions use: 'speakers=FL 45 15|FR
	   345 15'.  Descriptions with unrecognised channel names are ignored.

       lfegain
	   Set custom gain for LFE channels. Value is in dB. Default is 0.

       Examples

       ·   Using ClubFritz6 sofa file:

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

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

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

       ·   Similar as above but with custom speaker positions for front left,
	   front right, back left and back right and also with custom gain:

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

   stereotools
       This filter has some handy utilities to manage stereo signals, for
       converting M/S stereo recordings to L/R signal while having control
       over the parameters or spreading the stereo image of master track.

       The filter accepts the following options:

       level_in
	   Set input level before filtering for both channels. Defaults is 1.
	   Allowed range is from 0.015625 to 64.

       level_out
	   Set output level after filtering for both channels. Defaults is 1.
	   Allowed range is from 0.015625 to 64.

       balance_in
	   Set input balance between both channels. Default is 0.  Allowed
	   range is from -1 to 1.

       balance_out
	   Set output balance between both channels. Default is 0.  Allowed
	   range is from -1 to 1.

       softclip
	   Enable softclipping. Results in analog distortion instead of harsh
	   digital 0dB clipping. Disabled by default.

       mutel
	   Mute the left channel. Disabled by default.

       muter
	   Mute the right channel. Disabled by default.

       phasel
	   Change the phase of the left channel. Disabled by default.

       phaser
	   Change the phase of the right channel. Disabled by default.

       mode
	   Set stereo mode. Available values are:

	   lr>lr
	       Left/Right to Left/Right, this is default.

	   lr>ms
	       Left/Right to Mid/Side.

	   ms>lr
	       Mid/Side to Left/Right.

	   lr>ll
	       Left/Right to Left/Left.

	   lr>rr
	       Left/Right to Right/Right.

	   lr>l+r
	       Left/Right to Left + Right.

	   lr>rl
	       Left/Right to Right/Left.

	   ms>ll
	       Mid/Side to Left/Left.

	   ms>rr
	       Mid/Side to Right/Right.

       slev
	   Set level of side signal. Default is 1.  Allowed range is from
	   0.015625 to 64.

       sbal
	   Set balance of side signal. Default is 0.  Allowed range is from -1
	   to 1.

       mlev
	   Set level of the middle signal. Default is 1.  Allowed range is
	   from 0.015625 to 64.

       mpan
	   Set middle signal pan. Default is 0. Allowed range is from -1 to 1.

       base
	   Set stereo base between mono and inversed channels. Default is 0.
	   Allowed range is from -1 to 1.

       delay
	   Set delay in milliseconds how much to delay left from right channel
	   and vice versa. Default is 0. Allowed range is from -20 to 20.

       sclevel
	   Set S/C level. Default is 1. Allowed range is from 1 to 100.

       phase
	   Set the stereo phase in degrees. Default is 0. Allowed range is
	   from 0 to 360.

       bmode_in, bmode_out
	   Set balance mode for balance_in/balance_out option.

	   Can be one of the following:

	   balance
	       Classic balance mode. Attenuate one channel at time.  Gain is
	       raised up to 1.

	   amplitude
	       Similar as classic mode above but gain is raised up to 2.

	   power
	       Equal power distribution, from -6dB to +6dB range.

       Examples

       ·   Apply karaoke like effect:

		   stereotools=mlev=0.015625

       ·   Convert M/S signal to L/R:

		   "stereotools=mode=ms>lr"

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

       The filter accepts the following options:

       delay
	   Time in milliseconds of the delay of left signal into right and
	   vice versa.	Default is 20 milliseconds.

       feedback
	   Amount of gain in delayed signal into right and vice versa. Gives a
	   delay effect of left signal in right output and vice versa which
	   gives widening effect. Default is 0.3.

       crossfeed
	   Cross feed of left into right with inverted phase. This helps in
	   suppressing the mono. If the value is 1 it will cancel all the
	   signal common to both channels. Default is 0.3.

       drymix
	   Set level of input signal of original channel. Default is 0.8.

   superequalizer
       Apply 18 band equalizer.

       The filter accepts the following options:

       1b  Set 65Hz band gain.

       2b  Set 92Hz band gain.

       3b  Set 131Hz band gain.

       4b  Set 185Hz band gain.

       5b  Set 262Hz band gain.

       6b  Set 370Hz band gain.

       7b  Set 523Hz band gain.

       8b  Set 740Hz band gain.

       9b  Set 1047Hz band gain.

       10b Set 1480Hz band gain.

       11b Set 2093Hz band gain.

       12b Set 2960Hz band gain.

       13b Set 4186Hz band gain.

       14b Set 5920Hz band gain.

       15b Set 8372Hz band gain.

       16b Set 11840Hz band gain.

       17b Set 16744Hz band gain.

       18b Set 20000Hz band gain.

   surround
       Apply audio surround upmix filter.

       This filter allows to produce multichannel output from audio stream.

       The filter accepts the following options:

       chl_out
	   Set output channel layout. By default, this is 5.1.

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

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

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

       level_in
	   Set input volume level. By default, this is 1.

       level_out
	   Set output volume level. By default, this is 1.

       lfe Enable LFE channel output if output channel layout has it. By
	   default, this is enabled.

       lfe_low
	   Set LFE low cut off frequency. By default, this is 128 Hz.

       lfe_high
	   Set LFE high cut off frequency. By default, this is 256 Hz.

       fc_in
	   Set front center input volume. By default, this is 1.

       fc_out
	   Set front center output volume. By default, this is 1.

       lfe_in
	   Set LFE input volume. By default, this is 1.

       lfe_out
	   Set LFE output volume. By default, this is 1.

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

       The filter accepts the following options:

       gain, g
	   Give the gain at whichever is the lower of ~22 kHz and the Nyquist
	   frequency. Its useful range is about -20 (for a large cut) to +20
	   (for a large boost). Beware of clipping when using a positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 3000 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Determine how steep is the filter's shelf transition.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

   tremolo
       Sinusoidal amplitude modulation.

       The filter accepts the following options:

       f   Modulation frequency in Hertz. Modulation frequencies in the
	   subharmonic range (20 Hz or lower) will result in a tremolo effect.
	   This filter may also be used as a ring modulator by specifying a
	   modulation frequency higher than 20 Hz.  Range is 0.1 - 20000.0.
	   Default value is 5.0 Hz.

       d   Depth of modulation as a percentage. Range is 0.0 - 1.0.  Default
	   value is 0.5.

   vibrato
       Sinusoidal phase modulation.

       The filter accepts the following options:

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

       d   Depth of modulation as a percentage. Range is 0.0 - 1.0.  Default
	   value is 0.5.

   volume
       Adjust the input audio volume.

       It accepts the following parameters:

       volume
	   Set audio volume expression.

	   Output values are clipped to the maximum value.

	   The output audio volume is given by the relation:

		   <output_volume> = <volume> * <input_volume>

	   The default value for volume is "1.0".

       precision
	   This parameter represents the mathematical precision.

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

	   fixed
	       8-bit fixed-point; this limits input sample format to U8, S16,
	       and S32.

	   float
	       32-bit floating-point; this limits input sample format to FLT.
	       (default)

	   double
	       64-bit floating-point; this limits input sample format to DBL.

       replaygain
	   Choose the behaviour on encountering ReplayGain side data in input
	   frames.

	   drop
	       Remove ReplayGain side data, ignoring its contents (the
	       default).

	   ignore
	       Ignore ReplayGain side data, but leave it in the frame.

	   track
	       Prefer the track gain, if present.

	   album
	       Prefer the album gain, if present.

       replaygain_preamp
	   Pre-amplification gain in dB to apply to the selected replaygain
	   gain.

	   Default value for replaygain_preamp is 0.0.

       eval
	   Set when the volume expression is evaluated.

	   It accepts the following values:

	   once
	       only evaluate expression once during the filter initialization,
	       or when the volume command is sent

	   frame
	       evaluate expression for each incoming frame

	   Default value is once.

       The volume expression can contain the following parameters.

       n   frame number (starting at zero)

       nb_channels
	   number of channels

       nb_consumed_samples
	   number of samples consumed by the filter

       nb_samples
	   number of samples in the current frame

       pos original frame position in the file

       pts frame PTS

       sample_rate
	   sample rate

       startpts
	   PTS at start of stream

       startt
	   time at start of stream

       t   frame time

       tb  timestamp timebase

       volume
	   last set volume value

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

       Commands

       This filter supports the following commands:

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

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

       replaygain_noclip
	   Prevent clipping by limiting the gain applied.

	   Default value for replaygain_noclip is 1.

       Examples

       ·   Halve the input audio volume:

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

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

		   volume=0.5

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

		   volume=volume=6dB:precision=fixed

       ·   Fade volume after time 10 with an annihilation period of 5 seconds:

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

   volumedetect
       Detect the volume of the input video.

       The filter has no parameters. The input is not modified. Statistics
       about the volume will be printed in the log when the input stream end
       is reached.

       In particular it will show the mean volume (root mean square), maximum
       volume (on a per-sample basis), and the beginning of a histogram of the
       registered volume values (from the maximum value to a cumulated 1/1000
       of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

	       [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
	       [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
	       [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
	       [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
	       [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
	       [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
	       [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
	       [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
	       [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means that:

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

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

       ·   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

       In other words, raising the volume by +4 dB does not cause any
       clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
       Below is a description of the currently available audio sources.

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

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in libavfilter/asrc_abuffer.h.

       It accepts the following parameters:

       time_base
	   The timebase which will be used for timestamps of submitted frames.
	   It must be either a floating-point number or in
	   numerator/denominator form.

       sample_rate
	   The sample rate of the incoming audio buffers.

       sample_fmt
	   The sample format of the incoming audio buffers.  Either a sample
	   format name or its corresponding integer representation from the
	   enum AVSampleFormat in libavutil/samplefmt.h

       channel_layout
	   The channel layout of the incoming audio buffers.  Either a channel
	   layout name from channel_layout_map in libavutil/channel_layout.c
	   or its corresponding integer representation from the AV_CH_LAYOUT_*
	   macros in libavutil/channel_layout.h

       channels
	   The number of channels of the incoming audio buffers.  If both
	   channels and channel_layout are specified, then they must be
	   consistent.

       Examples

	       abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

       will instruct the source to accept planar 16bit signed stereo at
       44100Hz.  Since the sample format with name "s16p" corresponds to the
       number 6 and the "stereo" channel layout corresponds to the value 0x3,
       this is equivalent to:

	       abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

   aevalsrc
       Generate an audio signal specified by an expression.

       This source accepts in input one or more expressions (one for each
       channel), which are evaluated and used to generate a corresponding
       audio signal.

       This source accepts the following options:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   In case the channel_layout option is not specified, the selected
	   channel layout depends on the number of provided expressions.
	   Otherwise the last specified expression is applied to the remaining
	   output channels.

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

       duration, d
	   Set the minimum duration of the sourced audio. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater than the
	   specified duration, as the generated audio is always cut at the end
	   of a complete frame.

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

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default to 1024.

       sample_rate, s
	   Specify the sample rate, default to 44100.

       Each expression in exprs can contain the following constants:

       n   number of the evaluated sample, starting from 0

       t   time of the evaluated sample expressed in seconds, starting from 0

       s   sample rate

       Examples

       ·   Generate silence:

		   aevalsrc=0

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

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

       ·   Generate a two channels signal, specify the channel layout (Front
	   Center + Back Center) explicitly:

		   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       ·   Generate white noise:

		   aevalsrc="-2+random(0)"

       ·   Generate an amplitude modulated signal:

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

       ·   Generate 2.5 Hz binaural beats on a 360 Hz carrier:

		   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

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

       This source accepts the following options:

       channel_layout, cl
	   Specifies the channel layout, and can be either an integer or a
	   string representing a channel layout. The default value of
	   channel_layout is "stereo".

	   Check the channel_layout_map definition in
	   libavutil/channel_layout.c for the mapping between strings and
	   channel layout values.

       sample_rate, r
	   Specifies the sample rate, and defaults to 44100.

       nb_samples, n
	   Set the number of samples per requested frames.

       Examples

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

		   anullsrc=r=48000:cl=4

       ·   Do the same operation with a more obvious syntax:

		   anullsrc=r=48000:cl=mono

       All the parameters need to be explicitly defined.

   flite
       Synthesize a voice utterance using the libflite library.

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

       Note that the flite library is not thread-safe.

       The filter accepts the following options:

       list_voices
	   If set to 1, list the names of the available voices and exit
	   immediately. Default value is 0.

       nb_samples, n
	   Set the maximum number of samples per frame. Default value is 512.

       textfile
	   Set the filename containing the text to speak.

       text
	   Set the text to speak.

       voice, v
	   Set the voice to use for the speech synthesis. Default value is
	   "kal". See also the list_voices option.

       Examples

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

		   flite=textfile=speech.txt

       ·   Read the specified text selecting the "slt" voice:

		   flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Input text to ffmpeg:

		   ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Make ffplay speak the specified text, using "flite" and the "lavfi"
	   device:

		   ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

       For more information about libflite, check:
       <http://www.speech.cs.cmu.edu/flite/>

   anoisesrc
       Generate a noise audio signal.

       The filter accepts the following options:

       sample_rate, r
	   Specify the sample rate. Default value is 48000 Hz.

       amplitude, a
	   Specify the amplitude (0.0 - 1.0) of the generated audio stream.
	   Default value is 1.0.

       duration, d
	   Specify the duration of the generated audio stream. Not specifying
	   this option results in noise with an infinite length.

       color, colour, c
	   Specify the color of noise. Available noise colors are white, pink,
	   brown, blue and violet. Default color is white.

       seed, s
	   Specify a value used to seed the PRNG.

       nb_samples, n
	   Set the number of samples per each output frame, default is 1024.

       Examples

       ·   Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate
	   and an amplitude of 0.5:

		   anoisesrc=d=60:c=pink:r=44100:a=0.5

   sine
       Generate an audio signal made of a sine wave with amplitude 1/8.

       The audio signal is bit-exact.

       The filter accepts the following options:

       frequency, f
	   Set the carrier frequency. Default is 440 Hz.

       beep_factor, b
	   Enable a periodic beep every second with frequency beep_factor
	   times the carrier frequency. Default is 0, meaning the beep is
	   disabled.

       sample_rate, r
	   Specify the sample rate, default is 44100.

       duration, d
	   Specify the duration of the generated audio stream.

       samples_per_frame
	   Set the number of samples per output frame.

	   The expression can contain the following constants:

	   n   The (sequential) number of the output audio frame, starting
	       from 0.

	   pts The PTS (Presentation TimeStamp) of the output audio frame,
	       expressed in TB units.

	   t   The PTS of the output audio frame, expressed in seconds.

	   TB  The timebase of the output audio frames.

	   Default is 1024.

       Examples

       ·   Generate a simple 440 Hz sine wave:

		   sine

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

		   sine=220:4:d=5
		   sine=f=220:b=4:d=5
		   sine=frequency=220:beep_factor=4:duration=5

       ·   Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602"
	   NTSC pattern:

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

AUDIO SINKS
       Below is a description of the currently available audio sinks.

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

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

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

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

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

       Below is a description of the currently available video filters.

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

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

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

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

       Since this filter is designed for reconstruction, it operates on frame
       sequences without considering timestamps, and terminates when either
       input reaches end of stream. This will cause problems if your encoding
       pipeline drops frames. If you're trying to apply an image as an overlay
       to a video stream, consider the overlay filter instead.

   ass
       Same as the subtitles filter, except that it doesn't require libavcodec
       and libavformat to work. On the other hand, it is limited to ASS
       (Advanced Substation Alpha) subtitles files.

       This filter accepts the following option in addition to the common
       options from the subtitles filter:

       shaping
	   Set the shaping engine

	   Available values are:

	   auto
	       The default libass shaping engine, which is the best available.

	   simple
	       Fast, font-agnostic shaper that can do only substitutions

	   complex
	       Slower shaper using OpenType for substitutions and positioning

	   The default is "auto".

   atadenoise
       Apply an Adaptive Temporal Averaging Denoiser to the video input.

       The filter accepts the following options:

       0a  Set threshold A for 1st plane. Default is 0.02.  Valid range is 0
	   to 0.3.

       0b  Set threshold B for 1st plane. Default is 0.04.  Valid range is 0
	   to 5.

       1a  Set threshold A for 2nd plane. Default is 0.02.  Valid range is 0
	   to 0.3.

       1b  Set threshold B for 2nd plane. Default is 0.04.  Valid range is 0
	   to 5.

       2a  Set threshold A for 3rd plane. Default is 0.02.  Valid range is 0
	   to 0.3.

       2b  Set threshold B for 3rd plane. Default is 0.04.  Valid range is 0
	   to 5.

	   Threshold A is designed to react on abrupt changes in the input
	   signal and threshold B is designed to react on continuous changes
	   in the input signal.

       s   Set number of frames filter will use for averaging. Default is 33.
	   Must be odd number in range [5, 129].

       p   Set what planes of frame filter will use for averaging. Default is
	   all.

   avgblur
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
	   Set horizontal kernel size.

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

       sizeY
	   Set vertical kernel size, if zero it will be same as "sizeX".
	   Default is 0.

   bbox
       Compute the bounding box for the non-black pixels in the input frame
       luminance plane.

       This filter computes the bounding box containing all the pixels with a
       luminance value greater than the minimum allowed value.	The parameters
       describing the bounding box are printed on the filter log.

       The filter accepts the following option:

       min_val
	   Set the minimal luminance value. Default is 16.

   bitplanenoise
       Show and measure bit plane noise.

       The filter accepts the following options:

       bitplane
	   Set which plane to analyze. Default is 1.

       filter
	   Filter out noisy pixels from "bitplane" set above.  Default is
	   disabled.

   blackdetect
       Detect video intervals that are (almost) completely black. Can be
       useful to detect chapter transitions, commercials, or invalid
       recordings. Output lines contains the time for the start, end and
       duration of the detected black interval expressed in seconds.

       In order to display the output lines, you need to set the loglevel at
       least to the AV_LOG_INFO value.

       The filter accepts the following options:

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

	   Default value is 2.0.

       picture_black_ratio_th, pic_th
	   Set the threshold for considering a picture "black".  Express the
	   minimum value for the ratio:

		   <nb_black_pixels> / <nb_pixels>

	   for which a picture is considered black.  Default value is 0.98.

       pixel_black_th, pix_th
	   Set the threshold for considering a pixel "black".

	   The threshold expresses the maximum pixel luminance value for which
	   a pixel is considered "black". The provided value is scaled
	   according to the following equation:

		   <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

	   luminance_range_size and luminance_minimum_value depend on the
	   input video format, the range is [0-255] for YUV full-range formats
	   and [16-235] for YUV non full-range formats.

	   Default value is 0.10.

       The following example sets the maximum pixel threshold to the minimum
       value, and detects only black intervals of 2 or more seconds:

	       blackdetect=d=2:pix_th=0.00

   blackframe
       Detect frames that are (almost) completely black. Can be useful to
       detect chapter transitions or commercials. Output lines consist of the
       frame number of the detected frame, the percentage of blackness, the
       position in the file if known or -1 and the timestamp in seconds.

       In order to display the output lines, you need to set the loglevel at
       least to the AV_LOG_INFO value.

       This filter exports frame metadata "lavfi.blackframe.pblack".  The
       value represents the percentage of pixels in the picture that are below
       the threshold value.

       It accepts the following parameters:

       amount
	   The percentage of the pixels that have to be below the threshold;
	   it defaults to 98.

       threshold, thresh
	   The threshold below which a pixel value is considered black; it
	   defaults to 32.

   blend, tblend
       Blend two video frames into each other.

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

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

       A description of the accepted options follows.

       c0_mode
       c1_mode
       c2_mode
       c3_mode
       all_mode
	   Set blend mode for specific pixel component or all pixel components
	   in case of all_mode. Default value is "normal".

	   Available values for component modes are:

	   addition
	   grainmerge
	   and
	   average
	   burn
	   darken
	   difference
	   grainextract
	   divide
	   dodge
	   freeze
	   exclusion
	   extremity
	   glow
	   hardlight
	   hardmix
	   heat
	   lighten
	   linearlight
	   multiply
	   multiply128
	   negation
	   normal
	   or
	   overlay
	   phoenix
	   pinlight
	   reflect
	   screen
	   softlight
	   subtract
	   vividlight
	   xor
       c0_opacity
       c1_opacity
       c2_opacity
       c3_opacity
       all_opacity
	   Set blend opacity for specific pixel component or all pixel
	   components in case of all_opacity. Only used in combination with
	   pixel component blend modes.

       c0_expr
       c1_expr
       c2_expr
       c3_expr
       all_expr
	   Set blend expression for specific pixel component or all pixel
	   components in case of all_expr. Note that related mode options will
	   be ignored if those are set.

	   The expressions can use the following variables:

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

	   X
	   Y   the coordinates of the current sample

	   W
	   H   the width and height of currently filtered plane

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

	   T   Time of the current frame, expressed in seconds.

	   TOP, A
	       Value of pixel component at current location for first video
	       frame (top layer).

	   BOTTOM, B
	       Value of pixel component at current location for second video
	       frame (bottom layer).

       The "blend" filter also supports the framesync options.

       Examples

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

		   blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

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

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

       ·   Apply 1x1 checkerboard effect:

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

       ·   Apply uncover left effect:

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

       ·   Apply uncover down effect:

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

       ·   Apply uncover up-left effect:

		   blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

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

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

       ·   Display differences between the current and the previous frame:

		   tblend=all_mode=grainextract

   boxblur
       Apply a boxblur algorithm to the input video.

       It accepts the following parameters:

       luma_radius, lr
       luma_power, lp
       chroma_radius, cr
       chroma_power, cp
       alpha_radius, ar
       alpha_power, ap

       A description of the accepted options follows.

       luma_radius, lr
       chroma_radius, cr
       alpha_radius, ar
	   Set an expression for the box radius in pixels used for blurring
	   the corresponding input plane.

	   The radius value must be a non-negative number, and must not be
	   greater than the value of the expression "min(w,h)/2" for the luma
	   and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

	   Default value for luma_radius is "2". If not specified,
	   chroma_radius and alpha_radius default to the corresponding value
	   set for luma_radius.

	   The expressions can contain the following constants:

	   w
	   h   The input width and height in pixels.

	   cw
	   ch  The input chroma image width and height in pixels.

	   hsub
	   vsub
	       The horizontal and vertical chroma subsample values. For
	       example, for the pixel format "yuv422p", hsub is 2 and vsub is
	       1.

       luma_power, lp
       chroma_power, cp
       alpha_power, ap
	   Specify how many times the boxblur filter is applied to the
	   corresponding plane.

	   Default value for luma_power is 2. If not specified, chroma_power
	   and alpha_power default to the corresponding value set for
	   luma_power.

	   A value of 0 will disable the effect.

       Examples

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

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

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

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

       ·   Set the luma and chroma radii to a fraction of the video dimension:

		   boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

   bwdif
       Deinterlace the input video ("bwdif" stands for "Bob Weaver
       Deinterlacing Filter").

       Motion adaptive deinterlacing based on yadif with the use of w3fdif and
       cubic interpolation algorithms.	It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It accepts one of the following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   The default value is "send_field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first will be
	   assumed.

       deint
	   Specify which frames to deinterlace. Accept one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   chromakey
       YUV colorspace color/chroma keying.

       The filter accepts the following options:

       color
	   The color which will be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01 matches only the exact key color, while 1.0 matches
	   everything.

       blend
	   Blend percentage.

	   0.0 makes pixels either fully transparent, or not transparent at
	   all.

	   Higher values result in semi-transparent pixels, with a higher
	   transparency the more similar the pixels color is to the key color.

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

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

       Examples

       ·   Make every green pixel in the input image transparent:

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

       ·   Overlay a greenscreen-video on top of a static black background.

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

   ciescope
       Display CIE color diagram with pixels overlaid onto it.

       The filter accepts the following options:

       system
	   Set color system.

	   ntsc, 470m
	   ebu, 470bg
	   smpte
	   240m
	   apple
	   widergb
	   cie1931
	   rec709, hdtv
	   uhdtv, rec2020
       cie Set CIE system.

	   xyy
	   ucs
	   luv
       gamuts
	   Set what gamuts to draw.

	   See "system" option for available values.

       size, s
	   Set ciescope size, by default set to 512.

       intensity, i
	   Set intensity used to map input pixel values to CIE diagram.

       contrast
	   Set contrast used to draw tongue colors that are out of active
	   color system gamut.

       corrgamma
	   Correct gamma displayed on scope, by default enabled.

       showwhite
	   Show white point on CIE diagram, by default disabled.

       gamma
	   Set input gamma. Used only with XYZ input color space.

   codecview
       Visualize information exported by some codecs.

       Some codecs can export information through frames using side-data or
       other means. For example, some MPEG based codecs export motion vectors
       through the export_mvs flag in the codec flags2 option.

       The filter accepts the following option:

       mv  Set motion vectors to visualize.

	   Available flags for mv are:

	   pf  forward predicted MVs of P-frames

	   bf  forward predicted MVs of B-frames

	   bb  backward predicted MVs of B-frames

       qp  Display quantization parameters using the chroma planes.

       mv_type, mvt
	   Set motion vectors type to visualize. Includes MVs from all frames
	   unless specified by frame_type option.

	   Available flags for mv_type are:

	   fp  forward predicted MVs

	   bp  backward predicted MVs

       frame_type, ft
	   Set frame type to visualize motion vectors of.

	   Available flags for frame_type are:

	   if  intra-coded frames (I-frames)

	   pf  predicted frames (P-frames)

	   bf  bi-directionally predicted frames (B-frames)

       Examples

       ·   Visualize forward predicted MVs of all frames using ffplay:

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

       ·   Visualize multi-directionals MVs of P and B-Frames using ffplay:

		   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb

   colorbalance
       Modify intensity of primary colors (red, green and blue) of input
       frames.

       The filter allows an input frame to be adjusted in the shadows,
       midtones or highlights regions for the red-cyan, green-magenta or blue-
       yellow balance.

       A positive adjustment value shifts the balance towards the primary
       color, a negative value towards the complementary color.

       The filter accepts the following options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

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

       Examples

       ·   Add red color cast to shadows:

		   colorbalance=rs=.3

   colorkey
       RGB colorspace color keying.

       The filter accepts the following options:

       color
	   The color which will be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01 matches only the exact key color, while 1.0 matches
	   everything.

       blend
	   Blend percentage.

	   0.0 makes pixels either fully transparent, or not transparent at
	   all.

	   Higher values result in semi-transparent pixels, with a higher
	   transparency the more similar the pixels color is to the key color.

       Examples

       ·   Make every green pixel in the input image transparent:

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

       ·   Overlay a greenscreen-video on top of a static background image.

		   ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

   colorlevels
       Adjust video input frames using levels.

       The filter accepts the following options:

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

       rimax
       gimax
       bimax
       aimax
	   Adjust red, green, blue and alpha input white point.  Allowed
	   ranges for options are "[-1.0, 1.0]". Defaults are 1.

	   Input levels are used to lighten highlights (bright tones), darken
	   shadows (dark tones), change the balance of bright and dark tones.

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

       romax
       gomax
       bomax
       aomax
	   Adjust red, green, blue and alpha output white point.  Allowed
	   ranges for options are "[0, 1.0]". Defaults are 1.

	   Output levels allows manual selection of a constrained output level
	   range.

       Examples

       ·   Make video output darker:

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

       ·   Increase contrast:

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

       ·   Make video output lighter:

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

       ·   Increase brightness:

		   colorlevels=romin=0.5:gomin=0.5:bomin=0.5

   colorchannelmixer
       Adjust video input frames by re-mixing color channels.

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

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

       The filter accepts the following options:

       rr
       rg
       rb
       ra  Adjust contribution of input red, green, blue and alpha channels
	   for output red channel.  Default is 1 for rr, and 0 for rg, rb and
	   ra.

       gr
       gg
       gb
       ga  Adjust contribution of input red, green, blue and alpha channels
	   for output green channel.  Default is 1 for gg, and 0 for gr, gb
	   and ga.

       br
       bg
       bb
       ba  Adjust contribution of input red, green, blue and alpha channels
	   for output blue channel.  Default is 1 for bb, and 0 for br, bg and
	   ba.

       ar
       ag
       ab
       aa  Adjust contribution of input red, green, blue and alpha channels
	   for output alpha channel.  Default is 1 for aa, and 0 for ar, ag
	   and ab.

	   Allowed ranges for options are "[-2.0, 2.0]".

       Examples

       ·   Convert source to grayscale:

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

       ·   Simulate sepia tones:

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

   colormatrix
       Convert color matrix.

       The filter accepts the following options:

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

	   The accepted values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt601
	       BT.601

	   bt470
	       BT.470

	   bt470bg
	       BT.470BG

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       For example to convert from BT.601 to SMPTE-240M, use the command:

	       colormatrix=bt601:smpte240m

   colorspace
       Convert colorspace, transfer characteristics or color primaries.  Input
       video needs to have an even size.

       The filter accepts the following options:

       all Specify all color properties at once.

	   The accepted values are:

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   bt601-6-525
	       BT.601-6 525

	   bt601-6-625
	       BT.601-6 625

	   bt709
	       BT.709

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       space
	   Specify output colorspace.

	   The accepted values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   ycgco
	       YCgCo

	   bt2020ncl
	       BT.2020 with non-constant luminance

       trc Specify output transfer characteristics.

	   The accepted values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   gamma22
	       Constant gamma of 2.2

	   gamma28
	       Constant gamma of 2.8

	   smpte170m
	       SMPTE-170M, BT.601-6 625 or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   srgb
	       SRGB

	   iec61966-2-1
	       iec61966-2-1

	   iec61966-2-4
	       iec61966-2-4

	   xvycc
	       xvycc

	   bt2020-10
	       BT.2020 for 10-bits content

	   bt2020-12
	       BT.2020 for 12-bits content

       primaries
	   Specify output color primaries.

	   The accepted values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   film
	       film

	   smpte431
	       SMPTE-431

	   smpte432
	       SMPTE-432

	   bt2020
	       BT.2020

	   jedec-p22
	       JEDEC P22 phosphors

       range
	   Specify output color range.

	   The accepted values are:

	   tv  TV (restricted) range

	   mpeg
	       MPEG (restricted) range

	   pc  PC (full) range

	   jpeg
	       JPEG (full) range

       format
	   Specify output color format.

	   The accepted values are:

	   yuv420p
	       YUV 4:2:0 planar 8-bits

	   yuv420p10
	       YUV 4:2:0 planar 10-bits

	   yuv420p12
	       YUV 4:2:0 planar 12-bits

	   yuv422p
	       YUV 4:2:2 planar 8-bits

	   yuv422p10
	       YUV 4:2:2 planar 10-bits

	   yuv422p12
	       YUV 4:2:2 planar 12-bits

	   yuv444p
	       YUV 4:4:4 planar 8-bits

	   yuv444p10
	       YUV 4:4:4 planar 10-bits

	   yuv444p12
	       YUV 4:4:4 planar 12-bits

       fast
	   Do a fast conversion, which skips gamma/primary correction. This
	   will take significantly less CPU, but will be mathematically
	   incorrect. To get output compatible with that produced by the
	   colormatrix filter, use fast=1.

       dither
	   Specify dithering mode.

	   The accepted values are:

	   none
	       No dithering

	   fsb Floyd-Steinberg dithering

       wpadapt
	   Whitepoint adaptation mode.

	   The accepted values are:

	   bradford
	       Bradford whitepoint adaptation

	   vonkries
	       von Kries whitepoint adaptation

	   identity
	       identity whitepoint adaptation (i.e. no whitepoint adaptation)

       iall
	   Override all input properties at once. Same accepted values as all.

       ispace
	   Override input colorspace. Same accepted values as space.

       iprimaries
	   Override input color primaries. Same accepted values as primaries.

       itrc
	   Override input transfer characteristics. Same accepted values as
	   trc.

       irange
	   Override input color range. Same accepted values as range.

       The filter converts the transfer characteristics, color space and color
       primaries to the specified user values. The output value, if not
       specified, is set to a default value based on the "all" property. If
       that property is also not specified, the filter will log an error. The
       output color range and format default to the same value as the input
       color range and format. The input transfer characteristics, color
       space, color primaries and color range should be set on the input data.
       If any of these are missing, the filter will log an error and no
       conversion will take place.

       For example to convert the input to SMPTE-240M, use the command:

	       colorspace=smpte240m

   convolution
       Apply convolution 3x3 or 5x5 filter.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set matrix for each plane.  Matrix is sequence of 9 or 25 signed
	   integers.

       0rdiv
       1rdiv
       2rdiv
       3rdiv
	   Set multiplier for calculated value for each plane.

       0bias
       1bias
       2bias
       3bias
	   Set bias for each plane. This value is added to the result of the
	   multiplication.  Useful for making the overall image brighter or
	   darker. Default is 0.0.

       Examples

       ·   Apply sharpen:

		   convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"

       ·   Apply blur:

		   convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"

       ·   Apply edge enhance:

		   convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"

       ·   Apply edge detect:

		   convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"

       ·   Apply laplacian edge detector which includes diagonals:

		   convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"

       ·   Apply emboss:

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

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

       The filter accepts the following options:

       planes
	   Set which planes to process.

       impulse
	   Set which impulse video frames will be processed, can be first or
	   all. Default is all.

       The "convolve" filter also supports the framesync options.

   copy
       Copy the input video source unchanged to the output. This is mainly
       useful for testing purposes.

   coreimage
       Video filtering on GPU using Apple's CoreImage API on OSX.

       Hardware acceleration is based on an OpenGL context. Usually, this
       means it is processed by video hardware. However, software-based OpenGL
       implementations exist which means there is no guarantee for hardware
       processing. It depends on the respective OSX.

       There are many filters and image generators provided by Apple that come
       with a large variety of options. The filter has to be referenced by its
       name along with its options.

       The coreimage filter accepts the following options:

       list_filters
	   List all available filters and generators along with all their
	   respective options as well as possible minimum and maximum values
	   along with the default values.

		   list_filters=true

       filter
	   Specify all filters by their respective name and options.  Use
	   list_filters to determine all valid filter names and options.
	   Numerical options are specified by a float value and are
	   automatically clamped to their respective value range.  Vector and
	   color options have to be specified by a list of space separated
	   float values. Character escaping has to be done.  A special option
	   name "default" is available to use default options for a filter.

	   It is required to specify either "default" or at least one of the
	   filter options.  All omitted options are used with their default
	   values.  The syntax of the filter string is as follows:

		   filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]

       output_rect
	   Specify a rectangle where the output of the filter chain is copied
	   into the input image. It is given by a list of space separated
	   float values:

		   output_rect=x\ y\ width\ height

	   If not given, the output rectangle equals the dimensions of the
	   input image.  The output rectangle is automatically cropped at the
	   borders of the input image. Negative values are valid for each
	   component.

		   output_rect=25\ 25\ 100\ 100

       Several filters can be chained for successive processing without GPU-
       HOST transfers allowing for fast processing of complex filter chains.
       Currently, only filters with zero (generators) or exactly one (filters)
       input image and one output image are supported. Also, transition
       filters are not yet usable as intended.

       Some filters generate output images with additional padding depending
       on the respective filter kernel. The padding is automatically removed
       to ensure the filter output has the same size as the input image.

       For image generators, the size of the output image is determined by the
       previous output image of the filter chain or the input image of the
       whole filterchain, respectively. The generators do not use the pixel
       information of this image to generate their output. However, the
       generated output is blended onto this image, resulting in partial or
       complete coverage of the output image.

       The coreimagesrc video source can be used for generating input images
       which are directly fed into the filter chain. By using it, providing
       input images by another video source or an input video is not required.

       Examples

       ·   List all filters available:

		   coreimage=list_filters=true

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

		   coreimage=filter=CIBoxBlur@default

       ·   Use a filter chain with CISepiaTone at default values and
	   CIVignetteEffect with its center at 100x100 and a radius of 50
	   pixels:

		   coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50

       ·   Use nullsrc and CIQRCodeGenerator to create a QR code for the
	   FFmpeg homepage, given as complete and escaped command-line for
	   Apple's standard bash shell:

		   ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

   crop
       Crop the input video to given dimensions.

       It accepts the following parameters:

       w, out_w
	   The width of the output video. It defaults to "iw".	This
	   expression is evaluated only once during the filter configuration,
	   or when the w or out_w command is sent.

       h, out_h
	   The height of the output video. It defaults to "ih".  This
	   expression is evaluated only once during the filter configuration,
	   or when the h or out_h command is sent.

       x   The horizontal position, in the input video, of the left edge of
	   the output video. It defaults to "(in_w-out_w)/2".  This expression
	   is evaluated per-frame.

       y   The vertical position, in the input video, of the top edge of the
	   output video.  It defaults to "(in_h-out_h)/2".  This expression is
	   evaluated per-frame.

       keep_aspect
	   If set to 1 will force the output display aspect ratio to be the
	   same of the input, by changing the output sample aspect ratio. It
	   defaults to 0.

       exact
	   Enable exact cropping. If enabled, subsampled videos will be
	   cropped at exact width/height/x/y as specified and will not be
	   rounded to nearest smaller value.  It defaults to 0.

       The out_w, out_h, x, y parameters are expressions containing the
       following constants:

       x
       y   The computed values for x and y. They are evaluated for each new
	   frame.

       in_w
       in_h
	   The input width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (cropped) width and height.

       ow
       oh  These are the same as out_w and out_h.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       n   The number of the input frame, starting from 0.

       pos the position in the file of the input frame, NAN if unknown

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
	   is unknown.

       The expression for out_w may depend on the value of out_h, and the
       expression for out_h may depend on out_w, but they cannot depend on x
       and y, as x and y are evaluated after out_w and out_h.

       The x and y parameters specify the expressions for the position of the
       top-left corner of the output (non-cropped) area. They are evaluated
       for each frame. If the evaluated value is not valid, it is approximated
       to the nearest valid value.

       The expression for x may depend on y, and the expression for y may
       depend on x.

       Examples

       ·   Crop area with size 100x100 at position (12,34).

		   crop=100:100:12:34

	   Using named options, the example above becomes:

		   crop=w=100:h=100:x=12:y=34

       ·   Crop the central input area with size 100x100:

		   crop=100:100

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

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

       ·   Crop the input video central square:

		   crop=out_w=in_h
		   crop=in_h

       ·   Delimit the rectangle with the top-left corner placed at position
	   100:100 and the right-bottom corner corresponding to the right-
	   bottom corner of the input image.

		   crop=in_w-100:in_h-100:100:100

       ·   Crop 10 pixels from the left and right borders, and 20 pixels from
	   the top and bottom borders

		   crop=in_w-2*10:in_h-2*20

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

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

       ·   Crop height for getting Greek harmony:

		   crop=in_w:1/PHI*in_w

       ·   Apply trembling effect:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

       ·   Apply erratic camera effect depending on timestamp:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"

       ·   Set x depending on the value of y:

		   crop=in_w/2:in_h/2:y:10+10*sin(n/10)

       Commands

       This filter supports the following commands:

       w, out_w
       h, out_h
       x
       y   Set width/height of the output video and the horizontal/vertical
	   position in the input video.  The command accepts the same syntax
	   of the corresponding option.

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

   cropdetect
       Auto-detect the crop size.

       It calculates the necessary cropping parameters and prints the
       recommended parameters via the logging system. The detected dimensions
       correspond to the non-black area of the input video.

       It accepts the following parameters:

       limit
	   Set higher black value threshold, which can be optionally specified
	   from nothing (0) to everything (255 for 8-bit based formats). An
	   intensity value greater to the set value is considered non-black.
	   It defaults to 24.  You can also specify a value between 0.0 and
	   1.0 which will be scaled depending on the bitdepth of the pixel
	   format.

       round
	   The value which the width/height should be divisible by. It
	   defaults to 16. The offset is automatically adjusted to center the
	   video. Use 2 to get only even dimensions (needed for 4:2:2 video).
	   16 is best when encoding to most video codecs.

       reset_count, reset
	   Set the counter that determines after how many frames cropdetect
	   will reset the previously detected largest video area and start
	   over to detect the current optimal crop area. Default value is 0.

	   This can be useful when channel logos distort the video area. 0
	   indicates 'never reset', and returns the largest area encountered
	   during playback.

   curves
       Apply color adjustments using curves.

       This filter is similar to the Adobe Photoshop and GIMP curves tools.
       Each component (red, green and blue) has its values defined by N key
       points tied from each other using a smooth curve. The x-axis represents
       the pixel values from the input frame, and the y-axis the new pixel
       values to be set for the output frame.

       By default, a component curve is defined by the two points (0;0) and
       (1;1). This creates a straight line where each original pixel value is
       "adjusted" to its own value, which means no change to the image.

       The filter allows you to redefine these two points and add some more. A
       new curve (using a natural cubic spline interpolation) will be define
       to pass smoothly through all these new coordinates. The new defined
       points needs to be strictly increasing over the x-axis, and their x and
       y values must be in the [0;1] interval.	If the computed curves
       happened to go outside the vector spaces, the values will be clipped
       accordingly.

       The filter accepts the following options:

       preset
	   Select one of the available color presets. This option can be used
	   in addition to the r, g, b parameters; in this case, the later
	   options takes priority on the preset values.  Available presets
	   are:

	   none
	   color_negative
	   cross_process
	   darker
	   increase_contrast
	   lighter
	   linear_contrast
	   medium_contrast
	   negative
	   strong_contrast
	   vintage

	   Default is "none".

       master, m
	   Set the master key points. These points will define a second pass
	   mapping. It is sometimes called a "luminance" or "value" mapping.
	   It can be used with r, g, b or all since it acts like a post-
	   processing LUT.

       red, r
	   Set the key points for the red component.

       green, g
	   Set the key points for the green component.

       blue, b
	   Set the key points for the blue component.

       all Set the key points for all components (not including master).  Can
	   be used in addition to the other key points component options. In
	   this case, the unset component(s) will fallback on this all
	   setting.

       psfile
	   Specify a Photoshop curves file (".acv") to import the settings
	   from.

       plot
	   Save Gnuplot script of the curves in specified file.

       To avoid some filtergraph syntax conflicts, each key points list need
       to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

       Examples

       ·   Increase slightly the middle level of blue:

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

       ·   Vintage effect:

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

	   Here we obtain the following coordinates for each components:

	   red "(0;0.11) (0.42;0.51) (1;0.95)"

	   green
	       "(0;0) (0.50;0.48) (1;1)"

	   blue
	       "(0;0.22) (0.49;0.44) (1;0.80)"

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

		   curves=preset=vintage

       ·   Or simply:

		   curves=vintage

       ·   Use a Photoshop preset and redefine the points of the green
	   component:

		   curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

       ·   Check out the curves of the "cross_process" profile using ffmpeg
	   and gnuplot:

		   ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
		   gnuplot -p /tmp/curves.plt

   datascope
       Video data analysis filter.

       This filter shows hexadecimal pixel values of part of video.

       The filter accepts the following options:

       size, s
	   Set output video size.

       x   Set x offset from where to pick pixels.

       y   Set y offset from where to pick pixels.

       mode
	   Set scope mode, can be one of the following:

	   mono
	       Draw hexadecimal pixel values with white color on black
	       background.

	   color
	       Draw hexadecimal pixel values with input video pixel color on
	       black background.

	   color2
	       Draw hexadecimal pixel values on color background picked from
	       input video, the text color is picked in such way so its always
	       visible.

       axis
	   Draw rows and columns numbers on left and top of video.

       opacity
	   Set background opacity.

   dctdnoiz
       Denoise frames using 2D DCT (frequency domain filtering).

       This filter is not designed for real time.

       The filter accepts the following options:

       sigma, s
	   Set the noise sigma constant.

	   This sigma defines a hard threshold of "3 * sigma"; every DCT
	   coefficient (absolute value) below this threshold with be dropped.

	   If you need a more advanced filtering, see expr.

	   Default is 0.

       overlap
	   Set number overlapping pixels for each block. Since the filter can
	   be slow, you may want to reduce this value, at the cost of a less
	   effective filter and the risk of various artefacts.

	   If the overlapping value doesn't permit processing the whole input
	   width or height, a warning will be displayed and according borders
	   won't be denoised.

	   Default value is blocksize-1, which is the best possible setting.

       expr, e
	   Set the coefficient factor expression.

	   For each coefficient of a DCT block, this expression will be
	   evaluated as a multiplier value for the coefficient.

	   If this is option is set, the sigma option will be ignored.

	   The absolute value of the coefficient can be accessed through the c
	   variable.

       n   Set the blocksize using the number of bits. "1<<n" defines the
	   blocksize, which is the width and height of the processed blocks.

	   The default value is 3 (8x8) and can be raised to 4 for a blocksize
	   of 16x16. Note that changing this setting has huge consequences on
	   the speed processing. Also, a larger block size does not
	   necessarily means a better de-noising.

       Examples

       Apply a denoise with a sigma of 4.5:

	       dctdnoiz=4.5

       The same operation can be achieved using the expression system:

	       dctdnoiz=e='gte(c, 4.5*3)'

       Violent denoise using a block size of "16x16":

	       dctdnoiz=15:n=4

   deband
       Remove banding artifacts from input video.  It works by replacing
       banded pixels with average value of referenced pixels.

       The filter accepts the following options:

       1thr
       2thr
       3thr
       4thr
	   Set banding detection threshold for each plane. Default is 0.02.
	   Valid range is 0.00003 to 0.5.  If difference between current pixel
	   and reference pixel is less than threshold, it will be considered
	   as banded.

       range, r
	   Banding detection range in pixels. Default is 16. If positive,
	   random number in range 0 to set value will be used. If negative,
	   exact absolute value will be used.  The range defines square of
	   four pixels around current pixel.

       direction, d
	   Set direction in radians from which four pixel will be compared. If
	   positive, random direction from 0 to set direction will be picked.
	   If negative, exact of absolute value will be picked. For example
	   direction 0, -PI or -2*PI radians will pick only pixels on same row
	   and -PI/2 will pick only pixels on same column.

       blur, b
	   If enabled, current pixel is compared with average value of all
	   four surrounding pixels. The default is enabled. If disabled
	   current pixel is compared with all four surrounding pixels. The
	   pixel is considered banded if only all four differences with
	   surrounding pixels are less than threshold.

       coupling, c
	   If enabled, current pixel is changed if and only if all pixel
	   components are banded, e.g. banding detection threshold is
	   triggered for all color components.	The default is disabled.

   decimate
       Drop duplicated frames at regular intervals.

       The filter accepts the following options:

       cycle
	   Set the number of frames from which one will be dropped. Setting
	   this to N means one frame in every batch of N frames will be
	   dropped.  Default is 5.

       dupthresh
	   Set the threshold for duplicate detection. If the difference metric
	   for a frame is less than or equal to this value, then it is
	   declared as duplicate. Default is 1.1

       scthresh
	   Set scene change threshold. Default is 15.

       blockx
       blocky
	   Set the size of the x and y-axis blocks used during metric
	   calculations.  Larger blocks give better noise suppression, but
	   also give worse detection of small movements. Must be a power of
	   two. Default is 32.

       ppsrc
	   Mark main input as a pre-processed input and activate clean source
	   input stream. This allows the input to be pre-processed with
	   various filters to help the metrics calculation while keeping the
	   frame selection lossless. When set to 1, the first stream is for
	   the pre-processed input, and the second stream is the clean source
	   from where the kept frames are chosen. Default is 0.

       chroma
	   Set whether or not chroma is considered in the metric calculations.
	   Default is 1.

   deflate
       Apply deflate effect to the video.

       This filter replaces the pixel by the local(3x3) average by taking into
       account only values lower than the pixel.

       It accepts the following options:

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

   deflicker
       Remove temporal frame luminance variations.

       It accepts the following options:

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

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

	   Available values are:

	   am  Arithmetic mean

	   gm  Geometric mean

	   hm  Harmonic mean

	   qm  Quadratic mean

	   cm  Cubic mean

	   pm  Power mean

	   median
	       Median

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

   dejudder
       Remove judder produced by partially interlaced telecined content.

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

       The option available in this filter is:

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

	   Accepts any integer greater than 1. Useful values are:

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

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

	   20  If a mixture of the two.

	   The default is 4.

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

       It accepts the following parameters:

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

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

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

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

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

       Examples

       ·   Set a rectangle covering the area with top left corner coordinates
	   0,0 and size 100x77, and a band of size 10:

		   delogo=x=0:y=0:w=100:h=77:band=10

   deshake
       Attempt to fix small changes in horizontal and/or vertical shift. This
       filter helps remove camera shake from hand-holding a camera, bumping a
       tripod, moving on a vehicle, etc.

       The filter accepts the following options:

       x
       y
       w
       h   Specify a rectangular area where to limit the search for motion
	   vectors.  If desired the search for motion vectors can be limited
	   to a rectangular area of the frame defined by its top left corner,
	   width and height. These parameters have the same meaning as the
	   drawbox filter which can be used to visualise the position of the
	   bounding box.

	   This is useful when simultaneous movement of subjects within the
	   frame might be confused for camera motion by the motion vector
	   search.

	   If any or all of x, y, w and h are set to -1 then the full frame is
	   used. This allows later options to be set without specifying the
	   bounding box for the motion vector search.

	   Default - search the whole frame.

       rx
       ry  Specify the maximum extent of movement in x and y directions in the
	   range 0-64 pixels. Default 16.

       edge
	   Specify how to generate pixels to fill blanks at the edge of the
	   frame. Available values are:

	   blank, 0
	       Fill zeroes at blank locations

	   original, 1
	       Original image at blank locations

	   clamp, 2
	       Extruded edge value at blank locations

	   mirror, 3
	       Mirrored edge at blank locations

	   Default value is mirror.

       blocksize
	   Specify the blocksize to use for motion search. Range 4-128 pixels,
	   default 8.

       contrast
	   Specify the contrast threshold for blocks. Only blocks with more
	   than the specified contrast (difference between darkest and
	   lightest pixels) will be considered. Range 1-255, default 125.

       search
	   Specify the search strategy. Available values are:

	   exhaustive, 0
	       Set exhaustive search

	   less, 1
	       Set less exhaustive search.

	   Default value is exhaustive.

       filename
	   If set then a detailed log of the motion search is written to the
	   specified file.

       opencl
	   If set to 1, specify using OpenCL capabilities, only available if
	   FFmpeg was configured with "--enable-opencl". Default value is 0.

   despill
       Remove unwanted contamination of foreground colors, caused by reflected
       color of greenscreen or bluescreen.

       This filter accepts the following options:

       type
	   Set what type of despill to use.

       mix Set how spillmap will be generated.

       expand
	   Set how much to get rid of still remaining spill.

       red Controls amount of red in spill area.

       green
	   Controls amount of green in spill area.  Should be -1 for
	   greenscreen.

       blue
	   Controls amount of blue in spill area.  Should be -1 for
	   bluescreen.

       brightness
	   Controls brightness of spill area, preserving colors.

       alpha
	   Modify alpha from generated spillmap.

   detelecine
       Apply an exact inverse of the telecine operation. It requires a
       predefined pattern specified using the pattern option which must be the
       same as that passed to the telecine filter.

       This filter accepts the following options:

       first_field
	   top, t
	       top field first

	   bottom, b
	       bottom field first The default value is "top".

       pattern
	   A string of numbers representing the pulldown pattern you wish to
	   apply.  The default value is 23.

       start_frame
	   A number representing position of the first frame with respect to
	   the telecine pattern. This is to be used if the stream is cut. The
	   default value is 0.

   dilation
       Apply dilation effect to the video.

       This filter replaces the pixel by the local(3x3) maximum.

       It accepts the following options:

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

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

	   Flags to local 3x3 coordinates maps like this:

	       1 2 3
	       4   5
	       6 7 8

   displace
       Displace pixels as indicated by second and third input stream.

       It takes three input streams and outputs one stream, the first input is
       the source, and second and third input are displacement maps.

       The second input specifies how much to displace pixels along the
       x-axis, while the third input specifies how much to displace pixels
       along the y-axis.  If one of displacement map streams terminates, last
       frame from that displacement map will be used.

       Note that once generated, displacements maps can be reused over and
       over again.

       A description of the accepted options follows.

       edge
	   Set displace behavior for pixels that are out of range.

	   Available values are:

	   blank
	       Missing pixels are replaced by black pixels.

	   smear
	       Adjacent pixels will spread out to replace missing pixels.

	   wrap
	       Out of range pixels are wrapped so they point to pixels of
	       other side.

	   mirror
	       Out of range pixels will be replaced with mirrored pixels.

	   Default is smear.

       Examples

       ·   Add ripple effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT

       ·   Add wave effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT

   drawbox
       Draw a colored box on the input image.

       It accepts the following parameters:

       x
       y   The expressions which specify the top left corner coordinates of
	   the box. It defaults to 0.

       width, w
       height, h
	   The expressions which specify the width and height of the box; if 0
	   they are interpreted as the input width and height. It defaults to
	   0.

       color, c
	   Specify the color of the box to write. For the general syntax of
	   this option, check the "Color" section in the ffmpeg-utils manual.
	   If the special value "invert" is used, the box edge color is the
	   same as the video with inverted luma.

       thickness, t
	   The expression which sets the thickness of the box edge. Default
	   value is 3.

	   See below for the list of accepted constants.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y offset coordinates where the box is drawn.

       w
       h   The width and height of the drawn box.

       t   The thickness of the drawn box.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

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

		   drawbox

       ·   Draw a box with color red and an opacity of 50%:

		   drawbox=10:20:200:60:red@0.5

	   The previous example can be specified as:

		   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       ·   Fill the box with pink color:

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

       ·   Draw a 2-pixel red 2.40:1 mask:

		   drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

   drawgrid
       Draw a grid on the input image.

       It accepts the following parameters:

       x
       y   The expressions which specify the coordinates of some point of grid
	   intersection (meant to configure offset). Both default to 0.

       width, w
       height, h
	   The expressions which specify the width and height of the grid
	   cell, if 0 they are interpreted as the input width and height,
	   respectively, minus "thickness", so image gets framed. Default to
	   0.

       color, c
	   Specify the color of the grid. For the general syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual. If
	   the special value "invert" is used, the grid color is the same as
	   the video with inverted luma.

       thickness, t
	   The expression which sets the thickness of the grid line. Default
	   value is 1.

	   See below for the list of accepted constants.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input grid cell width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y coordinates of some point of grid intersection (meant
	   to configure offset).

       w
       h   The width and height of the drawn cell.

       t   The thickness of the drawn cell.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

       ·   Draw a grid with cell 100x100 pixels, thickness 2 pixels, with
	   color red and an opacity of 50%:

		   drawgrid=width=100:height=100:thickness=2:color=red@0.5

       ·   Draw a white 3x3 grid with an opacity of 50%:

		   drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

   drawtext
       Draw a text string or text from a specified file on top of a video,
       using the libfreetype library.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-libfreetype".	To enable default font fallback and the font
       option you need to configure FFmpeg with "--enable-libfontconfig".  To
       enable the text_shaping option, you need to configure FFmpeg with
       "--enable-libfribidi".

       Syntax

       It accepts the following parameters:

       box Used to draw a box around text using the background color.  The
	   value must be either 1 (enable) or 0 (disable).  The default value
	   of box is 0.

       boxborderw
	   Set the width of the border to be drawn around the box using
	   boxcolor.  The default value of boxborderw is 0.

       boxcolor
	   The color to be used for drawing box around text. For the syntax of
	   this option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of boxcolor is "white".

       line_spacing
	   Set the line spacing in pixels of the border to be drawn around the
	   box using box.  The default value of line_spacing is 0.

       borderw
	   Set the width of the border to be drawn around the text using
	   bordercolor.  The default value of borderw is 0.

       bordercolor
	   Set the color to be used for drawing border around text. For the
	   syntax of this option, check the "Color" section in the ffmpeg-
	   utils manual.

	   The default value of bordercolor is "black".

       expansion
	   Select how the text is expanded. Can be either "none", "strftime"
	   (deprecated) or "normal" (default). See the drawtext_expansion,
	   Text expansion section below for details.

       basetime
	   Set a start time for the count. Value is in microseconds. Only
	   applied in the deprecated strftime expansion mode. To emulate in
	   normal expansion mode use the "pts" function, supplying the start
	   time (in seconds) as the second argument.

       fix_bounds
	   If true, check and fix text coords to avoid clipping.

       fontcolor
	   The color to be used for drawing fonts. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of fontcolor is "black".

       fontcolor_expr
	   String which is expanded the same way as text to obtain dynamic
	   fontcolor value. By default this option has empty value and is not
	   processed. When this option is set, it overrides fontcolor option.

       font
	   The font family to be used for drawing text. By default Sans.

       fontfile
	   The font file to be used for drawing text. The path must be
	   included.  This parameter is mandatory if the fontconfig support is
	   disabled.

       alpha
	   Draw the text applying alpha blending. The value can be a number
	   between 0.0 and 1.0.  The expression accepts the same variables x,
	   y as well.  The default value is 1.	Please see fontcolor_expr.

       fontsize
	   The font size to be used for drawing text.  The default value of
	   fontsize is 16.

       text_shaping
	   If set to 1, attempt to shape the text (for example, reverse the
	   order of right-to-left text and join Arabic characters) before
	   drawing it.	Otherwise, just draw the text exactly as given.  By
	   default 1 (if supported).

       ft_load_flags
	   The flags to be used for loading the fonts.

	   The flags map the corresponding flags supported by libfreetype, and
	   are a combination of the following values:

	   default
	   no_scale
	   no_hinting
	   render
	   no_bitmap
	   vertical_layout
	   force_autohint
	   crop_bitmap
	   pedantic
	   ignore_global_advance_width
	   no_recurse
	   ignore_transform
	   monochrome
	   linear_design
	   no_autohint

	   Default value is "default".

	   For more information consult the documentation for the FT_LOAD_*
	   libfreetype flags.

       shadowcolor
	   The color to be used for drawing a shadow behind the drawn text.
	   For the syntax of this option, check the "Color" section in the
	   ffmpeg-utils manual.

	   The default value of shadowcolor is "black".

       shadowx
       shadowy
	   The x and y offsets for the text shadow position with respect to
	   the position of the text. They can be either positive or negative
	   values. The default value for both is "0".

       start_number
	   The starting frame number for the n/frame_num variable. The default
	   value is "0".

       tabsize
	   The size in number of spaces to use for rendering the tab.  Default
	   value is 4.

       timecode
	   Set the initial timecode representation in "hh:mm:ss[:;.]ff"
	   format. It can be used with or without text parameter.
	   timecode_rate option must be specified.

       timecode_rate, rate, r
	   Set the timecode frame rate (timecode only).

       tc24hmax
	   If set to 1, the output of the timecode option will wrap around at
	   24 hours.  Default is 0 (disabled).

       text
	   The text string to be drawn. The text must be a sequence of UTF-8
	   encoded characters.	This parameter is mandatory if no file is
	   specified with the parameter textfile.

       textfile
	   A text file containing text to be drawn. The text must be a
	   sequence of UTF-8 encoded characters.

	   This parameter is mandatory if no text string is specified with the
	   parameter text.

	   If both text and textfile are specified, an error is thrown.

       reload
	   If set to 1, the textfile will be reloaded before each frame.  Be
	   sure to update it atomically, or it may be read partially, or even
	   fail.

       x
       y   The expressions which specify the offsets where text will be drawn
	   within the video frame. They are relative to the top/left border of
	   the output image.

	   The default value of x and y is "0".

	   See below for the list of accepted constants and functions.

       The parameters for x and y are expressions containing the following
       constants and functions:

       dar input display aspect ratio, it is the same as (w / h) * sar

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       line_h, lh
	   the height of each text line

       main_h, h, H
	   the input height

       main_w, w, W
	   the input width

       max_glyph_a, ascent
	   the maximum distance from the baseline to the highest/upper grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  It is a positive value, due to the grid's
	   orientation with the Y axis upwards.

       max_glyph_d, descent
	   the maximum distance from the baseline to the lowest grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  This is a negative value, due to the grid's
	   orientation, with the Y axis upwards.

       max_glyph_h
	   maximum glyph height, that is the maximum height for all the glyphs
	   contained in the rendered text, it is equivalent to ascent -
	   descent.

       max_glyph_w
	   maximum glyph width, that is the maximum width for all the glyphs
	   contained in the rendered text

       n   the number of input frame, starting from 0

       rand(min, max)
	   return a random number included between min and max

       sar The input sample aspect ratio.

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

       text_h, th
	   the height of the rendered text

       text_w, tw
	   the width of the rendered text

       x
       y   the x and y offset coordinates where the text is drawn.

	   These parameters allow the x and y expressions to refer each other,
	   so you can for example specify "y=x/dar".

       Text expansion

       If expansion is set to "strftime", the filter recognizes strftime()
       sequences in the provided text and expands them accordingly. Check the
       documentation of strftime(). This feature is deprecated.

       If expansion is set to "none", the text is printed verbatim.

       If expansion is set to "normal" (which is the default), the following
       expansion mechanism is used.

       The backslash character \, followed by any character, always expands to
       the second character.

       Sequences of the form "%{...}" are expanded. The text between the
       braces is a function name, possibly followed by arguments separated by
       ':'.  If the arguments contain special characters or delimiters (':' or
       '}'), they should be escaped.

       Note that they probably must also be escaped as the value for the text
       option in the filter argument string and as the filter argument in the
       filtergraph description, and possibly also for the shell, that makes up
       to four levels of escaping; using a text file avoids these problems.

       The following functions are available:

       expr, e
	   The expression evaluation result.

	   It must take one argument specifying the expression to be
	   evaluated, which accepts the same constants and functions as the x
	   and y values. Note that not all constants should be used, for
	   example the text size is not known when evaluating the expression,
	   so the constants text_w and text_h will have an undefined value.

       expr_int_format, eif
	   Evaluate the expression's value and output as formatted integer.

	   The first argument is the expression to be evaluated, just as for
	   the expr function.  The second argument specifies the output
	   format. Allowed values are x, X, d and u. They are treated exactly
	   as in the "printf" function.  The third parameter is optional and
	   sets the number of positions taken by the output.  It can be used
	   to add padding with zeros from the left.

       gmtime
	   The time at which the filter is running, expressed in UTC.  It can
	   accept an argument: a strftime() format string.

       localtime
	   The time at which the filter is running, expressed in the local
	   time zone.  It can accept an argument: a strftime() format string.

       metadata
	   Frame metadata. Takes one or two arguments.

	   The first argument is mandatory and specifies the metadata key.

	   The second argument is optional and specifies a default value, used
	   when the metadata key is not found or empty.

       n, frame_num
	   The frame number, starting from 0.

       pict_type
	   A 1 character description of the current picture type.

       pts The timestamp of the current frame.	It can take up to three
	   arguments.

	   The first argument is the format of the timestamp; it defaults to
	   "flt" for seconds as a decimal number with microsecond accuracy;
	   "hms" stands for a formatted [-]HH:MM:SS.mmm timestamp with
	   millisecond accuracy.  "gmtime" stands for the timestamp of the
	   frame formatted as UTC time; "localtime" stands for the timestamp
	   of the frame formatted as local time zone time.

	   The second argument is an offset added to the timestamp.

	   If the format is set to "localtime" or "gmtime", a third argument
	   may be supplied: a strftime() format string.  By default, YYYY-MM-
	   DD HH:MM:SS format will be used.

       Examples

       ·   Draw "Test Text" with font FreeSerif, using the default values for
	   the optional parameters.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

       ·   Draw 'Test Text' with font FreeSerif of size 24 at position x=100
	   and y=50 (counting from the top-left corner of the screen), text is
	   yellow with a red box around it. Both the text and the box have an
	   opacity of 20%.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
			     x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

	   Note that the double quotes are not necessary if spaces are not
	   used within the parameter list.

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

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

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

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

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

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

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

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

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

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

       ·   Show text for 1 second every 3 seconds:

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

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

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

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

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

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

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

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

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

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

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

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

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

       The filter accepts the following options:

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

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

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

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

       mode
	   Define the drawing mode.

	   wires
	       Draw white/gray wires on black background.

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

	   Default value is wires.

       Examples

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

		   edgedetect=low=0.1:high=0.4

       ·   Painting effect without thresholding:

		   edgedetect=mode=colormix:high=0

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

       The filter accepts the following options:

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

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

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

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

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

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

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

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

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

	   It accepts the following values:

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

	   frame
	       evaluate expressions for each incoming frame

	   Default value is init.

       The expressions accept the following parameters:

       n   frame count of the input frame starting from 0

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

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

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

       Commands

       The filter supports the following commands:

       contrast
	   Set the contrast expression.

       brightness
	   Set the brightness expression.

       saturation
	   Set the saturation expression.

       gamma
	   Set the gamma expression.

       gamma_r
	   Set the gamma_r expression.

       gamma_g
	   Set gamma_g expression.

       gamma_b
	   Set gamma_b expression.

       gamma_weight
	   Set gamma_weight expression.

	   The command accepts the same syntax of the corresponding option.

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

   erosion
       Apply erosion effect to the video.

       This filter replaces the pixel by the local(3x3) minimum.

       It accepts the following options:

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

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

	   Flags to local 3x3 coordinates maps like this:

	       1 2 3
	       4   5
	       6 7 8

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

       The filter accepts the following option:

       planes
	   Set plane(s) to extract.

	   Available values for planes are:

	   y
	   u
	   v
	   a
	   r
	   g
	   b

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

       Examples

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

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

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

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

       This filter accepts the following options.

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

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

       seed, s
	   Set a random seed, must be an integer included between 0 and
	   UINT32_MAX. If not specified, or if explicitly set to -1, the
	   filter will try to use a good random seed on a best effort basis.

       pal8
	   Set pal8 output pixel format. This option does not work with
	   codebook length greater than 256.

   fade
       Apply a fade-in/out effect to the input video.

       It accepts the following parameters:

       type, t
	   The effect type can be either "in" for a fade-in, or "out" for a
	   fade-out effect.  Default is "in".

       start_frame, s
	   Specify the number of the frame to start applying the fade effect
	   at. Default is 0.

       nb_frames, n
	   The number of frames that the fade effect lasts. At the end of the
	   fade-in effect, the output video will have the same intensity as
	   the input video.  At the end of the fade-out transition, the output
	   video will be filled with the selected color.  Default is 25.

       alpha
	   If set to 1, fade only alpha channel, if one exists on the input.
	   Default value is 0.

       start_time, st
	   Specify the timestamp (in seconds) of the frame to start to apply
	   the fade effect. If both start_frame and start_time are specified,
	   the fade will start at whichever comes last.  Default is 0.

       duration, d
	   The number of seconds for which the fade effect has to last. At the
	   end of the fade-in effect the output video will have the same
	   intensity as the input video, at the end of the fade-out transition
	   the output video will be filled with the selected color.  If both
	   duration and nb_frames are specified, duration is used. Default is
	   0 (nb_frames is used by default).

       color, c
	   Specify the color of the fade. Default is "black".

       Examples

       ·   Fade in the first 30 frames of video:

		   fade=in:0:30

	   The command above is equivalent to:

		   fade=t=in:s=0:n=30

       ·   Fade out the last 45 frames of a 200-frame video:

		   fade=out:155:45
		   fade=type=out:start_frame=155:nb_frames=45

       ·   Fade in the first 25 frames and fade out the last 25 frames of a
	   1000-frame video:

		   fade=in:0:25, fade=out:975:25

       ·   Make the first 5 frames yellow, then fade in from frame 5-24:

		   fade=in:5:20:color=yellow

       ·   Fade in alpha over first 25 frames of video:

		   fade=in:0:25:alpha=1

       ·   Make the first 5.5 seconds black, then fade in for 0.5 seconds:

		   fade=t=in:st=5.5:d=0.5

   fftfilt
       Apply arbitrary expressions to samples in frequency domain

       dc_Y
	   Adjust the dc value (gain) of the luma plane of the image. The
	   filter accepts an integer value in range 0 to 1000. The default
	   value is set to 0.

       dc_U
	   Adjust the dc value (gain) of the 1st chroma plane of the image.
	   The filter accepts an integer value in range 0 to 1000. The default
	   value is set to 0.

       dc_V
	   Adjust the dc value (gain) of the 2nd chroma plane of the image.
	   The filter accepts an integer value in range 0 to 1000. The default
	   value is set to 0.

       weight_Y
	   Set the frequency domain weight expression for the luma plane.

       weight_U
	   Set the frequency domain weight expression for the 1st chroma
	   plane.

       weight_V
	   Set the frequency domain weight expression for the 2nd chroma
	   plane.

       eval
	   Set when the expressions are evaluated.

	   It accepts the following values:

	   init
	       Only evaluate expressions once during the filter
	       initialization.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

	   The filter accepts the following variables:

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       N   The number of input frame, starting from 0.

       Examples

       ·   High-pass:

		   fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

       ·   Low-pass:

		   fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

       ·   Sharpen:

		   fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

       ·   Blur:

		   fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'

   field
       Extract a single field from an interlaced image using stride arithmetic
       to avoid wasting CPU time. The output frames are marked as non-
       interlaced.

       The filter accepts the following options:

       type
	   Specify whether to extract the top (if the value is 0 or "top") or
	   the bottom field (if the value is 1 or "bottom").

   fieldhint
       Create new frames by copying the top and bottom fields from surrounding
       frames supplied as numbers by the hint file.

       hint
	   Set file containing hints: absolute/relative frame numbers.

	   There must be one line for each frame in a clip. Each line must
	   contain two numbers separated by the comma, optionally followed by
	   "-" or "+".	Numbers supplied on each line of file can not be out
	   of [N-1,N+1] where N is current frame number for "absolute" mode or
	   out of [-1, 1] range for "relative" mode. First number tells from
	   which frame to pick up top field and second number tells from which
	   frame to pick up bottom field.

	   If optionally followed by "+" output frame will be marked as
	   interlaced, else if followed by "-" output frame will be marked as
	   progressive, else it will be marked same as input frame.  If line
	   starts with "#" or ";" that line is skipped.

       mode
	   Can be item "absolute" or "relative". Default is "absolute".

       Example of first several lines of "hint" file for "relative" mode:

	       0,0 - # first frame
	       1,0 - # second frame, use third's frame top field and second's frame bottom field
	       1,0 - # third frame, use fourth's frame top field and third's frame bottom field
	       1,0 -
	       0,0 -
	       0,0 -
	       1,0 -
	       1,0 -
	       1,0 -
	       0,0 -
	       0,0 -
	       1,0 -
	       1,0 -
	       1,0 -
	       0,0 -

   fieldmatch
       Field matching filter for inverse telecine. It is meant to reconstruct
       the progressive frames from a telecined stream. The filter does not
       drop duplicated frames, so to achieve a complete inverse telecine
       "fieldmatch" needs to be followed by a decimation filter such as
       decimate in the filtergraph.

       The separation of the field matching and the decimation is notably
       motivated by the possibility of inserting a de-interlacing filter
       fallback between the two.  If the source has mixed telecined and real
       interlaced content, "fieldmatch" will not be able to match fields for
       the interlaced parts.  But these remaining combed frames will be marked
       as interlaced, and thus can be de-interlaced by a later filter such as
       yadif before decimation.

       In addition to the various configuration options, "fieldmatch" can take
       an optional second stream, activated through the ppsrc option. If
       enabled, the frames reconstruction will be based on the fields and
       frames from this second stream. This allows the first input to be pre-
       processed in order to help the various algorithms of the filter, while
       keeping the output lossless (assuming the fields are matched properly).
       Typically, a field-aware denoiser, or brightness/contrast adjustments
       can help.

       Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth
       project) and VIVTC/VFM (VapourSynth project). The later is a light
       clone of TFM from which "fieldmatch" is based on. While the semantic
       and usage are very close, some behaviour and options names can differ.

       The decimate filter currently only works for constant frame rate input.
       If your input has mixed telecined (30fps) and progressive content with
       a lower framerate like 24fps use the following filterchain to produce
       the necessary cfr stream:
       "dejudder,fps=30000/1001,fieldmatch,decimate".

       The filter accepts the following options:

       order
	   Specify the assumed field order of the input stream. Available
	   values are:

	   auto
	       Auto detect parity (use FFmpeg's internal parity value).

	   bff Assume bottom field first.

	   tff Assume top field first.

	   Note that it is sometimes recommended not to trust the parity
	   announced by the stream.

	   Default value is auto.

       mode
	   Set the matching mode or strategy to use. pc mode is the safest in
	   the sense that it won't risk creating jerkiness due to duplicate
	   frames when possible, but if there are bad edits or blended fields
	   it will end up outputting combed frames when a good match might
	   actually exist. On the other hand, pcn_ub mode is the most risky in
	   terms of creating jerkiness, but will almost always find a good
	   frame if there is one. The other values are all somewhere in
	   between pc and pcn_ub in terms of risking jerkiness and creating
	   duplicate frames versus finding good matches in sections with bad
	   edits, orphaned fields, blended fields, etc.

	   More details about p/c/n/u/b are available in p/c/n/u/b meaning
	   section.

	   Available values are:

	   pc  2-way matching (p/c)

	   pc_n
	       2-way matching, and trying 3rd match if still combed (p/c + n)

	   pc_u
	       2-way matching, and trying 3rd match (same order) if still
	       combed (p/c + u)

	   pc_n_ub
	       2-way matching, trying 3rd match if still combed, and trying
	       4th/5th matches if still combed (p/c + n + u/b)

	   pcn 3-way matching (p/c/n)

	   pcn_ub
	       3-way matching, and trying 4th/5th matches if all 3 of the
	       original matches are detected as combed (p/c/n + u/b)

	   The parenthesis at the end indicate the matches that would be used
	   for that mode assuming order=tff (and field on auto or top).

	   In terms of speed pc mode is by far the fastest and pcn_ub is the
	   slowest.

	   Default value is pc_n.

       ppsrc
	   Mark the main input stream as a pre-processed input, and enable the
	   secondary input stream as the clean source to pick the fields from.
	   See the filter introduction for more details. It is similar to the
	   clip2 feature from VFM/TFM.

	   Default value is 0 (disabled).

       field
	   Set the field to match from. It is recommended to set this to the
	   same value as order unless you experience matching failures with
	   that setting. In certain circumstances changing the field that is
	   used to match from can have a large impact on matching performance.
	   Available values are:

	   auto
	       Automatic (same value as order).

	   bottom
	       Match from the bottom field.

	   top Match from the top field.

	   Default value is auto.

       mchroma
	   Set whether or not chroma is included during the match comparisons.
	   In most cases it is recommended to leave this enabled. You should
	   set this to 0 only if your clip has bad chroma problems such as
	   heavy rainbowing or other artifacts. Setting this to 0 could also
	   be used to speed things up at the cost of some accuracy.

	   Default value is 1.

       y0
       y1  These define an exclusion band which excludes the lines between y0
	   and y1 from being included in the field matching decision. An
	   exclusion band can be used to ignore subtitles, a logo, or other
	   things that may interfere with the matching. y0 sets the starting
	   scan line and y1 sets the ending line; all lines in between y0 and
	   y1 (including y0 and y1) will be ignored. Setting y0 and y1 to the
	   same value will disable the feature.  y0 and y1 defaults to 0.

       scthresh
	   Set the scene change detection threshold as a percentage of maximum
	   change on the luma plane. Good values are in the "[8.0, 14.0]"
	   range. Scene change detection is only relevant in case
	   combmatch=sc.  The range for scthresh is "[0.0, 100.0]".

	   Default value is 12.0.

       combmatch
	   When combatch is not none, "fieldmatch" will take into account the
	   combed scores of matches when deciding what match to use as the
	   final match. Available values are:

	   none
	       No final matching based on combed scores.

	   sc  Combed scores are only used when a scene change is detected.

	   full
	       Use combed scores all the time.

	   Default is sc.

       combdbg
	   Force "fieldmatch" to calculate the combed metrics for certain
	   matches and print them. This setting is known as micout in TFM/VFM
	   vocabulary.	Available values are:

	   none
	       No forced calculation.

	   pcn Force p/c/n calculations.

	   pcnub
	       Force p/c/n/u/b calculations.

	   Default value is none.

       cthresh
	   This is the area combing threshold used for combed frame detection.
	   This essentially controls how "strong" or "visible" combing must be
	   to be detected.  Larger values mean combing must be more visible
	   and smaller values mean combing can be less visible or strong and
	   still be detected. Valid settings are from "-1" (every pixel will
	   be detected as combed) to 255 (no pixel will be detected as
	   combed). This is basically a pixel difference value. A good range
	   is "[8, 12]".

	   Default value is 9.

       chroma
	   Sets whether or not chroma is considered in the combed frame
	   decision.  Only disable this if your source has chroma problems
	   (rainbowing, etc.) that are causing problems for the combed frame
	   detection with chroma enabled. Actually, using chroma=0 is usually
	   more reliable, except for the case where there is chroma only
	   combing in the source.

	   Default value is 0.

       blockx
       blocky
	   Respectively set the x-axis and y-axis size of the window used
	   during combed frame detection. This has to do with the size of the
	   area in which combpel pixels are required to be detected as combed
	   for a frame to be declared combed. See the combpel parameter
	   description for more info.  Possible values are any number that is
	   a power of 2 starting at 4 and going up to 512.

	   Default value is 16.

       combpel
	   The number of combed pixels inside any of the blocky by blockx size
	   blocks on the frame for the frame to be detected as combed. While
	   cthresh controls how "visible" the combing must be, this setting
	   controls "how much" combing there must be in any localized area (a
	   window defined by the blockx and blocky settings) on the frame.
	   Minimum value is 0 and maximum is "blocky x blockx" (at which point
	   no frames will ever be detected as combed). This setting is known
	   as MI in TFM/VFM vocabulary.

	   Default value is 80.

       p/c/n/u/b meaning

       p/c/n

       We assume the following telecined stream:

	       Top fields:     1 2 2 3 4
	       Bottom fields:  1 2 3 4 4

       The numbers correspond to the progressive frame the fields relate to.
       Here, the first two frames are progressive, the 3rd and 4th are combed,
       and so on.

       When "fieldmatch" is configured to run a matching from bottom
       (field=bottom) this is how this input stream get transformed:

	       Input stream:
			       T     1 2 2 3 4
			       B     1 2 3 4 4	 <-- matching reference

	       Matches: 	     c c n n c

	       Output stream:
			       T     1 2 3 4 4
			       B     1 2 3 4 4

       As a result of the field matching, we can see that some frames get
       duplicated.  To perform a complete inverse telecine, you need to rely
       on a decimation filter after this operation. See for instance the
       decimate filter.

       The same operation now matching from top fields (field=top) looks like
       this:

	       Input stream:
			       T     1 2 2 3 4	 <-- matching reference
			       B     1 2 3 4 4

	       Matches: 	     c c p p c

	       Output stream:
			       T     1 2 2 3 4
			       B     1 2 2 3 4

       In these examples, we can see what p, c and n mean; basically, they
       refer to the frame and field of the opposite parity:

       *<p matches the field of the opposite parity in the previous frame>
       *<c matches the field of the opposite parity in the current frame>
       *<n matches the field of the opposite parity in the next frame>

       u/b

       The u and b matching are a bit special in the sense that they match
       from the opposite parity flag. In the following examples, we assume
       that we are currently matching the 2nd frame (Top:2, bottom:2).
       According to the match, a 'x' is placed above and below each matched
       fields.

       With bottom matching (field=bottom):

	       Match:		c	  p	      n 	 b	    u

				x	x		x	 x	    x
		 Top	      1 2 2	1 2 2	    1 2 2      1 2 2	  1 2 2
		 Bottom       1 2 3	1 2 3	    1 2 3      1 2 3	  1 2 3
				x	  x	      x        x	      x

	       Output frames:
				2	   1	      2 	 2	    2
				2	   2	      2 	 1	    3

       With top matching (field=top):

	       Match:		c	  p	      n 	 b	    u

				x	  x	      x        x	      x
		 Top	      1 2 2	1 2 2	    1 2 2      1 2 2	  1 2 2
		 Bottom       1 2 3	1 2 3	    1 2 3      1 2 3	  1 2 3
				x	x		x	 x	    x

	       Output frames:
				2	   2	      2 	 1	    2
				2	   1	      3 	 2	    2

       Examples

       Simple IVTC of a top field first telecined stream:

	       fieldmatch=order=tff:combmatch=none, decimate

       Advanced IVTC, with fallback on yadif for still combed frames:

	       fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

   fieldorder
       Transform the field order of the input video.

       It accepts the following parameters:

       order
	   The output field order. Valid values are tff for top field first or
	   bff for bottom field first.

       The default value is tff.

       The transformation is done by shifting the picture content up or down
       by one line, and filling the remaining line with appropriate picture
       content.  This method is consistent with most broadcast field order
       converters.

       If the input video is not flagged as being interlaced, or it is already
       flagged as being of the required output field order, then this filter
       does not alter the incoming video.

       It is very useful when converting to or from PAL DV material, which is
       bottom field first.

       For example:

	       ffmpeg -i in.vob -vf "fieldorder=bff" out.dv

   fifo, afifo
       Buffer input images and send them when they are requested.

       It is mainly useful when auto-inserted by the libavfilter framework.

       It does not take parameters.

   find_rect
       Find a rectangular object

       It accepts the following options:

       object
	   Filepath of the object image, needs to be in gray8.

       threshold
	   Detection threshold, default is 0.5.

       mipmaps
	   Number of mipmaps, default is 3.

       xmin, ymin, xmax, ymax
	   Specifies the rectangle in which to search.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

		   ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

   cover_rect
       Cover a rectangular object

       It accepts the following options:

       cover
	   Filepath of the optional cover image, needs to be in yuv420.

       mode
	   Set covering mode.

	   It accepts the following values:

	   cover
	       cover it by the supplied image

	   blur
	       cover it by interpolating the surrounding pixels

	   Default value is blur.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

		   ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

   floodfill
       Flood area with values of same pixel components with another values.

       It accepts the following options:

       x   Set pixel x coordinate.

       y   Set pixel y coordinate.

       s0  Set source #0 component value.

       s1  Set source #1 component value.

       s2  Set source #2 component value.

       s3  Set source #3 component value.

       d0  Set destination #0 component value.

       d1  Set destination #1 component value.

       d2  Set destination #2 component value.

       d3  Set destination #3 component value.

   format
       Convert the input video to one of the specified pixel formats.
       Libavfilter will try to pick one that is suitable as input to the next
       filter.

       It accepts the following parameters:

       pix_fmts
	   A '|'-separated list of pixel format names, such as
	   "pix_fmts=yuv420p|monow|rgb24".

       Examples

       ·   Convert the input video to the yuv420p format

		   format=pix_fmts=yuv420p

	   Convert the input video to any of the formats in the list

		   format=pix_fmts=yuv420p|yuv444p|yuv410p

   fps
       Convert the video to specified constant frame rate by duplicating or
       dropping frames as necessary.

       It accepts the following parameters:

       fps The desired output frame rate. The default is 25.

       start_time
	   Assume the first PTS should be the given value, in seconds. This
	   allows for padding/trimming at the start of stream. By default, no
	   assumption is made about the first frame's expected PTS, so no
	   padding or trimming is done.  For example, this could be set to 0
	   to pad the beginning with duplicates of the first frame if a video
	   stream starts after the audio stream or to trim any frames with a
	   negative PTS.

       round
	   Timestamp (PTS) rounding method.

	   Possible values are:

	   zero
	       round towards 0

	   inf round away from 0

	   down
	       round towards -infinity

	   up  round towards +infinity

	   near
	       round to nearest

	   The default is "near".

       eof_action
	   Action performed when reading the last frame.

	   Possible values are:

	   round
	       Use same timestamp rounding method as used for other frames.

	   pass
	       Pass through last frame if input duration has not been reached
	       yet.

	   The default is "round".

       Alternatively, the options can be specified as a flat string:
       fps[:start_time[:round]].

       See also the setpts filter.

       Examples

       ·   A typical usage in order to set the fps to 25:

		   fps=fps=25

       ·   Sets the fps to 24, using abbreviation and rounding method to round
	   to nearest:

		   fps=fps=film:round=near

   framepack
       Pack two different video streams into a stereoscopic video, setting
       proper metadata on supported codecs. The two views should have the same
       size and framerate and processing will stop when the shorter video
       ends. Please note that you may conveniently adjust view properties with
       the scale and fps filters.

       It accepts the following parameters:

       format
	   The desired packing format. Supported values are:

	   sbs The views are next to each other (default).

	   tab The views are on top of each other.

	   lines
	       The views are packed by line.

	   columns
	       The views are packed by column.

	   frameseq
	       The views are temporally interleaved.

       Some examples:

	       # Convert left and right views into a frame-sequential video
	       ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

	       # Convert views into a side-by-side video with the same output resolution as the input
	       ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

   framerate
       Change the frame rate by interpolating new video output frames from the
       source frames.

       This filter is not designed to function correctly with interlaced
       media. If you wish to change the frame rate of interlaced media then
       you are required to deinterlace before this filter and re-interlace
       after this filter.

       A description of the accepted options follows.

       fps Specify the output frames per second. This option can also be
	   specified as a value alone. The default is 50.

       interp_start
	   Specify the start of a range where the output frame will be created
	   as a linear interpolation of two frames. The range is [0-255], the
	   default is 15.

       interp_end
	   Specify the end of a range where the output frame will be created
	   as a linear interpolation of two frames. The range is [0-255], the
	   default is 240.

       scene
	   Specify the level at which a scene change is detected as a value
	   between 0 and 100 to indicate a new scene; a low value reflects a
	   low probability for the current frame to introduce a new scene,
	   while a higher value means the current frame is more likely to be
	   one.  The default is 7.

       flags
	   Specify flags influencing the filter process.

	   Available value for flags is:

	   scene_change_detect, scd
	       Enable scene change detection using the value of the option
	       scene.  This flag is enabled by default.

   framestep
       Select one frame every N-th frame.

       This filter accepts the following option:

       step
	   Select frame after every "step" frames.  Allowed values are
	   positive integers higher than 0. Default value is 1.

   frei0r
       Apply a frei0r effect to the input video.

       To enable the compilation of this filter, you need to install the
       frei0r header and configure FFmpeg with "--enable-frei0r".

       It accepts the following parameters:

       filter_name
	   The name of the frei0r effect to load. If the environment variable
	   FREI0R_PATH is defined, the frei0r effect is searched for in each
	   of the directories specified by the colon-separated list in
	   FREI0R_PATH.  Otherwise, the standard frei0r paths are searched, in
	   this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/,
	   /usr/lib/frei0r-1/.

       filter_params
	   A '|'-separated list of parameters to pass to the frei0r effect.

       A frei0r effect parameter can be a boolean (its value is either "y" or
       "n"), a double, a color (specified as R/G/B, where R, G, and B are
       floating point numbers between 0.0 and 1.0, inclusive) or by a color
       description specified in the "Color" section in the ffmpeg-utils
       manual), a position (specified as X/Y, where X and Y are floating point
       numbers) and/or a string.

       The number and types of parameters depend on the loaded effect. If an
       effect parameter is not specified, the default value is set.

       Examples

       ·   Apply the distort0r effect, setting the first two double
	   parameters:

		   frei0r=filter_name=distort0r:filter_params=0.5|0.01

       ·   Apply the colordistance effect, taking a color as the first
	   parameter:

		   frei0r=colordistance:0.2/0.3/0.4
		   frei0r=colordistance:violet
		   frei0r=colordistance:0x112233

       ·   Apply the perspective effect, specifying the top left and top right
	   image positions:

		   frei0r=perspective:0.2/0.2|0.8/0.2

       For more information, see <http://frei0r.dyne.org>

   fspp
       Apply fast and simple postprocessing. It is a faster version of spp.

       It splits (I)DCT into horizontal/vertical passes. Unlike the simple
       post- processing filter, one of them is performed once per block, not
       per pixel.  This allows for much higher speed.

       The filter accepts the following options:

       quality
	   Set quality. This option defines the number of levels for
	   averaging. It accepts an integer in the range 4-5. Default value is
	   4.

       qp  Force a constant quantization parameter. It accepts an integer in
	   range 0-63.	If not set, the filter will use the QP from the video
	   stream (if available).

       strength
	   Set filter strength. It accepts an integer in range -15 to 32.
	   Lower values mean more details but also more artifacts, while
	   higher values make the image smoother but also blurrier. Default
	   value is 0 X PSNR optimal.

       use_bframe_qp
	   Enable the use of the QP from the B-Frames if set to 1. Using this
	   option may cause flicker since the B-Frames have often larger QP.
	   Default is 0 (not enabled).

   gblur
       Apply Gaussian blur filter.

       The filter accepts the following options:

       sigma
	   Set horizontal sigma, standard deviation of Gaussian blur. Default
	   is 0.5.

       steps
	   Set number of steps for Gaussian approximation. Defauls is 1.

       planes
	   Set which planes to filter. By default all planes are filtered.

       sigmaV
	   Set vertical sigma, if negative it will be same as "sigma".
	   Default is "-1".

   geq
       The filter accepts the following options:

       lum_expr, lum
	   Set the luminance expression.

       cb_expr, cb
	   Set the chrominance blue expression.

       cr_expr, cr
	   Set the chrominance red expression.

       alpha_expr, a
	   Set the alpha expression.

       red_expr, r
	   Set the red expression.

       green_expr, g
	   Set the green expression.

       blue_expr, b
	   Set the blue expression.

       The colorspace is selected according to the specified options. If one
       of the lum_expr, cb_expr, or cr_expr options is specified, the filter
       will automatically select a YCbCr colorspace. If one of the red_expr,
       green_expr, or blue_expr options is specified, it will select an RGB
       colorspace.

       If one of the chrominance expression is not defined, it falls back on
       the other one. If no alpha expression is specified it will evaluate to
       opaque value.  If none of chrominance expressions are specified, they
       will evaluate to the luminance expression.

       The expressions can use the following variables and functions:

       N   The sequential number of the filtered frame, starting from 0.

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       SW
       SH  Width and height scale depending on the currently filtered plane.
	   It is the ratio between the corresponding luma plane number of
	   pixels and the current plane ones. E.g. for YUV4:2:0 the values are
	   "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

       T   Time of the current frame, expressed in seconds.

       p(x, y)
	   Return the value of the pixel at location (x,y) of the current
	   plane.

       lum(x, y)
	   Return the value of the pixel at location (x,y) of the luminance
	   plane.

       cb(x, y)
	   Return the value of the pixel at location (x,y) of the blue-
	   difference chroma plane. Return 0 if there is no such plane.

       cr(x, y)
	   Return the value of the pixel at location (x,y) of the red-
	   difference chroma plane. Return 0 if there is no such plane.

       r(x, y)
       g(x, y)
       b(x, y)
	   Return the value of the pixel at location (x,y) of the
	   red/green/blue component. Return 0 if there is no such component.

       alpha(x, y)
	   Return the value of the pixel at location (x,y) of the alpha plane.
	   Return 0 if there is no such plane.

       For functions, if x and y are outside the area, the value will be
       automatically clipped to the closer edge.

       Examples

       ·   Flip the image horizontally:

		   geq=p(W-X\,Y)

       ·   Generate a bidimensional sine wave, with angle "PI/3" and a
	   wavelength of 100 pixels:

		   geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

       ·   Generate a fancy enigmatic moving light:

		   nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128

       ·   Generate a quick emboss effect:

		   format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

       ·   Modify RGB components depending on pixel position:

		   geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

       ·   Create a radial gradient that is the same size as the input (also
	   see the vignette filter):

		   geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

   gradfun
       Fix the banding artifacts that are sometimes introduced into nearly
       flat regions by truncation to 8-bit color depth.  Interpolate the
       gradients that should go where the bands are, and dither them.

       It is designed for playback only.  Do not use it prior to lossy
       compression, because compression tends to lose the dither and bring
       back the bands.

       It accepts the following parameters:

       strength
	   The maximum amount by which the filter will change any one pixel.
	   This is also the threshold for detecting nearly flat regions.
	   Acceptable values range from .51 to 64; the default value is 1.2.
	   Out-of-range values will be clipped to the valid range.

       radius
	   The neighborhood to fit the gradient to. A larger radius makes for
	   smoother gradients, but also prevents the filter from modifying the
	   pixels near detailed regions. Acceptable values are 8-32; the
	   default value is 16. Out-of-range values will be clipped to the
	   valid range.

       Alternatively, the options can be specified as a flat string:
       strength[:radius]

       Examples

       ·   Apply the filter with a 3.5 strength and radius of 8:

		   gradfun=3.5:8

       ·   Specify radius, omitting the strength (which will fall-back to the
	   default value):

		   gradfun=radius=8

   haldclut
       Apply a Hald CLUT to a video stream.

       First input is the video stream to process, and second one is the Hald
       CLUT.  The Hald CLUT input can be a simple picture or a complete video
       stream.

       The filter accepts the following options:

       shortest
	   Force termination when the shortest input terminates. Default is 0.

       repeatlast
	   Continue applying the last CLUT after the end of the stream. A
	   value of 0 disable the filter after the last frame of the CLUT is
	   reached.  Default is 1.

       "haldclut" also has the same interpolation options as lut3d (both
       filters share the same internals).

       More information about the Hald CLUT can be found on Eskil Steenberg's
       website (Hald CLUT author) at
       <http://www.quelsolaar.com/technology/clut.html>.

       Workflow examples

       Hald CLUT video stream

       Generate an identity Hald CLUT stream altered with various effects:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

       Note: make sure you use a lossless codec.

       Then use it with "haldclut" to apply it on some random stream:

	       ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

       The Hald CLUT will be applied to the 10 first seconds (duration of
       clut.nut), then the latest picture of that CLUT stream will be applied
       to the remaining frames of the "mandelbrot" stream.

       Hald CLUT with preview

       A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by
       "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select
       the biggest possible square starting at the top left of the picture.
       The remaining padding pixels (bottom or right) will be ignored. This
       area can be used to add a preview of the Hald CLUT.

       Typically, the following generated Hald CLUT will be supported by the
       "haldclut" filter:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
		  pad=iw+320 [padded_clut];
		  smptebars=s=320x256, split [a][b];
		  [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
		  [main][b] overlay=W-320" -frames:v 1 clut.png

       It contains the original and a preview of the effect of the CLUT: SMPTE
       color bars are displayed on the right-top, and below the same color
       bars processed by the color changes.

       Then, the effect of this Hald CLUT can be visualized with:

	       ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

   hflip
       Flip the input video horizontally.

       For example, to horizontally flip the input video with ffmpeg:

	       ffmpeg -i in.avi -vf "hflip" out.avi

   histeq
       This filter applies a global color histogram equalization on a per-
       frame basis.

       It can be used to correct video that has a compressed range of pixel
       intensities.  The filter redistributes the pixel intensities to
       equalize their distribution across the intensity range. It may be
       viewed as an "automatically adjusting contrast filter". This filter is
       useful only for correcting degraded or poorly captured source video.

       The filter accepts the following options:

       strength
	   Determine the amount of equalization to be applied.	As the
	   strength is reduced, the distribution of pixel intensities more-
	   and-more approaches that of the input frame. The value must be a
	   float number in the range [0,1] and defaults to 0.200.

       intensity
	   Set the maximum intensity that can generated and scale the output
	   values appropriately.  The strength should be set as desired and
	   then the intensity can be limited if needed to avoid washing-out.
	   The value must be a float number in the range [0,1] and defaults to
	   0.210.

       antibanding
	   Set the antibanding level. If enabled the filter will randomly vary
	   the luminance of output pixels by a small amount to avoid banding
	   of the histogram. Possible values are "none", "weak" or "strong".
	   It defaults to "none".

   histogram
       Compute and draw a color distribution histogram for the input video.

       The computed histogram is a representation of the color component
       distribution in an image.

       Standard histogram displays the color components distribution in an
       image.  Displays color graph for each color component. Shows
       distribution of the Y, U, V, A or R, G, B components, depending on
       input format, in the current frame. Below each graph a color component
       scale meter is shown.

       The filter accepts the following options:

       level_height
	   Set height of level. Default value is 200.  Allowed range is [50,
	   2048].

       scale_height
	   Set height of color scale. Default value is 12.  Allowed range is
	   [0, 40].

       display_mode
	   Set display mode.  It accepts the following values:

	   stack
	       Per color component graphs are placed below each other.

	   parade
	       Per color component graphs are placed side by side.

	   overlay
	       Presents information identical to that in the "parade", except
	       that the graphs representing color components are superimposed
	       directly over one another.

	   Default is "stack".

       levels_mode
	   Set mode. Can be either "linear", or "logarithmic".	Default is
	   "linear".

       components
	   Set what color components to display.  Default is 7.

       fgopacity
	   Set foreground opacity. Default is 0.7.

       bgopacity
	   Set background opacity. Default is 0.5.

       Examples

       ·   Calculate and draw histogram:

		   ffplay -i input -vf histogram

   hqdn3d
       This is a high precision/quality 3d denoise filter. It aims to reduce
       image noise, producing smooth images and making still images really
       still. It should enhance compressibility.

       It accepts the following optional parameters:

       luma_spatial
	   A non-negative floating point number which specifies spatial luma
	   strength.  It defaults to 4.0.

       chroma_spatial
	   A non-negative floating point number which specifies spatial chroma
	   strength.  It defaults to 3.0*luma_spatial/4.0.

       luma_tmp
	   A floating point number which specifies luma temporal strength. It
	   defaults to 6.0*luma_spatial/4.0.

       chroma_tmp
	   A floating point number which specifies chroma temporal strength.
	   It defaults to luma_tmp*chroma_spatial/luma_spatial.

   hwdownload
       Download hardware frames to system memory.

       The input must be in hardware frames, and the output a non-hardware
       format.	Not all formats will be supported on the output - it may be
       necessary to insert an additional format filter immediately following
       in the graph to get the output in a supported format.

   hwmap
       Map hardware frames to system memory or to another device.

       This filter has several different modes of operation; which one is used
       depends on the input and output formats:

       ·   Hardware frame input, normal frame output

	   Map the input frames to system memory and pass them to the output.
	   If the original hardware frame is later required (for example,
	   after overlaying something else on part of it), the hwmap filter
	   can be used again in the next mode to retrieve it.

       ·   Normal frame input, hardware frame output

	   If the input is actually a software-mapped hardware frame, then
	   unmap it - that is, return the original hardware frame.

	   Otherwise, a device must be provided.  Create new hardware surfaces
	   on that device for the output, then map them back to the software
	   format at the input and give those frames to the preceding filter.
	   This will then act like the hwupload filter, but may be able to
	   avoid an additional copy when the input is already in a compatible
	   format.

       ·   Hardware frame input and output

	   A device must be supplied for the output, either directly or with
	   the derive_device option.  The input and output devices must be of
	   different types and compatible - the exact meaning of this is
	   system-dependent, but typically it means that they must refer to
	   the same underlying hardware context (for example, refer to the
	   same graphics card).

	   If the input frames were originally created on the output device,
	   then unmap to retrieve the original frames.

	   Otherwise, map the frames to the output device - create new
	   hardware frames on the output corresponding to the frames on the
	   input.

       The following additional parameters are accepted:

       mode
	   Set the frame mapping mode.	Some combination of:

	   read
	       The mapped frame should be readable.

	   write
	       The mapped frame should be writeable.

	   overwrite
	       The mapping will always overwrite the entire frame.

	       This may improve performance in some cases, as the original
	       contents of the frame need not be loaded.

	   direct
	       The mapping must not involve any copying.

	       Indirect mappings to copies of frames are created in some cases
	       where either direct mapping is not possible or it would have
	       unexpected properties.  Setting this flag ensures that the
	       mapping is direct and will fail if that is not possible.

	   Defaults to read+write if not specified.

       derive_device type
	   Rather than using the device supplied at initialisation, instead
	   derive a new device of type type from the device the input frames
	   exist on.

       reverse
	   In a hardware to hardware mapping, map in reverse - create frames
	   in the sink and map them back to the source.  This may be necessary
	   in some cases where a mapping in one direction is required but only
	   the opposite direction is supported by the devices being used.

	   This option is dangerous - it may break the preceding filter in
	   undefined ways if there are any additional constraints on that
	   filter's output.  Do not use it without fully understanding the
	   implications of its use.

   hwupload
       Upload system memory frames to hardware surfaces.

       The device to upload to must be supplied when the filter is
       initialised.  If using ffmpeg, select the appropriate device with the
       -filter_hw_device option.

   hwupload_cuda
       Upload system memory frames to a CUDA device.

       It accepts the following optional parameters:

       device
	   The number of the CUDA device to use

   hqx
       Apply a high-quality magnification filter designed for pixel art. This
       filter was originally created by Maxim Stepin.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for
	   "hq4x".  Default is 3.

   hstack
       Stack input videos horizontally.

       All streams must be of same pixel format and of same height.

       Note that this filter is faster than using overlay and pad filter to
       create same output.

       The filter accept the following option:

       inputs
	   Set number of input streams. Default is 2.

       shortest
	   If set to 1, force the output to terminate when the shortest input
	   terminates. Default value is 0.

   hue
       Modify the hue and/or the saturation of the input.

       It accepts the following parameters:

       h   Specify the hue angle as a number of degrees. It accepts an
	   expression, and defaults to "0".

       s   Specify the saturation in the [-10,10] range. It accepts an
	   expression and defaults to "1".

       H   Specify the hue angle as a number of radians. It accepts an
	   expression, and defaults to "0".

       b   Specify the brightness in the [-10,10] range. It accepts an
	   expression and defaults to "0".

       h and H are mutually exclusive, and can't be specified at the same
       time.

       The b, h, H and s option values are expressions containing the
       following constants:

       n   frame count of the input frame starting from 0

       pts presentation timestamp of the input frame expressed in time base
	   units

       r   frame rate of the input video, NAN if the input frame rate is
	   unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is
	   unknown

       tb  time base of the input video

       Examples

       ·   Set the hue to 90 degrees and the saturation to 1.0:

		   hue=h=90:s=1

       ·   Same command but expressing the hue in radians:

		   hue=H=PI/2:s=1

       ·   Rotate hue and make the saturation swing between 0 and 2 over a
	   period of 1 second:

		   hue="H=2*PI*t: s=sin(2*PI*t)+1"

       ·   Apply a 3 seconds saturation fade-in effect starting at 0:

		   hue="s=min(t/3\,1)"

	   The general fade-in expression can be written as:

		   hue="s=min(0\, max((t-START)/DURATION\, 1))"

       ·   Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

		   hue="s=max(0\, min(1\, (8-t)/3))"

	   The general fade-out expression can be written as:

		   hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

       Commands

       This filter supports the following commands:

       b
       s
       h
       H   Modify the hue and/or the saturation and/or brightness of the input
	   video.  The command accepts the same syntax of the corresponding
	   option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   hysteresis
       Grow first stream into second stream by connecting components.  This
       makes it possible to build more robust edge masks.

       This filter accepts the following options:

       planes
	   Set which planes will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       threshold
	   Set threshold which is used in filtering. If pixel component value
	   is higher than this value filter algorithm for connecting
	   components is activated.  By default value is 0.

   idet
       Detect video interlacing type.

       This filter tries to detect if the input frames are interlaced,
       progressive, top or bottom field first. It will also try to detect
       fields that are repeated between adjacent frames (a sign of telecine).

       Single frame detection considers only immediately adjacent frames when
       classifying each frame.	Multiple frame detection incorporates the
       classification history of previous frames.

       The filter will log these metadata values:

       single.current_frame
	   Detected type of current frame using single-frame detection. One
	   of: ``tff'' (top field first), ``bff'' (bottom field first),
	   ``progressive'', or ``undetermined''

       single.tff
	   Cumulative number of frames detected as top field first using
	   single-frame detection.

       multiple.tff
	   Cumulative number of frames detected as top field first using
	   multiple-frame detection.

       single.bff
	   Cumulative number of frames detected as bottom field first using
	   single-frame detection.

       multiple.current_frame
	   Detected type of current frame using multiple-frame detection. One
	   of: ``tff'' (top field first), ``bff'' (bottom field first),
	   ``progressive'', or ``undetermined''

       multiple.bff
	   Cumulative number of frames detected as bottom field first using
	   multiple-frame detection.

       single.progressive
	   Cumulative number of frames detected as progressive using single-
	   frame detection.

       multiple.progressive
	   Cumulative number of frames detected as progressive using multiple-
	   frame detection.

       single.undetermined
	   Cumulative number of frames that could not be classified using
	   single-frame detection.

       multiple.undetermined
	   Cumulative number of frames that could not be classified using
	   multiple-frame detection.

       repeated.current_frame
	   Which field in the current frame is repeated from the last. One of
	   ``neither'', ``top'', or ``bottom''.

       repeated.neither
	   Cumulative number of frames with no repeated field.

       repeated.top
	   Cumulative number of frames with the top field repeated from the
	   previous frame's top field.

       repeated.bottom
	   Cumulative number of frames with the bottom field repeated from the
	   previous frame's bottom field.

       The filter accepts the following options:

       intl_thres
	   Set interlacing threshold.

       prog_thres
	   Set progressive threshold.

       rep_thres
	   Threshold for repeated field detection.

       half_life
	   Number of frames after which a given frame's contribution to the
	   statistics is halved (i.e., it contributes only 0.5 to its
	   classification). The default of 0 means that all frames seen are
	   given full weight of 1.0 forever.

       analyze_interlaced_flag
	   When this is not 0 then idet will use the specified number of
	   frames to determine if the interlaced flag is accurate, it will not
	   count undetermined frames.  If the flag is found to be accurate it
	   will be used without any further computations, if it is found to be
	   inaccurate it will be cleared without any further computations.
	   This allows inserting the idet filter as a low computational method
	   to clean up the interlaced flag

   il
       Deinterleave or interleave fields.

       This filter allows one to process interlaced images fields without
       deinterlacing them. Deinterleaving splits the input frame into 2 fields
       (so called half pictures). Odd lines are moved to the top half of the
       output image, even lines to the bottom half.  You can process (filter)
       them independently and then re-interleave them.

       The filter accepts the following options:

       luma_mode, l
       chroma_mode, c
       alpha_mode, a
	   Available values for luma_mode, chroma_mode and alpha_mode are:

	   none
	       Do nothing.

	   deinterleave, d
	       Deinterleave fields, placing one above the other.

	   interleave, i
	       Interleave fields. Reverse the effect of deinterleaving.

	   Default value is "none".

       luma_swap, ls
       chroma_swap, cs
       alpha_swap, as
	   Swap luma/chroma/alpha fields. Exchange even & odd lines. Default
	   value is 0.

   inflate
       Apply inflate effect to the video.

       This filter replaces the pixel by the local(3x3) average by taking into
       account only values higher than the pixel.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit the maximum change for each plane, default is 65535.  If 0,
	   plane will remain unchanged.

   interlace
       Simple interlacing filter from progressive contents. This interleaves
       upper (or lower) lines from odd frames with lower (or upper) lines from
       even frames, halving the frame rate and preserving image height.

		  Original	  Original	       New Frame
		  Frame 'j'	 Frame 'j+1'		 (tff)
		 ==========	 ===========	   ==================
		   Line 0  -------------------->    Frame 'j' Line 0
		   Line 1	   Line 1  ---->   Frame 'j+1' Line 1
		   Line 2 --------------------->    Frame 'j' Line 2
		   Line 3	   Line 3  ---->   Frame 'j+1' Line 3
		    ... 	    ... 		  ...
	       New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

       It accepts the following optional parameters:

       scan
	   This determines whether the interlaced frame is taken from the even
	   (tff - default) or odd (bff) lines of the progressive frame.

       lowpass
	   Vertical lowpass filter to avoid twitter interlacing and reduce
	   moire patterns.

	   0, off
	       Disable vertical lowpass filter

	   1, linear
	       Enable linear filter (default)

	   2, complex
	       Enable complex filter. This will slightly less reduce twitter
	       and moire but better retain detail and subjective sharpness
	       impression.

   kerndeint
       Deinterlace input video by applying Donald Graft's adaptive kernel
       deinterling. Work on interlaced parts of a video to produce progressive
       frames.

       The description of the accepted parameters follows.

       thresh
	   Set the threshold which affects the filter's tolerance when
	   determining if a pixel line must be processed. It must be an
	   integer in the range [0,255] and defaults to 10. A value of 0 will
	   result in applying the process on every pixels.

       map Paint pixels exceeding the threshold value to white if set to 1.
	   Default is 0.

       order
	   Set the fields order. Swap fields if set to 1, leave fields alone
	   if 0. Default is 0.

       sharp
	   Enable additional sharpening if set to 1. Default is 0.

       twoway
	   Enable twoway sharpening if set to 1. Default is 0.

       Examples

       ·   Apply default values:

		   kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

       ·   Enable additional sharpening:

		   kerndeint=sharp=1

       ·   Paint processed pixels in white:

		   kerndeint=map=1

   lenscorrection
       Correct radial lens distortion

       This filter can be used to correct for radial distortion as can result
       from the use of wide angle lenses, and thereby re-rectify the image. To
       find the right parameters one can use tools available for example as
       part of opencv or simply trial-and-error.  To use opencv use the
       calibration sample (under samples/cpp) from the opencv sources and
       extract the k1 and k2 coefficients from the resulting matrix.

       Note that effectively the same filter is available in the open-source
       tools Krita and Digikam from the KDE project.

       In contrast to the vignette filter, which can also be used to
       compensate lens errors, this filter corrects the distortion of the
       image, whereas vignette corrects the brightness distribution, so you
       may want to use both filters together in certain cases, though you will
       have to take care of ordering, i.e. whether vignetting should be
       applied before or after lens correction.

       Options

       The filter accepts the following options:

       cx  Relative x-coordinate of the focal point of the image, and thereby
	   the center of the distortion. This value has a range [0,1] and is
	   expressed as fractions of the image width.

       cy  Relative y-coordinate of the focal point of the image, and thereby
	   the center of the distortion. This value has a range [0,1] and is
	   expressed as fractions of the image height.

       k1  Coefficient of the quadratic correction term. 0.5 means no
	   correction.

       k2  Coefficient of the double quadratic correction term. 0.5 means no
	   correction.

       The formula that generates the correction is:

       r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

       where r_0 is halve of the image diagonal and r_src and r_tgt are the
       distances from the focal point in the source and target images,
       respectively.

   libvmaf
       Obtain the average VMAF (Video Multi-Method Assessment Fusion) score
       between two input videos.

       This filter takes two input videos.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average VMAF score is printed through the logging system.

       It requires Netflix's vmaf library (libvmaf) as a pre-requisite.  After
       installing the library it can be enabled using: "./configure
       --enable-libvmaf".  If no model path is specified it uses the default
       model: "vmaf_v0.6.1.pkl".

       On the below examples the input file main.mpg being processed is
       compared with the reference file ref.mpg.

       The filter has following options:

       model_path
	   Set the model path which is to be used for SVM.  Default value:
	   "vmaf_v0.6.1.pkl"

       log_path
	   Set the file path to be used to store logs.

       log_fmt
	   Set the format of the log file (xml or json).

       enable_transform
	   Enables transform for computing vmaf.

       phone_model
	   Invokes the phone model which will generate VMAF scores higher than
	   in the regular model, which is more suitable for laptop, TV, etc.
	   viewing conditions.

       psnr
	   Enables computing psnr along with vmaf.

       ssim
	   Enables computing ssim along with vmaf.

       ms_ssim
	   Enables computing ms_ssim along with vmaf.

       pool
	   Set the pool method to be used for computing vmaf.

       This filter also supports the framesync options.

       For example:

	       ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -

       Example with options:

	       ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -

   limiter
       Limits the pixel components values to the specified range [min, max].

       The filter accepts the following options:

       min Lower bound. Defaults to the lowest allowed value for the input.

       max Upper bound. Defaults to the highest allowed value for the input.

       planes
	   Specify which planes will be processed. Defaults to all available.

   loop
       Loop video frames.

       The filter accepts the following options:

       loop
	   Set the number of loops.

       size
	   Set maximal size in number of frames.

       start
	   Set first frame of loop.

   lut3d
       Apply a 3D LUT to an input video.

       The filter accepts the following options:

       file
	   Set the 3D LUT file name.

	   Currently supported formats:

	   3dl AfterEffects

	   cube
	       Iridas

	   dat DaVinci

	   m3d Pandora

       interp
	   Select interpolation mode.

	   Available values are:

	   nearest
	       Use values from the nearest defined point.

	   trilinear
	       Interpolate values using the 8 points defining a cube.

	   tetrahedral
	       Interpolate values using a tetrahedron.

       This filter also supports the framesync options.

   lumakey
       Turn certain luma values into transparency.

       The filter accepts the following options:

       threshold
	   Set the luma which will be used as base for transparency.  Default
	   value is 0.

       tolerance
	   Set the range of luma values to be keyed out.  Default value is 0.

       softness
	   Set the range of softness. Default value is 0.  Use this to control
	   gradual transition from zero to full transparency.

   lut, lutrgb, lutyuv
       Compute a look-up table for binding each pixel component input value to
       an output value, and apply it to the input video.

       lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB
       input video.

       These filters accept the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
	   component

       r   set red component expression

       g   set green component expression

       b   set blue component expression

       a   alpha component expression

       y   set Y/luminance component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

       Each of them specifies the expression to use for computing the lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c* options depends on the
       format in input.

       The lut filter requires either YUV or RGB pixel formats in input,
       lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       clipval
	   The input value, clipped to the minval-maxval range.

       maxval
	   The maximum value for the pixel component.

       minval
	   The minimum value for the pixel component.

       negval
	   The negated value for the pixel component value, clipped to the
	   minval-maxval range; it corresponds to the expression
	   "maxval-clipval+minval".

       clip(val)
	   The computed value in val, clipped to the minval-maxval range.

       gammaval(gamma)
	   The computed gamma correction value of the pixel component value,
	   clipped to the minval-maxval range. It corresponds to the
	   expression
	   "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "val".

       Examples

       ·   Negate input video:

		   lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
		   lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

	   The above is the same as:

		   lutrgb="r=negval:g=negval:b=negval"
		   lutyuv="y=negval:u=negval:v=negval"

       ·   Negate luminance:

		   lutyuv=y=negval

       ·   Remove chroma components, turning the video into a graytone image:

		   lutyuv="u=128:v=128"

       ·   Apply a luma burning effect:

		   lutyuv="y=2*val"

       ·   Remove green and blue components:

		   lutrgb="g=0:b=0"

       ·   Set a constant alpha channel value on input:

		   format=rgba,lutrgb=a="maxval-minval/2"

       ·   Correct luminance gamma by a factor of 0.5:

		   lutyuv=y=gammaval(0.5)

       ·   Discard least significant bits of luma:

		   lutyuv=y='bitand(val, 128+64+32)'

       ·   Technicolor like effect:

		   lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'

   lut2, tlut2
       The "lut2" filter takes two input streams and outputs one stream.

       The "tlut2" (time lut2) filter takes two consecutive frames from one
       single stream.

       This filter accepts the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
	   component

       Each of them specifies the expression to use for computing the lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c* options depends on the
       format in inputs.

       The expressions can contain the following constants:

       w
       h   The input width and height.

       x   The first input value for the pixel component.

       y   The second input value for the pixel component.

       bdx The first input video bit depth.

       bdy The second input video bit depth.

       All expressions default to "x".

       Examples

       ·   Highlight differences between two RGB video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

       ·   Highlight differences between two YUV video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

       ·   Show max difference between two video streams:

		   lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'

   maskedclamp
       Clamp the first input stream with the second input and third input
       stream.

       Returns the value of first stream to be between second input stream -
       "undershoot" and third input stream + "overshoot".

       This filter accepts the following options:

       undershoot
	   Default value is 0.

       overshoot
	   Default value is 0.

       planes
	   Set which planes will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

   maskedmerge
       Merge the first input stream with the second input stream using per
       pixel weights in the third input stream.

       A value of 0 in the third stream pixel component means that pixel
       component from first stream is returned unchanged, while maximum value
       (eg. 255 for 8-bit videos) means that pixel component from second
       stream is returned unchanged. Intermediate values define the amount of
       merging between both input stream's pixel components.

       This filter accepts the following options:

       planes
	   Set which planes will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

   mcdeint
       Apply motion-compensation deinterlacing.

       It needs one field per frame as input and must thus be used together
       with yadif=1/3 or equivalent.

       This filter accepts the following options:

       mode
	   Set the deinterlacing mode.

	   It accepts one of the following values:

	   fast
	   medium
	   slow
	       use iterative motion estimation

	   extra_slow
	       like slow, but use multiple reference frames.

	   Default value is fast.

       parity
	   Set the picture field parity assumed for the input video. It must
	   be one of the following values:

	   0, tff
	       assume top field first

	   1, bff
	       assume bottom field first

	   Default value is bff.

       qp  Set per-block quantization parameter (QP) used by the internal
	   encoder.

	   Higher values should result in a smoother motion vector field but
	   less optimal individual vectors. Default value is 1.

   mergeplanes
       Merge color channel components from several video streams.

       The filter accepts up to 4 input streams, and merge selected input
       planes to the output video.

       This filter accepts the following options:

       mapping
	   Set input to output plane mapping. Default is 0.

	   The mappings is specified as a bitmap. It should be specified as a
	   hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the
	   mapping for the first plane of the output stream. 'A' sets the
	   number of the input stream to use (from 0 to 3), and 'a' the plane
	   number of the corresponding input to use (from 0 to 3). The rest of
	   the mappings is similar, 'Bb' describes the mapping for the output
	   stream second plane, 'Cc' describes the mapping for the output
	   stream third plane and 'Dd' describes the mapping for the output
	   stream fourth plane.

       format
	   Set output pixel format. Default is "yuva444p".

       Examples

       ·   Merge three gray video streams of same width and height into single
	   video stream:

		   [a0][a1][a2]mergeplanes=0x001020:yuv444p

       ·   Merge 1st yuv444p stream and 2nd gray video stream into yuva444p
	   video stream:

		   [a0][a1]mergeplanes=0x00010210:yuva444p

       ·   Swap Y and A plane in yuva444p stream:

		   format=yuva444p,mergeplanes=0x03010200:yuva444p

       ·   Swap U and V plane in yuv420p stream:

		   format=yuv420p,mergeplanes=0x000201:yuv420p

       ·   Cast a rgb24 clip to yuv444p:

		   format=rgb24,mergeplanes=0x000102:yuv444p

   mestimate
       Estimate and export motion vectors using block matching algorithms.
       Motion vectors are stored in frame side data to be used by other
       filters.

       This filter accepts the following options:

       method
	   Specify the motion estimation method. Accepts one of the following
	   values:

	   esa Exhaustive search algorithm.

	   tss Three step search algorithm.

	   tdls
	       Two dimensional logarithmic search algorithm.

	   ntss
	       New three step search algorithm.

	   fss Four step search algorithm.

	   ds  Diamond search algorithm.

	   hexbs
	       Hexagon-based search algorithm.

	   epzs
	       Enhanced predictive zonal search algorithm.

	   umh Uneven multi-hexagon search algorithm.

	   Default value is esa.

       mb_size
	   Macroblock size. Default 16.

       search_param
	   Search parameter. Default 7.

   midequalizer
       Apply Midway Image Equalization effect using two video streams.

       Midway Image Equalization adjusts a pair of images to have the same
       histogram, while maintaining their dynamics as much as possible. It's
       useful for e.g. matching exposures from a pair of stereo cameras.

       This filter has two inputs and one output, which must be of same pixel
       format, but may be of different sizes. The output of filter is first
       input adjusted with midway histogram of both inputs.

       This filter accepts the following option:

       planes
	   Set which planes to process. Default is 15, which is all available
	   planes.

   minterpolate
       Convert the video to specified frame rate using motion interpolation.

       This filter accepts the following options:

       fps Specify the output frame rate. This can be rational e.g.
	   "60000/1001". Frames are dropped if fps is lower than source fps.
	   Default 60.

       mi_mode
	   Motion interpolation mode. Following values are accepted:

	   dup Duplicate previous or next frame for interpolating new ones.

	   blend
	       Blend source frames. Interpolated frame is mean of previous and
	       next frames.

	   mci Motion compensated interpolation. Following options are
	       effective when this mode is selected:

	       mc_mode
		   Motion compensation mode. Following values are accepted:

		   obmc
		       Overlapped block motion compensation.

		   aobmc
		       Adaptive overlapped block motion compensation. Window
		       weighting coefficients are controlled adaptively
		       according to the reliabilities of the neighboring
		       motion vectors to reduce oversmoothing.

		   Default mode is obmc.

	       me_mode
		   Motion estimation mode. Following values are accepted:

		   bidir
		       Bidirectional motion estimation. Motion vectors are
		       estimated for each source frame in both forward and
		       backward directions.

		   bilat
		       Bilateral motion estimation. Motion vectors are
		       estimated directly for interpolated frame.

		   Default mode is bilat.

	       me  The algorithm to be used for motion estimation. Following
		   values are accepted:

		   esa Exhaustive search algorithm.

		   tss Three step search algorithm.

		   tdls
		       Two dimensional logarithmic search algorithm.

		   ntss
		       New three step search algorithm.

		   fss Four step search algorithm.

		   ds  Diamond search algorithm.

		   hexbs
		       Hexagon-based search algorithm.

		   epzs
		       Enhanced predictive zonal search algorithm.

		   umh Uneven multi-hexagon search algorithm.

		   Default algorithm is epzs.

	       mb_size
		   Macroblock size. Default 16.

	       search_param
		   Motion estimation search parameter. Default 32.

	       vsbmc
		   Enable variable-size block motion compensation. Motion
		   estimation is applied with smaller block sizes at object
		   boundaries in order to make the them less blur. Default is
		   0 (disabled).

       scd Scene change detection method. Scene change leads motion vectors to
	   be in random direction. Scene change detection replace interpolated
	   frames by duplicate ones. May not be needed for other modes.
	   Following values are accepted:

	   none
	       Disable scene change detection.

	   fdiff
	       Frame difference. Corresponding pixel values are compared and
	       if it satisfies scd_threshold scene change is detected.

	   Default method is fdiff.

       scd_threshold
	   Scene change detection threshold. Default is 5.0.

   mpdecimate
       Drop frames that do not differ greatly from the previous frame in order
       to reduce frame rate.

       The main use of this filter is for very-low-bitrate encoding (e.g.
       streaming over dialup modem), but it could in theory be used for fixing
       movies that were inverse-telecined incorrectly.

       A description of the accepted options follows.

       max Set the maximum number of consecutive frames which can be dropped
	   (if positive), or the minimum interval between dropped frames (if
	   negative). If the value is 0, the frame is dropped disregarding the
	   number of previous sequentially dropped frames.

	   Default value is 0.

       hi
       lo
       frac
	   Set the dropping threshold values.

	   Values for hi and lo are for 8x8 pixel blocks and represent actual
	   pixel value differences, so a threshold of 64 corresponds to 1 unit
	   of difference for each pixel, or the same spread out differently
	   over the block.

	   A frame is a candidate for dropping if no 8x8 blocks differ by more
	   than a threshold of hi, and if no more than frac blocks (1 meaning
	   the whole image) differ by more than a threshold of lo.

	   Default value for hi is 64*12, default value for lo is 64*5, and
	   default value for frac is 0.33.

   negate
       Negate input video.

       It accepts an integer in input; if non-zero it negates the alpha
       component (if available). The default value in input is 0.

   nlmeans
       Denoise frames using Non-Local Means algorithm.

       Each pixel is adjusted by looking for other pixels with similar
       contexts. This context similarity is defined by comparing their
       surrounding patches of size pxp. Patches are searched in an area of rxr
       around the pixel.

       Note that the research area defines centers for patches, which means
       some patches will be made of pixels outside that research area.

       The filter accepts the following options.

       s   Set denoising strength.

       p   Set patch size.

       pc  Same as p but for chroma planes.

	   The default value is 0 and means automatic.

       r   Set research size.

       rc  Same as r but for chroma planes.

	   The default value is 0 and means automatic.

   nnedi
       Deinterlace video using neural network edge directed interpolation.

       This filter accepts the following options:

       weights
	   Mandatory option, without binary file filter can not work.
	   Currently file can be found here:
	   https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

       deint
	   Set which frames to deinterlace, by default it is "all".  Can be
	   "all" or "interlaced".

       field
	   Set mode of operation.

	   Can be one of the following:

	   af  Use frame flags, both fields.

	   a   Use frame flags, single field.

	   t   Use top field only.

	   b   Use bottom field only.

	   tf  Use both fields, top first.

	   bf  Use both fields, bottom first.

       planes
	   Set which planes to process, by default filter process all frames.

       nsize
	   Set size of local neighborhood around each pixel, used by the
	   predictor neural network.

	   Can be one of the following:

	   s8x6
	   s16x6
	   s32x6
	   s48x6
	   s8x4
	   s16x4
	   s32x4
       nns Set the number of neurons in predictor neural network.  Can be one
	   of the following:

	   n16
	   n32
	   n64
	   n128
	   n256
       qual
	   Controls the number of different neural network predictions that
	   are blended together to compute the final output value. Can be
	   "fast", default or "slow".

       etype
	   Set which set of weights to use in the predictor.  Can be one of
	   the following:

	   a   weights trained to minimize absolute error

	   s   weights trained to minimize squared error

       pscrn
	   Controls whether or not the prescreener neural network is used to
	   decide which pixels should be processed by the predictor neural
	   network and which can be handled by simple cubic interpolation.
	   The prescreener is trained to know whether cubic interpolation will
	   be sufficient for a pixel or whether it should be predicted by the
	   predictor nn.  The computational complexity of the prescreener nn
	   is much less than that of the predictor nn. Since most pixels can
	   be handled by cubic interpolation, using the prescreener generally
	   results in much faster processing.  The prescreener is pretty
	   accurate, so the difference between using it and not using it is
	   almost always unnoticeable.

	   Can be one of the following:

	   none
	   original
	   new

	   Default is "new".

       fapprox
	   Set various debugging flags.

   noformat
       Force libavfilter not to use any of the specified pixel formats for the
       input to the next filter.

       It accepts the following parameters:

       pix_fmts
	   A '|'-separated list of pixel format names, such as
	   pix_fmts=yuv420p|monow|rgb24".

       Examples

       ·   Force libavfilter to use a format different from yuv420p for the
	   input to the vflip filter:

		   noformat=pix_fmts=yuv420p,vflip

       ·   Convert the input video to any of the formats not contained in the
	   list:

		   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
	   Set noise seed for specific pixel component or all pixel components
	   in case of all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
	   Set noise strength for specific pixel component or all pixel
	   components in case all_strength. Default value is 0. Allowed range
	   is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
	   Set pixel component flags or set flags for all components if
	   all_flags.  Available values for component flags are:

	   a   averaged temporal noise (smoother)

	   p   mix random noise with a (semi)regular pattern

	   t   temporal noise (noise pattern changes between frames)

	   u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

	       noise=alls=20:allf=t+u

   null
       Pass the video source unchanged to the output.

   ocr
       Optical Character Recognition

       This filter uses Tesseract for optical character recognition.

       It accepts the following options:

       datapath
	   Set datapath to tesseract data. Default is to use whatever was set
	   at installation.

       language
	   Set language, default is "eng".

       whitelist
	   Set character whitelist.

       blacklist
	   Set character blacklist.

       The filter exports recognized text as the frame metadata
       "lavfi.ocr.text".

   ocv
       Apply a video transform using libopencv.

       To enable this filter, install the libopencv library and headers and
       configure FFmpeg with "--enable-libopencv".

       It accepts the following parameters:

       filter_name
	   The name of the libopencv filter to apply.

       filter_params
	   The parameters to pass to the libopencv filter. If not specified,
	   the default values are assumed.

       Refer to the official libopencv documentation for more precise
       information:
       <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

       Several libopencv filters are supported; see the following subsections.

       dilate

       Dilate an image by using a specific structuring element.  It
       corresponds to the libopencv function "cvDilate".

       It accepts the parameters: struct_el|nb_iterations.

       struct_el represents a structuring element, and has the syntax:
       colsxrows+anchor_xxanchor_y/shape

       cols and rows represent the number of columns and rows of the
       structuring element, anchor_x and anchor_y the anchor point, and shape
       the shape for the structuring element. shape must be "rect", "cross",
       "ellipse", or "custom".

       If the value for shape is "custom", it must be followed by a string of
       the form "=filename". The file with name filename is assumed to
       represent a binary image, with each printable character corresponding
       to a bright pixel. When a custom shape is used, cols and rows are
       ignored, the number or columns and rows of the read file are assumed
       instead.

       The default value for struct_el is "3x3+0x0/rect".

       nb_iterations specifies the number of times the transform is applied to
       the image, and defaults to 1.

       Some examples:

	       # Use the default values
	       ocv=dilate

	       # Dilate using a structuring element with a 5x5 cross, iterating two times
	       ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

	       # Read the shape from the file diamond.shape, iterating two times.
	       # The file diamond.shape may contain a pattern of characters like this
	       #   *
	       #  ***
	       # *****
	       #  ***
	       #   *
	       # The specified columns and rows are ignored
	       # but the anchor point coordinates are not
	       ocv=dilate:0x0+2x2/custom=diamond.shape|2

       erode

       Erode an image by using a specific structuring element.	It corresponds
       to the libopencv function "cvErode".

       It accepts the parameters: struct_el:nb_iterations, with the same
       syntax and semantics as the dilate filter.

       smooth

       Smooth the input video.

       The filter takes the following parameters:
       type|param1|param2|param3|param4.

       type is the type of smooth filter to apply, and must be one of the
       following values: "blur", "blur_no_scale", "median", "gaussian", or
       "bilateral". The default value is "gaussian".

       The meaning of param1, param2, param3, and param4 depend on the smooth
       type. param1 and param2 accept integer positive values or 0. param3 and
       param4 accept floating point values.

       The default value for param1 is 3. The default value for the other
       parameters is 0.

       These parameters correspond to the parameters assigned to the libopencv
       function "cvSmooth".

   oscilloscope
       2D Video Oscilloscope.

       Useful to measure spatial impulse, step responses, chroma delays, etc.

       It accepts the following parameters:

       x   Set scope center x position.

       y   Set scope center y position.

       s   Set scope size, relative to frame diagonal.

       t   Set scope tilt/rotation.

       o   Set trace opacity.

       tx  Set trace center x position.

       ty  Set trace center y position.

       tw  Set trace width, relative to width of frame.

       th  Set trace height, relative to height of frame.

       c   Set which components to trace. By default it traces first three
	   components.

       g   Draw trace grid. By default is enabled.

       st  Draw some statistics. By default is enabled.

       sc  Draw scope. By default is enabled.

       Examples

       ·   Inspect full first row of video frame.

		   oscilloscope=x=0.5:y=0:s=1

       ·   Inspect full last row of video frame.

		   oscilloscope=x=0.5:y=1:s=1

       ·   Inspect full 5th line of video frame of height 1080.

		   oscilloscope=x=0.5:y=5/1080:s=1

       ·   Inspect full last column of video frame.

		   oscilloscope=x=1:y=0.5:s=1:t=1

   overlay
       Overlay one video on top of another.

       It takes two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.

       It accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and y coordinates of the overlaid
	   video on the main video. Default value is "0" for both expressions.
	   In case the expression is invalid, it is set to a huge value
	   (meaning that the overlay will not be displayed within the output
	   visible area).

       eof_action
	   See framesync.

       eval
	   Set when the expressions for x, and y are evaluated.

	   It accepts the following values:

	   init
	       only evaluate expressions once during the filter initialization
	       or when a command is processed

	   frame
	       evaluate expressions for each incoming frame

	   Default value is frame.

       shortest
	   See framesync.

       format
	   Set the format for the output video.

	   It accepts the following values:

	   yuv420
	       force YUV420 output

	   yuv422
	       force YUV422 output

	   yuv444
	       force YUV444 output

	   rgb force packed RGB output

	   gbrp
	       force planar RGB output

	   auto
	       automatically pick format

	   Default value is yuv420.

       repeatlast
	   See framesync.

       The x, and y expressions can contain the following parameters.

       main_w, W
       main_h, H
	   The main input width and height.

       overlay_w, w
       overlay_h, h
	   The overlay input width and height.

       x
       y   The computed values for x and y. They are evaluated for each new
	   frame.

       hsub
       vsub
	   horizontal and vertical chroma subsample values of the output
	   format. For example for the pixel format "yuv422p" hsub is 2 and
	   vsub is 1.

       n   the number of input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       t   The timestamp, expressed in seconds. It's NAN if the input
	   timestamp is unknown.

       This filter also supports the framesync options.

       Note that the n, pos, t variables are available only when evaluation is
       done per frame, and will evaluate to NAN when eval is set to init.

       Be aware that frames are taken from each input video in timestamp
       order, hence, if their initial timestamps differ, it is a good idea to
       pass the two inputs through a setpts=PTS-STARTPTS filter to have them
       begin in the same zero timestamp, as the example for the movie filter
       does.

       You can chain together more overlays but you should test the efficiency
       of such approach.

       Commands

       This filter supports the following commands:

       x
       y   Modify the x and y of the overlay input.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

       Examples

       ·   Draw the overlay at 10 pixels from the bottom right corner of the
	   main video:

		   overlay=main_w-overlay_w-10:main_h-overlay_h-10

	   Using named options the example above becomes:

		   overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

       ·   Insert a transparent PNG logo in the bottom left corner of the
	   input, using the ffmpeg tool with the "-filter_complex" option:

		   ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

       ·   Insert 2 different transparent PNG logos (second logo on bottom
	   right corner) using the ffmpeg tool:

		   ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

       ·   Add a transparent color layer on top of the main video; "WxH" must
	   specify the size of the main input to the overlay filter:

		   color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

       ·   Play an original video and a filtered version (here with the
	   deshake filter) side by side using the ffplay tool:

		   ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

	   The above command is the same as:

		   ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

       ·   Make a sliding overlay appearing from the left to the right top
	   part of the screen starting since time 2:

		   overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

       ·   Compose output by putting two input videos side to side:

		   ffmpeg -i left.avi -i right.avi -filter_complex "
		   nullsrc=size=200x100 [background];
		   [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
		   [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
		   [background][left]	    overlay=shortest=1	     [background+left];
		   [background+left][right] overlay=shortest=1:x=100 [left+right]
		   "

       ·   Mask 10-20 seconds of a video by applying the delogo filter to a
	   section

		   ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
		   -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
		   masked.avi

       ·   Chain several overlays in cascade:

		   nullsrc=s=200x200 [bg];
		   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
		   [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
		   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
		   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
		   [in3] null,	     [mid2] overlay=100:100 [out0]

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
	   Set depth.

	   Larger depth values will denoise lower frequency components more,
	   but slow down filtering.

	   Must be an int in the range 8-16, default is 8.

       luma_strength, ls
	   Set luma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
	   Set chroma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add paddings to the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following parameters:

       width, w
       height, h
	   Specify an expression for the size of the output image with the
	   paddings added. If the value for width or height is 0, the
	   corresponding input size is used for the output.

	   The width expression can reference the value set by the height
	   expression, and vice versa.

	   The default value of width and height is 0.

       x
       y   Specify the offsets to place the input image at within the padded
	   area, with respect to the top/left border of the output image.

	   The x expression can reference the value set by the y expression,
	   and vice versa.

	   The default value of x and y is 0.

	   If x or y evaluate to a negative number, they'll be changed so the
	   input image is centered on the padded area.

       color
	   Specify the color of the padded area. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of color is "black".

       eval
	   Specify when to evaluate  width, height, x and y expression.

	   It accepts the following values:

	   init
	       Only evaluate expressions once during the filter initialization
	       or when a command is processed.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

       aspect
	   Pad to aspect instead to a resolution.

       The value for the width, height, x, and y options are expressions
       containing the following constants:

       in_w
       in_h
	   The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output width and height (the size of the padded area), as
	   specified by the width and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN
	   if not yet specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
	   The horizontal and vertical chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   Add paddings with the color "violet" to the input video. The output
	   video size is 640x480, and the top-left corner of the input video
	   is placed at column 0, row 40

		   pad=640:480:0:40:violet

	   The example above is equivalent to the following command:

		   pad=width=640:height=480:x=0:y=40:color=violet

       ·   Pad the input to get an output with dimensions increased by 3/2,
	   and put the input video at the center of the padded area:

		   pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

       ·   Pad the input to get a squared output with size equal to the
	   maximum value between the input width and height, and put the input
	   video at the center of the padded area:

		   pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

       ·   Pad the input to get a final w/h ratio of 16:9:

		   pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

       ·   In case of anamorphic video, in order to set the output display
	   aspect correctly, it is necessary to use sar in the expression,
	   according to the relation:

		   (ih * X / ih) * sar = output_dar
		   X = output_dar / sar

	   Thus the previous example needs to be modified to:

		   pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

       ·   Double the output size and put the input video in the bottom-right
	   corner of the output padded area:

		   pad="2*iw:2*ih:ow-iw:oh-ih"

   palettegen
       Generate one palette for a whole video stream.

       It accepts the following options:

       max_colors
	   Set the maximum number of colors to quantize in the palette.  Note:
	   the palette will still contain 256 colors; the unused palette
	   entries will be black.

       reserve_transparent
	   Create a palette of 255 colors maximum and reserve the last one for
	   transparency. Reserving the transparency color is useful for GIF
	   optimization.  If not set, the maximum of colors in the palette
	   will be 256. You probably want to disable this option for a
	   standalone image.  Set by default.

       stats_mode
	   Set statistics mode.

	   It accepts the following values:

	   full
	       Compute full frame histograms.

	   diff
	       Compute histograms only for the part that differs from previous
	       frame. This might be relevant to give more importance to the
	       moving part of your input if the background is static.

	   single
	       Compute new histogram for each frame.

	   Default value is full.

       The filter also exports the frame metadata "lavfi.color_quant_ratio"
       ("nb_color_in / nb_color_out") which you can use to evaluate the degree
       of color quantization of the palette. This information is also visible
       at info logging level.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

		   ffmpeg -i input.mkv -vf palettegen palette.png

   paletteuse
       Use a palette to downsample an input video stream.

       The filter takes two inputs: one video stream and a palette. The
       palette must be a 256 pixels image.

       It accepts the following options:

       dither
	   Select dithering mode. Available algorithms are:

	   bayer
	       Ordered 8x8 bayer dithering (deterministic)

	   heckbert
	       Dithering as defined by Paul Heckbert in 1982 (simple error
	       diffusion).  Note: this dithering is sometimes considered
	       "wrong" and is included as a reference.

	   floyd_steinberg
	       Floyd and Steingberg dithering (error diffusion)

	   sierra2
	       Frankie Sierra dithering v2 (error diffusion)

	   sierra2_4a
	       Frankie Sierra dithering v2 "Lite" (error diffusion)

	   Default is sierra2_4a.

       bayer_scale
	   When bayer dithering is selected, this option defines the scale of
	   the pattern (how much the crosshatch pattern is visible). A low
	   value means more visible pattern for less banding, and higher value
	   means less visible pattern at the cost of more banding.

	   The option must be an integer value in the range [0,5]. Default is
	   2.

       diff_mode
	   If set, define the zone to process

	   rectangle
	       Only the changing rectangle will be reprocessed. This is
	       similar to GIF cropping/offsetting compression mechanism. This
	       option can be useful for speed if only a part of the image is
	       changing, and has use cases such as limiting the scope of the
	       error diffusal dither to the rectangle that bounds the moving
	       scene (it leads to more deterministic output if the scene
	       doesn't change much, and as a result less moving noise and
	       better GIF compression).

	   Default is none.

       new Take new palette for each output frame.

       Examples

       ·   Use a palette (generated for example with palettegen) to encode a
	   GIF using ffmpeg:

		   ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

   perspective
       Correct perspective of video not recorded perpendicular to the screen.

       A description of the accepted parameters follows.

       x0
       y0
       x1
       y1
       x2
       y2
       x3
       y3  Set coordinates expression for top left, top right, bottom left and
	   bottom right corners.  Default values are "0:0:W:0:0:H:W:H" with
	   which perspective will remain unchanged.  If the "sense" option is
	   set to "source", then the specified points will be sent to the
	   corners of the destination. If the "sense" option is set to
	   "destination", then the corners of the source will be sent to the
	   specified coordinates.

	   The expressions can use the following variables:

	   W
	   H   the width and height of video frame.

	   in  Input frame count.

	   on  Output frame count.

       interpolation
	   Set interpolation for perspective correction.

	   It accepts the following values:

	   linear
	   cubic

	   Default value is linear.

       sense
	   Set interpretation of coordinate options.

	   It accepts the following values:

	   0, source
	       Send point in the source specified by the given coordinates to
	       the corners of the destination.

	   1, destination
	       Send the corners of the source to the point in the destination
	       specified by the given coordinates.

	       Default value is source.

       eval
	   Set when the expressions for coordinates x0,y0,...x3,y3 are
	   evaluated.

	   It accepts the following values:

	   init
	       only evaluate expressions once during the filter initialization
	       or when a command is processed

	   frame
	       evaluate expressions for each incoming frame

	   Default value is init.

   phase
       Delay interlaced video by one field time so that the field order
       changes.

       The intended use is to fix PAL movies that have been captured with the
       opposite field order to the film-to-video transfer.

       A description of the accepted parameters follows.

       mode
	   Set phase mode.

	   It accepts the following values:

	   t   Capture field order top-first, transfer bottom-first.  Filter
	       will delay the bottom field.

	   b   Capture field order bottom-first, transfer top-first.  Filter
	       will delay the top field.

	   p   Capture and transfer with the same field order. This mode only
	       exists for the documentation of the other options to refer to,
	       but if you actually select it, the filter will faithfully do
	       nothing.

	   a   Capture field order determined automatically by field flags,
	       transfer opposite.  Filter selects among t and b modes on a
	       frame by frame basis using field flags. If no field information
	       is available, then this works just like u.

	   u   Capture unknown or varying, transfer opposite.  Filter selects
	       among t and b on a frame by frame basis by analyzing the images
	       and selecting the alternative that produces best match between
	       the fields.

	   T   Capture top-first, transfer unknown or varying.	Filter selects
	       among t and p using image analysis.

	   B   Capture bottom-first, transfer unknown or varying.  Filter
	       selects among b and p using image analysis.

	   A   Capture determined by field flags, transfer unknown or varying.
	       Filter selects among t, b and p using field flags and image
	       analysis. If no field information is available, then this works
	       just like U. This is the default mode.

	   U   Both capture and transfer unknown or varying.  Filter selects
	       among t, b and p using image analysis only.

   pixdesctest
       Pixel format descriptor test filter, mainly useful for internal
       testing. The output video should be equal to the input video.

       For example:

	       format=monow, pixdesctest

       can be used to test the monowhite pixel format descriptor definition.

   pixscope
       Display sample values of color channels. Mainly useful for checking
       color and levels. Minimum supported resolution is 640x480.

       The filters accept the following options:

       x   Set scope X position, relative offset on X axis.

       y   Set scope Y position, relative offset on Y axis.

       w   Set scope width.

       h   Set scope height.

       o   Set window opacity. This window also holds statistics about pixel
	   area.

       wx  Set window X position, relative offset on X axis.

       wy  Set window Y position, relative offset on Y axis.

   pp
       Enable the specified chain of postprocessing subfilters using
       libpostproc. This library should be automatically selected with a GPL
       build ("--enable-gpl").	Subfilters must be separated by '/' and can be
       disabled by prepending a '-'.  Each subfilter and some options have a
       short and a long name that can be used interchangeably, i.e. dr/dering
       are the same.

       The filters accept the following options:

       subfilters
	   Set postprocessing subfilters string.

       All subfilters share common options to determine their scope:

       a/autoq
	   Honor the quality commands for this subfilter.

       c/chrom
	   Do chrominance filtering, too (default).

       y/nochrom
	   Do luminance filtering only (no chrominance).

       n/noluma
	   Do chrominance filtering only (no luminance).

       These options can be appended after the subfilter name, separated by a
       '|'.

       Available subfilters are:

       hb/hdeblock[|difference[|flatness]]
	   Horizontal deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       vb/vdeblock[|difference[|flatness]]
	   Vertical deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       ha/hadeblock[|difference[|flatness]]
	   Accurate horizontal deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       va/vadeblock[|difference[|flatness]]
	   Accurate vertical deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       The horizontal and vertical deblocking filters share the difference and
       flatness values so you cannot set different horizontal and vertical
       thresholds.

       h1/x1hdeblock
	   Experimental horizontal deblocking filter

       v1/x1vdeblock
	   Experimental vertical deblocking filter

       dr/dering
	   Deringing filter

       tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
       reducer
	   threshold1
	       larger -> stronger filtering

	   threshold2
	       larger -> stronger filtering

	   threshold3
	       larger -> stronger filtering

       al/autolevels[:f/fullyrange], automatic brightness / contrast
       correction
	   f/fullyrange
	       Stretch luminance to "0-255".

       lb/linblenddeint
	   Linear blend deinterlacing filter that deinterlaces the given block
	   by filtering all lines with a "(1 2 1)" filter.

       li/linipoldeint
	   Linear interpolating deinterlacing filter that deinterlaces the
	   given block by linearly interpolating every second line.

       ci/cubicipoldeint
	   Cubic interpolating deinterlacing filter deinterlaces the given
	   block by cubically interpolating every second line.

       md/mediandeint
	   Median deinterlacing filter that deinterlaces the given block by
	   applying a median filter to every second line.

       fd/ffmpegdeint
	   FFmpeg deinterlacing filter that deinterlaces the given block by
	   filtering every second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
	   Vertically applied FIR lowpass deinterlacing filter that
	   deinterlaces the given block by filtering all lines with a "(-1 2 6
	   2 -1)" filter.

       fq/forceQuant[|quantizer]
	   Overrides the quantizer table from the input with the constant
	   quantizer you specify.

	   quantizer
	       Quantizer to use

       de/default
	   Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
	   Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       ·   Apply horizontal and vertical deblocking, deringing and automatic
	   brightness/contrast:

		   pp=hb/vb/dr/al

       ·   Apply default filters without brightness/contrast correction:

		   pp=de/-al

       ·   Apply default filters and temporal denoiser:

		   pp=default/tmpnoise|1|2|3

       ·   Apply deblocking on luminance only, and switch vertical deblocking
	   on or off automatically depending on available CPU time:

		   pp=hb|y/vb|a

   pp7
       Apply Postprocessing filter 7. It is variant of the spp filter, similar
       to spp = 6 with 7 point DCT, where only the center sample is used after
       IDCT.

       The filter accepts the following options:

       qp  Force a constant quantization parameter. It accepts an integer in
	   range 0 to 63. If not set, the filter will use the QP from the
	   video stream (if available).

       mode
	   Set thresholding mode. Available modes are:

	   hard
	       Set hard thresholding.

	   soft
	       Set soft thresholding (better de-ringing effect, but likely
	       blurrier).

	   medium
	       Set medium thresholding (good results, default).

   premultiply
       Apply alpha premultiply effect to input video stream using first plane
       of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       inplace
	   Do not require 2nd input for processing, instead use alpha plane
	   from input stream.

   prewitt
       Apply prewitt operator to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

   pseudocolor
       Alter frame colors in video with pseudocolors.

       This filter accept the following options:

       c0  set pixel first component expression

       c1  set pixel second component expression

       c2  set pixel third component expression

       c3  set pixel fourth component expression, corresponds to the alpha
	   component

       i   set component to use as base for altering colors

       Each of them specifies the expression to use for computing the lookup
       table for the corresponding pixel component values.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       ymin, umin, vmin, amin
	   The minimum allowed component value.

       ymax, umax, vmax, amax
	   The maximum allowed component value.

       All expressions default to "val".

       Examples

       ·   Change too high luma values to gradient:

		   pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"

   psnr
       Obtain the average, maximum and minimum PSNR (Peak Signal to Noise
       Ratio) between two input videos.

       This filter takes in input two input videos, the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the PSNR.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average PSNR is printed through the logging system.

       The filter stores the accumulated MSE (mean squared error) of each
       frame, and at the end of the processing it is averaged across all
       frames equally, and the following formula is applied to obtain the
       PSNR:

	       PSNR = 10*log10(MAX^2/MSE)

       Where MAX is the average of the maximum values of each component of the
       image.

       The description of the accepted parameters follows.

       stats_file, f
	   If specified the filter will use the named file to save the PSNR of
	   each individual frame. When filename equals "-" the data is sent to
	   standard output.

       stats_version
	   Specifies which version of the stats file format to use. Details of
	   each format are written below.  Default value is 1.

       stats_add_max
	   Determines whether the max value is output to the stats log.
	   Default value is 0.	Requires stats_version >= 2. If this is set
	   and stats_version < 2, the filter will return an error.

       This filter also supports the framesync options.

       The file printed if stats_file is selected, contains a sequence of
       key/value pairs of the form key:value for each compared couple of
       frames.

       If a stats_version greater than 1 is specified, a header line precedes
       the list of per-frame-pair stats, with key value pairs following the
       frame format with the following parameters:

       psnr_log_version
	   The version of the log file format. Will match stats_version.

       fields
	   A comma separated list of the per-frame-pair parameters included in
	   the log.

       A description of each shown per-frame-pair parameter follows:

       n   sequential number of the input frame, starting from 1

       mse_avg
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames, averaged over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames for the component specified by the suffix.

       psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
	   Peak Signal to Noise ratio of the compared frames for the component
	   specified by the suffix.

       max_avg, max_y, max_u, max_v
	   Maximum allowed value for each channel, and average over all
	   channels.

       For example:

	       movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
	       [main][ref] psnr="stats_file=stats.log" [out]

       On this example the input file being processed is compared with the
       reference file ref_movie.mpg. The PSNR of each individual frame is
       stored in stats.log.

   pullup
       Pulldown reversal (inverse telecine) filter, capable of handling mixed
       hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps
       progressive content.

       The pullup filter is designed to take advantage of future context in
       making its decisions. This filter is stateless in the sense that it
       does not lock onto a pattern to follow, but it instead looks forward to
       the following fields in order to identify matches and rebuild
       progressive frames.

       To produce content with an even framerate, insert the fps filter after
       pullup, use "fps=24000/1001" if the input frame rate is 29.97fps,
       "fps=24" for 30fps and the (rare) telecined 25fps input.

       The filter accepts the following options:

       jl
       jr
       jt
       jb  These options set the amount of "junk" to ignore at the left,
	   right, top, and bottom of the image, respectively. Left and right
	   are in units of 8 pixels, while top and bottom are in units of 2
	   lines.  The default is 8 pixels on each side.

       sb  Set the strict breaks. Setting this option to 1 will reduce the
	   chances of filter generating an occasional mismatched frame, but it
	   may also cause an excessive number of frames to be dropped during
	   high motion sequences.  Conversely, setting it to -1 will make
	   filter match fields more easily.  This may help processing of video
	   where there is slight blurring between the fields, but may also
	   cause there to be interlaced frames in the output.  Default value
	   is 0.

       mp  Set the metric plane to use. It accepts the following values:

	   l   Use luma plane.

	   u   Use chroma blue plane.

	   v   Use chroma red plane.

	   This option may be set to use chroma plane instead of the default
	   luma plane for doing filter's computations. This may improve
	   accuracy on very clean source material, but more likely will
	   decrease accuracy, especially if there is chroma noise (rainbow
	   effect) or any grayscale video.  The main purpose of setting mp to
	   a chroma plane is to reduce CPU load and make pullup usable in
	   realtime on slow machines.

       For best results (without duplicated frames in the output file) it is
       necessary to change the output frame rate. For example, to inverse
       telecine NTSC input:

	       ffmpeg -i input -vf pullup -r 24000/1001 ...

   qp
       Change video quantization parameters (QP).

       The filter accepts the following option:

       qp  Set expression for quantization parameter.

       The expression is evaluated through the eval API and can contain, among
       others, the following constants:

       known
	   1 if index is not 129, 0 otherwise.

       qp  Sequential index starting from -129 to 128.

       Examples

       ·   Some equation like:

		   qp=2+2*sin(PI*qp)

   random
       Flush video frames from internal cache of frames into a random order.
       No frame is discarded.  Inspired by frei0r nervous filter.

       frames
	   Set size in number of frames of internal cache, in range from 2 to
	   512. Default is 30.

       seed
	   Set seed for random number generator, must be an integer included
	   between 0 and "UINT32_MAX". If not specified, or if explicitly set
	   to less than 0, the filter will try to use a good random seed on a
	   best effort basis.

   readeia608
       Read closed captioning (EIA-608) information from the top lines of a
       video frame.

       This filter adds frame metadata for "lavfi.readeia608.X.cc" and
       "lavfi.readeia608.X.line", where "X" is the number of the identified
       line with EIA-608 data (starting from 0). A description of each
       metadata value follows:

       lavfi.readeia608.X.cc
	   The two bytes stored as EIA-608 data (printed in hexadecimal).

       lavfi.readeia608.X.line
	   The number of the line on which the EIA-608 data was identified and
	   read.

       This filter accepts the following options:

       scan_min
	   Set the line to start scanning for EIA-608 data. Default is 0.

       scan_max
	   Set the line to end scanning for EIA-608 data. Default is 29.

       mac Set minimal acceptable amplitude change for sync codes detection.
	   Default is 0.2. Allowed range is "[0.001 - 1]".

       spw Set the ratio of width reserved for sync code detection.  Default
	   is 0.27. Allowed range is "[0.01 - 0.7]".

       mhd Set the max peaks height difference for sync code detection.
	   Default is 0.1. Allowed range is "[0.0 - 0.5]".

       mpd Set max peaks period difference for sync code detection.  Default
	   is 0.1. Allowed range is "[0.0 - 0.5]".

       msd Set the first two max start code bits differences.  Default is
	   0.02. Allowed range is "[0.0 - 0.5]".

       bhd Set the minimum ratio of bits height compared to 3rd start code
	   bit.  Default is 0.75. Allowed range is "[0.01 - 1]".

       th_w
	   Set the white color threshold. Default is 0.35. Allowed range is
	   "[0.1 - 1]".

       th_b
	   Set the black color threshold. Default is 0.15. Allowed range is
	   "[0.0 - 0.5]".

       chp Enable checking the parity bit. In the event of a parity error, the
	   filter will output 0x00 for that character. Default is false.

       Examples

       ·   Output a csv with presentation time and the first two lines of
	   identified EIA-608 captioning data.

		   ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv

   readvitc
       Read vertical interval timecode (VITC) information from the top lines
       of a video frame.

       The filter adds frame metadata key "lavfi.readvitc.tc_str" with the
       timecode value, if a valid timecode has been detected. Further metadata
       key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
       data has been found or not.

       This filter accepts the following options:

       scan_max
	   Set the maximum number of lines to scan for VITC data. If the value
	   is set to "-1" the full video frame is scanned. Default is 45.

       thr_b
	   Set the luma threshold for black. Accepts float numbers in the
	   range [0.0,1.0], default value is 0.2. The value must be equal or
	   less than "thr_w".

       thr_w
	   Set the luma threshold for white. Accepts float numbers in the
	   range [0.0,1.0], default value is 0.6. The value must be equal or
	   greater than "thr_b".

       Examples

       ·   Detect and draw VITC data onto the video frame; if no valid VITC is
	   detected, draw "--:--:--:--" as a placeholder:

		   ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

   remap
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x, y)
       position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
       out of range, zero value for pixel will be used for destination pixel.

       Xmap and Ymap input video streams must be of same dimensions. Output
       video stream will have Xmap/Ymap video stream dimensions.  Xmap and
       Ymap input video streams are 16bit depth, single channel.

   removegrain
       The removegrain filter is a spatial denoiser for progressive video.

       m0  Set mode for the first plane.

       m1  Set mode for the second plane.

       m2  Set mode for the third plane.

       m3  Set mode for the fourth plane.

       Range of mode is from 0 to 24. Description of each mode follows:

       0   Leave input plane unchanged. Default.

       1   Clips the pixel with the minimum and maximum of the 8 neighbour
	   pixels.

       2   Clips the pixel with the second minimum and maximum of the 8
	   neighbour pixels.

       3   Clips the pixel with the third minimum and maximum of the 8
	   neighbour pixels.

       4   Clips the pixel with the fourth minimum and maximum of the 8
	   neighbour pixels.  This is equivalent to a median filter.

       5   Line-sensitive clipping giving the minimal change.

       6   Line-sensitive clipping, intermediate.

       7   Line-sensitive clipping, intermediate.

       8   Line-sensitive clipping, intermediate.

       9   Line-sensitive clipping on a line where the neighbours pixels are
	   the closest.

       10  Replaces the target pixel with the closest neighbour.

       11  [1 2 1] horizontal and vertical kernel blur.

       12  Same as mode 11.

       13  Bob mode, interpolates top field from the line where the neighbours
	   pixels are the closest.

       14  Bob mode, interpolates bottom field from the line where the
	   neighbours pixels are the closest.

       15  Bob mode, interpolates top field. Same as 13 but with a more
	   complicated interpolation formula.

       16  Bob mode, interpolates bottom field. Same as 14 but with a more
	   complicated interpolation formula.

       17  Clips the pixel with the minimum and maximum of respectively the
	   maximum and minimum of each pair of opposite neighbour pixels.

       18  Line-sensitive clipping using opposite neighbours whose greatest
	   distance from the current pixel is minimal.

       19  Replaces the pixel with the average of its 8 neighbours.

       20  Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

       21  Clips pixels using the averages of opposite neighbour.

       22  Same as mode 21 but simpler and faster.

       23  Small edge and halo removal, but reputed useless.

       24  Similar as 23.

   removelogo
       Suppress a TV station logo, using an image file to determine which
       pixels comprise the logo. It works by filling in the pixels that
       comprise the logo with neighboring pixels.

       The filter accepts the following options:

       filename, f
	   Set the filter bitmap file, which can be any image format supported
	   by libavformat. The width and height of the image file must match
	   those of the video stream being processed.

       Pixels in the provided bitmap image with a value of zero are not
       considered part of the logo, non-zero pixels are considered part of the
       logo. If you use white (255) for the logo and black (0) for the rest,
       you will be safe. For making the filter bitmap, it is recommended to
       take a screen capture of a black frame with the logo visible, and then
       using a threshold filter followed by the erode filter once or twice.

       If needed, little splotches can be fixed manually. Remember that if
       logo pixels are not covered, the filter quality will be much reduced.
       Marking too many pixels as part of the logo does not hurt as much, but
       it will increase the amount of blurring needed to cover over the image
       and will destroy more information than necessary, and extra pixels will
       slow things down on a large logo.

   repeatfields
       This filter uses the repeat_field flag from the Video ES headers and
       hard repeats fields based on its value.

   reverse
       Reverse a video clip.

       Warning: This filter requires memory to buffer the entire clip, so
       trimming is suggested.

       Examples

       ·   Take the first 5 seconds of a clip, and reverse it.

		   trim=end=5,reverse

   roberts
       Apply roberts cross operator to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

   rotate
       Rotate video by an arbitrary angle expressed in radians.

       The filter accepts the following options:

       A description of the optional parameters follows.

       angle, a
	   Set an expression for the angle by which to rotate the input video
	   clockwise, expressed as a number of radians. A negative value will
	   result in a counter-clockwise rotation. By default it is set to
	   "0".

	   This expression is evaluated for each frame.

       out_w, ow
	   Set the output width expression, default value is "iw".  This
	   expression is evaluated just once during configuration.

       out_h, oh
	   Set the output height expression, default value is "ih".  This
	   expression is evaluated just once during configuration.

       bilinear
	   Enable bilinear interpolation if set to 1, a value of 0 disables
	   it. Default value is 1.

       fillcolor, c
	   Set the color used to fill the output area not covered by the
	   rotated image. For the general syntax of this option, check the
	   "Color" section in the ffmpeg-utils manual. If the special value
	   "none" is selected then no background is printed (useful for
	   example if the background is never shown).

	   Default value is "black".

       The expressions for the angle and the output size can contain the
       following constants and functions:

       n   sequential number of the input frame, starting from 0. It is always
	   NAN before the first frame is filtered.

       t   time in seconds of the input frame, it is set to 0 when the filter
	   is configured. It is always NAN before the first frame is filtered.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_w, iw
       in_h, ih
	   the input video width and height

       out_w, ow
       out_h, oh
	   the output width and height, that is the size of the padded area as
	   specified by the width and height expressions

       rotw(a)
       roth(a)
	   the minimal width/height required for completely containing the
	   input video rotated by a radians.

	   These are only available when computing the out_w and out_h
	   expressions.

       Examples

       ·   Rotate the input by PI/6 radians clockwise:

		   rotate=PI/6

       ·   Rotate the input by PI/6 radians counter-clockwise:

		   rotate=-PI/6

       ·   Rotate the input by 45 degrees clockwise:

		   rotate=45*PI/180

       ·   Apply a constant rotation with period T, starting from an angle of
	   PI/3:

		   rotate=PI/3+2*PI*t/T

       ·   Make the input video rotation oscillating with a period of T
	   seconds and an amplitude of A radians:

		   rotate=A*sin(2*PI/T*t)

       ·   Rotate the video, output size is chosen so that the whole rotating
	   input video is always completely contained in the output:

		   rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

       ·   Rotate the video, reduce the output size so that no background is
	   ever shown:

		   rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

       Commands

       The filter supports the following commands:

       a, angle
	   Set the angle expression.  The command accepts the same syntax of
	   the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   sab
       Apply Shape Adaptive Blur.

       The filter accepts the following options:

       luma_radius, lr
	   Set luma blur filter strength, must be a value in range 0.1-4.0,
	   default value is 1.0. A greater value will result in a more blurred
	   image, and in slower processing.

       luma_pre_filter_radius, lpfr
	   Set luma pre-filter radius, must be a value in the 0.1-2.0 range,
	   default value is 1.0.

       luma_strength, ls
	   Set luma maximum difference between pixels to still be considered,
	   must be a value in the 0.1-100.0 range, default value is 1.0.

       chroma_radius, cr
	   Set chroma blur filter strength, must be a value in range -0.9-4.0.
	   A greater value will result in a more blurred image, and in slower
	   processing.

       chroma_pre_filter_radius, cpfr
	   Set chroma pre-filter radius, must be a value in the -0.9-2.0
	   range.

       chroma_strength, cs
	   Set chroma maximum difference between pixels to still be
	   considered, must be a value in the -0.9-100.0 range.

       Each chroma option value, if not explicitly specified, is set to the
       corresponding luma option value.

   scale
       Scale (resize) the input video, using the libswscale library.

       The scale filter forces the output display aspect ratio to be the same
       of the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the scale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options, or any of the options
       supported by the libswscale scaler.

       See the ffmpeg-scaler manual for the complete list of scaler options.

       width, w
       height, h
	   Set the output video dimension expression. Default value is the
	   input dimension.

	   If the width or w value is 0, the input width is used for the
	   output. If the height or h value is 0, the input height is used for
	   the output.

	   If one and only one of the values is -n with n >= 1, the scale
	   filter will use a value that maintains the aspect ratio of the
	   input image, calculated from the other specified dimension. After
	   that it will, however, make sure that the calculated dimension is
	   divisible by n and adjust the value if necessary.

	   If both values are -n with n >= 1, the behavior will be identical
	   to both values being set to 0 as previously detailed.

	   See below for the list of accepted constants for use in the
	   dimension expression.

       eval
	   Specify when to evaluate width and height expression. It accepts
	   the following values:

	   init
	       Only evaluate expressions once during the filter initialization
	       or when a command is processed.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

       interl
	   Set the interlacing mode. It accepts the following values:

	   1   Force interlaced aware scaling.

	   0   Do not apply interlaced scaling.

	   -1  Select interlaced aware scaling depending on whether the source
	       frames are flagged as interlaced or not.

	   Default value is 0.

       flags
	   Set libswscale scaling flags. See the ffmpeg-scaler manual for the
	   complete list of values. If not explicitly specified the filter
	   applies the default flags.

       param0, param1
	   Set libswscale input parameters for scaling algorithms that need
	   them. See the ffmpeg-scaler manual for the complete documentation.
	   If not explicitly specified the filter applies empty parameters.

       size, s
	   Set the video size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual.

       in_color_matrix
       out_color_matrix
	   Set in/output YCbCr color space type.

	   This allows the autodetected value to be overridden as well as
	   allows forcing a specific value used for the output and encoder.

	   If not specified, the color space type depends on the pixel format.

	   Possible values:

	   auto
	       Choose automatically.

	   bt709
	       Format conforming to International Telecommunication Union
	       (ITU) Recommendation BT.709.

	   fcc Set color space conforming to the United States Federal
	       Communications Commission (FCC) Code of Federal Regulations
	       (CFR) Title 47 (2003) 73.682 (a).

	   bt601
	       Set color space conforming to:

	       ·   ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

	       ·   ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

	       ·   Society of Motion Picture and Television Engineers (SMPTE)
		   ST 170:2004

	   smpte240m
	       Set color space conforming to SMPTE ST 240:1999.

       in_range
       out_range
	   Set in/output YCbCr sample range.

	   This allows the autodetected value to be overridden as well as
	   allows forcing a specific value used for the output and encoder. If
	   not specified, the range depends on the pixel format. Possible
	   values:

	   auto
	       Choose automatically.

	   jpeg/full/pc
	       Set full range (0-255 in case of 8-bit luma).

	   mpeg/tv
	       Set "MPEG" range (16-235 in case of 8-bit luma).

       force_original_aspect_ratio
	   Enable decreasing or increasing output video width or height if
	   necessary to keep the original aspect ratio. Possible values:

	   disable
	       Scale the video as specified and disable this feature.

	   decrease
	       The output video dimensions will automatically be decreased if
	       needed.

	   increase
	       The output video dimensions will automatically be increased if
	       needed.

	   One useful instance of this option is that when you know a specific
	   device's maximum allowed resolution, you can use this to limit the
	   output video to that, while retaining the aspect ratio. For
	   example, device A allows 1280x720 playback, and your video is
	   1920x800. Using this option (set it to decrease) and specifying
	   1280x720 to the command line makes the output 1280x533.

	   Please note that this is a different thing than specifying -1 for w
	   or h, you still need to specify the output resolution for this
	   option to work.

       The values of the w and h options are expressions containing the
       following constants:

       in_w
       in_h
	   The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
	   horizontal and vertical input chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
	   horizontal and vertical output chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   Scale the input video to a size of 200x100

		   scale=w=200:h=100

	   This is equivalent to:

		   scale=200:100

	   or:

		   scale=200x100

       ·   Specify a size abbreviation for the output size:

		   scale=qcif

	   which can also be written as:

		   scale=size=qcif

       ·   Scale the input to 2x:

		   scale=w=2*iw:h=2*ih

       ·   The above is the same as:

		   scale=2*in_w:2*in_h

       ·   Scale the input to 2x with forced interlaced scaling:

		   scale=2*iw:2*ih:interl=1

       ·   Scale the input to half size:

		   scale=w=iw/2:h=ih/2

       ·   Increase the width, and set the height to the same size:

		   scale=3/2*iw:ow

       ·   Seek Greek harmony:

		   scale=iw:1/PHI*iw
		   scale=ih*PHI:ih

       ·   Increase the height, and set the width to 3/2 of the height:

		   scale=w=3/2*oh:h=3/5*ih

       ·   Increase the size, making the size a multiple of the chroma
	   subsample values:

		   scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

       ·   Increase the width to a maximum of 500 pixels, keeping the same
	   aspect ratio as the input:

		   scale=w='min(500\, iw*3/2):h=-1'

       Commands

       This filter supports the following commands:

       width, w
       height, h
	   Set the output video dimension expression.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   scale_npp
       Use the NVIDIA Performance Primitives (libnpp) to perform scaling
       and/or pixel format conversion on CUDA video frames. Setting the output
       width and height works in the same way as for the scale filter.

       The following additional options are accepted:

       format
	   The pixel format of the output CUDA frames. If set to the string
	   "same" (the default), the input format will be kept. Note that
	   automatic format negotiation and conversion is not yet supported
	   for hardware frames

       interp_algo
	   The interpolation algorithm used for resizing. One of the
	   following:

	   nn  Nearest neighbour.

	   linear
	   cubic
	   cubic2p_bspline
	       2-parameter cubic (B=1, C=0)

	   cubic2p_catmullrom
	       2-parameter cubic (B=0, C=1/2)

	   cubic2p_b05c03
	       2-parameter cubic (B=1/2, C=3/10)

	   super
	       Supersampling

	   lanczos

   scale2ref
       Scale (resize) the input video, based on a reference video.

       See the scale filter for available options, scale2ref supports the same
       but uses the reference video instead of the main input as basis.
       scale2ref also supports the following additional constants for the w
       and h options:

       main_w
       main_h
	   The main input video's width and height

       main_a
	   The same as main_w / main_h

       main_sar
	   The main input video's sample aspect ratio

       main_dar, mdar
	   The main input video's display aspect ratio. Calculated from
	   "(main_w / main_h) * main_sar".

       main_hsub
       main_vsub
	   The main input video's horizontal and vertical chroma subsample
	   values.  For example for the pixel format "yuv422p" hsub is 2 and
	   vsub is 1.

       Examples

       ·   Scale a subtitle stream (b) to match the main video (a) in size
	   before overlaying

		   'scale2ref[b][a];[a][b]overlay'

   selectivecolor
       Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of
       colors (such as "reds", "yellows", "greens", "cyans", ...). The
       adjustment range is defined by the "purity" of the color (that is, how
       saturated it already is).

       This filter is similar to the Adobe Photoshop Selective Color tool.

       The filter accepts the following options:

       correction_method
	   Select color correction method.

	   Available values are:

	   absolute
	       Specified adjustments are applied "as-is" (added/subtracted to
	       original pixel component value).

	   relative
	       Specified adjustments are relative to the original component
	       value.

	   Default is "absolute".

       reds
	   Adjustments for red pixels (pixels where the red component is the
	   maximum)

       yellows
	   Adjustments for yellow pixels (pixels where the blue component is
	   the minimum)

       greens
	   Adjustments for green pixels (pixels where the green component is
	   the maximum)

       cyans
	   Adjustments for cyan pixels (pixels where the red component is the
	   minimum)

       blues
	   Adjustments for blue pixels (pixels where the blue component is the
	   maximum)

       magentas
	   Adjustments for magenta pixels (pixels where the green component is
	   the minimum)

       whites
	   Adjustments for white pixels (pixels where all components are
	   greater than 128)

       neutrals
	   Adjustments for all pixels except pure black and pure white

       blacks
	   Adjustments for black pixels (pixels where all components are
	   lesser than 128)

       psfile
	   Specify a Photoshop selective color file (".asv") to import the
	   settings from.

       All the adjustment settings (reds, yellows, ...) accept up to 4 space
       separated floating point adjustment values in the [-1,1] range,
       respectively to adjust the amount of cyan, magenta, yellow and black
       for the pixels of its range.

       Examples

       ·   Increase cyan by 50% and reduce yellow by 33% in every green areas,
	   and increase magenta by 27% in blue areas:

		   selectivecolor=greens=.5 0 -.33 0:blues=0 .27

       ·   Use a Photoshop selective color preset:

		   selectivecolor=psfile=MySelectiveColorPresets/Misty.asv

   separatefields
       The "separatefields" takes a frame-based video input and splits each
       frame into its components fields, producing a new half height clip with
       twice the frame rate and twice the frame count.

       This filter use field-dominance information in frame to decide which of
       each pair of fields to place first in the output.  If it gets it wrong
       use setfield filter before "separatefields" filter.

   setdar, setsar
       The "setdar" filter sets the Display Aspect Ratio for the filter output
       video.

       This is done by changing the specified Sample (aka Pixel) Aspect Ratio,
       according to the following equation:

	       <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

       Keep in mind that the "setdar" filter does not modify the pixel
       dimensions of the video frame. Also, the display aspect ratio set by
       this filter may be changed by later filters in the filterchain, e.g. in
       case of scaling or if another "setdar" or a "setsar" filter is applied.

       The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the
       filter output video.

       Note that as a consequence of the application of this filter, the
       output display aspect ratio will change according to the equation
       above.

       Keep in mind that the sample aspect ratio set by the "setsar" filter
       may be changed by later filters in the filterchain, e.g. if another
       "setsar" or a "setdar" filter is applied.

       It accepts the following parameters:

       r, ratio, dar ("setdar" only), sar ("setsar" only)
	   Set the aspect ratio used by the filter.

	   The parameter can be a floating point number string, an expression,
	   or a string of the form num:den, where num and den are the
	   numerator and denominator of the aspect ratio. If the parameter is
	   not specified, it is assumed the value "0".	In case the form
	   "num:den" is used, the ":" character should be escaped.

       max Set the maximum integer value to use for expressing numerator and
	   denominator when reducing the expressed aspect ratio to a rational.
	   Default value is 100.

       The parameter sar is an expression containing the following constants:

       E, PI, PHI
	   These are approximated values for the mathematical constants e
	   (Euler's number), pi (Greek pi), and phi (the golden ratio).

       w, h
	   The input width and height.

       a   These are the same as w / h.

       sar The input sample aspect ratio.

       dar The input display aspect ratio. It is the same as (w / h) * sar.

       hsub, vsub
	   Horizontal and vertical chroma subsample values. For example, for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   To change the display aspect ratio to 16:9, specify one of the
	   following:

		   setdar=dar=1.77777
		   setdar=dar=16/9

       ·   To change the sample aspect ratio to 10:11, specify:

		   setsar=sar=10/11

       ·   To set a display aspect ratio of 16:9, and specify a maximum
	   integer value of 1000 in the aspect ratio reduction, use the
	   command:

		   setdar=ratio=16/9:max=1000

   setfield
       Force field for the output video frame.

       The "setfield" filter marks the interlace type field for the output
       frames. It does not change the input frame, but only sets the
       corresponding property, which affects how the frame is treated by
       following filters (e.g. "fieldorder" or "yadif").

       The filter accepts the following options:

       mode
	   Available values are:

	   auto
	       Keep the same field property.

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

   showinfo
       Show a line containing various information for each input video frame.
       The input video is not modified.

       The shown line contains a sequence of key/value pairs of the form
       key:value.

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The Presentation TimeStamp of the input frame, expressed as a
	   number of time base units. The time base unit depends on the filter
	   input pad.

       pts_time
	   The Presentation TimeStamp of the input frame, expressed as a
	   number of seconds.

       pos The position of the frame in the input stream, or -1 if this
	   information is unavailable and/or meaningless (for example in case
	   of synthetic video).

       fmt The pixel format name.

       sar The sample aspect ratio of the input frame, expressed in the form
	   num/den.

       s   The size of the input frame. For the syntax of this option, check
	   the "Video size" section in the ffmpeg-utils manual.

       i   The type of interlaced mode ("P" for "progressive", "T" for top
	   field first, "B" for bottom field first).

       iskey
	   This is 1 if the frame is a key frame, 0 otherwise.

       type
	   The picture type of the input frame ("I" for an I-frame, "P" for a
	   P-frame, "B" for a B-frame, or "?" for an unknown type).  Also
	   refer to the documentation of the "AVPictureType" enum and of the
	   "av_get_picture_type_char" function defined in libavutil/avutil.h.

       checksum
	   The Adler-32 checksum (printed in hexadecimal) of all the planes of
	   the input frame.

       plane_checksum
	   The Adler-32 checksum (printed in hexadecimal) of each plane of the
	   input frame, expressed in the form "[c0 c1 c2 c3]".

   showpalette
       Displays the 256 colors palette of each frame. This filter is only
       relevant for pal8 pixel format frames.

       It accepts the following option:

       s   Set the size of the box used to represent one palette color entry.
	   Default is 30 (for a "30x30" pixel box).

   shuffleframes
       Reorder and/or duplicate and/or drop video frames.

       It accepts the following parameters:

       mapping
	   Set the destination indexes of input frames.  This is space or '|'
	   separated list of indexes that maps input frames to output frames.
	   Number of indexes also sets maximal value that each index may have.
	   '-1' index have special meaning and that is to drop frame.

       The first frame has the index 0. The default is to keep the input
       unchanged.

       Examples

       ·   Swap second and third frame of every three frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT

       ·   Swap 10th and 1st frame of every ten frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT

   shuffleplanes
       Reorder and/or duplicate video planes.

       It accepts the following parameters:

       map0
	   The index of the input plane to be used as the first output plane.

       map1
	   The index of the input plane to be used as the second output plane.

       map2
	   The index of the input plane to be used as the third output plane.

       map3
	   The index of the input plane to be used as the fourth output plane.

       The first plane has the index 0. The default is to keep the input
       unchanged.

       Examples

       ·   Swap the second and third planes of the input:

		   ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

   signalstats
       Evaluate various visual metrics that assist in determining issues
       associated with the digitization of analog video media.

       By default the filter will log these metadata values:

       YMIN
	   Display the minimal Y value contained within the input frame.
	   Expressed in range of [0-255].

       YLOW
	   Display the Y value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       YAVG
	   Display the average Y value within the input frame. Expressed in
	   range of [0-255].

       YHIGH
	   Display the Y value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       YMAX
	   Display the maximum Y value contained within the input frame.
	   Expressed in range of [0-255].

       UMIN
	   Display the minimal U value contained within the input frame.
	   Expressed in range of [0-255].

       ULOW
	   Display the U value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       UAVG
	   Display the average U value within the input frame. Expressed in
	   range of [0-255].

       UHIGH
	   Display the U value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       UMAX
	   Display the maximum U value contained within the input frame.
	   Expressed in range of [0-255].

       VMIN
	   Display the minimal V value contained within the input frame.
	   Expressed in range of [0-255].

       VLOW
	   Display the V value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       VAVG
	   Display the average V value within the input frame. Expressed in
	   range of [0-255].

       VHIGH
	   Display the V value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       VMAX
	   Display the maximum V value contained within the input frame.
	   Expressed in range of [0-255].

       SATMIN
	   Display the minimal saturation value contained within the input
	   frame.  Expressed in range of [0-~181.02].

       SATLOW
	   Display the saturation value at the 10% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATAVG
	   Display the average saturation value within the input frame.
	   Expressed in range of [0-~181.02].

       SATHIGH
	   Display the saturation value at the 90% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATMAX
	   Display the maximum saturation value contained within the input
	   frame.  Expressed in range of [0-~181.02].

       HUEMED
	   Display the median value for hue within the input frame. Expressed
	   in range of [0-360].

       HUEAVG
	   Display the average value for hue within the input frame. Expressed
	   in range of [0-360].

       YDIF
	   Display the average of sample value difference between all values
	   of the Y plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       UDIF
	   Display the average of sample value difference between all values
	   of the U plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       VDIF
	   Display the average of sample value difference between all values
	   of the V plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       YBITDEPTH
	   Display bit depth of Y plane in current frame.  Expressed in range
	   of [0-16].

       UBITDEPTH
	   Display bit depth of U plane in current frame.  Expressed in range
	   of [0-16].

       VBITDEPTH
	   Display bit depth of V plane in current frame.  Expressed in range
	   of [0-16].

       The filter accepts the following options:

       stat
       out stat specify an additional form of image analysis.  out output
	   video with the specified type of pixel highlighted.

	   Both options accept the following values:

	   tout
	       Identify temporal outliers pixels. A temporal outlier is a
	       pixel unlike the neighboring pixels of the same field. Examples
	       of temporal outliers include the results of video dropouts,
	       head clogs, or tape tracking issues.

	   vrep
	       Identify vertical line repetition. Vertical line repetition
	       includes similar rows of pixels within a frame. In born-digital
	       video vertical line repetition is common, but this pattern is
	       uncommon in video digitized from an analog source. When it
	       occurs in video that results from the digitization of an analog
	       source it can indicate concealment from a dropout compensator.

	   brng
	       Identify pixels that fall outside of legal broadcast range.

       color, c
	   Set the highlight color for the out option. The default color is
	   yellow.

       Examples

       ·   Output data of various video metrics:

		   ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

       ·   Output specific data about the minimum and maximum values of the Y
	   plane per frame:

		   ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

       ·   Playback video while highlighting pixels that are outside of
	   broadcast range in red.

		   ffplay example.mov -vf signalstats="out=brng:color=red"

       ·   Playback video with signalstats metadata drawn over the frame.

		   ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

	   The contents of signalstat_drawtext.txt used in the command are:

		   time %{pts:hms}
		   Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
		   U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
		   V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
		   saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

   signature
       Calculates the MPEG-7 Video Signature. The filter can handle more than
       one input. In this case the matching between the inputs can be
       calculated additionally.  The filter always passes through the first
       input. The signature of each stream can be written into a file.

       It accepts the following options:

       detectmode
	   Enable or disable the matching process.

	   Available values are:

	   off Disable the calculation of a matching (default).

	   full
	       Calculate the matching for the whole video and output whether
	       the whole video matches or only parts.

	   fast
	       Calculate only until a matching is found or the video ends.
	       Should be faster in some cases.

       nb_inputs
	   Set the number of inputs. The option value must be a non negative
	   integer.  Default value is 1.

       filename
	   Set the path to which the output is written. If there is more than
	   one input, the path must be a prototype, i.e. must contain %d or
	   %0nd (where n is a positive integer), that will be replaced with
	   the input number. If no filename is specified, no output will be
	   written. This is the default.

       format
	   Choose the output format.

	   Available values are:

	   binary
	       Use the specified binary representation (default).

	   xml Use the specified xml representation.

       th_d
	   Set threshold to detect one word as similar. The option value must
	   be an integer greater than zero. The default value is 9000.

       th_dc
	   Set threshold to detect all words as similar. The option value must
	   be an integer greater than zero. The default value is 60000.

       th_xh
	   Set threshold to detect frames as similar. The option value must be
	   an integer greater than zero. The default value is 116.

       th_di
	   Set the minimum length of a sequence in frames to recognize it as
	   matching sequence. The option value must be a non negative integer
	   value.  The default value is 0.

       th_it
	   Set the minimum relation, that matching frames to all frames must
	   have.  The option value must be a double value between 0 and 1. The
	   default value is 0.5.

       Examples

       ·   To calculate the signature of an input video and store it in
	   signature.bin:

		   ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -

       ·   To detect whether two videos match and store the signatures in XML
	   format in signature0.xml and signature1.xml:

		   ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -

   smartblur
       Blur the input video without impacting the outlines.

       It accepts the following options:

       luma_radius, lr
	   Set the luma radius. The option value must be a float number in the
	   range [0.1,5.0] that specifies the variance of the gaussian filter
	   used to blur the image (slower if larger). Default value is 1.0.

       luma_strength, ls
	   Set the luma strength. The option value must be a float number in
	   the range [-1.0,1.0] that configures the blurring. A value included
	   in [0.0,1.0] will blur the image whereas a value included in
	   [-1.0,0.0] will sharpen the image. Default value is 1.0.

       luma_threshold, lt
	   Set the luma threshold used as a coefficient to determine whether a
	   pixel should be blurred or not. The option value must be an integer
	   in the range [-30,30]. A value of 0 will filter all the image, a
	   value included in [0,30] will filter flat areas and a value
	   included in [-30,0] will filter edges. Default value is 0.

       chroma_radius, cr
	   Set the chroma radius. The option value must be a float number in
	   the range [0.1,5.0] that specifies the variance of the gaussian
	   filter used to blur the image (slower if larger). Default value is
	   luma_radius.

       chroma_strength, cs
	   Set the chroma strength. The option value must be a float number in
	   the range [-1.0,1.0] that configures the blurring. A value included
	   in [0.0,1.0] will blur the image whereas a value included in
	   [-1.0,0.0] will sharpen the image. Default value is luma_strength.

       chroma_threshold, ct
	   Set the chroma threshold used as a coefficient to determine whether
	   a pixel should be blurred or not. The option value must be an
	   integer in the range [-30,30]. A value of 0 will filter all the
	   image, a value included in [0,30] will filter flat areas and a
	   value included in [-30,0] will filter edges. Default value is
	   luma_threshold.

       If a chroma option is not explicitly set, the corresponding luma value
       is set.

   ssim
       Obtain the SSIM (Structural SImilarity Metric) between two input
       videos.

       This filter takes in input two input videos, the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the SSIM.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The filter stores the calculated SSIM of each frame.

       The description of the accepted parameters follows.

       stats_file, f
	   If specified the filter will use the named file to save the SSIM of
	   each individual frame. When filename equals "-" the data is sent to
	   standard output.

       The file printed if stats_file is selected, contains a sequence of
       key/value pairs of the form key:value for each compared couple of
       frames.

       A description of each shown parameter follows:

       n   sequential number of the input frame, starting from 1

       Y, U, V, R, G, B
	   SSIM of the compared frames for the component specified by the
	   suffix.

       All SSIM of the compared frames for the whole frame.

       dB  Same as above but in dB representation.

       This filter also supports the framesync options.

       For example:

	       movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
	       [main][ref] ssim="stats_file=stats.log" [out]

       On this example the input file being processed is compared with the
       reference file ref_movie.mpg. The SSIM of each individual frame is
       stored in stats.log.

       Another example with both psnr and ssim at same time:

	       ffmpeg -i main.mpg -i ref.mpg -lavfi  "ssim;[0:v][1:v]psnr" -f null -

   stereo3d
       Convert between different stereoscopic image formats.

       The filters accept the following options:

       in  Set stereoscopic image format of input.

	   Available values for input image formats are:

	   sbsl
	       side by side parallel (left eye left, right eye right)

	   sbsr
	       side by side crosseye (right eye left, left eye right)

	   sbs2l
	       side by side parallel with half width resolution (left eye
	       left, right eye right)

	   sbs2r
	       side by side crosseye with half width resolution (right eye
	       left, left eye right)

	   abl above-below (left eye above, right eye below)

	   abr above-below (right eye above, left eye below)

	   ab2l
	       above-below with half height resolution (left eye above, right
	       eye below)

	   ab2r
	       above-below with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	   irl interleaved rows (left eye has top row, right eye starts on
	       next row)

	   irr interleaved rows (right eye has top row, left eye starts on
	       next row)

	   icl interleaved columns, left eye first

	   icr interleaved columns, right eye first

	       Default value is sbsl.

       out Set stereoscopic image format of output.

	   sbsl
	       side by side parallel (left eye left, right eye right)

	   sbsr
	       side by side crosseye (right eye left, left eye right)

	   sbs2l
	       side by side parallel with half width resolution (left eye
	       left, right eye right)

	   sbs2r
	       side by side crosseye with half width resolution (right eye
	       left, left eye right)

	   abl above-below (left eye above, right eye below)

	   abr above-below (right eye above, left eye below)

	   ab2l
	       above-below with half height resolution (left eye above, right
	       eye below)

	   ab2r
	       above-below with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	   irl interleaved rows (left eye has top row, right eye starts on
	       next row)

	   irr interleaved rows (right eye has top row, left eye starts on
	       next row)

	   arbg
	       anaglyph red/blue gray (red filter on left eye, blue filter on
	       right eye)

	   argg
	       anaglyph red/green gray (red filter on left eye, green filter
	       on right eye)

	   arcg
	       anaglyph red/cyan gray (red filter on left eye, cyan filter on
	       right eye)

	   arch
	       anaglyph red/cyan half colored (red filter on left eye, cyan
	       filter on right eye)

	   arcc
	       anaglyph red/cyan color (red filter on left eye, cyan filter on
	       right eye)

	   arcd
	       anaglyph red/cyan color optimized with the least squares
	       projection of dubois (red filter on left eye, cyan filter on
	       right eye)

	   agmg
	       anaglyph green/magenta gray (green filter on left eye, magenta
	       filter on right eye)

	   agmh
	       anaglyph green/magenta half colored (green filter on left eye,
	       magenta filter on right eye)

	   agmc
	       anaglyph green/magenta colored (green filter on left eye,
	       magenta filter on right eye)

	   agmd
	       anaglyph green/magenta color optimized with the least squares
	       projection of dubois (green filter on left eye, magenta filter
	       on right eye)

	   aybg
	       anaglyph yellow/blue gray (yellow filter on left eye, blue
	       filter on right eye)

	   aybh
	       anaglyph yellow/blue half colored (yellow filter on left eye,
	       blue filter on right eye)

	   aybc
	       anaglyph yellow/blue colored (yellow filter on left eye, blue
	       filter on right eye)

	   aybd
	       anaglyph yellow/blue color optimized with the least squares
	       projection of dubois (yellow filter on left eye, blue filter on
	       right eye)

	   ml  mono output (left eye only)

	   mr  mono output (right eye only)

	   chl checkerboard, left eye first

	   chr checkerboard, right eye first

	   icl interleaved columns, left eye first

	   icr interleaved columns, right eye first

	   hdmi
	       HDMI frame pack

	   Default value is arcd.

       Examples

       ·   Convert input video from side by side parallel to anaglyph
	   yellow/blue dubois:

		   stereo3d=sbsl:aybd

       ·   Convert input video from above below (left eye above, right eye
	   below) to side by side crosseye.

		   stereo3d=abl:sbsr

   streamselect, astreamselect
       Select video or audio streams.

       The filter accepts the following options:

       inputs
	   Set number of inputs. Default is 2.

       map Set input indexes to remap to outputs.

       Commands

       The "streamselect" and "astreamselect" filter supports the following
       commands:

       map Set input indexes to remap to outputs.

       Examples

       ·   Select first 5 seconds 1st stream and rest of time 2nd stream:

		   sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0

       ·   Same as above, but for audio:

		   asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0

   sobel
       Apply sobel operator to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

   spp
       Apply a simple postprocessing filter that compresses and decompresses
       the image at several (or - in the case of quality level 6 - all) shifts
       and average the results.

       The filter accepts the following options:

       quality
	   Set quality. This option defines the number of levels for
	   averaging. It accepts an integer in the range 0-6. If set to 0, the
	   filter will have no effect. A value of 6 means the higher quality.
	   For each increment of that value the speed drops by a factor of
	   approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If not set, the filter
	   will use the QP from the video stream (if available).

       mode
	   Set thresholding mode. Available modes are:

	   hard
	       Set hard thresholding (default).

	   soft
	       Set soft thresholding (better de-ringing effect, but likely
	       blurrier).

       use_bframe_qp
	   Enable the use of the QP from the B-Frames if set to 1. Using this
	   option may cause flicker since the B-Frames have often larger QP.
	   Default is 0 (not enabled).

   subtitles
       Draw subtitles on top of input video using the libass library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libass". This filter also requires a build with libavcodec
       and libavformat to convert the passed subtitles file to ASS (Advanced
       Substation Alpha) subtitles format.

       The filter accepts the following options:

       filename, f
	   Set the filename of the subtitle file to read. It must be
	   specified.

       original_size
	   Specify the size of the original video, the video for which the ASS
	   file was composed. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual.  Due to a misdesign in
	   ASS aspect ratio arithmetic, this is necessary to correctly scale
	   the fonts if the aspect ratio has been changed.

       fontsdir
	   Set a directory path containing fonts that can be used by the
	   filter.  These fonts will be used in addition to whatever the font
	   provider uses.

       alpha
	   Process alpha channel, by default alpha channel is untouched.

       charenc
	   Set subtitles input character encoding. "subtitles" filter only.
	   Only useful if not UTF-8.

       stream_index, si
	   Set subtitles stream index. "subtitles" filter only.

       force_style
	   Override default style or script info parameters of the subtitles.
	   It accepts a string containing ASS style format "KEY=VALUE" couples
	   separated by ",".

       If the first key is not specified, it is assumed that the first value
       specifies the filename.

       For example, to render the file sub.srt on top of the input video, use
       the command:

	       subtitles=sub.srt

       which is equivalent to:

	       subtitles=filename=sub.srt

       To render the default subtitles stream from file video.mkv, use:

	       subtitles=video.mkv

       To render the second subtitles stream from that file, use:

	       subtitles=video.mkv:si=1

       To make the subtitles stream from sub.srt appear in transparent green
       "DejaVu Serif", use:

	       subtitles=sub.srt:force_style='FontName=DejaVu Serif,PrimaryColour=&HAA00FF00'

   super2xsai
       Scale the input by 2x and smooth using the Super2xSaI (Scale and
       Interpolate) pixel art scaling algorithm.

       Useful for enlarging pixel art images without reducing sharpness.

   swaprect
       Swap two rectangular objects in video.

       This filter accepts the following options:

       w   Set object width.

       h   Set object height.

       x1  Set 1st rect x coordinate.

       y1  Set 1st rect y coordinate.

       x2  Set 2nd rect x coordinate.

       y2  Set 2nd rect y coordinate.

	   All expressions are evaluated once for each frame.

       The all options are expressions containing the following constants:

       w
       h   The input width and height.

       a   same as w / h

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (w / h) * sar

       n   The number of the input frame, starting from 0.

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
	   is unknown.

       pos the position in the file of the input frame, NAN if unknown

   swapuv
       Swap U & V plane.

   telecine
       Apply telecine process to the video.

       This filter accepts the following options:

       first_field
	   top, t
	       top field first

	   bottom, b
	       bottom field first The default value is "top".

       pattern
	   A string of numbers representing the pulldown pattern you wish to
	   apply.  The default value is 23.

	       Some typical patterns:

	       NTSC output (30i):
	       27.5p: 32222
	       24p: 23 (classic)
	       24p: 2332 (preferred)
	       20p: 33
	       18p: 334
	       16p: 3444

	       PAL output (25i):
	       27.5p: 12222
	       24p: 222222222223 ("Euro pulldown")
	       16.67p: 33
	       16p: 33333334

   threshold
       Apply threshold effect to video stream.

       This filter needs four video streams to perform thresholding.  First
       stream is stream we are filtering.  Second stream is holding threshold
       values, third stream is holding min values, and last, fourth stream is
       holding max values.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       For example if first stream pixel's component value is less then
       threshold value of pixel component from 2nd threshold stream, third
       stream value will picked, otherwise fourth stream pixel component value
       will be picked.

       Using color source filter one can perform various types of
       thresholding:

       Examples

       ·   Binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi

       ·   Inverted binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi

       ·   Truncate binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi

       ·   Threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi

       ·   Inverted threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi

   thumbnail
       Select the most representative frame in a given sequence of consecutive
       frames.

       The filter accepts the following options:

       n   Set the frames batch size to analyze; in a set of n frames, the
	   filter will pick one of them, and then handle the next batch of n
	   frames until the end. Default is 100.

       Since the filter keeps track of the whole frames sequence, a bigger n
       value will result in a higher memory usage, so a high value is not
       recommended.

       Examples

       ·   Extract one picture each 50 frames:

		   thumbnail=50

       ·   Complete example of a thumbnail creation with ffmpeg:

		   ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

   tile
       Tile several successive frames together.

       The filter accepts the following options:

       layout
	   Set the grid size (i.e. the number of lines and columns). For the
	   syntax of this option, check the "Video size" section in the
	   ffmpeg-utils manual.

       nb_frames
	   Set the maximum number of frames to render in the given area. It
	   must be less than or equal to wxh. The default value is 0, meaning
	   all the area will be used.

       margin
	   Set the outer border margin in pixels.

       padding
	   Set the inner border thickness (i.e. the number of pixels between
	   frames). For more advanced padding options (such as having
	   different values for the edges), refer to the pad video filter.

       color
	   Specify the color of the unused area. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual. The
	   default value of color is "black".

       Examples

       ·   Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a
	   movie:

		   ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

	   The -vsync 0 is necessary to prevent ffmpeg from duplicating each
	   output frame to accommodate the originally detected frame rate.

       ·   Display 5 pictures in an area of "3x2" frames, with 7 pixels
	   between them, and 2 pixels of initial margin, using mixed flat and
	   named options:

		   tile=3x2:nb_frames=5:padding=7:margin=2

   tinterlace
       Perform various types of temporal field interlacing.

       Frames are counted starting from 1, so the first input frame is
       considered odd.

       The filter accepts the following options:

       mode
	   Specify the mode of the interlacing. This option can also be
	   specified as a value alone. See below for a list of values for this
	   option.

	   Available values are:

	   merge, 0
	       Move odd frames into the upper field, even into the lower
	       field, generating a double height frame at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444

	   drop_even, 1
	       Only output odd frames, even frames are dropped, generating a
	       frame with unchanged height at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111			       33333
		       11111			       33333
		       11111			       33333
		       11111			       33333

	   drop_odd, 2
	       Only output even frames, odd frames are dropped, generating a
	       frame with unchanged height at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
				       22222			       44444
				       22222			       44444
				       22222			       44444
				       22222			       44444

	   pad, 3
	       Expand each frame to full height, but pad alternate lines with
	       black, generating a frame with double height at the same input
	       frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444

	   interleave_top, 4
	       Interleave the upper field from odd frames with the lower field
	       from even frames, generating a frame with unchanged height at
	       half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-

		       Output:
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444

	   interleave_bottom, 5
	       Interleave the lower field from odd frames with the upper field
	       from even frames, generating a frame with unchanged height at
	       half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444

		       Output:
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333

	   interlacex2, 6
	       Double frame rate with unchanged height. Frames are inserted
	       each containing the second temporal field from the previous
	       input frame and the first temporal field from the next input
	       frame. This mode relies on the top_field_first flag. Useful for
	       interlaced video displays with no field synchronisation.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
			11111		22222		33333		44444
		       11111	       22222	       33333	       44444
			11111		22222		33333		44444

		       Output:
		       11111   22222   22222   33333   33333   44444   44444
			11111	11111	22222	22222	33333	33333	44444
		       11111   22222   22222   33333   33333   44444   44444
			11111	11111	22222	22222	33333	33333	44444

	   mergex2, 7
	       Move odd frames into the upper field, even into the lower
	       field, generating a double height frame at same frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444

	   Numeric values are deprecated but are accepted for backward
	   compatibility reasons.

	   Default mode is "merge".

       flags
	   Specify flags influencing the filter process.

	   Available value for flags is:

	   low_pass_filter, vlfp
	       Enable linear vertical low-pass filtering in the filter.
	       Vertical low-pass filtering is required when creating an
	       interlaced destination from a progressive source which contains
	       high-frequency vertical detail. Filtering will reduce interlace
	       'twitter' and Moire patterning.

	   complex_filter, cvlfp
	       Enable complex vertical low-pass filtering.  This will slightly
	       less reduce interlace 'twitter' and Moire patterning but better
	       retain detail and subjective sharpness impression.

	   Vertical low-pass filtering can only be enabled for mode
	   interleave_top and interleave_bottom.

   tonemap
       Tone map colors from different dynamic ranges.

       This filter expects data in single precision floating point, as it
       needs to operate on (and can output) out-of-range values. Another
       filter, such as zscale, is needed to convert the resulting frame to a
       usable format.

       The tonemapping algorithms implemented only work on linear light, so
       input data should be linearized beforehand (and possibly correctly
       tagged).

	       ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT

       Options

       The filter accepts the following options.

       tonemap
	   Set the tone map algorithm to use.

	   Possible values are:

	   none
	       Do not apply any tone map, only desaturate overbright pixels.

	   clip
	       Hard-clip any out-of-range values. Use it for perfect color
	       accuracy for in-range values, while distorting out-of-range
	       values.

	   linear
	       Stretch the entire reference gamut to a linear multiple of the
	       display.

	   gamma
	       Fit a logarithmic transfer between the tone curves.

	   reinhard
	       Preserve overall image brightness with a simple curve, using
	       nonlinear contrast, which results in flattening details and
	       degrading color accuracy.

	   hable
	       Preserve both dark and bright details better than reinhard, at
	       the cost of slightly darkening everything. Use it when detail
	       preservation is more important than color and brightness
	       accuracy.

	   mobius
	       Smoothly map out-of-range values, while retaining contrast and
	       colors for in-range material as much as possible. Use it when
	       color accuracy is more important than detail preservation.

	   Default is none.

       param
	   Tune the tone mapping algorithm.

	   This affects the following algorithms:

	   none
	       Ignored.

	   linear
	       Specifies the scale factor to use while stretching.  Default to
	       1.0.

	   gamma
	       Specifies the exponent of the function.	Default to 1.8.

	   clip
	       Specify an extra linear coefficient to multiply into the signal
	       before clipping.  Default to 1.0.

	   reinhard
	       Specify the local contrast coefficient at the display peak.
	       Default to 0.5, which means that in-gamut values will be about
	       half as bright as when clipping.

	   hable
	       Ignored.

	   mobius
	       Specify the transition point from linear to mobius transform.
	       Every value below this point is guaranteed to be mapped 1:1.
	       The higher the value, the more accurate the result will be, at
	       the cost of losing bright details.  Default to 0.3, which due
	       to the steep initial slope still preserves in-range colors
	       fairly accurately.

       desat
	   Apply desaturation for highlights that exceed this level of
	   brightness. The higher the parameter, the more color information
	   will be preserved. This setting helps prevent unnaturally blown-out
	   colors for super-highlights, by (smoothly) turning into white
	   instead. This makes images feel more natural, at the cost of
	   reducing information about out-of-range colors.

	   The default of 2.0 is somewhat conservative and will mostly just
	   apply to skies or directly sunlit surfaces. A setting of 0.0
	   disables this option.

	   This option works only if the input frame has a supported color
	   tag.

       peak
	   Override signal/nominal/reference peak with this value. Useful when
	   the embedded peak information in display metadata is not reliable
	   or when tone mapping from a lower range to a higher range.

   transpose
       Transpose rows with columns in the input video and optionally flip it.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   0, 4, cclock_flip
	       Rotate by 90 degrees counterclockwise and vertically flip
	       (default), that is:

		       L.R     L.l
		       . . ->  . .
		       l.r     R.r

	   1, 5, clock
	       Rotate by 90 degrees clockwise, that is:

		       L.R     l.L
		       . . ->  . .
		       l.r     r.R

	   2, 6, cclock
	       Rotate by 90 degrees counterclockwise, that is:

		       L.R     R.r
		       . . ->  . .
		       l.r     L.l

	   3, 7, clock_flip
	       Rotate by 90 degrees clockwise and vertically flip, that is:

		       L.R     r.R
		       . . ->  . .
		       l.r     l.L

	   For values between 4-7, the transposition is only done if the input
	   video geometry is portrait and not landscape. These values are
	   deprecated, the "passthrough" option should be used instead.

	   Numerical values are deprecated, and should be dropped in favor of
	   symbolic constants.

       passthrough
	   Do not apply the transposition if the input geometry matches the
	   one specified by the specified value. It accepts the following
	   values:

	   none
	       Always apply transposition.

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

	   Default value is "none".

       For example to rotate by 90 degrees clockwise and preserve portrait
       layout:

	       transpose=dir=1:passthrough=portrait

       The command above can also be specified as:

	       transpose=1:portrait

   trim
       Trim the input so that the output contains one continuous subpart of
       the input.

       It accepts the following parameters:

       start
	   Specify the time of the start of the kept section, i.e. the frame
	   with the timestamp start will be the first frame in the output.

       end Specify the time of the first frame that will be dropped, i.e. the
	   frame immediately preceding the one with the timestamp end will be
	   the last frame in the output.

       start_pts
	   This is the same as start, except this option sets the start
	   timestamp in timebase units instead of seconds.

       end_pts
	   This is the same as end, except this option sets the end timestamp
	   in timebase units instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_frame
	   The number of the first frame that should be passed to the output.

       end_frame
	   The number of the first frame that should be dropped.

       start, end, and duration are expressed as time duration specifications;
       see the Time duration section in the ffmpeg-utils(1) manual for the
       accepted syntax.

       Note that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the _frame variants simply
       count the frames that pass through the filter. Also note that this
       filter does not modify the timestamps. If you wish for the output
       timestamps to start at zero, insert a setpts filter after the trim
       filter.

       If multiple start or end options are set, this filter tries to be
       greedy and keep all the frames that match at least one of the specified
       constraints. To keep only the part that matches all the constraints at
       once, chain multiple trim filters.

       The defaults are such that all the input is kept. So it is possible to
       set e.g.  just the end values to keep everything before the specified
       time.

       Examples:

       ·   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -vf trim=60:120

       ·   Keep only the first second:

		   ffmpeg -i INPUT -vf trim=duration=1

   unpremultiply
       Apply alpha unpremultiply effect to input video stream using first
       plane of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

	   If the format has 1 or 2 components, then luma is bit 0.  If the
	   format has 3 or 4 components: for RGB formats bit 0 is green, bit 1
	   is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
	   chroma-U and bit 2 is chroma-V.  If present, the alpha channel is
	   always the last bit.

       inplace
	   Do not require 2nd input for processing, instead use alpha plane
	   from input stream.

   unsharp
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
	   Set the luma matrix horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       luma_msize_y, ly
	   Set the luma matrix vertical size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       luma_amount, la
	   Set the luma effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 1.0.

       chroma_msize_x, cx
	   Set the chroma matrix horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       chroma_msize_y, cy
	   Set the chroma matrix vertical size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 0.0.

       opencl
	   If set to 1, specify using OpenCL capabilities, only available if
	   FFmpeg was configured with "--enable-opencl". Default value is 0.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       ·   Apply strong luma sharpen effect:

		   unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

       ·   Apply a strong blur of both luma and chroma parameters:

		   unsharp=7:7:-2:7:7:-2

   uspp
       Apply ultra slow/simple postprocessing filter that compresses and
       decompresses the image at several (or - in the case of quality level 8
       - all) shifts and average the results.

       The way this differs from the behavior of spp is that uspp actually
       encodes & decodes each case with libavcodec Snow, whereas spp uses a
       simplified intra only 8x8 DCT similar to MJPEG.

       The filter accepts the following options:

       quality
	   Set quality. This option defines the number of levels for
	   averaging. It accepts an integer in the range 0-8. If set to 0, the
	   filter will have no effect. A value of 8 means the higher quality.
	   For each increment of that value the speed drops by a factor of
	   approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If not set, the filter
	   will use the QP from the video stream (if available).

   vaguedenoiser
       Apply a wavelet based denoiser.

       It transforms each frame from the video input into the wavelet domain,
       using Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to
       the obtained coefficients. It does an inverse wavelet transform after.
       Due to wavelet properties, it should give a nice smoothed result, and
       reduced noise, without blurring picture features.

       This filter accepts the following options:

       threshold
	   The filtering strength. The higher, the more filtered the video
	   will be.  Hard thresholding can use a higher threshold than soft
	   thresholding before the video looks overfiltered. Default value is
	   2.

       method
	   The filtering method the filter will use.

	   It accepts the following values:

	   hard
	       All values under the threshold will be zeroed.

	   soft
	       All values under the threshold will be zeroed. All values above
	       will be reduced by the threshold.

	   garrote
	       Scales or nullifies coefficients - intermediary between (more)
	       soft and (less) hard thresholding.

	   Default is garrote.

       nsteps
	   Number of times, the wavelet will decompose the picture. Picture
	   can't be decomposed beyond a particular point (typically, 8 for a
	   640x480 frame - as 2^9 = 512 > 480). Valid values are integers
	   between 1 and 32. Default value is 6.

       percent
	   Partial of full denoising (limited coefficients shrinking), from 0
	   to 100. Default value is 85.

       planes
	   A list of the planes to process. By default all planes are
	   processed.

   vectorscope
       Display 2 color component values in the two dimensional graph (which is
       called a vectorscope).

       This filter accepts the following options:

       mode, m
	   Set vectorscope mode.

	   It accepts the following values:

	   gray
	       Gray values are displayed on graph, higher brightness means
	       more pixels have same component color value on location in
	       graph. This is the default mode.

	   color
	       Gray values are displayed on graph. Surrounding pixels values
	       which are not present in video frame are drawn in gradient of 2
	       color components which are set by option "x" and "y". The 3rd
	       color component is static.

	   color2
	       Actual color components values present in video frame are
	       displayed on graph.

	   color3
	       Similar as color2 but higher frequency of same values "x" and
	       "y" on graph increases value of another color component, which
	       is luminance by default values of "x" and "y".

	   color4
	       Actual colors present in video frame are displayed on graph. If
	       two different colors map to same position on graph then color
	       with higher value of component not present in graph is picked.

	   color5
	       Gray values are displayed on graph. Similar to "color" but with
	       3rd color component picked from radial gradient.

       x   Set which color component will be represented on X-axis. Default is
	   1.

       y   Set which color component will be represented on Y-axis. Default is
	   2.

       intensity, i
	   Set intensity, used by modes: gray, color, color3 and color5 for
	   increasing brightness of color component which represents frequency
	   of (X, Y) location in graph.

       envelope, e
	   none
	       No envelope, this is default.

	   instant
	       Instant envelope, even darkest single pixel will be clearly
	       highlighted.

	   peak
	       Hold maximum and minimum values presented in graph over time.
	       This way you can still spot out of range values without
	       constantly looking at vectorscope.

	   peak+instant
	       Peak and instant envelope combined together.

       graticule, g
	   Set what kind of graticule to draw.

	   none
	   green
	   color
       opacity, o
	   Set graticule opacity.

       flags, f
	   Set graticule flags.

	   white
	       Draw graticule for white point.

	   black
	       Draw graticule for black point.

	   name
	       Draw color points short names.

       bgopacity, b
	   Set background opacity.

       lthreshold, l
	   Set low threshold for color component not represented on X or Y
	   axis.  Values lower than this value will be ignored. Default is 0.
	   Note this value is multiplied with actual max possible value one
	   pixel component can have. So for 8-bit input and low threshold
	   value of 0.1 actual threshold is 0.1 * 255 = 25.

       hthreshold, h
	   Set high threshold for color component not represented on X or Y
	   axis.  Values higher than this value will be ignored. Default is 1.
	   Note this value is multiplied with actual max possible value one
	   pixel component can have. So for 8-bit input and high threshold
	   value of 0.9 actual threshold is 0.9 * 255 = 230.

       colorspace, c
	   Set what kind of colorspace to use when drawing graticule.

	   auto
	   601
	   709

	   Default is auto.

   vidstabdetect
       Analyze video stabilization/deshaking. Perform pass 1 of 2, see
       vidstabtransform for pass 2.

       This filter generates a file with relative translation and rotation
       transform information about subsequent frames, which is then used by
       the vidstabtransform filter.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libvidstab".

       This filter accepts the following options:

       result
	   Set the path to the file used to write the transforms information.
	   Default value is transforms.trf.

       shakiness
	   Set how shaky the video is and how quick the camera is. It accepts
	   an integer in the range 1-10, a value of 1 means little shakiness,
	   a value of 10 means strong shakiness. Default value is 5.

       accuracy
	   Set the accuracy of the detection process. It must be a value in
	   the range 1-15. A value of 1 means low accuracy, a value of 15
	   means high accuracy. Default value is 15.

       stepsize
	   Set stepsize of the search process. The region around minimum is
	   scanned with 1 pixel resolution. Default value is 6.

       mincontrast
	   Set minimum contrast. Below this value a local measurement field is
	   discarded. Must be a floating point value in the range 0-1. Default
	   value is 0.3.

       tripod
	   Set reference frame number for tripod mode.

	   If enabled, the motion of the frames is compared to a reference
	   frame in the filtered stream, identified by the specified number.
	   The idea is to compensate all movements in a more-or-less static
	   scene and keep the camera view absolutely still.

	   If set to 0, it is disabled. The frames are counted starting from
	   1.

       show
	   Show fields and transforms in the resulting frames. It accepts an
	   integer in the range 0-2. Default value is 0, which disables any
	   visualization.

       Examples

       ·   Use default values:

		   vidstabdetect

       ·   Analyze strongly shaky movie and put the results in file
	   mytransforms.trf:

		   vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

       ·   Visualize the result of internal transformations in the resulting
	   video:

		   vidstabdetect=show=1

       ·   Analyze a video with medium shakiness using ffmpeg:

		   ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

   vidstabtransform
       Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
       1.

       Read a file with transform information for each frame and
       apply/compensate them. Together with the vidstabdetect filter this can
       be used to deshake videos. See also
       <http://public.hronopik.de/vid.stab>. It is important to also use the
       unsharp filter, see below.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libvidstab".

       Options

       input
	   Set path to the file used to read the transforms. Default value is
	   transforms.trf.

       smoothing
	   Set the number of frames (value*2 + 1) used for lowpass filtering
	   the camera movements. Default value is 10.

	   For example a number of 10 means that 21 frames are used (10 in the
	   past and 10 in the future) to smoothen the motion in the video. A
	   larger value leads to a smoother video, but limits the acceleration
	   of the camera (pan/tilt movements). 0 is a special case where a
	   static camera is simulated.

       optalgo
	   Set the camera path optimization algorithm.

	   Accepted values are:

	   gauss
	       gaussian kernel low-pass filter on camera motion (default)

	   avg averaging on transformations

       maxshift
	   Set maximal number of pixels to translate frames. Default value is
	   -1, meaning no limit.

       maxangle
	   Set maximal angle in radians (degree*PI/180) to rotate frames.
	   Default value is -1, meaning no limit.

       crop
	   Specify how to deal with borders that may be visible due to
	   movement compensation.

	   Available values are:

	   keep
	       keep image information from previous frame (default)

	   black
	       fill the border black

       invert
	   Invert transforms if set to 1. Default value is 0.

       relative
	   Consider transforms as relative to previous frame if set to 1,
	   absolute if set to 0. Default value is 0.

       zoom
	   Set percentage to zoom. A positive value will result in a zoom-in
	   effect, a negative value in a zoom-out effect. Default value is 0
	   (no zoom).

       optzoom
	   Set optimal zooming to avoid borders.

	   Accepted values are:

	   0   disabled

	   1   optimal static zoom value is determined (only very strong
	       movements will lead to visible borders) (default)

	   2   optimal adaptive zoom value is determined (no borders will be
	       visible), see zoomspeed

	   Note that the value given at zoom is added to the one calculated
	   here.

       zoomspeed
	   Set percent to zoom maximally each frame (enabled when optzoom is
	   set to 2). Range is from 0 to 5, default value is 0.25.

       interpol
	   Specify type of interpolation.

	   Available values are:

	   no  no interpolation

	   linear
	       linear only horizontal

	   bilinear
	       linear in both directions (default)

	   bicubic
	       cubic in both directions (slow)

       tripod
	   Enable virtual tripod mode if set to 1, which is equivalent to
	   "relative=0:smoothing=0". Default value is 0.

	   Use also "tripod" option of vidstabdetect.

       debug
	   Increase log verbosity if set to 1. Also the detected global
	   motions are written to the temporary file global_motions.trf.
	   Default value is 0.

       Examples

       ·   Use ffmpeg for a typical stabilization with default values:

		   ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

	   Note the use of the unsharp filter which is always recommended.

       ·   Zoom in a bit more and load transform data from a given file:

		   vidstabtransform=zoom=5:input="mytransforms.trf"

       ·   Smoothen the video even more:

		   vidstabtransform=smoothing=30

   vflip
       Flip the input video vertically.

       For example, to vertically flip a video with ffmpeg:

	       ffmpeg -i in.avi -vf "vflip" out.avi

   vignette
       Make or reverse a natural vignetting effect.

       The filter accepts the following options:

       angle, a
	   Set lens angle expression as a number of radians.

	   The value is clipped in the "[0,PI/2]" range.

	   Default value: "PI/5"

       x0
       y0  Set center coordinates expressions. Respectively "w/2" and "h/2" by
	   default.

       mode
	   Set forward/backward mode.

	   Available modes are:

	   forward
	       The larger the distance from the central point, the darker the
	       image becomes.

	   backward
	       The larger the distance from the central point, the brighter
	       the image becomes.  This can be used to reverse a vignette
	       effect, though there is no automatic detection to extract the
	       lens angle and other settings (yet). It can also be used to
	       create a burning effect.

	   Default value is forward.

       eval
	   Set evaluation mode for the expressions (angle, x0, y0).

	   It accepts the following values:

	   init
	       Evaluate expressions only once during the filter
	       initialization.

	   frame
	       Evaluate expressions for each incoming frame. This is way
	       slower than the init mode since it requires all the scalers to
	       be re-computed, but it allows advanced dynamic expressions.

	   Default value is init.

       dither
	   Set dithering to reduce the circular banding effects. Default is 1
	   (enabled).

       aspect
	   Set vignette aspect. This setting allows one to adjust the shape of
	   the vignette.  Setting this value to the SAR of the input will make
	   a rectangular vignetting following the dimensions of the video.

	   Default is "1/1".

       Expressions

       The alpha, x0 and y0 expressions can contain the following parameters.

       w
       h   input width and height

       n   the number of input frame, starting from 0

       pts the PTS (Presentation TimeStamp) time of the filtered video frame,
	   expressed in TB units, NAN if undefined

       r   frame rate of the input video, NAN if the input frame rate is
	   unknown

       t   the PTS (Presentation TimeStamp) of the filtered video frame,
	   expressed in seconds, NAN if undefined

       tb  time base of the input video

       Examples

       ·   Apply simple strong vignetting effect:

		   vignette=PI/4

       ·   Make a flickering vignetting:

		   vignette='PI/4+random(1)*PI/50':eval=frame

   vmafmotion
       Obtain the average vmaf motion score of a video.  It is one of the
       component filters of VMAF.

       The obtained average motion score is printed through the logging
       system.

       In the below example the input file ref.mpg is being processed and
       score is computed.

	       ffmpeg -i ref.mpg -lavfi vmafmotion -f null -

   vstack
       Stack input videos vertically.

       All streams must be of same pixel format and of same width.

       Note that this filter is faster than using overlay and pad filter to
       create same output.

       The filter accept the following option:

       inputs
	   Set number of input streams. Default is 2.

       shortest
	   If set to 1, force the output to terminate when the shortest input
	   terminates. Default value is 0.

   w3fdif
       Deinterlace the input video ("w3fdif" stands for "Weston 3 Field
       Deinterlacing Filter").

       Based on the process described by Martin Weston for BBC R&D, and
       implemented based on the de-interlace algorithm written by Jim
       Easterbrook for BBC R&D, the Weston 3 field deinterlacing filter uses
       filter coefficients calculated by BBC R&D.

       There are two sets of filter coefficients, so called "simple": and
       "complex". Which set of filter coefficients is used can be set by
       passing an optional parameter:

       filter
	   Set the interlacing filter coefficients. Accepts one of the
	   following values:

	   simple
	       Simple filter coefficient set.

	   complex
	       More-complex filter coefficient set.

	   Default value is complex.

       deint
	   Specify which frames to deinterlace. Accept one of the following
	   values:

	   all Deinterlace all frames,

	   interlaced
	       Only deinterlace frames marked as interlaced.

	   Default value is all.

   waveform
       Video waveform monitor.

       The waveform monitor plots color component intensity. By default
       luminance only. Each column of the waveform corresponds to a column of
       pixels in the source video.

       It accepts the following options:

       mode, m
	   Can be either "row", or "column". Default is "column".  In row
	   mode, the graph on the left side represents color component value 0
	   and the right side represents value = 255. In column mode, the top
	   side represents color component value = 0 and bottom side
	   represents value = 255.

       intensity, i
	   Set intensity. Smaller values are useful to find out how many
	   values of the same luminance are distributed across input
	   rows/columns.  Default value is 0.04. Allowed range is [0, 1].

       mirror, r
	   Set mirroring mode. 0 means unmirrored, 1 means mirrored.  In
	   mirrored mode, higher values will be represented on the left side
	   for "row" mode and at the top for "column" mode. Default is 1
	   (mirrored).

       display, d
	   Set display mode.  It accepts the following values:

	   overlay
	       Presents information identical to that in the "parade", except
	       that the graphs representing color components are superimposed
	       directly over one another.

	       This display mode makes it easier to spot relative differences
	       or similarities in overlapping areas of the color components
	       that are supposed to be identical, such as neutral whites,
	       grays, or blacks.

	   stack
	       Display separate graph for the color components side by side in
	       "row" mode or one below the other in "column" mode.

	   parade
	       Display separate graph for the color components side by side in
	       "column" mode or one below the other in "row" mode.

	       Using this display mode makes it easy to spot color casts in
	       the highlights and shadows of an image, by comparing the
	       contours of the top and the bottom graphs of each waveform.
	       Since whites, grays, and blacks are characterized by exactly
	       equal amounts of red, green, and blue, neutral areas of the
	       picture should display three waveforms of roughly equal
	       width/height. If not, the correction is easy to perform by
	       making level adjustments the three waveforms.

	   Default is "stack".

       components, c
	   Set which color components to display. Default is 1, which means
	   only luminance or red color component if input is in RGB
	   colorspace. If is set for example to 7 it will display all 3 (if)
	   available color components.

       envelope, e
	   none
	       No envelope, this is default.

	   instant
	       Instant envelope, minimum and maximum values presented in graph
	       will be easily visible even with small "step" value.

	   peak
	       Hold minimum and maximum values presented in graph across time.
	       This way you can still spot out of range values without
	       constantly looking at waveforms.

	   peak+instant
	       Peak and instant envelope combined together.

       filter, f
	   lowpass
	       No filtering, this is default.

	   flat
	       Luma and chroma combined together.

	   aflat
	       Similar as above, but shows difference between blue and red
	       chroma.

	   chroma
	       Displays only chroma.

	   color
	       Displays actual color value on waveform.

	   acolor
	       Similar as above, but with luma showing frequency of chroma
	       values.

       graticule, g
	   Set which graticule to display.

	   none
	       Do not display graticule.

	   green
	       Display green graticule showing legal broadcast ranges.

       opacity, o
	   Set graticule opacity.

       flags, fl
	   Set graticule flags.

	   numbers
	       Draw numbers above lines. By default enabled.

	   dots
	       Draw dots instead of lines.

       scale, s
	   Set scale used for displaying graticule.

	   digital
	   millivolts
	   ire

	   Default is digital.

       bgopacity, b
	   Set background opacity.

   weave, doubleweave
       The "weave" takes a field-based video input and join each two
       sequential fields into single frame, producing a new double height clip
       with half the frame rate and half the frame count.

       The "doubleweave" works same as "weave" but without halving frame rate
       and frame count.

       It accepts the following option:

       first_field
	   Set first field. Available values are:

	   top, t
	       Set the frame as top-field-first.

	   bottom, b
	       Set the frame as bottom-field-first.

       Examples

       ·   Interlace video using select and separatefields filter:

		   separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave

   xbr
       Apply the xBR high-quality magnification filter which is designed for
       pixel art. It follows a set of edge-detection rules, see
       <http://www.libretro.com/forums/viewtopic.php?f=6&t=134>.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "2xBR", 3 for "3xBR" and 4 for
	   "4xBR".  Default is 3.

   yadif
       Deinterlace the input video ("yadif" means "yet another deinterlacing
       filter").

       It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It accepts one of the following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   2, send_frame_nospatial
	       Like "send_frame", but it skips the spatial interlacing check.

	   3, send_field_nospatial
	       Like "send_field", but it skips the spatial interlacing check.

	   The default value is "send_frame".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first will be
	   assumed.

       deint
	   Specify which frames to deinterlace. Accept one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   zoompan
       Apply Zoom & Pan effect.

       This filter accepts the following options:

       zoom, z
	   Set the zoom expression. Default is 1.

       x
       y   Set the x and y expression. Default is 0.

       d   Set the duration expression in number of frames.  This sets for how
	   many number of frames effect will last for single input image.

       s   Set the output image size, default is 'hd720'.

       fps Set the output frame rate, default is '25'.

       Each expression can contain the following constants:

       in_w, iw
	   Input width.

       in_h, ih
	   Input height.

       out_w, ow
	   Output width.

       out_h, oh
	   Output height.

       in  Input frame count.

       on  Output frame count.

       x
       y   Last calculated 'x' and 'y' position from 'x' and 'y' expression
	   for current input frame.

       px
       py  'x' and 'y' of last output frame of previous input frame or 0 when
	   there was not yet such frame (first input frame).

       zoom
	   Last calculated zoom from 'z' expression for current input frame.

       pzoom
	   Last calculated zoom of last output frame of previous input frame.

       duration
	   Number of output frames for current input frame. Calculated from
	   'd' expression for each input frame.

       pduration
	   number of output frames created for previous input frame

       a   Rational number: input width / input height

       sar sample aspect ratio

       dar display aspect ratio

       Examples

       ·   Zoom-in up to 1.5 and pan at same time to some spot near center of
	   picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

       ·   Zoom-in up to 1.5 and pan always at center of picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       ·   Same as above but without pausing:

		   zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

   zscale
       Scale (resize) the input video, using the z.lib library:
       https://github.com/sekrit-twc/zimg.

       The zscale filter forces the output display aspect ratio to be the same
       as the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the zscale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options.

       width, w
       height, h
	   Set the output video dimension expression. Default value is the
	   input dimension.

	   If the width or w value is 0, the input width is used for the
	   output. If the height or h value is 0, the input height is used for
	   the output.

	   If one and only one of the values is -n with n >= 1, the zscale
	   filter will use a value that maintains the aspect ratio of the
	   input image, calculated from the other specified dimension. After
	   that it will, however, make sure that the calculated dimension is
	   divisible by n and adjust the value if necessary.

	   If both values are -n with n >= 1, the behavior will be identical
	   to both values being set to 0 as previously detailed.

	   See below for the list of accepted constants for use in the
	   dimension expression.

       size, s
	   Set the video size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual.

       dither, d
	   Set the dither type.

	   Possible values are:

	   none
	   ordered
	   random
	   error_diffusion

	   Default is none.

       filter, f
	   Set the resize filter type.

	   Possible values are:

	   point
	   bilinear
	   bicubic
	   spline16
	   spline36
	   lanczos

	   Default is bilinear.

       range, r
	   Set the color range.

	   Possible values are:

	   input
	   limited
	   full

	   Default is same as input.

       primaries, p
	   Set the color primaries.

	   Possible values are:

	   input
	   709
	   unspecified
	   170m
	   240m
	   2020

	   Default is same as input.

       transfer, t
	   Set the transfer characteristics.

	   Possible values are:

	   input
	   709
	   unspecified
	   601
	   linear
	   2020_10
	   2020_12
	   smpte2084
	   iec61966-2-1
	   arib-std-b67

	   Default is same as input.

       matrix, m
	   Set the colorspace matrix.

	   Possible value are:

	   input
	   709
	   unspecified
	   470bg
	   170m
	   2020_ncl
	   2020_cl

	   Default is same as input.

       rangein, rin
	   Set the input color range.

	   Possible values are:

	   input
	   limited
	   full

	   Default is same as input.

       primariesin, pin
	   Set the input color primaries.

	   Possible values are:

	   input
	   709
	   unspecified
	   170m
	   240m
	   2020

	   Default is same as input.

       transferin, tin
	   Set the input transfer characteristics.

	   Possible values are:

	   input
	   709
	   unspecified
	   601
	   linear
	   2020_10
	   2020_12

	   Default is same as input.

       matrixin, min
	   Set the input colorspace matrix.

	   Possible value are:

	   input
	   709
	   unspecified
	   470bg
	   170m
	   2020_ncl
	   2020_cl
       chromal, c
	   Set the output chroma location.

	   Possible values are:

	   input
	   left
	   center
	   topleft
	   top
	   bottomleft
	   bottom
       chromalin, cin
	   Set the input chroma location.

	   Possible values are:

	   input
	   left
	   center
	   topleft
	   top
	   bottomleft
	   bottom
       npl Set the nominal peak luminance.

       The values of the w and h options are expressions containing the
       following constants:

       in_w
       in_h
	   The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
	   horizontal and vertical input chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
	   horizontal and vertical output chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

VIDEO SOURCES
       Below is a description of the currently available video sources.

   buffer
       Buffer video frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in libavfilter/vsrc_buffer.h.

       It accepts the following parameters:

       video_size
	   Specify the size (width and height) of the buffered video frames.
	   For the syntax of this option, check the "Video size" section in
	   the ffmpeg-utils manual.

       width
	   The input video width.

       height
	   The input video height.

       pix_fmt
	   A string representing the pixel format of the buffered video
	   frames.  It may be a number corresponding to a pixel format, or a
	   pixel format name.

       time_base
	   Specify the timebase assumed by the timestamps of the buffered
	   frames.

       frame_rate
	   Specify the frame rate expected for the video stream.

       pixel_aspect, sar
	   The sample (pixel) aspect ratio of the input video.

       sws_param
	   Specify the optional parameters to be used for the scale filter
	   which is automatically inserted when an input change is detected in
	   the input size or format.

       hw_frames_ctx
	   When using a hardware pixel format, this should be a reference to
	   an AVHWFramesContext describing input frames.

       For example:

	       buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

       will instruct the source to accept video frames with size 320x240 and
       with format "yuv410p", assuming 1/24 as the timestamps timebase and
       square pixels (1:1 sample aspect ratio).  Since the pixel format with
       name "yuv410p" corresponds to the number 6 (check the enum
       AVPixelFormat definition in libavutil/pixfmt.h), this example
       corresponds to:

	       buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

       Alternatively, the options can be specified as a flat string, but this
       syntax is deprecated:

       width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]

   cellauto
       Create a pattern generated by an elementary cellular automaton.

       The initial state of the cellular automaton can be defined through the
       filename and pattern options. If such options are not specified an
       initial state is created randomly.

       At each new frame a new row in the video is filled with the result of
       the cellular automaton next generation. The behavior when the whole
       frame is filled is defined by the scroll option.

       This source accepts the following options:

       filename, f
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified file.  In the file, each non-whitespace
	   character is considered an alive cell, a newline will terminate the
	   row, and further characters in the file will be ignored.

       pattern, p
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified string.

	   Each non-whitespace character in the string is considered an alive
	   cell, a newline will terminate the row, and further characters in
	   the string will be ignored.

       rate, r
	   Set the video rate, that is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial cellular automaton row.
	   It is a floating point number value ranging from 0 to 1, defaults
	   to 1/PHI.

	   This option is ignored when a file or a pattern is specified.

       random_seed, seed
	   Set the seed for filling randomly the initial row, must be an
	   integer included between 0 and UINT32_MAX. If not specified, or if
	   explicitly set to -1, the filter will try to use a good random seed
	   on a best effort basis.

       rule
	   Set the cellular automaton rule, it is a number ranging from 0 to
	   255.  Default value is 110.

       size, s
	   Set the size of the output video. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If filename or pattern is specified, the size is set by default to
	   the width of the specified initial state row, and the height is set
	   to width * PHI.

	   If size is set, it must contain the width of the specified pattern
	   string, and the specified pattern will be centered in the larger
	   row.

	   If a filename or a pattern string is not specified, the size value
	   defaults to "320x518" (used for a randomly generated initial
	   state).

       scroll
	   If set to 1, scroll the output upward when all the rows in the
	   output have been already filled. If set to 0, the new generated row
	   will be written over the top row just after the bottom row is
	   filled.  Defaults to 1.

       start_full, full
	   If set to 1, completely fill the output with generated rows before
	   outputting the first frame.	This is the default behavior, for
	   disabling set the value to 0.

       stitch
	   If set to 1, stitch the left and right row edges together.  This is
	   the default behavior, for disabling set the value to 0.

       Examples

       ·   Read the initial state from pattern, and specify an output of size
	   200x400.

		   cellauto=f=pattern:s=200x400

       ·   Generate a random initial row with a width of 200 cells, with a
	   fill ratio of 2/3:

		   cellauto=ratio=2/3:s=200x200

       ·   Create a pattern generated by rule 18 starting by a single alive
	   cell centered on an initial row with width 100:

		   cellauto=p=@s=100x400:full=0:rule=18

       ·   Specify a more elaborated initial pattern:

		   cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

   coreimagesrc
       Video source generated on GPU using Apple's CoreImage API on OSX.

       This video source is a specialized version of the coreimage video
       filter.	Use a core image generator at the beginning of the applied
       filterchain to generate the content.

       The coreimagesrc video source accepts the following options:

       list_generators
	   List all available generators along with all their respective
	   options as well as possible minimum and maximum values along with
	   the default values.

		   list_generators=true

       size, s
	   Specify the size of the sourced video. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   The default value is "320x240".

       rate, r
	   Specify the frame rate of the sourced video, as the number of
	   frames generated per second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default value
	   is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
	   Set the duration of the sourced video. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       Additionally, all options of the coreimage video filter are accepted.
       A complete filterchain can be used for further processing of the
       generated input without CPU-HOST transfer. See coreimage documentation
       and examples for details.

       Examples

       ·   Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage,
	   given as complete and escaped command-line for Apple's standard
	   bash shell:

		   ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

	   This example is equivalent to the QRCode example of coreimage
	   without the need for a nullsrc video source.

   mandelbrot
       Generate a Mandelbrot set fractal, and progressively zoom towards the
       point specified with start_x and start_y.

       This source accepts the following options:

       end_pts
	   Set the terminal pts value. Default value is 400.

       end_scale
	   Set the terminal scale value.  Must be a floating point value.
	   Default value is 0.3.

       inner
	   Set the inner coloring mode, that is the algorithm used to draw the
	   Mandelbrot fractal internal region.

	   It shall assume one of the following values:

	   black
	       Set black mode.

	   convergence
	       Show time until convergence.

	   mincol
	       Set color based on point closest to the origin of the
	       iterations.

	   period
	       Set period mode.

	   Default value is mincol.

       bailout
	   Set the bailout value. Default value is 10.0.

       maxiter
	   Set the maximum of iterations performed by the rendering algorithm.
	   Default value is 7189.

       outer
	   Set outer coloring mode.  It shall assume one of following values:

	   iteration_count
	       Set iteration cound mode.

	   normalized_iteration_count
	       set normalized iteration count mode.

	   Default value is normalized_iteration_count.

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       size, s
	   Set frame size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual. Default value is
	   "640x480".

       start_scale
	   Set the initial scale value. Default value is 3.0.

       start_x
	   Set the initial x position. Must be a floating point value between
	   -100 and 100. Default value is
	   -0.743643887037158704752191506114774.

       start_y
	   Set the initial y position. Must be a floating point value between
	   -100 and 100. Default value is
	   -0.131825904205311970493132056385139.

   mptestsrc
       Generate various test patterns, as generated by the MPlayer test
       filter.

       The size of the generated video is fixed, and is 256x256.  This source
       is useful in particular for testing encoding features.

       This source accepts the following options:

       rate, r
	   Specify the frame rate of the sourced video, as the number of
	   frames generated per second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default value
	   is "25".

       duration, d
	   Set the duration of the sourced video. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       test, t
	   Set the number or the name of the test to perform. Supported tests
	   are:

	   dc_luma
	   dc_chroma
	   freq_luma
	   freq_chroma
	   amp_luma
	   amp_chroma
	   cbp
	   mv
	   ring1
	   ring2
	   all

	   Default value is "all", which will cycle through the list of all
	   tests.

       Some examples:

	       mptestsrc=t=dc_luma

       will generate a "dc_luma" test pattern.

   frei0r_src
       Provide a frei0r source.

       To enable compilation of this filter you need to install the frei0r
       header and configure FFmpeg with "--enable-frei0r".

       This source accepts the following parameters:

       size
	   The size of the video to generate. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       framerate
	   The framerate of the generated video. It may be a string of the
	   form num/den or a frame rate abbreviation.

       filter_name
	   The name to the frei0r source to load. For more information
	   regarding frei0r and how to set the parameters, read the frei0r
	   section in the video filters documentation.

       filter_params
	   A '|'-separated list of parameters to pass to the frei0r source.

       For example, to generate a frei0r partik0l source with size 200x200 and
       frame rate 10 which is overlaid on the overlay filter main input:

	       frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

   life
       Generate a life pattern.

       This source is based on a generalization of John Conway's life game.

       The sourced input represents a life grid, each pixel represents a cell
       which can be in one of two possible states, alive or dead. Every cell
       interacts with its eight neighbours, which are the cells that are
       horizontally, vertically, or diagonally adjacent.

       At each interaction the grid evolves according to the adopted rule,
       which specifies the number of neighbor alive cells which will make a
       cell stay alive or born. The rule option allows one to specify the rule
       to adopt.

       This source accepts the following options:

       filename, f
	   Set the file from which to read the initial grid state. In the
	   file, each non-whitespace character is considered an alive cell,
	   and newline is used to delimit the end of each row.

	   If this option is not specified, the initial grid is generated
	   randomly.

       rate, r
	   Set the video rate, that is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial random grid. It is a
	   floating point number value ranging from 0 to 1, defaults to 1/PHI.
	   It is ignored when a file is specified.

       random_seed, seed
	   Set the seed for filling the initial random grid, must be an
	   integer included between 0 and UINT32_MAX. If not specified, or if
	   explicitly set to -1, the filter will try to use a good random seed
	   on a best effort basis.

       rule
	   Set the life rule.

	   A rule can be specified with a code of the kind "SNS/BNB", where NS
	   and NB are sequences of numbers in the range 0-8, NS specifies the
	   number of alive neighbor cells which make a live cell stay alive,
	   and NB the number of alive neighbor cells which make a dead cell to
	   become alive (i.e. to "born").  "s" and "b" can be used in place of
	   "S" and "B", respectively.

	   Alternatively a rule can be specified by an 18-bits integer. The 9
	   high order bits are used to encode the next cell state if it is
	   alive for each number of neighbor alive cells, the low order bits
	   specify the rule for "borning" new cells. Higher order bits encode
	   for an higher number of neighbor cells.  For example the number
	   6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born
	   rule of 9, which corresponds to "S23/B03".

	   Default value is "S23/B3", which is the original Conway's game of
	   life rule, and will keep a cell alive if it has 2 or 3 neighbor
	   alive cells, and will born a new cell if there are three alive
	   cells around a dead cell.

       size, s
	   Set the size of the output video. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If filename is specified, the size is set by default to the same
	   size of the input file. If size is set, it must contain the size
	   specified in the input file, and the initial grid defined in that
	   file is centered in the larger resulting area.

	   If a filename is not specified, the size value defaults to
	   "320x240" (used for a randomly generated initial grid).

       stitch
	   If set to 1, stitch the left and right grid edges together, and the
	   top and bottom edges also. Defaults to 1.

       mold
	   Set cell mold speed. If set, a dead cell will go from death_color
	   to mold_color with a step of mold. mold can have a value from 0 to
	   255.

       life_color
	   Set the color of living (or new born) cells.

       death_color
	   Set the color of dead cells. If mold is set, this is the first
	   color used to represent a dead cell.

       mold_color
	   Set mold color, for definitely dead and moldy cells.

	   For the syntax of these 3 color options, check the "Color" section
	   in the ffmpeg-utils manual.

       Examples

       ·   Read a grid from pattern, and center it on a grid of size 300x300
	   pixels:

		   life=f=pattern:s=300x300

       ·   Generate a random grid of size 200x200, with a fill ratio of 2/3:

		   life=ratio=2/3:s=200x200

       ·   Specify a custom rule for evolving a randomly generated grid:

		   life=rule=S14/B34

       ·   Full example with slow death effect (mold) using ffplay:

		   ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

   allrgb, allyuv, color, haldclutsrc, nullsrc, rgbtestsrc, smptebars,
       smptehdbars, testsrc, testsrc2, yuvtestsrc
       The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

       The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

       The "color" source provides an uniformly colored input.

       The "haldclutsrc" source provides an identity Hald CLUT. See also
       haldclut filter.

       The "nullsrc" source returns unprocessed video frames. It is mainly
       useful to be employed in analysis / debugging tools, or as the source
       for filters which ignore the input data.

       The "rgbtestsrc" source generates an RGB test pattern useful for
       detecting RGB vs BGR issues. You should see a red, green and blue
       stripe from top to bottom.

       The "smptebars" source generates a color bars pattern, based on the
       SMPTE Engineering Guideline EG 1-1990.

       The "smptehdbars" source generates a color bars pattern, based on the
       SMPTE RP 219-2002.

       The "testsrc" source generates a test video pattern, showing a color
       pattern, a scrolling gradient and a timestamp. This is mainly intended
       for testing purposes.

       The "testsrc2" source is similar to testsrc, but supports more pixel
       formats instead of just "rgb24". This allows using it as an input for
       other tests without requiring a format conversion.

       The "yuvtestsrc" source generates an YUV test pattern. You should see a
       y, cb and cr stripe from top to bottom.

       The sources accept the following parameters:

       alpha
	   Specify the alpha (opacity) of the background, only available in
	   the "testsrc2" source. The value must be between 0 (fully
	   transparent) and 255 (fully opaque, the default).

       color, c
	   Specify the color of the source, only available in the "color"
	   source. For the syntax of this option, check the "Color" section in
	   the ffmpeg-utils manual.

       level
	   Specify the level of the Hald CLUT, only available in the
	   "haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
	   by "N*N*N" pixels to be used as identity matrix for 3D lookup
	   tables. Each component is coded on a "1/(N*N)" scale.

       size, s
	   Specify the size of the sourced video. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   The default value is "320x240".

	   This option is not available with the "haldclutsrc" filter.

       rate, r
	   Specify the frame rate of the sourced video, as the number of
	   frames generated per second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default value
	   is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
	   Set the duration of the sourced video. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       decimals, n
	   Set the number of decimals to show in the timestamp, only available
	   in the "testsrc" source.

	   The displayed timestamp value will correspond to the original
	   timestamp value multiplied by the power of 10 of the specified
	   value. Default value is 0.

       For example the following:

	       testsrc=duration=5.3:size=qcif:rate=10

       will generate a video with a duration of 5.3 seconds, with size 176x144
       and a frame rate of 10 frames per second.

       The following graph description will generate a red source with an
       opacity of 0.2, with size "qcif" and a frame rate of 10 frames per
       second.

	       color=c=red@0.2:s=qcif:r=10

       If the input content is to be ignored, "nullsrc" can be used. The
       following command generates noise in the luminance plane by employing
       the "geq" filter:

	       nullsrc=s=256x256, geq=random(1)*255:128:128

       Commands

       The "color" source supports the following commands:

       c, color
	   Set the color of the created image. Accepts the same syntax of the
	   corresponding color option.

VIDEO SINKS
       Below is a description of the currently available video sinks.

   buffersink
       Buffer video frames, and make them available to the end of the filter
       graph.

       This sink is mainly intended for programmatic use, in particular
       through the interface defined in libavfilter/buffersink.h or the
       options system.

       It accepts a pointer to an AVBufferSinkContext structure, which defines
       the incoming buffers' formats, to be passed as the opaque parameter to
       "avfilter_init_filter" for initialization.

   nullsink
       Null video sink: do absolutely nothing with the input video. It is
       mainly useful as a template and for use in analysis / debugging tools.

MULTIMEDIA FILTERS
       Below is a description of the currently available multimedia filters.

   abitscope
       Convert input audio to a video output, displaying the audio bit scope.

       The filter accepts the following options:

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "1024x256".

       colors
	   Specify list of colors separated by space or by '|' which will be
	   used to draw channels. Unrecognized or missing colors will be
	   replaced by white color.

   ahistogram
       Convert input audio to a video output, displaying the volume histogram.

       The filter accepts the following options:

       dmode
	   Specify how histogram is calculated.

	   It accepts the following values:

	   single
	       Use single histogram for all channels.

	   separate
	       Use separate histogram for each channel.

	   Default is "single".

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "hd720".

       scale
	   Set display scale.

	   It accepts the following values:

	   log logarithmic

	   sqrt
	       square root

	   cbrt
	       cubic root

	   lin linear

	   rlog
	       reverse logarithmic

	   Default is "log".

       ascale
	   Set amplitude scale.

	   It accepts the following values:

	   log logarithmic

	   lin linear

	   Default is "log".

       acount
	   Set how much frames to accumulate in histogram.  Defauls is 1.
	   Setting this to -1 accumulates all frames.

       rheight
	   Set histogram ratio of window height.

       slide
	   Set sonogram sliding.

	   It accepts the following values:

	   replace
	       replace old rows with new ones.

	   scroll
	       scroll from top to bottom.

	   Default is "replace".

   aphasemeter
       Convert input audio to a video output, displaying the audio phase.

       The filter accepts the following options:

       rate, r
	   Set the output frame rate. Default value is 25.

       size, s
	   Set the video size for the output. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.  Default
	   value is "800x400".

       rc
       gc
       bc  Specify the red, green, blue contrast. Default values are 2, 7 and
	   1.  Allowed range is "[0, 255]".

       mpc Set color which will be used for drawing median phase. If color is
	   "none" which is default, no median phase value will be drawn.

       video
	   Enable video output. Default is enabled.

       The filter also exports the frame metadata "lavfi.aphasemeter.phase"
       which represents mean phase of current audio frame. Value is in range
       "[-1, 1]".  The "-1" means left and right channels are completely out
       of phase and 1 means channels are in phase.

   avectorscope
       Convert input audio to a video output, representing the audio vector
       scope.

       The filter is used to measure the difference between channels of stereo
       audio stream. A monoaural signal, consisting of identical left and
       right signal, results in straight vertical line. Any stereo separation
       is visible as a deviation from this line, creating a Lissajous figure.
       If the straight (or deviation from it) but horizontal line appears this
       indicates that the left and right channels are out of phase.

       The filter accepts the following options:

       mode, m
	   Set the vectorscope mode.

	   Available values are:

	   lissajous
	       Lissajous rotated by 45 degrees.

	   lissajous_xy
	       Same as above but not rotated.

	   polar
	       Shape resembling half of circle.

	   Default value is lissajous.

       size, s
	   Set the video size for the output. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.  Default
	   value is "400x400".

       rate, r
	   Set the output frame rate. Default value is 25.

       rc
       gc
       bc
       ac  Specify the red, green, blue and alpha contrast. Default values are
	   40, 160, 80 and 255.  Allowed range is "[0, 255]".

       rf
       gf
       bf
       af  Specify the red, green, blue and alpha fade. Default values are 15,
	   10, 5 and 5.  Allowed range is "[0, 255]".

       zoom
	   Set the zoom factor. Default value is 1. Allowed range is "[0,
	   10]".  Values lower than 1 will auto adjust zoom factor to maximal
	   possible value.

       draw
	   Set the vectorscope drawing mode.

	   Available values are:

	   dot Draw dot for each sample.

	   line
	       Draw line between previous and current sample.

	   Default value is dot.

       scale
	   Specify amplitude scale of audio samples.

	   Available values are:

	   lin Linear.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   log Logarithmic.

       Examples

       ·   Complete example using ffplay:

		   ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
				[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'

   bench, abench
       Benchmark part of a filtergraph.

       The filter accepts the following options:

       action
	   Start or stop a timer.

	   Available values are:

	   start
	       Get the current time, set it as frame metadata (using the key
	       "lavfi.bench.start_time"), and forward the frame to the next
	       filter.

	   stop
	       Get the current time and fetch the "lavfi.bench.start_time"
	       metadata from the input frame metadata to get the time
	       difference. Time difference, average, maximum and minimum time
	       (respectively "t", "avg", "max" and "min") are then printed.
	       The timestamps are expressed in seconds.

       Examples

       ·   Benchmark selectivecolor filter:

		   bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop

   concat
       Concatenate audio and video streams, joining them together one after
       the other.

       The filter works on segments of synchronized video and audio streams.
       All segments must have the same number of streams of each type, and
       that will also be the number of streams at output.

       The filter accepts the following options:

       n   Set the number of segments. Default is 2.

       v   Set the number of output video streams, that is also the number of
	   video streams in each segment. Default is 1.

       a   Set the number of output audio streams, that is also the number of
	   audio streams in each segment. Default is 0.

       unsafe
	   Activate unsafe mode: do not fail if segments have a different
	   format.

       The filter has v+a outputs: first v video outputs, then a audio
       outputs.

       There are nx(v+a) inputs: first the inputs for the first segment, in
       the same order as the outputs, then the inputs for the second segment,
       etc.

       Related streams do not always have exactly the same duration, for
       various reasons including codec frame size or sloppy authoring. For
       that reason, related synchronized streams (e.g. a video and its audio
       track) should be concatenated at once. The concat filter will use the
       duration of the longest stream in each segment (except the last one),
       and if necessary pad shorter audio streams with silence.

       For this filter to work correctly, all segments must start at timestamp
       0.

       All corresponding streams must have the same parameters in all
       segments; the filtering system will automatically select a common pixel
       format for video streams, and a common sample format, sample rate and
       channel layout for audio streams, but other settings, such as
       resolution, must be converted explicitly by the user.

       Different frame rates are acceptable but will result in variable frame
       rate at output; be sure to configure the output file to handle it.

       Examples

       ·   Concatenate an opening, an episode and an ending, all in bilingual
	   version (video in stream 0, audio in streams 1 and 2):

		   ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
		     '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
		      concat=n=3:v=1:a=2 [v] [a1] [a2]' \
		     -map '[v]' -map '[a1]' -map '[a2]' output.mkv

       ·   Concatenate two parts, handling audio and video separately, using
	   the (a)movie sources, and adjusting the resolution:

		   movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
		   movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
		   [v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]

	   Note that a desync will happen at the stitch if the audio and video
	   streams do not have exactly the same duration in the first file.

   drawgraph, adrawgraph
       Draw a graph using input video or audio metadata.

       It accepts the following parameters:

       m1  Set 1st frame metadata key from which metadata values will be used
	   to draw a graph.

       fg1 Set 1st foreground color expression.

       m2  Set 2nd frame metadata key from which metadata values will be used
	   to draw a graph.

       fg2 Set 2nd foreground color expression.

       m3  Set 3rd frame metadata key from which metadata values will be used
	   to draw a graph.

       fg3 Set 3rd foreground color expression.

       m4  Set 4th frame metadata key from which metadata values will be used
	   to draw a graph.

       fg4 Set 4th foreground color expression.

       min Set minimal value of metadata value.

       max Set maximal value of metadata value.

       bg  Set graph background color. Default is white.

       mode
	   Set graph mode.

	   Available values for mode is:

	   bar
	   dot
	   line

	   Default is "line".

       slide
	   Set slide mode.

	   Available values for slide is:

	   frame
	       Draw new frame when right border is reached.

	   replace
	       Replace old columns with new ones.

	   scroll
	       Scroll from right to left.

	   rscroll
	       Scroll from left to right.

	   picture
	       Draw single picture.

	   Default is "frame".

       size
	   Set size of graph video. For the syntax of this option, check the
	   "Video size" section in the ffmpeg-utils manual.  The default value
	   is "900x256".

	   The foreground color expressions can use the following variables:

	   MIN Minimal value of metadata value.

	   MAX Maximal value of metadata value.

	   VAL Current metadata key value.

	   The color is defined as 0xAABBGGRR.

       Example using metadata from signalstats filter:

	       signalstats,drawgraph=lavfi.signalstats.YAVG:min=0:max=255

       Example using metadata from ebur128 filter:

	       ebur128=metadata=1,adrawgraph=lavfi.r128.M:min=-120:max=5

   ebur128
       EBU R128 scanner filter. This filter takes an audio stream as input and
       outputs it unchanged. By default, it logs a message at a frequency of
       10Hz with the Momentary loudness (identified by "M"), Short-term
       loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

       The filter also has a video output (see the video option) with a real
       time graph to observe the loudness evolution. The graphic contains the
       logged message mentioned above, so it is not printed anymore when this
       option is set, unless the verbose logging is set. The main graphing
       area contains the short-term loudness (3 seconds of analysis), and the
       gauge on the right is for the momentary loudness (400 milliseconds).

       More information about the Loudness Recommendation EBU R128 on
       <http://tech.ebu.ch/loudness>.

       The filter accepts the following options:

       video
	   Activate the video output. The audio stream is passed unchanged
	   whether this option is set or no. The video stream will be the
	   first output stream if activated. Default is 0.

       size
	   Set the video size. This option is for video only. For the syntax
	   of this option, check the "Video size" section in the ffmpeg-utils
	   manual.  Default and minimum resolution is "640x480".

       meter
	   Set the EBU scale meter. Default is 9. Common values are 9 and 18,
	   respectively for EBU scale meter +9 and EBU scale meter +18. Any
	   other integer value between this range is allowed.

       metadata
	   Set metadata injection. If set to 1, the audio input will be
	   segmented into 100ms output frames, each of them containing various
	   loudness information in metadata.  All the metadata keys are
	   prefixed with "lavfi.r128.".

	   Default is 0.

       framelog
	   Force the frame logging level.

	   Available values are:

	   info
	       information logging level

	   verbose
	       verbose logging level

	   By default, the logging level is set to info. If the video or the
	   metadata options are set, it switches to verbose.

       peak
	   Set peak mode(s).

	   Available modes can be cumulated (the option is a "flag" type).
	   Possible values are:

	   none
	       Disable any peak mode (default).

	   sample
	       Enable sample-peak mode.

	       Simple peak mode looking for the higher sample value. It logs a
	       message for sample-peak (identified by "SPK").

	   true
	       Enable true-peak mode.

	       If enabled, the peak lookup is done on an over-sampled version
	       of the input stream for better peak accuracy. It logs a message
	       for true-peak.  (identified by "TPK") and true-peak per frame
	       (identified by "FTPK").	This mode requires a build with
	       "libswresample".

       dualmono
	   Treat mono input files as "dual mono". If a mono file is intended
	   for playback on a stereo system, its EBU R128 measurement will be
	   perceptually incorrect.  If set to "true", this option will
	   compensate for this effect.	Multi-channel input files are not
	   affected by this option.

       panlaw
	   Set a specific pan law to be used for the measurement of dual mono
	   files.  This parameter is optional, and has a default value of
	   -3.01dB.

       Examples

       ·   Real-time graph using ffplay, with a EBU scale meter +18:

		   ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"

       ·   Run an analysis with ffmpeg:

		   ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -

   interleave, ainterleave
       Temporally interleave frames from several inputs.

       "interleave" works with video inputs, "ainterleave" with audio.

       These filters read frames from several inputs and send the oldest
       queued frame to the output.

       Input streams must have well defined, monotonically increasing frame
       timestamp values.

       In order to submit one frame to output, these filters need to enqueue
       at least one frame for each input, so they cannot work in case one
       input is not yet terminated and will not receive incoming frames.

       For example consider the case when one input is a "select" filter which
       always drops input frames. The "interleave" filter will keep reading
       from that input, but it will never be able to send new frames to output
       until the input sends an end-of-stream signal.

       Also, depending on inputs synchronization, the filters will drop frames
       in case one input receives more frames than the other ones, and the
       queue is already filled.

       These filters accept the following options:

       nb_inputs, n
	   Set the number of different inputs, it is 2 by default.

       Examples

       ·   Interleave frames belonging to different streams using ffmpeg:

		   ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

       ·   Add flickering blur effect:

		   select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

   metadata, ametadata
       Manipulate frame metadata.

       This filter accepts the following options:

       mode
	   Set mode of operation of the filter.

	   Can be one of the following:

	   select
	       If both "value" and "key" is set, select frames which have such
	       metadata. If only "key" is set, select every frame that has
	       such key in metadata.

	   add Add new metadata "key" and "value". If key is already available
	       do nothing.

	   modify
	       Modify value of already present key.

	   delete
	       If "value" is set, delete only keys that have such value.
	       Otherwise, delete key. If "key" is not set, delete all metadata
	       values in the frame.

	   print
	       Print key and its value if metadata was found. If "key" is not
	       set print all metadata values available in frame.

       key Set key used with all modes. Must be set for all modes except
	   "print" and "delete".

       value
	   Set metadata value which will be used. This option is mandatory for
	   "modify" and "add" mode.

       function
	   Which function to use when comparing metadata value and "value".

	   Can be one of following:

	   same_str
	       Values are interpreted as strings, returns true if metadata
	       value is same as "value".

	   starts_with
	       Values are interpreted as strings, returns true if metadata
	       value starts with the "value" option string.

	   less
	       Values are interpreted as floats, returns true if metadata
	       value is less than "value".

	   equal
	       Values are interpreted as floats, returns true if "value" is
	       equal with metadata value.

	   greater
	       Values are interpreted as floats, returns true if metadata
	       value is greater than "value".

	   expr
	       Values are interpreted as floats, returns true if expression
	       from option "expr" evaluates to true.

       expr
	   Set expression which is used when "function" is set to "expr".  The
	   expression is evaluated through the eval API and can contain the
	   following constants:

	   VALUE1
	       Float representation of "value" from metadata key.

	   VALUE2
	       Float representation of "value" as supplied by user in "value"
	       option.

       file
	   If specified in "print" mode, output is written to the named file.
	   Instead of plain filename any writable url can be specified.
	   Filename ``-'' is a shorthand for standard output. If "file" option
	   is not set, output is written to the log with AV_LOG_INFO loglevel.

       Examples

       ·   Print all metadata values for frames with key
	   "lavfi.singnalstats.YDIF" with values between 0 and 1.

		   signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

       ·   Print silencedetect output to file metadata.txt.

		   silencedetect,ametadata=mode=print:file=metadata.txt

       ·   Direct all metadata to a pipe with file descriptor 4.

		   metadata=mode=print:file='pipe\:4'

   perms, aperms
       Set read/write permissions for the output frames.

       These filters are mainly aimed at developers to test direct path in the
       following filter in the filtergraph.

       The filters accept the following options:

       mode
	   Select the permissions mode.

	   It accepts the following values:

	   none
	       Do nothing. This is the default.

	   ro  Set all the output frames read-only.

	   rw  Set all the output frames directly writable.

	   toggle
	       Make the frame read-only if writable, and writable if read-
	       only.

	   random
	       Set each output frame read-only or writable randomly.

       seed
	   Set the seed for the random mode, must be an integer included
	   between 0 and "UINT32_MAX". If not specified, or if explicitly set
	   to "-1", the filter will try to use a good random seed on a best
	   effort basis.

       Note: in case of auto-inserted filter between the permission filter and
       the following one, the permission might not be received as expected in
       that following filter. Inserting a format or aformat filter before the
       perms/aperms filter can avoid this problem.

   realtime, arealtime
       Slow down filtering to match real time approximately.

       These filters will pause the filtering for a variable amount of time to
       match the output rate with the input timestamps.  They are similar to
       the re option to "ffmpeg".

       They accept the following options:

       limit
	   Time limit for the pauses. Any pause longer than that will be
	   considered a timestamp discontinuity and reset the timer. Default
	   is 2 seconds.

   select, aselect
       Select frames to pass in output.

       This filter accepts the following options:

       expr, e
	   Set expression, which is evaluated for each input frame.

	   If the expression is evaluated to zero, the frame is discarded.

	   If the evaluation result is negative or NaN, the frame is sent to
	   the first output; otherwise it is sent to the output with index
	   "ceil(val)-1", assuming that the input index starts from 0.

	   For example a value of 1.2 corresponds to the output with index
	   "ceil(1.2)-1 = 2-1 = 1", that is the second output.

       outputs, n
	   Set the number of outputs. The output to which to send the selected
	   frame is based on the result of the evaluation. Default value is 1.

       The expression can contain the following constants:

       n   The (sequential) number of the filtered frame, starting from 0.

       selected_n
	   The (sequential) number of the selected frame, starting from 0.

       prev_selected_n
	   The sequential number of the last selected frame. It's NAN if
	   undefined.

       TB  The timebase of the input timestamps.

       pts The PTS (Presentation TimeStamp) of the filtered video frame,
	   expressed in TB units. It's NAN if undefined.

       t   The PTS of the filtered video frame, expressed in seconds. It's NAN
	   if undefined.

       prev_pts
	   The PTS of the previously filtered video frame. It's NAN if
	   undefined.

       prev_selected_pts
	   The PTS of the last previously filtered video frame. It's NAN if
	   undefined.

       prev_selected_t
	   The PTS of the last previously selected video frame. It's NAN if
	   undefined.

       start_pts
	   The PTS of the first video frame in the video. It's NAN if
	   undefined.

       start_t
	   The time of the first video frame in the video. It's NAN if
	   undefined.

       pict_type (video only)
	   The type of the filtered frame. It can assume one of the following
	   values:

	   I
	   P
	   B
	   S
	   SI
	   SP
	   BI
       interlace_type (video only)
	   The frame interlace type. It can assume one of the following
	   values:

	   PROGRESSIVE
	       The frame is progressive (not interlaced).

	   TOPFIRST
	       The frame is top-field-first.

	   BOTTOMFIRST
	       The frame is bottom-field-first.

       consumed_sample_n (audio only)
	   the number of selected samples before the current frame

       samples_n (audio only)
	   the number of samples in the current frame

       sample_rate (audio only)
	   the input sample rate

       key This is 1 if the filtered frame is a key-frame, 0 otherwise.

       pos the position in the file of the filtered frame, -1 if the
	   information is not available (e.g. for synthetic video)

       scene (video only)
	   value between 0 and 1 to indicate a new scene; a low value reflects
	   a low probability for the current frame to introduce a new scene,
	   while a higher value means the current frame is more likely to be
	   one (see the example below)

       concatdec_select
	   The concat demuxer can select only part of a concat input file by
	   setting an inpoint and an outpoint, but the output packets may not
	   be entirely contained in the selected interval. By using this
	   variable, it is possible to skip frames generated by the concat
	   demuxer which are not exactly contained in the selected interval.

	   This works by comparing the frame pts against the
	   lavf.concat.start_time and the lavf.concat.duration packet metadata
	   values which are also present in the decoded frames.

	   The concatdec_select variable is -1 if the frame pts is at least
	   start_time and either the duration metadata is missing or the frame
	   pts is less than start_time + duration, 0 otherwise, and NaN if the
	   start_time metadata is missing.

	   That basically means that an input frame is selected if its pts is
	   within the interval set by the concat demuxer.

       The default value of the select expression is "1".

       Examples

       ·   Select all frames in input:

		   select

	   The example above is the same as:

		   select=1

       ·   Skip all frames:

		   select=0

       ·   Select only I-frames:

		   select='eq(pict_type\,I)'

       ·   Select one frame every 100:

		   select='not(mod(n\,100))'

       ·   Select only frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)

       ·   Select only I-frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)*eq(pict_type\,I)

       ·   Select frames with a minimum distance of 10 seconds:

		   select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

       ·   Use aselect to select only audio frames with samples number > 100:

		   aselect='gt(samples_n\,100)'

       ·   Create a mosaic of the first scenes:

		   ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png

	   Comparing scene against a value between 0.3 and 0.5 is generally a
	   sane choice.

       ·   Send even and odd frames to separate outputs, and compose them:

		   select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h

       ·   Select useful frames from an ffconcat file which is using inpoints
	   and outpoints but where the source files are not intra frame only.

		   ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi

   sendcmd, asendcmd
       Send commands to filters in the filtergraph.

       These filters read commands to be sent to other filters in the
       filtergraph.

       "sendcmd" must be inserted between two video filters, "asendcmd" must
       be inserted between two audio filters, but apart from that they act the
       same way.

       The specification of commands can be provided in the filter arguments
       with the commands option, or in a file specified by the filename
       option.

       These filters accept the following options:

       commands, c
	   Set the commands to be read and sent to the other filters.

       filename, f
	   Set the filename of the commands to be read and sent to the other
	   filters.

       Commands syntax

       A commands description consists of a sequence of interval
       specifications, comprising a list of commands to be executed when a
       particular event related to that interval occurs. The occurring event
       is typically the current frame time entering or leaving a given time
       interval.

       An interval is specified by the following syntax:

	       <START>[-<END>] <COMMANDS>;

       The time interval is specified by the START and END times.  END is
       optional and defaults to the maximum time.

       The current frame time is considered within the specified interval if
       it is included in the interval [START, END), that is when the time is
       greater or equal to START and is lesser than END.

       COMMANDS consists of a sequence of one or more command specifications,
       separated by ",", relating to that interval.  The syntax of a command
       specification is given by:

	       [<FLAGS>] <TARGET> <COMMAND> <ARG>

       FLAGS is optional and specifies the type of events relating to the time
       interval which enable sending the specified command, and must be a non-
       null sequence of identifier flags separated by "+" or "|" and enclosed
       between "[" and "]".

       The following flags are recognized:

       enter
	   The command is sent when the current frame timestamp enters the
	   specified interval. In other words, the command is sent when the
	   previous frame timestamp was not in the given interval, and the
	   current is.

       leave
	   The command is sent when the current frame timestamp leaves the
	   specified interval. In other words, the command is sent when the
	   previous frame timestamp was in the given interval, and the current
	   is not.

       If FLAGS is not specified, a default value of "[enter]" is assumed.

       TARGET specifies the target of the command, usually the name of the
       filter class or a specific filter instance name.

       COMMAND specifies the name of the command for the target filter.

       ARG is optional and specifies the optional list of argument for the
       given COMMAND.

       Between one interval specification and another, whitespaces, or
       sequences of characters starting with "#" until the end of line, are
       ignored and can be used to annotate comments.

       A simplified BNF description of the commands specification syntax
       follows:

	       <COMMAND_FLAG>  ::= "enter" | "leave"
	       <COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
	       <COMMAND>       ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
	       <COMMANDS>      ::= <COMMAND> [,<COMMANDS>]
	       <INTERVAL>      ::= <START>[-<END>] <COMMANDS>
	       <INTERVALS>     ::= <INTERVAL>[;<INTERVALS>]

       Examples

       ·   Specify audio tempo change at second 4:

		   asendcmd=c='4.0 atempo tempo 1.5',atempo

       ·   Target a specific filter instance:

		   asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my

       ·   Specify a list of drawtext and hue commands in a file.

		   # show text in the interval 5-10
		   5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
			    [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';

		   # desaturate the image in the interval 15-20
		   15.0-20.0 [enter] hue s 0,
			     [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
			     [leave] hue s 1,
			     [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';

		   # apply an exponential saturation fade-out effect, starting from time 25
		   25 [enter] hue s exp(25-t)

	   A filtergraph allowing to read and process the above command list
	   stored in a file test.cmd, can be specified with:

		   sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

   setpts, asetpts
       Change the PTS (presentation timestamp) of the input frames.

       "setpts" works on video frames, "asetpts" on audio frames.

       This filter accepts the following options:

       expr
	   The expression which is evaluated for each frame to construct its
	   timestamp.

       The expression is evaluated through the eval API and can contain the
       following constants:

       FRAME_RATE
	   frame rate, only defined for constant frame-rate video

       PTS The presentation timestamp in input

       N   The count of the input frame for video or the number of consumed
	   samples, not including the current frame for audio, starting from
	   0.

       NB_CONSUMED_SAMPLES
	   The number of consumed samples, not including the current frame
	   (only audio)

       NB_SAMPLES, S
	   The number of samples in the current frame (only audio)

       SAMPLE_RATE, SR
	   The audio sample rate.

       STARTPTS
	   The PTS of the first frame.

       STARTT
	   the time in seconds of the first frame

       INTERLACED
	   State whether the current frame is interlaced.

       T   the time in seconds of the current frame

       POS original position in the file of the frame, or undefined if
	   undefined for the current frame

       PREV_INPTS
	   The previous input PTS.

       PREV_INT
	   previous input time in seconds

       PREV_OUTPTS
	   The previous output PTS.

       PREV_OUTT
	   previous output time in seconds

       RTCTIME
	   The wallclock (RTC) time in microseconds. This is deprecated, use
	   time(0) instead.

       RTCSTART
	   The wallclock (RTC) time at the start of the movie in microseconds.

       TB  The timebase of the input timestamps.

       Examples

       ·   Start counting PTS from zero

		   setpts=PTS-STARTPTS

       ·   Apply fast motion effect:

		   setpts=0.5*PTS

       ·   Apply slow motion effect:

		   setpts=2.0*PTS

       ·   Set fixed rate of 25 frames per second:

		   setpts=N/(25*TB)

       ·   Set fixed rate 25 fps with some jitter:

		   setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

       ·   Apply an offset of 10 seconds to the input PTS:

		   setpts=PTS+10/TB

       ·   Generate timestamps from a "live source" and rebase onto the
	   current timebase:

		   setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

       ·   Generate timestamps by counting samples:

		   asetpts=N/SR/TB

   settb, asettb
       Set the timebase to use for the output frames timestamps.  It is mainly
       useful for testing timebase configuration.

       It accepts the following parameters:

       expr, tb
	   The expression which is evaluated into the output timebase.

       The value for tb is an arithmetic expression representing a rational.
       The expression can contain the constants "AVTB" (the default timebase),
       "intb" (the input timebase) and "sr" (the sample rate, audio only).
       Default value is "intb".

       Examples

       ·   Set the timebase to 1/25:

		   settb=expr=1/25

       ·   Set the timebase to 1/10:

		   settb=expr=0.1

       ·   Set the timebase to 1001/1000:

		   settb=1+0.001

       ·   Set the timebase to 2*intb:

		   settb=2*intb

       ·   Set the default timebase value:

		   settb=AVTB

   showcqt
       Convert input audio to a video output representing frequency spectrum
       logarithmically using Brown-Puckette constant Q transform algorithm
       with direct frequency domain coefficient calculation (but the transform
       itself is not really constant Q, instead the Q factor is actually
       variable/clamped), with musical tone scale, from E0 to D#10.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. It must be even. For the
	   syntax of this option, check the "Video size" section in the
	   ffmpeg-utils manual.  Default value is "1920x1080".

       fps, rate, r
	   Set the output frame rate. Default value is 25.

       bar_h
	   Set the bargraph height. It must be even. Default value is "-1"
	   which computes the bargraph height automatically.

       axis_h
	   Set the axis height. It must be even. Default value is "-1" which
	   computes the axis height automatically.

       sono_h
	   Set the sonogram height. It must be even. Default value is "-1"
	   which computes the sonogram height automatically.

       fullhd
	   Set the fullhd resolution. This option is deprecated, use size, s
	   instead. Default value is 1.

       sono_v, volume
	   Specify the sonogram volume expression. It can contain variables:

	   bar_v
	       the bar_v evaluated expression

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   a_weighting(f)
	       A-weighting of equal loudness

	   b_weighting(f)
	       B-weighting of equal loudness

	   c_weighting(f)
	       C-weighting of equal loudness.

	   Default value is 16.

       bar_v, volume2
	   Specify the bargraph volume expression. It can contain variables:

	   sono_v
	       the sono_v evaluated expression

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   a_weighting(f)
	       A-weighting of equal loudness

	   b_weighting(f)
	       B-weighting of equal loudness

	   c_weighting(f)
	       C-weighting of equal loudness.

	   Default value is "sono_v".

       sono_g, gamma
	   Specify the sonogram gamma. Lower gamma makes the spectrum more
	   contrast, higher gamma makes the spectrum having more range.
	   Default value is 3.	Acceptable range is "[1, 7]".

       bar_g, gamma2
	   Specify the bargraph gamma. Default value is 1. Acceptable range is
	   "[1, 7]".

       bar_t
	   Specify the bargraph transparency level. Lower value makes the
	   bargraph sharper.  Default value is 1. Acceptable range is "[0,
	   1]".

       timeclamp, tc
	   Specify the transform timeclamp. At low frequency, there is trade-
	   off between accuracy in time domain and frequency domain. If
	   timeclamp is lower, event in time domain is represented more
	   accurately (such as fast bass drum), otherwise event in frequency
	   domain is represented more accurately (such as bass guitar).
	   Acceptable range is "[0.002, 1]". Default value is 0.17.

       attack
	   Set attack time in seconds. The default is 0 (disabled). Otherwise,
	   it limits future samples by applying asymmetric windowing in time
	   domain, useful when low latency is required. Accepted range is "[0,
	   1]".

       basefreq
	   Specify the transform base frequency. Default value is
	   20.01523126408007475, which is frequency 50 cents below E0.
	   Acceptable range is "[10, 100000]".

       endfreq
	   Specify the transform end frequency. Default value is
	   20495.59681441799654, which is frequency 50 cents above D#10.
	   Acceptable range is "[10, 100000]".

       coeffclamp
	   This option is deprecated and ignored.

       tlength
	   Specify the transform length in time domain. Use this option to
	   control accuracy trade-off between time domain and frequency domain
	   at every frequency sample.  It can contain variables:

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option.

	   Default value is "384*tc/(384+tc*f)".

       count
	   Specify the transform count for every video frame. Default value is
	   6.  Acceptable range is "[1, 30]".

       fcount
	   Specify the transform count for every single pixel. Default value
	   is 0, which makes it computed automatically. Acceptable range is
	   "[0, 10]".

       fontfile
	   Specify font file for use with freetype to draw the axis. If not
	   specified, use embedded font. Note that drawing with font file or
	   embedded font is not implemented with custom basefreq and endfreq,
	   use axisfile option instead.

       font
	   Specify fontconfig pattern. This has lower priority than fontfile.
	   The : in the pattern may be replaced by | to avoid unnecessary
	   escaping.

       fontcolor
	   Specify font color expression. This is arithmetic expression that
	   should return integer value 0xRRGGBB. It can contain variables:

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   midi(f)
	       midi number of frequency f, some midi numbers: E0(16), C1(24),
	       C2(36), A4(69)

	   r(x), g(x), b(x)
	       red, green, and blue value of intensity x.

	   Default value is "st(0, (midi(f)-59.5)/12); st(1,
	   if(between(ld(0),0,1), 0.5-0.5*cos(2*PI*ld(0)), 0)); r(1-ld(1)) +
	   b(ld(1))".

       axisfile
	   Specify image file to draw the axis. This option override fontfile
	   and fontcolor option.

       axis, text
	   Enable/disable drawing text to the axis. If it is set to 0, drawing
	   to the axis is disabled, ignoring fontfile and axisfile option.
	   Default value is 1.

       csp Set colorspace. The accepted values are:

	   unspecified
	       Unspecified (default)

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   bt2020ncl
	       BT.2020 with non-constant luminance

       cscheme
	   Set spectrogram color scheme. This is list of floating point values
	   with format "left_r|left_g|left_b|right_r|right_g|right_b".	The
	   default is "1|0.5|0|0|0.5|1".

       Examples

       ·   Playing audio while showing the spectrum:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

       ·   Same as above, but with frame rate 30 fps:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'

       ·   Playing at 1280x720:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'

       ·   Disable sonogram display:

		   sono_h=0

       ·   A1 and its harmonics: A1, A2, (near)E3, A3:

		   ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
				    asplit[a][out1]; [a] showcqt [out0]'

       ·   Same as above, but with more accuracy in frequency domain:

		   ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
				    asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'

       ·   Custom volume:

		   bar_v=10:sono_v=bar_v*a_weighting(f)

       ·   Custom gamma, now spectrum is linear to the amplitude.

		   bar_g=2:sono_g=2

       ·   Custom tlength equation:

		   tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'

       ·   Custom fontcolor and fontfile, C-note is colored green, others are
	   colored blue:

		   fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf

       ·   Custom font using fontconfig:

		   font='Courier New,Monospace,mono|bold'

       ·   Custom frequency range with custom axis using image file:

		   axisfile=myaxis.png:basefreq=40:endfreq=10000

   showfreqs
       Convert input audio to video output representing the audio power
       spectrum.  Audio amplitude is on Y-axis while frequency is on X-axis.

       The filter accepts the following options:

       size, s
	   Specify size of video. For the syntax of this option, check the
	   "Video size" section in the ffmpeg-utils manual.  Default is
	   "1024x512".

       mode
	   Set display mode.  This set how each frequency bin will be
	   represented.

	   It accepts the following values:

	   line
	   bar
	   dot

	   Default is "bar".

       ascale
	   Set amplitude scale.

	   It accepts the following values:

	   lin Linear scale.

	   sqrt
	       Square root scale.

	   cbrt
	       Cubic root scale.

	   log Logarithmic scale.

	   Default is "log".

       fscale
	   Set frequency scale.

	   It accepts the following values:

	   lin Linear scale.

	   log Logarithmic scale.

	   rlog
	       Reverse logarithmic scale.

	   Default is "lin".

       win_size
	   Set window size.

	   It accepts the following values:

	   w16
	   w32
	   w64
	   w128
	   w256
	   w512
	   w1024
	   w2048
	   w4096
	   w8192
	   w16384
	   w32768
	   w65536

	   Default is "w2048"

       win_func
	   Set windowing function.

	   It accepts the following values:

	   rect
	   bartlett
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson

	   Default is "hanning".

       overlap
	   Set window overlap. In range "[0, 1]". Default is 1, which means
	   optimal overlap for selected window function will be picked.

       averaging
	   Set time averaging. Setting this to 0 will display current maximal
	   peaks.  Default is 1, which means time averaging is disabled.

       colors
	   Specify list of colors separated by space or by '|' which will be
	   used to draw channel frequencies. Unrecognized or missing colors
	   will be replaced by white color.

       cmode
	   Set channel display mode.

	   It accepts the following values:

	   combined
	   separate

	   Default is "combined".

       minamp
	   Set minimum amplitude used in "log" amplitude scaler.

   showspectrum
       Convert input audio to a video output, representing the audio frequency
       spectrum.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "640x512".

       slide
	   Specify how the spectrum should slide along the window.

	   It accepts the following values:

	   replace
	       the samples start again on the left when they reach the right

	   scroll
	       the samples scroll from right to left

	   fullframe
	       frames are only produced when the samples reach the right

	   rscroll
	       the samples scroll from left to right

	   Default value is "replace".

       mode
	   Specify display mode.

	   It accepts the following values:

	   combined
	       all channels are displayed in the same row

	   separate
	       all channels are displayed in separate rows

	   Default value is combined.

       color
	   Specify display color mode.

	   It accepts the following values:

	   channel
	       each channel is displayed in a separate color

	   intensity
	       each channel is displayed using the same color scheme

	   rainbow
	       each channel is displayed using the rainbow color scheme

	   moreland
	       each channel is displayed using the moreland color scheme

	   nebulae
	       each channel is displayed using the nebulae color scheme

	   fire
	       each channel is displayed using the fire color scheme

	   fiery
	       each channel is displayed using the fiery color scheme

	   fruit
	       each channel is displayed using the fruit color scheme

	   cool
	       each channel is displayed using the cool color scheme

	   Default value is channel.

       scale
	   Specify scale used for calculating intensity color values.

	   It accepts the following values:

	   lin linear

	   sqrt
	       square root, default

	   cbrt
	       cubic root

	   log logarithmic

	   4thrt
	       4th root

	   5thrt
	       5th root

	   Default value is sqrt.

       saturation
	   Set saturation modifier for displayed colors. Negative values
	   provide alternative color scheme. 0 is no saturation at all.
	   Saturation must be in [-10.0, 10.0] range.  Default value is 1.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson

	   Default value is "hann".

       orientation
	   Set orientation of time vs frequency axis. Can be "vertical" or
	   "horizontal". Default is "vertical".

       overlap
	   Set ratio of overlap window. Default value is 0.  When value is 1
	   overlap is set to recommended size for specific window function
	   currently used.

       gain
	   Set scale gain for calculating intensity color values.  Default
	   value is 1.

       data
	   Set which data to display. Can be "magnitude", default or "phase".

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

       The usage is very similar to the showwaves filter; see the examples in
       that section.

       Examples

       ·   Large window with logarithmic color scaling:

		   showspectrum=s=1280x480:scale=log

       ·   Complete example for a colored and sliding spectrum per channel
	   using ffplay:

		   ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
				[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'

   showspectrumpic
       Convert input audio to a single video frame, representing the audio
       frequency spectrum.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "4096x2048".

       mode
	   Specify display mode.

	   It accepts the following values:

	   combined
	       all channels are displayed in the same row

	   separate
	       all channels are displayed in separate rows

	   Default value is combined.

       color
	   Specify display color mode.

	   It accepts the following values:

	   channel
	       each channel is displayed in a separate color

	   intensity
	       each channel is displayed using the same color scheme

	   rainbow
	       each channel is displayed using the rainbow color scheme

	   moreland
	       each channel is displayed using the moreland color scheme

	   nebulae
	       each channel is displayed using the nebulae color scheme

	   fire
	       each channel is displayed using the fire color scheme

	   fiery
	       each channel is displayed using the fiery color scheme

	   fruit
	       each channel is displayed using the fruit color scheme

	   cool
	       each channel is displayed using the cool color scheme

	   Default value is intensity.

       scale
	   Specify scale used for calculating intensity color values.

	   It accepts the following values:

	   lin linear

	   sqrt
	       square root, default

	   cbrt
	       cubic root

	   log logarithmic

	   4thrt
	       4th root

	   5thrt
	       5th root

	   Default value is log.

       saturation
	   Set saturation modifier for displayed colors. Negative values
	   provide alternative color scheme. 0 is no saturation at all.
	   Saturation must be in [-10.0, 10.0] range.  Default value is 1.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson

	   Default value is "hann".

       orientation
	   Set orientation of time vs frequency axis. Can be "vertical" or
	   "horizontal". Default is "vertical".

       gain
	   Set scale gain for calculating intensity color values.  Default
	   value is 1.

       legend
	   Draw time and frequency axes and legends. Default is enabled.

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

       Examples

       ·   Extract an audio spectrogram of a whole audio track in a 1024x1024
	   picture using ffmpeg:

		   ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png

   showvolume
       Convert input audio volume to a video output.

       The filter accepts the following options:

       rate, r
	   Set video rate.

       b   Set border width, allowed range is [0, 5]. Default is 1.

       w   Set channel width, allowed range is [80, 8192]. Default is 400.

       h   Set channel height, allowed range is [1, 900]. Default is 20.

       f   Set fade, allowed range is [0.001, 1]. Default is 0.95.

       c   Set volume color expression.

	   The expression can use the following variables:

	   VOLUME
	       Current max volume of channel in dB.

	   PEAK
	       Current peak.

	   CHANNEL
	       Current channel number, starting from 0.

       t   If set, displays channel names. Default is enabled.

       v   If set, displays volume values. Default is enabled.

       o   Set orientation, can be "horizontal" or "vertical", default is
	   "horizontal".

       s   Set step size, allowed range s [0, 5]. Default is 0, which means
	   step is disabled.

   showwaves
       Convert input audio to a video output, representing the samples waves.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "600x240".

       mode
	   Set display mode.

	   Available values are:

	   point
	       Draw a point for each sample.

	   line
	       Draw a vertical line for each sample.

	   p2p Draw a point for each sample and a line between them.

	   cline
	       Draw a centered vertical line for each sample.

	   Default value is "point".

       n   Set the number of samples which are printed on the same column. A
	   larger value will decrease the frame rate. Must be a positive
	   integer. This option can be set only if the value for rate is not
	   explicitly specified.

       rate, r
	   Set the (approximate) output frame rate. This is done by setting
	   the option n. Default value is "25".

       split_channels
	   Set if channels should be drawn separately or overlap. Default
	   value is 0.

       colors
	   Set colors separated by '|' which are going to be used for drawing
	   of each channel.

       scale
	   Set amplitude scale.

	   Available values are:

	   lin Linear.

	   log Logarithmic.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   Default is linear.

       Examples

       ·   Output the input file audio and the corresponding video
	   representation at the same time:

		   amovie=a.mp3,asplit[out0],showwaves[out1]

       ·   Create a synthetic signal and show it with showwaves, forcing a
	   frame rate of 30 frames per second:

		   aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

   showwavespic
       Convert input audio to a single video frame, representing the samples
       waves.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "600x240".

       split_channels
	   Set if channels should be drawn separately or overlap. Default
	   value is 0.

       colors
	   Set colors separated by '|' which are going to be used for drawing
	   of each channel.

       scale
	   Set amplitude scale.

	   Available values are:

	   lin Linear.

	   log Logarithmic.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   Default is linear.

       Examples

       ·   Extract a channel split representation of the wave form of a whole
	   audio track in a 1024x800 picture using ffmpeg:

		   ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png

   sidedata, asidedata
       Delete frame side data, or select frames based on it.

       This filter accepts the following options:

       mode
	   Set mode of operation of the filter.

	   Can be one of the following:

	   select
	       Select every frame with side data of "type".

	   delete
	       Delete side data of "type". If "type" is not set, delete all
	       side data in the frame.

       type
	   Set side data type used with all modes. Must be set for "select"
	   mode. For the list of frame side data types, refer to the
	   "AVFrameSideDataType" enum in libavutil/frame.h. For example, to
	   choose "AV_FRAME_DATA_PANSCAN" side data, you must specify
	   "PANSCAN".

   spectrumsynth
       Sythesize audio from 2 input video spectrums, first input stream
       represents magnitude across time and second represents phase across
       time.  The filter will transform from frequency domain as displayed in
       videos back to time domain as presented in audio output.

       This filter is primarily created for reversing processed showspectrum
       filter outputs, but can synthesize sound from other spectrograms too.
       But in such case results are going to be poor if the phase data is not
       available, because in such cases phase data need to be recreated,
       usually its just recreated from random noise.  For best results use
       gray only output ("channel" color mode in showspectrum filter) and
       "log" scale for magnitude video and "lin" scale for phase video. To
       produce phase, for 2nd video, use "data" option. Inputs videos should
       generally use "fullframe" slide mode as that saves resources needed for
       decoding video.

       The filter accepts the following options:

       sample_rate
	   Specify sample rate of output audio, the sample rate of audio from
	   which spectrum was generated may differ.

       channels
	   Set number of channels represented in input video spectrums.

       scale
	   Set scale which was used when generating magnitude input spectrum.
	   Can be "lin" or "log". Default is "log".

       slide
	   Set slide which was used when generating inputs spectrums.  Can be
	   "replace", "scroll", "fullframe" or "rscroll".  Default is
	   "fullframe".

       win_func
	   Set window function used for resynthesis.

       overlap
	   Set window overlap. In range "[0, 1]". Default is 1, which means
	   optimal overlap for selected window function will be picked.

       orientation
	   Set orientation of input videos. Can be "vertical" or "horizontal".
	   Default is "vertical".

       Examples

       ·   First create magnitude and phase videos from audio, assuming audio
	   is stereo with 44100 sample rate, then resynthesize videos back to
	   audio with spectrumsynth:

		   ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
		   ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
		   ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac

   split, asplit
       Split input into several identical outputs.

       "asplit" works with audio input, "split" with video.

       The filter accepts a single parameter which specifies the number of
       outputs. If unspecified, it defaults to 2.

       Examples

       ·   Create two separate outputs from the same input:

		   [in] split [out0][out1]

       ·   To create 3 or more outputs, you need to specify the number of
	   outputs, like in:

		   [in] asplit=3 [out0][out1][out2]

       ·   Create two separate outputs from the same input, one cropped and
	   one padded:

		   [in] split [splitout1][splitout2];
		   [splitout1] crop=100:100:0:0    [cropout];
		   [splitout2] pad=200:200:100:100 [padout];

       ·   Create 5 copies of the input audio with ffmpeg:

		   ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

   zmq, azmq
       Receive commands sent through a libzmq client, and forward them to
       filters in the filtergraph.

       "zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted
       between two video filters, "azmq" between two audio filters.

       To enable these filters you need to install the libzmq library and
       headers and configure FFmpeg with "--enable-libzmq".

       For more information about libzmq see: <http://www.zeromq.org/>

       The "zmq" and "azmq" filters work as a libzmq server, which receives
       messages sent through a network interface defined by the bind_address
       option.

       The received message must be in the form:

	       <TARGET> <COMMAND> [<ARG>]

       TARGET specifies the target of the command, usually the name of the
       filter class or a specific filter instance name.

       COMMAND specifies the name of the command for the target filter.

       ARG is optional and specifies the optional argument list for the given
       COMMAND.

       Upon reception, the message is processed and the corresponding command
       is injected into the filtergraph. Depending on the result, the filter
       will send a reply to the client, adopting the format:

	       <ERROR_CODE> <ERROR_REASON>
	       <MESSAGE>

       MESSAGE is optional.

       Examples

       Look at tools/zmqsend for an example of a zmq client which can be used
       to send commands processed by these filters.

       Consider the following filtergraph generated by ffplay

	       ffplay -dumpgraph 1 -f lavfi "
	       color=s=100x100:c=red  [l];
	       color=s=100x100:c=blue [r];
	       nullsrc=s=200x100, zmq [bg];
	       [bg][l]	 overlay      [bg+l];
	       [bg+l][r] overlay=x=100 "

       To change the color of the left side of the video, the following
       command can be used:

	       echo Parsed_color_0 c yellow | tools/zmqsend

       To change the right side:

	       echo Parsed_color_1 c pink | tools/zmqsend

MULTIMEDIA SOURCES
       Below is a description of the currently available multimedia sources.

   amovie
       This is the same as movie source, except it selects an audio stream by
       default.

   movie
       Read audio and/or video stream(s) from a movie container.

       It accepts the following parameters:

       filename
	   The name of the resource to read (not necessarily a file; it can
	   also be a device or a stream accessed through some protocol).

       format_name, f
	   Specifies the format assumed for the movie to read, and can be
	   either the name of a container or an input device. If not
	   specified, the format is guessed from movie_name or by probing.

       seek_point, sp
	   Specifies the seek point in seconds. The frames will be output
	   starting from this seek point. The parameter is evaluated with
	   "av_strtod", so the numerical value may be suffixed by an IS
	   postfix. The default value is "0".

       streams, s
	   Specifies the streams to read. Several streams can be specified,
	   separated by "+". The source will then have as many outputs, in the
	   same order. The syntax is explained in the ``Stream specifiers''
	   section in the ffmpeg manual. Two special names, "dv" and "da"
	   specify respectively the default (best suited) video and audio
	   stream. Default is "dv", or "da" if the filter is called as
	   "amovie".

       stream_index, si
	   Specifies the index of the video stream to read. If the value is
	   -1, the most suitable video stream will be automatically selected.
	   The default value is "-1". Deprecated. If the filter is called
	   "amovie", it will select audio instead of video.

       loop
	   Specifies how many times to read the stream in sequence.  If the
	   value is 0, the stream will be looped infinitely.  Default value is
	   "1".

	   Note that when the movie is looped the source timestamps are not
	   changed, so it will generate non monotonically increasing
	   timestamps.

       discontinuity
	   Specifies the time difference between frames above which the point
	   is considered a timestamp discontinuity which is removed by
	   adjusting the later timestamps.

       It allows overlaying a second video on top of the main input of a
       filtergraph, as shown in this graph:

	       input -----------> deltapts0 --> overlay --> output
						   ^
						   |
	       movie --> scale--> deltapts1 -------+

       Examples

       ·   Skip 3.2 seconds from the start of the AVI file in.avi, and overlay
	   it on top of the input labelled "in":

		   movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
		   [in] setpts=PTS-STARTPTS [main];
		   [main][over] overlay=16:16 [out]

       ·   Read from a video4linux2 device, and overlay it on top of the input
	   labelled "in":

		   movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
		   [in] setpts=PTS-STARTPTS [main];
		   [main][over] overlay=16:16 [out]

       ·   Read the first video stream and the audio stream with id 0x81 from
	   dvd.vob; the video is connected to the pad named "video" and the
	   audio is connected to the pad named "audio":

		   movie=dvd.vob:s=v:0+#0x81 [video] [audio]

       Commands

       Both movie and amovie support the following commands:

       seek
	   Perform seek using "av_seek_frame".	The syntax is: seek
	   stream_index|timestamp|flags

	   ·   stream_index: If stream_index is -1, a default stream is
	       selected, and timestamp is automatically converted from
	       AV_TIME_BASE units to the stream specific time_base.

	   ·   timestamp: Timestamp in AVStream.time_base units or, if no
	       stream is specified, in AV_TIME_BASE units.

	   ·   flags: Flags which select direction and seeking mode.

       get_duration
	   Get movie duration in AV_TIME_BASE units.

SEE ALSO
       ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), ffmpeg-utils(1),
       ffmpeg-scaler(1), ffmpeg-resampler(1), ffmpeg-codecs(1),
       ffmpeg-bitstream-filters(1), ffmpeg-formats(1), ffmpeg-devices(1),
       ffmpeg-protocols(1), ffmpeg-filters(1)

AUTHORS
       The FFmpeg developers.

       For details about the authorship, see the Git history of the project
       (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
       the FFmpeg source directory, or browsing the online repository at
       <http://source.ffmpeg.org>.

       Maintainers for the specific components are listed in the file
       MAINTAINERS in the source code tree.

								 FFMPEG-ALL(1)
