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

NAME
       ffmpeg - ffmpeg media converter

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

DESCRIPTION
       ffmpeg is a universal media converter. It can read a wide variety of
       inputs - including live grabbing/recording devices - filter, and
       transcode them into a plethora of output formats.

       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.

       Some simple examples follow.

       o   Convert an input media file to a different format, by re-encoding
	   media streams:

		   ffmpeg -i input.avi output.mp4

       o   Set the video bitrate of the output file to 64 kbit/s:

		   ffmpeg -i input.avi -b:v 64k -bufsize 64k output.mp4

       o   Force the frame rate of the output file to 24 fps:

		   ffmpeg -i input.avi -r 24 output.mp4

       o   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.mp4

       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.

   Loopback decoders
       While decoders are normally associated with demuxer streams, it is also
       possible to create "loopback" decoders that decode the output from some
       encoder and allow it to be fed back to complex filtergraphs. This is
       done with the "-dec" directive, which takes as a parameter the index of
       the output stream that should be decoded. Every such directive creates
       a new loopback decoder, indexed with successive integers starting at
       zero. These indices should then be used to refer to loopback decoders
       in complex filtergraph link labels, as described in the documentation
       for -filter_complex.

       Decoding AVOptions can be passed to loopback decoders by placing them
       before "-dec", analogously to input/output options.

       E.g. the following example:

	       ffmpeg -i INPUT					      \
		 -map 0:v:0 -c:v libx264 -crf 45 -f null -	      \
		 -threads 3 -dec 0:0				      \
		 -filter_complex '[0:v][dec:0]hstack[stack]'	      \
		 -map '[stack]' -c:v ffv1 OUTPUT

       reads an input video and

       o   (line 2) encodes it with "libx264" at low quality;

       o   (line 3) decodes this encoded stream using 3 threads;

       o   (line 4) places decoded video side by side with the original input
	   video;

       o   (line 5) combined video is then losslessly encoded and written into
	   OUTPUT.

STREAM SELECTION
       ffmpeg provides the "-map" option for manual control of stream
       selection in each output file. Users can skip "-map" and let ffmpeg
       perform automatic stream selection as described below. The "-vn / -an /
       -sn / -dn" options can be used to skip inclusion of video, audio,
       subtitle and data streams respectively, whether manually mapped or
       automatically selected, except for those streams which are outputs of
       complex filtergraphs.

   Description
       The sub-sections that follow describe the various rules that are
       involved in stream selection.  The examples that follow next show how
       these rules are applied in practice.

       While every effort is made to accurately reflect the behavior of the
       program, FFmpeg is under continuous development and the code may have
       changed since the time of this writing.

       Automatic stream selection

       In the absence of any map options for a particular output file, ffmpeg
       inspects the output format to check which type of streams can be
       included in it, viz. video, audio and/or subtitles. For each acceptable
       stream type, ffmpeg will pick one stream, when available, from among
       all the inputs.

       It will select that stream based upon the following criteria:

       o   for video, it is the stream with the highest resolution,

       o   for audio, it is the stream with the most channels,

       o   for subtitles, it is the first subtitle stream found but there's a
	   caveat.  The output format's default subtitle encoder can be either
	   text-based or image-based, and only a subtitle stream of the same
	   type will be chosen.

       In the case where several streams of the same type rate equally, the
       stream with the lowest index is chosen.

       Data or attachment streams are not automatically selected and can only
       be included using "-map".

       Manual stream selection

       When "-map" is used, only user-mapped streams are included in that
       output file, with one possible exception for filtergraph outputs
       described below.

       Complex filtergraphs

       If there are any complex filtergraph output streams with unlabeled
       pads, they will be added to the first output file. This will lead to a
       fatal error if the stream type is not supported by the output format.
       In the absence of the map option, the inclusion of these streams leads
       to the automatic stream selection of their types being skipped. If map
       options are present, these filtergraph streams are included in addition
       to the mapped streams.

       Complex filtergraph output streams with labeled pads must be mapped
       once and exactly once.

       Stream handling

       Stream handling is independent of stream selection, with an exception
       for subtitles described below. Stream handling is set via the "-codec"
       option addressed to streams within a specific output file. In
       particular, codec options are applied by ffmpeg after the stream
       selection process and thus do not influence the latter. If no "-codec"
       option is specified for a stream type, ffmpeg will select the default
       encoder registered by the output file muxer.

       An exception exists for subtitles. If a subtitle encoder is specified
       for an output file, the first subtitle stream found of any type, text
       or image, will be included. ffmpeg does not validate if the specified
       encoder can convert the selected stream or if the converted stream is
       acceptable within the output format. This applies generally as well:
       when the user sets an encoder manually, the stream selection process
       cannot check if the encoded stream can be muxed into the output file.
       If it cannot, ffmpeg will abort and all output files will fail to be
       processed.

   Examples
       The following examples illustrate the behavior, quirks and limitations
       of ffmpeg's stream selection methods.

       They assume the following three input files.

	       input file 'A.avi'
		     stream 0: video 640x360
		     stream 1: audio 2 channels

	       input file 'B.mp4'
		     stream 0: video 1920x1080
		     stream 1: audio 2 channels
		     stream 2: subtitles (text)
		     stream 3: audio 5.1 channels
		     stream 4: subtitles (text)

	       input file 'C.mkv'
		     stream 0: video 1280x720
		     stream 1: audio 2 channels
		     stream 2: subtitles (image)

       Example: automatic stream selection

	       ffmpeg -i A.avi -i B.mp4 out1.mkv out2.wav -map 1:a -c:a copy out3.mov

       There are three output files specified, and for the first two, no
       "-map" options are set, so ffmpeg will select streams for these two
       files automatically.

       out1.mkv is a Matroska container file and accepts video, audio and
       subtitle streams, so ffmpeg will try to select one of each type.For
       video, it will select "stream 0" from B.mp4, which has the highest
       resolution among all the input video streams.For audio, it will select
       "stream 3" from B.mp4, since it has the greatest number of channels.For
       subtitles, it will select "stream 2" from B.mp4, which is the first
       subtitle stream from among A.avi and B.mp4.

       out2.wav accepts only audio streams, so only "stream 3" from B.mp4 is
       selected.

       For out3.mov, since a "-map" option is set, no automatic stream
       selection will occur. The "-map 1:a" option will select all audio
       streams from the second input B.mp4. No other streams will be included
       in this output file.

       For the first two outputs, all included streams will be transcoded. The
       encoders chosen will be the default ones registered by each output
       format, which may not match the codec of the selected input streams.

       For the third output, codec option for audio streams has been set to
       "copy", so no decoding-filtering-encoding operations will occur, or can
       occur.  Packets of selected streams shall be conveyed from the input
       file and muxed within the output file.

       Example: automatic subtitles selection

	       ffmpeg -i C.mkv out1.mkv -c:s dvdsub -an out2.mkv

       Although out1.mkv is a Matroska container file which accepts subtitle
       streams, only a video and audio stream shall be selected. The subtitle
       stream of C.mkv is image-based and the default subtitle encoder of the
       Matroska muxer is text-based, so a transcode operation for the
       subtitles is expected to fail and hence the stream isn't selected.
       However, in out2.mkv, a subtitle encoder is specified in the command
       and so, the subtitle stream is selected, in addition to the video
       stream. The presence of "-an" disables audio stream selection for
       out2.mkv.

       Example: unlabeled filtergraph outputs

	       ffmpeg -i A.avi -i C.mkv -i B.mp4 -filter_complex "overlay" out1.mp4 out2.srt

       A filtergraph is setup here using the "-filter_complex" option and
       consists of a single video filter. The "overlay" filter requires
       exactly two video inputs, but none are specified, so the first two
       available video streams are used, those of A.avi and C.mkv. The output
       pad of the filter has no label and so is sent to the first output file
       out1.mp4. Due to this, automatic selection of the video stream is
       skipped, which would have selected the stream in B.mp4. The audio
       stream with most channels viz. "stream 3" in B.mp4, is chosen
       automatically. No subtitle stream is chosen however, since the MP4
       format has no default subtitle encoder registered, and the user hasn't
       specified a subtitle encoder.

       The 2nd output file, out2.srt, only accepts text-based subtitle
       streams. So, even though the first subtitle stream available belongs to
       C.mkv, it is image-based and hence skipped.  The selected stream,
       "stream 2" in B.mp4, is the first text-based subtitle stream.

       Example: labeled filtergraph outputs

	       ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
		      -map '[outv]' -an        out1.mp4 \
					       out2.mkv \
		      -map '[outv]' -map 1:a:0 out3.mkv

       The above command will fail, as the output pad labelled "[outv]" has
       been mapped twice.  None of the output files shall be processed.

	       ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0[outv];overlay;aresample" \
		      -an	 out1.mp4 \
				 out2.mkv \
		      -map 1:a:0 out3.mkv

       This command above will also fail as the hue filter output has a label,
       "[outv]", and hasn't been mapped anywhere.

       The command should be modified as follows,

	       ffmpeg -i A.avi -i B.mp4 -i C.mkv -filter_complex "[1:v]hue=s=0,split=2[outv1][outv2];overlay;aresample" \
		       -map '[outv1]' -an	 out1.mp4 \
						 out2.mkv \
		       -map '[outv2]' -map 1:a:0 out3.mkv

       The video stream from B.mp4 is sent to the hue filter, whose output is
       cloned once using the split filter, and both outputs labelled. Then a
       copy each is mapped to the first and third output files.

       The overlay filter, requiring two video inputs, uses the first two
       unused video streams. Those are the streams from A.avi and C.mkv. The
       overlay output isn't labelled, so it is sent to the first output file
       out1.mp4, regardless of the presence of the "-map" option.

       The aresample filter is sent the first unused audio stream, that of
       A.avi. Since this filter output is also unlabelled, it too is mapped to
       the first output file. The presence of "-an" only suppresses automatic
       or manual stream selection of audio streams, not outputs sent from
       filtergraphs. Both these mapped streams shall be ordered before the
       mapped stream in out1.mp4.

       The video, audio and subtitle streams mapped to "out2.mkv" are entirely
       determined by automatic stream selection.

       out3.mkv consists of the cloned video output from the hue filter and
       the first audio stream from B.mp4.

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.

       Options that take arguments support a special syntax where the argument
       given on the command line is interpreted as a path to the file from
       which the actual argument value is loaded. To use this feature, add a
       forward slash '/' immediately before the option name (after the leading
       dash). E.g.

	       ffmpeg -i INPUT -/filter:v filter.script OUTPUT

       will load a filtergraph description from the file named filter.script.

   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. If stream_index is
	   used as an additional stream specifier (see below), then it selects
	   stream number stream_index from the matching streams. Stream
	   numbering is based on the order of the streams as detected by
	   libavformat except when a stream group specifier or program ID is
	   also specified. In this case it is based on the ordering of the
	   streams in the group or program.

       stream_type[:additional_stream_specifier]
	   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
	   additional_stream_specifier is used, then it matches streams which
	   both have this type and match the additional_stream_specifier.
	   Otherwise, it matches all streams of the specified type.

       g:group_specifier[:additional_stream_specifier]
	   Matches streams which are in the group with the specifier
	   group_specifier.  if additional_stream_specifier is used, then it
	   matches streams which both are part of the group and match the
	   additional_stream_specifier.  group_specifier may be one of the
	   following:

	   group_index
	       Match the stream with this group index.

	   #group_id or i:group_id
	       Match the stream with this group id.

       p:program_id[:additional_stream_specifier]
	   Matches streams which are in the program with the id program_id. If
	   additional_stream_specifier is used, then it matches streams which
	   both are part of the program and match the
	   additional_stream_specifier.

       #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 named filter_name.
	       Use the -filters option to get a list of all filters.

	   bsf=bitstream_filter_name
	       Print detailed information about the bitstream filter named
	       bitstream_filter_name.  Use the -bsfs option to get a list of
	       all bitstream filters.

	   protocol=protocol_name
	       Print detailed information about the protocol named
	       protocol_name.  Use the -protocols option to get a list of all
	       protocols.

       -version
	   Show version.

       -buildconf
	   Show the build configuration, one option per line.

       -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.

       -dispositions
	   Show stream dispositions.

       -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 [flags+]loglevel | -v [flags+]loglevel
	   Set logging level and flags used by the library.

	   The optional flags prefix can consist of the following values:

	   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.

	   level
	       Indicates that log output should add a "[level]" prefix to each
	       message line. This can be used as an alternative to log
	       coloring, e.g. when dumping the log to file.

	   Flags can also be used alone by adding a '+'/'-' prefix to
	   set/reset a single flag without affecting other flags or changing
	   loglevel. When setting both flags and loglevel, a '+' separator is
	   expected between the last flags value and before 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

	   For example to enable repeated log output, add the "level" prefix,
	   and set loglevel to "verbose":

		   ffmpeg -loglevel repeat+level+verbose -i input output

	   Another example that enables repeated log output without affecting
	   current state of "level" prefix flag or loglevel:

		   ffmpeg [...] -loglevel +repeat

	   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 can be forced setting the environment
	   variable AV_LOG_FORCE_COLOR.

       -report
	   Dump full command line and log 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 debug".

	   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
       -cpucount count (global)
	   Override detection of CPU count. This option is intended for
	   testing. Do not use it unless you know what you're doing.

		   ffmpeg -cpucount 2

       -max_alloc bytes
	   Set the maximum size limit for allocating a block on the heap by
	   ffmpeg's family of malloc functions. Exercise extreme caution when
	   using this option. Don't use if you do not understand the full
	   consequence of doing so.  Default is INT_MAX.

   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:

	       ffmpeg -i multichannel.mxf -map 0:v:0 -map 0:a:0 -map 0:a:0 -c:a:0 ac3 -b:a:0 640k -ac:a:1 2 -c:a:1 aac -b:2 128k out.mp4

       In the above example, a multichannel audio stream is mapped twice for
       output.	The first instance is encoded with codec ac3 and bitrate 640k.
       The second instance is downmixed to 2 channels and encoded with codec
       aac. A bitrate of 128k is specified for it using absolute index of the
       output stream.

       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.

       -recast_media (global)
	   Allow forcing a decoder of a different media type than the one
	   detected or designated by the demuxer. Useful for decoding media
	   data muxed as data streams.

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

       -isync input_index (input)
	   Assign an input as a sync source.

	   This will take the difference between the start times of the target
	   and reference inputs and offset the timestamps of the target file
	   by that difference. The source timestamps of the two inputs should
	   derive from the same clock source for expected results. If "copyts"
	   is set then "start_at_zero" must also be set. If either of the
	   inputs has no starting timestamp then no sync adjustment is made.

	   Acceptable values are those that refer to a valid ffmpeg input
	   index. If the sync reference is the target index itself or -1, then
	   no adjustment is made to target timestamps. A sync reference may
	   not itself be synced to any other input.

	   Default value is -1.

       -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.

       -itsscale scale (input,per-stream)
	   Rescale input timestamps. scale should be a floating point number.

       -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.

	   By default, the disposition is copied from the input stream, unless
	   the output stream this option applies to is fed by a complex
	   filtergraph - in that case the disposition is unset by default.

	   value is a sequence of items separated by '+' or '-'. The first
	   item may also be prefixed with '+' or '-', in which case this
	   option modifies the default value. Otherwise (the first item is not
	   prefixed) this options overrides the default value. A '+' prefix
	   adds the given disposition, '-' removes it. It is also possible to
	   clear the disposition by setting it to 0.

	   If no "-disposition" options were specified for an output file,
	   ffmpeg will automatically set the 'default' disposition on the
	   first stream of each type, when there are multiple streams of this
	   type in the output file and no stream of that type is already
	   marked as default.

	   The "-dispositions" option lists the known dispositions.

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

		   ffmpeg -i in.mkv -c copy -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 in.mkv -c copy -disposition:s:0 0 -disposition:s:1 default out.mkv

	   To add an embedded cover/thumbnail:

		   ffmpeg -i in.mp4 -i IMAGE -map 0 -map 1 -c copy -c:v:1 png -disposition:v:1 attached_pic out.mp4

	   Not all muxers support embedded thumbnails, and those who do, only
	   support a few formats, like JPEG or PNG.

       -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.

       -stream_group
       [map=input_file_id=stream_group][type=type:]st=stream[:st=stream][:stg=stream_group][:id=stream_group_id...]
       (output)
	   Creates a stream group of the specified type and stream_group_id,
	   or by mapping an input group, adding the specified stream(s) and/or
	   previously defined stream_group(s) to it.

	   type can be one of the following:

	   iamf_audio_element
	       Groups streams that belong to the same IAMF Audio Element

	       For this group type, the following options are available

	       audio_element_type
		   The Audio Element type. The following values are supported:

		   channel
		       Scalable channel audio representation

		   scene
		       Ambisonics representation

	       demixing
		   Demixing information used to reconstruct a scalable channel
		   audio representation.  This option must be separated from
		   the rest with a ',', and takes the following key=value
		   options

		   parameter_id
		       An identifier parameters blocks in frames may refer to

		   dmixp_mode
		       A pre-defined combination of demixing parameters

	       recon_gain
		   Recon gain information used to reconstruct a scalable
		   channel audio representation.  This option must be
		   separated from the rest with a ',', and takes the following
		   key=value options

		   parameter_id
		       An identifier parameters blocks in frames may refer to

	       layer
		   A layer defining a Channel Layout in the Audio Element.
		   This option must be separated from the rest with a ','.
		   Several ',' separated entries can be defined, and at least
		   one must be set.

		   It takes the following ":"-separated key=value options

		   ch_layout
		       The layer's channel layout

		   flags
		       The following flags are available:

		       recon_gain
			   Wether to signal if recon_gain is present as
			   metadata in parameter blocks within frames

		   output_gain
		   output_gain_flags
		       Which channels output_gain applies to. The following
		       flags are available:

		       FL
		       FR
		       BL
		       BR
		       TFL
		       TFR
		   ambisonics_mode
		       The ambisonics mode. This has no effect if
		       audio_element_type is set to channel.

		       The following values are supported:

		       mono
			   Each ambisonics channel is coded as an individual
			   mono stream in the group

	       default_w
		   Default weight value

	   iamf_mix_presentation
	       Groups streams that belong to all IAMF Audio Element the same
	       IAMF Mix Presentation references

	       For this group type, the following options are available

	       submix
		   A sub-mix within the Mix Presentation.  This option must be
		   separated from the rest with a ','. Several ',' separated
		   entries can be defined, and at least one must be set.

		   It takes the following ":"-separated key=value options

		   parameter_id
		       An identifier parameters blocks in frames may refer to,
		       for post-processing the mixed audio signal to generate
		       the audio signal for playback

		   parameter_rate
		       The sample rate duration fields in parameters blocks in
		       frames that refer to this parameter_id are expressed as

		   default_mix_gain
		       Default mix gain value to apply when there are no
		       parameter blocks sharing the same parameter_id for a
		       given frame

		   element
		       References an Audio Element used in this Mix
		       Presentation to generate the final output audio signal
		       for playback.  This option must be separated from the
		       rest with a '|'. Several '|' separated entries can be
		       defined, and at least one must be set.

		       It takes the following ":"-separated key=value options:

		       stg The stream_group_id for an Audio Element which this
			   sub-mix refers to

		       parameter_id
			   An identifier parameters blocks in frames may refer
			   to, for applying any processing to the referenced
			   and rendered Audio Element before being summed with
			   other processed Audio Elements

		       parameter_rate
			   The sample rate duration fields in parameters
			   blocks in frames that refer to this parameter_id
			   are expressed as

		       default_mix_gain
			   Default mix gain value to apply when there are no
			   parameter blocks sharing the same parameter_id for
			   a given frame

		       annotations
			   A key=value string describing the sub-mix element
			   where "key" is a string conforming to BCP-47 that
			   specifies the language for the "value" string.
			   "key" must be the same as the one in the mix's
			   annotations

		       headphones_rendering_mode
			   Indicates whether the input channel-based Audio
			   Element is rendered to stereo loudspeakers or
			   spatialized with a binaural renderer when played
			   back on headphones.	This has no effect if the
			   referenced Audio Element's audio_element_type is
			   set to channel.

			   The following values are supported:

			   stereo
			   binaural
		   layout
		       Specifies the layouts for this sub-mix on which the
		       loudness information was measured.  This option must be
		       separated from the rest with a '|'. Several '|'
		       separated entries can be defined, and at least one must
		       be set.

		       It takes the following ":"-separated key=value options:

		       layout_type
			   loudspeakers
			       The layout follows the loudspeaker sound system
			       convention of ITU-2051-3.

			   binaural
			       The layout is binaural.

		       sound_system
			   Channel layout matching one of Sound Systems A to J
			   of ITU-2051-3, plus 7.1.2 and 3.1.2 This has no
			   effect if layout_type is set to binaural.

		       integrated_loudness
			   The program integrated loudness information, as
			   defined in ITU-1770-4.

		       digital_peak
			   The digital (sampled) peak value of the audio
			   signal, as defined in ITU-1770-4.

		       true_peak
			   The true peak of the audio signal, as defined in
			   ITU-1770-4.

		       dialog_anchored_loudness
			   The Dialogue loudness information, as defined in
			   ITU-1770-4.

		       album_anchored_loudness
			   The Album loudness information, as defined in
			   ITU-1770-4.

	       annotations
		   A key=value string string describing the mix where "key" is
		   a string conforming to BCP-47 that specifies the language
		   for the "value" string. "key" must be the same as the ones
		   in all sub-mix element's annotationss

	   E.g. to create an scalable 5.1 IAMF file from several WAV input
	   files

		   ffmpeg -i front.wav -i back.wav -i center.wav -i lfe.wav
		   -map 0:0 -map 1:0 -map 2:0 -map 3:0 -c:a opus
		   -stream_group type=iamf_audio_element:id=1:st=0:st=1:st=2:st=3,
		   demixing=parameter_id=998,
		   recon_gain=parameter_id=101,
		   layer=ch_layout=stereo,
		   layer=ch_layout=5.1,
		   -stream_group type=iamf_mix_presentation:id=2:stg=0:annotations=en-us=Mix_Presentation,
		   submix=parameter_id=100:parameter_rate=48000|element=stg=0:parameter_id=100:annotations=en-us=Scalable_Submix|layout=sound_system=stereo|layout=sound_system=5.1
		   -streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.iamf

	   To copy the two stream groups (Audio Element and Mix Presentation)
	   from an input IAMF file with four streams into an mp4 output

		   ffmpeg -i input.iamf -c:a copy -stream_group map=0=0:st=0:st=1:st=2:st=3 -stream_group map=0=1:stg=0
		   -streamid 0:0 -streamid 1:1 -streamid 2:2 -streamid 3:3 output.mp4

       -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

	   The parameters set for each target are as follows.

	   VCD

		   <pal>:
		   -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
		   -s 352x288 -r 25
		   -codec:v mpeg1video -g 15 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
		   -ar 44100 -ac 2
		   -codec:a mp2 -b:a 224k

		   <ntsc>:
		   -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
		   -s 352x240 -r 30000/1001
		   -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
		   -ar 44100 -ac 2
		   -codec:a mp2 -b:a 224k

		   <film>:
		   -f vcd -muxrate 1411200 -muxpreload 0.44 -packetsize 2324
		   -s 352x240 -r 24000/1001
		   -codec:v mpeg1video -g 18 -b:v 1150k -maxrate:v 1150k -minrate:v 1150k -bufsize:v 327680
		   -ar 44100 -ac 2
		   -codec:a mp2 -b:a 224k

	   SVCD

		   <pal>:
		   -f svcd -packetsize 2324
		   -s 480x576 -pix_fmt yuv420p -r 25
		   -codec:v mpeg2video -g 15 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
		   -ar 44100
		   -codec:a mp2 -b:a 224k

		   <ntsc>:
		   -f svcd -packetsize 2324
		   -s 480x480 -pix_fmt yuv420p -r 30000/1001
		   -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
		   -ar 44100
		   -codec:a mp2 -b:a 224k

		   <film>:
		   -f svcd -packetsize 2324
		   -s 480x480 -pix_fmt yuv420p -r 24000/1001
		   -codec:v mpeg2video -g 18 -b:v 2040k -maxrate:v 2516k -minrate:v 0 -bufsize:v 1835008 -scan_offset 1
		   -ar 44100
		   -codec:a mp2 -b:a 224k

	   DVD

		   <pal>:
		   -f dvd -muxrate 10080k -packetsize 2048
		   -s 720x576 -pix_fmt yuv420p -r 25
		   -codec:v mpeg2video -g 15 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
		   -ar 48000
		   -codec:a ac3 -b:a 448k

		   <ntsc>:
		   -f dvd -muxrate 10080k -packetsize 2048
		   -s 720x480 -pix_fmt yuv420p -r 30000/1001
		   -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
		   -ar 48000
		   -codec:a ac3 -b:a 448k

		   <film>:
		   -f dvd -muxrate 10080k -packetsize 2048
		   -s 720x480 -pix_fmt yuv420p -r 24000/1001
		   -codec:v mpeg2video -g 18 -b:v 6000k -maxrate:v 9000k -minrate:v 0 -bufsize:v 1835008
		   -ar 48000
		   -codec:a ac3 -b:a 448k

	   DV

		   <pal>:
		   -f dv
		   -s 720x576 -pix_fmt yuv420p -r 25
		   -ar 48000 -ac 2

		   <ntsc>:
		   -f dv
		   -s 720x480 -pix_fmt yuv411p -r 30000/1001
		   -ar 48000 -ac 2

		   <film>:
		   -f dv
		   -s 720x480 -pix_fmt yuv411p -r 24000/1001
		   -ar 48000 -ac 2

	   The "dv50" target is identical to the "dv" target except that the
	   pixel format set is "yuv422p" for all three standards.

	   Any user-set value for a parameter above will override the target
	   preset value. In that case, the output may not comply with the
	   target standard.

       -dn (input/output)
	   As an input option, blocks all data streams of a file from being
	   filtered or being automatically selected or mapped for any output.
	   See "-discard" option to disable streams individually.

	   As an output option, disables data recording i.e. automatic
	   selection or mapping of any data stream. For full manual control
	   see the "-map" option.

       -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.

       -reinit_filter[:stream_specifier] integer (input,per-stream)
	   This boolean option determines if the filtergraph(s) to which this
	   stream is fed gets reinitialized when input frame parameters change
	   mid-stream. This option is enabled by default as most video and all
	   audio filters cannot handle deviation in input frame properties.
	   Upon reinitialization, existing filter state is lost, like e.g. the
	   frame count "n" reference available in some filters. Any frames
	   buffered at time of reinitialization are lost.  The properties
	   where a change triggers reinitialization are, for video, frame
	   resolution or pixel format; for audio, sample format, sample rate,
	   channel count or channel layout.

       -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".

       -stats_period time (global)
	   Set period at which encoding progress/statistics are updated.
	   Default is 0.5 seconds.

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

	   Progress information is written periodically 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".

	   The update period is set using "-stats_period".

       -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/latency 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.

   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:

	   video encoding
	       Duplicate or drop frames right before encoding them to achieve
	       constant output frame rate fps.

	   video streamcopy
	       Indicate to the muxer that fps is the stream frame rate. No
	       data is dropped or duplicated in this case. This may produce
	       invalid files if fps does not match the actual stream frame
	       rate as determined by packet timestamps.  See also the "setts"
	       bitstream filter.

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

	   Clamps output frame rate when output framerate is auto-set and is
	   higher than this value.  Useful in batch processing or when input
	   framerate is wrongly detected as very high.	It cannot be set
	   together with "-r". It is ignored during streamcopy.

       -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.

       -display_rotation[:stream_specifier] rotation (input,per-stream)
	   Set video rotation metadata.

	   rotation is a decimal number specifying the amount in degree by
	   which the video should be rotated counter-clockwise before being
	   displayed.

	   This option overrides the rotation/display transform metadata
	   stored in the file, if any. When the video is being transcoded
	   (rather than copied) and "-autorotate" is enabled, the video will
	   be rotated at the filtering stage. Otherwise, the metadata will be
	   written into the output file if the muxer supports it.

	   If the "-display_hflip" and/or "-display_vflip" options are given,
	   they are applied after the rotation specified by this option.

       -display_hflip[:stream_specifier] (input,per-stream)
	   Set whether on display the image should be horizontally flipped.

	   See the "-display_rotation" option for more details.

       -display_vflip[:stream_specifier] (input,per-stream)
	   Set whether on display the image should be vertically flipped.

	   See the "-display_rotation" option for more details.

       -vn (input/output)
	   As an input option, blocks all video streams of a file from being
	   filtered or being automatically selected or mapped for any output.
	   See "-discard" option to disable streams individually.

	   As an output option, disables video recording i.e. automatic
	   selection or mapping of any video stream. For full manual control
	   see the "-map" option.

       -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.

       -autorotate
	   Automatically rotate the video according to file metadata. Enabled
	   by default, use -noautorotate to disable it.

       -autoscale
	   Automatically scale the video according to the resolution of first
	   frame.  Enabled by default, use -noautoscale to disable it. When
	   autoscale is disabled, all output frames of filter graph might not
	   be in the same resolution and may be inadequate for some
	   encoder/muxer. Therefore, it is not recommended to disable it
	   unless you really know what you are doing.  Disable autoscale at
	   your own risk.

   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 default flags for the libswscale library. These flags are used
	   by automatically inserted "scale" filters and those within simple
	   filtergraphs, if not overridden within the filtergraph definition.

	   See the ffmpeg-scaler manual for a list of scaler options.

       -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.

       -vstats
	   Dump video coding statistics to vstats_HHMMSS.log. See the vstats
	   file format section for the format description.

       -vstats_file file
	   Dump video coding statistics to file. See the vstats file format
	   section for the format description.

       -vstats_version file
	   Specify which version of the vstats format to use. Default is 2.
	   See the vstats file format section for the format description.

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

       -force_key_frames[:stream_specifier] time[,time...] (output,per-stream)
       -force_key_frames[:stream_specifier] expr:expr (output,per-stream)
       -force_key_frames[:stream_specifier] source (output,per-stream)
	   force_key_frames can take arguments of the following form:

	   time[,time...]
	       If the argument consists of timestamps, ffmpeg will round the
	       specified times to the nearest output timestamp as per the
	       encoder time base and force a keyframe at the first frame
	       having timestamp equal or greater than the computed timestamp.
	       Note that if the encoder time base is too coarse, then the
	       keyframes may be forced on frames with timestamps lower than
	       the specified time.  The default encoder time base is the
	       inverse of the output framerate but may be set otherwise via
	       "-enc_time_base".

	       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

	   expr:expr
	       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.

	       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))

	   source
	       If the argument is "source", ffmpeg will force a key frame if
	       the current frame being encoded is marked as a key frame in its
	       source.	In cases where this particular source frame has to be
	       dropped, enforce the next available frame to become a key frame
	       instead.

	   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.

	       The following options are recognized:

	       primary_ctx
		   If set to 1, uses the primary device context instead of
		   creating a new one.

	       Examples:

	       -init_hw_device cuda:1
		   Choose the second device on the system.

	       -init_hw_device cuda:0,primary_ctx=1
		   Choose the first device and use the primary device context.

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

	   d3d11va
	       device is the number of the Direct3D 11 display adapter.  If
	       not specified, it will attempt to use the default Direct3D 11
	       display adapter or the first Direct3D 11 display adapter whose
	       hardware VendorId is specified by vendor_id.

	       Examples:

	       -init_hw_device d3d11va
		   Create a d3d11va device on the default Direct3D 11 display
		   adapter.

	       -init_hw_device d3d11va:1
		   Create a d3d11va device on the Direct3D 11 display adapter
		   specified by index 1.

	       -init_hw_device d3d11va:,vendor_id=0x8086
		   Create a d3d11va device on the first Direct3D 11 display
		   adapter whose hardware VendorId is 0x8086.

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

	       The following options are recognized:

	       kernel_driver
		   When device is not specified, use this option to specify
		   the name of the kernel driver associated with the desired
		   device. This option is available only when the hardware
		   acceleration method drm and vaapi are enabled.

	       Examples:

	       -init_hw_device vaapi
		   Create a vaapi device on the default device.

	       -init_hw_device vaapi:/dev/dri/renderD129
		   Create a vaapi device on DRM render node
		   /dev/dri/renderD129.

	       -init_hw_device vaapi:1
		   Create a vaapi device on DirectX adapter 1.

	       -init_hw_device vaapi:,kernel_driver=i915
		   Create a vaapi device on a device associated with kernel
		   driver i915.

	   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 d3d11va or vaapi) and
	       then deriving a QSV device from that.)

	       The following options are recognized:

	       child_device
		   Specify a DRM render node on Linux or DirectX adapter on
		   Windows.

	       child_device_type
		   Choose platform-appropriate subdevice type. On Windows
		   d3d11va is used as default subdevice type when
		   "--enable-libvpl" is specified at configuration time, dxva2
		   is used as default subdevice type when "--enable-libmfx" is
		   specified at configuration time. On Linux user can use
		   vaapi only as subdevice type.

	       Examples:

	       -init_hw_device qsv:hw,child_device=/dev/dri/renderD129
		   Create a QSV device with MFX_IMPL_HARDWARE on DRM render
		   node /dev/dri/renderD129.

	       -init_hw_device qsv:hw,child_device=1
		   Create a QSV device with MFX_IMPL_HARDWARE on DirectX
		   adapter 1.

	       -init_hw_device qsv:hw,child_device_type=d3d11va
		   Choose the GPU subdevice with type d3d11va and create QSV
		   device with MFX_IMPL_HARDWARE.

	       -init_hw_device qsv:hw,child_device_type=dxva2
		   Choose the GPU subdevice with type dxva2 and create QSV
		   device with MFX_IMPL_HARDWARE.

	       -init_hw_device qsv:hw,child_device=1,child_device_type=d3d11va
		   Create a QSV device with MFX_IMPL_HARDWARE on DirectX
		   adapter 1 with subdevice type d3d11va.

	       -init_hw_device vaapi=va:/dev/dri/renderD129 -init_hw_device
	       qsv=hw1@va
		   Create a VAAPI device called va on /dev/dri/renderD129,
		   then derive a QSV device called hw1 from device va.

	   opencl
	       device selects the platform and device as
	       platform_index.device_index.

	       The set of devices can also be filtered using the key-value
	       pairs to find only devices matching particular platform or
	       device strings.

	       The strings usable as filters are:

	       platform_profile
	       platform_version
	       platform_name
	       platform_vendor
	       platform_extensions
	       device_name
	       device_vendor
	       driver_version
	       device_version
	       device_profile
	       device_extensions
	       device_type

	       The indices and filters must together uniquely select a device.

	       Examples:

	       -init_hw_device opencl:0.1
		   Choose the second device on the first platform.

	       -init_hw_device opencl:,device_name=Foo9000
		   Choose the device with a name containing the string
		   Foo9000.

	       -init_hw_device
	       opencl:1,device_type=gpu,device_extensions=cl_khr_fp16
		   Choose the GPU device on the second platform supporting the
		   cl_khr_fp16 extension.

	   vulkan
	       If device is an integer, it selects the device by its index in
	       a system-dependent list of devices.  If device is any other
	       string, it selects the first device with a name containing that
	       string as a substring.

	       The following options are recognized:

	       debug
		   If set to 1, enables the validation layer, if installed.

	       linear_images
		   If set to 1, images allocated by the hwcontext will be
		   linear and locally mappable.

	       instance_extensions
		   A plus separated list of additional instance extensions to
		   enable.

	       device_extensions
		   A plus separated list of additional device extensions to
		   enable.

	       Examples:

	       -init_hw_device vulkan:1
		   Choose the second device on the system.

	       -init_hw_device vulkan:RADV
		   Choose the first device with a name containing the string
		   RADV.

	       -init_hw_device
	       vulkan:0,instance_extensions=VK_KHR_wayland_surface+VK_KHR_xcb_surface
		   Choose the first device and enable the Wayland and XCB
		   instance extensions.

       -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.

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

	   dxva2
	       Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

	   d3d11va
	       Use D3D11VA (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 components enabled in this build of
	   ffmpeg.  Actual runtime availability depends on the hardware and
	   its suitable driver being installed.

       -fix_sub_duration_heartbeat[:stream_specifier]
	   Set a specific output video stream as the heartbeat stream
	   according to which to split and push through currently in-progress
	   subtitle upon receipt of a random access packet.

	   This lowers the latency of subtitles for which the end packet or
	   the following subtitle has not yet been received. As a drawback,
	   this will most likely lead to duplication of subtitle events in
	   order to cover the full duration, so when dealing with use cases
	   where latency of when the subtitle event is passed on to output is
	   not relevant this option should not be utilized.

	   Requires -fix_sub_duration to be set for the relevant input
	   subtitle stream for this to have any effect, as well as for the
	   input subtitle stream having to be directly mapped to the same
	   output in which the heartbeat stream resides.

   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 (input/output)
	   As an input option, blocks all audio streams of a file from being
	   filtered or being automatically selected or mapped for any output.
	   See "-discard" option to disable streams individually.

	   As an output option, disables audio recording i.e. automatic
	   selection or mapping of any audio stream. For full manual control
	   see the "-map" option.

       -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".

       -ch_layout[:stream_specifier] layout (input/output,per-stream)
	   Alias for "-channel_layout".

       -channel_layout[:stream_specifier] layout (input/output,per-stream)
	   Set the audio channel layout. For output streams it is set by
	   default to the input channel layout. For input streams it overrides
	   the channel layout of the input. Not all decoders respect the
	   overridden channel layout. This option also sets the channel layout
	   for audio grabbing devices and raw demuxers and is mapped to the
	   corresponding demuxer option.

       -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. Using the
	   "-channel_layout" option to explicitly specify an input layout also
	   disables guessing.

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

       -sn (input/output)
	   As an input option, blocks all subtitle streams of a file from
	   being filtered or being automatically selected or mapped for any
	   output. See "-discard" option to disable streams individually.

	   As an output option, disables subtitle recording i.e. automatic
	   selection or mapping of any subtitle stream. For full manual
	   control see the "-map" option.

   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][?] | [linklabel] (output)
	   Create one or more streams in the output file. This option has two
	   forms for specifying the data source(s): the first selects one or
	   more streams from some input file (specified with "-i"), the second
	   takes an output from some complex filtergraph (specified with
	   "-filter_complex").

	   In the first form, an output stream is created for every stream
	   from the input file with the index input_file_id. If
	   stream_specifier is given, only those streams that match the
	   specifier are used (see the Stream specifiers section for the
	   stream_specifier syntax).

	   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.

	   This option may be specified multiple times, each adding more
	   streams to the output file. Any given input stream may also be
	   mapped any number of times as a source for different output
	   streams, e.g. in order to use different encoding options and/or
	   filters. The streams are created in the output in the same order in
	   which the "-map" options are given on the commandline.

	   Using this option disables the default mappings for this output
	   file.

	   Examples:

	   map everything
	       To map ALL streams from the first input file to output

		       ffmpeg -i INPUT -map 0 output

	   select specific stream
	       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 second input stream in INPUT to the (single)
	       output stream in out.wav.

	   create multiple streams
	       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

	   create multiple streams 2
	       To select all video and the third audio stream from an input
	       file:

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

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

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

	   optional map
	       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

	   map by language
	       To pick the English audio stream:

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

       -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_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 real,
	   system and user 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 real,
	   system and user time used in various steps (audio/video
	   encode/decode).

       -timelimit duration (global)
	   Exit after ffmpeg has been running for duration seconds in CPU user
	   time.

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

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

       -readrate speed (input)
	   Limit input read speed.

	   Its value is a floating-point positive number which represents the
	   maximum duration of media, in seconds, that should be ingested in
	   one second of wallclock time.  Default value is zero and represents
	   no imposed limitation on speed of ingestion.  Value 1 represents
	   real-time speed and is equivalent to "-re".

	   Mainly used to simulate a capture device or live input stream (e.g.
	   when reading from a file).  Should not be used with a low value
	   when input is an actual capture device or live stream as it may
	   cause packet loss.

	   It is useful for when flow speed of output packets is important,
	   such as live streaming.

       -re (input)
	   Read input at native frame rate. This is equivalent to setting
	   "-readrate 1".

       -readrate_initial_burst seconds
	   Set an initial read burst time, in seconds, after which
	   -re/-readrate will be enforced.

       -vsync parameter (global)
       -fps_mode[:stream_specifier] parameter (output,per-stream)
	   Set video sync method / framerate mode. vsync is applied to all
	   output video streams but can be overridden for a stream by setting
	   fps_mode. vsync is deprecated and will be removed in the future.

	   For compatibility reasons some of the values for vsync can be
	   specified as numbers (shown in parentheses in the following table).

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

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

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

	   auto (-1)
	       Chooses between cfr and vfr 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.

       -apad parameters (output,per-stream)
	   Pad the output audio stream(s). This is the same as applying "-af
	   apad".  Argument is a string of filter parameters composed the same
	   as with the "apad" filter.  "-shortest" must be set for this output
	   for the option to take effect.

       -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 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.

	   demux
	       Use the timebase from the demuxer.

	   filter
	       Use the timebase from the filtergraph.

	   a positive number
	       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 decimal number (e.g. 0.04166, 2.0833e-5)

	   Default value is 0.

       -bitexact (input/output)
	   Enable bitexact mode for (de)muxer and (de/en)coder

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

	   Note that this option may require buffering frames, which
	   introduces extra latency. The maximum amount of this latency may be
	   controlled with the "-shortest_buf_duration" option.

       -shortest_buf_duration duration (output)
	   The "-shortest" option may require buffering potentially large
	   amounts of data when at least one of the streams is "sparse" (i.e.
	   has large gaps between frames X this is typically the case for
	   subtitles).

	   This option controls the maximum duration of buffered frames in
	   seconds.  Larger values may allow the "-shortest" option to produce
	   more accurate results, but increase memory use and latency.

	   The default value is 10 seconds.

       -dts_delta_threshold threshold
	   Timestamp discontinuity delta threshold, expressed as a decimal
	   number of seconds.

	   The timestamp discontinuity correction enabled by this option is
	   only applied to input formats accepting timestamp discontinuity
	   (for which the "AVFMT_TS_DISCONT" flag is enabled), e.g. MPEG-TS
	   and HLS, and is automatically disabled when employing the "-copyts"
	   option (unless wrapping is detected).

	   If a timestamp discontinuity is detected whose absolute value is
	   greater than threshold, ffmpeg will remove the discontinuity by
	   decreasing/increasing the current DTS and PTS by the corresponding
	   delta value.

	   The default value is 10.

       -dts_error_threshold threshold
	   Timestamp error delta threshold, expressed as a decimal number of
	   seconds.

	   The timestamp correction enabled by this option is only applied to
	   input formats not accepting timestamp discontinuity (for which the
	   "AVFMT_TS_DISCONT" flag is not enabled).

	   If a timestamp discontinuity is detected whose absolute value is
	   greater than threshold, ffmpeg will drop the PTS/DTS timestamp
	   value.

	   The default value is "3600*30" (30 hours), which is arbitrarily
	   picked and quite conservative.

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

       -muxpreload seconds (output)
	   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 (input/output,per-stream)
	   Apply bitstream filters to matching streams. The filters are
	   applied to each packet as it is received from the demuxer (when
	   used as an input option) or before it is sent to the muxer (when
	   used as an output option).

	   bitstream_filters is a comma-separated list of bitstream filter
	   specifications, each of the form

		   <filter>[=<optname0>=<optval0>:<optname1>=<optval1>:...]

	   Any of the ',=:' characters that are to be a part of an option
	   value need to be escaped with a backslash.

	   Use the "-bsfs" option to get the list of bitstream filters.

	   E.g.

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

	   applies the "h264_mp4toannexb" bitstream filter (which converts
	   MP4-encapsulated H.264 stream to Annex B) to the input video
	   stream.

	   On the other hand,

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

	   applies the "mov2textsub" bitstream filter (which extracts text
	   from MOV subtitles) to the output subtitle stream. Note, however,
	   that since both examples use "-c copy", it matters little whether
	   the filters are applied on input or output - that would change if
	   transcoding was happening.

       -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.
	   This option may be specified multiple times - each use creates a
	   new complex filtergraph.

	   Inputs to a complex filtergraph may come from different source
	   types, distinguished by the format of the corresponding link label:

	   o   To connect an input stream, use "[file_index:stream_specifier]"
	       (i.e. the same syntax as -map). If stream_specifier matches
	       multiple streams, the first one will be used.

	   o   To connect a loopback decoder use [dec:dec_idx], where dec_idx
	       is the index of the loopback decoder to be connected to given
	       input.

	   o   To connect an output from another complex filtergraph, use its
	       link label. E.g the following example:

		       ffmpeg -i input.mkv \
			 -filter_complex '[0:v]scale=size=hd1080,split=outputs=2[for_enc][orig_scaled]' \
			 -c:v libx264 -map '[for_enc]' output.mkv \
			 -dec 0:0 \
			 -filter_complex '[dec:0][orig_scaled]hstack[stacked]' \
			 -map '[stacked]' -c:v ffv1 comparison.mkv

	       reads an input video and

	       o   (line 2) uses a complex filtergraph with one input and two
		   outputs to scale the video to 1920x1080 and duplicate the
		   result to both outputs;

	       o   (line 3) encodes one scaled output with "libx264" and
		   writes the result to output.mkv;

	       o   (line 4) decodes this encoded stream with a loopback
		   decoder;

	       o   (line 5) places the output of the loopback decoder (i.e.
		   the "libx264"-encoded video) side by side with the scaled
		   original input;

	       o   (line 6) combined video is then losslessly encoded and
		   written into comparison.mkv.

	       Note that the two filtergraphs cannot be combined into one,
	       because then there would be a cycle in the transcoding pipeline
	       (filtergraph output goes to encoding, from there to decoding,
	       then back to the same graph), and such cycles are not allowed.

	   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

	   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)

	   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.

       -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/output)
	   For 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; setting this value can force ffmpeg to use a
	   separate input thread and read packets as soon as they arrive. By
	   default ffmpeg only does this if multiple inputs are specified.

	   For output, this option specified the maximum number of packets
	   that may be queued to each muxing thread.

       -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 from streams.  Any
	   input stream can be fully discarded, using value "all" whereas
	   selective discarding of frames from a stream occurs at the demuxer
	   and is not supported by all demuxers.

	   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.

	   empty_output_stream
	       No packets were passed to the muxer in some of the output
	       streams.

       -max_error_rate (global)
	   Set fraction of decoding frame failures across all inputs which
	   when crossed ffmpeg will return exit code 69. Crossing this
	   threshold does not terminate processing. Range is a floating-point
	   number between 0 to 1. Default is 2/3.

       -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.

       -muxing_queue_data_threshold bytes (output,per-stream)
	   This is a minimum threshold until which the muxing queue size is
	   not taken into account. Defaults to 50 megabytes per stream, and is
	   based on the overall size of packets passed to the muxer.

       -auto_conversion_filters (global)
	   Enable automatically inserting format conversion filters in all
	   filter graphs, including those defined by -vf, -af, -filter_complex
	   and -lavfi. If filter format negotiation requires a conversion, the
	   initialization of the filters will fail.  Conversions can still be
	   performed by inserting the relevant conversion filter (scale,
	   aresample) in the graph.  On by default, to explicitly disable it
	   you need to specify "-noauto_conversion_filters".

       -bits_per_raw_sample[:stream_specifier] value (output,per-stream)
	   Declare the number of bits per raw sample in the given output
	   stream to be value. Note that this option sets the information
	   provided to the encoder/muxer, it does not change the stream to
	   conform to this value. Setting values that do not match the stream
	   properties may result in encoding failures or invalid output files.

       -stats_enc_pre[:stream_specifier] path (output,per-stream)
       -stats_enc_post[:stream_specifier] path (output,per-stream)
       -stats_mux_pre[:stream_specifier] path (output,per-stream)
	   Write per-frame encoding information about the matching streams
	   into the file given by path.

	   -stats_enc_pre writes information about raw video or audio frames
	   right before they are sent for encoding, while -stats_enc_post
	   writes information about encoded packets as they are received from
	   the encoder.  -stats_mux_pre writes information about packets just
	   as they are about to be sent to the muxer. Every frame or packet
	   produces one line in the specified file. The format of this line is
	   controlled by -stats_enc_pre_fmt / -stats_enc_post_fmt /
	   -stats_mux_pre_fmt.

	   When stats for multiple streams are written into a single file, the
	   lines corresponding to different streams will be interleaved. The
	   precise order of this interleaving is not specified and not
	   guaranteed to remain stable between different invocations of the
	   program, even with the same options.

       -stats_enc_pre_fmt[:stream_specifier] format_spec (output,per-stream)
       -stats_enc_post_fmt[:stream_specifier] format_spec (output,per-stream)
       -stats_mux_pre_fmt[:stream_specifier] format_spec (output,per-stream)
	   Specify the format for the lines written with -stats_enc_pre /
	   -stats_enc_post / -stats_mux_pre.

	   format_spec is a string that may contain directives of the form
	   {fmt}. format_spec is backslash-escaped --- use \{, \}, and \\ to
	   write a literal {, }, or \, respectively, into the output.

	   The directives given with fmt may be one of the following:

	   fidx
	       Index of the output file.

	   sidx
	       Index of the output stream in the file.

	   n   Frame number. Pre-encoding: number of frames sent to the
	       encoder so far.	Post-encoding: number of packets received from
	       the encoder so far.  Muxing: number of packets submitted to the
	       muxer for this stream so far.

	   ni  Input frame number. Index of the input frame (i.e. output by a
	       decoder) that corresponds to this output frame or packet. -1 if
	       unavailable.

	   tb  Timebase in which this frame/packet's timestamps are expressed,
	       as a rational number num/den. Note that encoder and muxer may
	       use different timebases.

	   tbi Timebase for ptsi, as a rational number num/den. Available when
	       ptsi is available, 0/1 otherwise.

	   pts Presentation timestamp of the frame or packet, as an integer.
	       Should be multiplied by the timebase to compute presentation
	       time.

	   ptsi
	       Presentation timestamp of the input frame (see ni), as an
	       integer. Should be multiplied by tbi to compute presentation
	       time. Printed as (2^63 - 1 = 9223372036854775807) when not
	       available.

	   t   Presentation time of the frame or packet, as a decimal number.
	       Equal to pts multiplied by tb.

	   ti  Presentation time of the input frame (see ni), as a decimal
	       number. Equal to ptsi multiplied by tbi. Printed as inf when
	       not available.

	   dts (packet)
	       Decoding timestamp of the packet, as an integer. Should be
	       multiplied by the timebase to compute presentation time.

	   dt (packet)
	       Decoding time of the frame or packet, as a decimal number.
	       Equal to dts multiplied by tb.

	   sn (frame,audio)
	       Number of audio samples sent to the encoder so far.

	   samp (frame,audio)
	       Number of audio samples in the frame.

	   size (packet)
	       Size of the encoded packet in bytes.

	   br (packet)
	       Current bitrate in bits per second.

	   abr (packet)
	       Average bitrate for the whole stream so far, in bits per
	       second, -1 if it cannot be determined at this point.

	   key (packet)
	       Character 'K' if the packet contains a keyframe, character 'N'
	       otherwise.

	   Directives tagged with packet may only be used with
	   -stats_enc_post_fmt and -stats_mux_pre_fmt.

	   Directives tagged with frame may only be used with
	   -stats_enc_pre_fmt.

	   Directives tagged with audio may only be used with audio streams.

	   The default format strings are:

	   pre-encoding
	       {fidx} {sidx} {n} {t}

	   post-encoding
	       {fidx} {sidx} {n} {t}

	   In the future, new items may be added to the end of the default
	   formatting strings. Users who depend on the format staying exactly
	   the same, should prescribe it manually.

	   Note that stats for different streams written into the same file
	   may have different formats.

   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.

   vstats file format
       The "-vstats" and "-vstats_file" options enable generation of a file
       containing statistics about the generated video outputs.

       The "-vstats_version" option controls the format version of the
       generated file.

       With version 1 the format is:

	       frame= <FRAME> q= <FRAME_QUALITY> PSNR= <PSNR> f_size= <FRAME_SIZE> s_size= <STREAM_SIZE>kB time= <TIMESTAMP> br= <BITRATE>kbits/s avg_br= <AVERAGE_BITRATE>kbits/s

       With version 2 the format is:

	       out= <OUT_FILE_INDEX> st= <OUT_FILE_STREAM_INDEX> frame= <FRAME_NUMBER> q= <FRAME_QUALITY>f PSNR= <PSNR> f_size= <FRAME_SIZE> s_size= <STREAM_SIZE>kB time= <TIMESTAMP> br= <BITRATE>kbits/s avg_br= <AVERAGE_BITRATE>kbits/s

       The value corresponding to each key is described below:

       avg_br
	   average bitrate expressed in Kbits/s

       br  bitrate expressed in Kbits/s

       frame
	   number of encoded frame

       out out file index

       PSNR
	   Peak Signal to Noise Ratio

       q   quality of the frame

       f_size
	   encoded packet size expressed as number of bytes

       s_size
	   stream size expressed in KiB

       st  out file stream index

       time
	   time of the packet

       type
	   picture type

       See also the -stats_enc options for an alternative way to show encoding
       statistics.

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:

       o   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.

       o   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.

       o   You can output to a raw YUV420P file:

		   ffmpeg -i mydivx.avi hugefile.yuv

       o   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.

       o   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.

       o   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.

       o   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".

       o   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

       o   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.

       o   To force CBR video output:

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

       o   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:

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

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

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

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

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

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

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

       Examples

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

		   Crime d\'Amour

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

		   'Crime d'\''Amour'

       o   Include leading or trailing whitespaces using quoting:

		   '  this string starts and ends with whitespaces  '

       o   Escaping and quoting can be mixed together:

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

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

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

   Date
       The accepted syntax is:

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

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

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

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

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

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

       or

	       [-]<S>+[.<m>...][s|ms|us]

       S expresses the number of seconds, with the optional decimal part m.
       The optional literal suffixes s, ms or us indicate to interpret the
       value as seconds, milliseconds or microseconds, respectively.

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

       0.2 0.2 seconds

       200ms
	   200 milliseconds, that's 0.2s

       200000us
	   200000 microseconds, that's 0.2s

       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

       3.1.2
	   FL+FR+FC+LFE+TFL+TFR

       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

       5.1.2
	   FL+FR+FC+LFE+BL+BR+TFL+TFR

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

       cube
	   FL+FR+BL+BR+TFL+TFR+TBL+TBR

       5.1.4
	   FL+FR+FC+LFE+BL+BR+TFL+TFR+TBL+TBR

       7.1.2
	   FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR

       7.1.4
	   FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBL+TBR

       7.2.3
	   FL+FR+FC+LFE+BL+BR+SL+SR+TFL+TFR+TBC+LFE2

       9.1.4
	   FL+FR+FC+LFE+BL+BR+FLC+FRC+SL+SR+TFL+TFR+TBL+TBR

       hexadecagonal
	   FL+FR+FC+BL+BR+BC+SL+SR+WL+WR+TBL+TBR+TBC+TFC+TFL+TFR

       downmix
	   DL+DR

       22.2
	   FL+FR+FC+LFE+BL+BR+FLC+FRC+BC+SL+SR+TC+TFL+TFC+TFR+TBL+TBC+TBR+LFE2+TSL+TSR+BFC+BFL+BFR

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

       o   the name of a single channel (e.g. FL, FR, FC, LFE, etc.), each
	   optionally containing a custom name after a '@', (e.g. FL@Left,
	   FR@Right, FC@Center, LFE@Low_Frequency, etc.)

       A standard channel layout can be specified by the following:

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

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

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

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

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

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

       See also the function "av_channel_layout_from_string" 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: "+", "-".

       Some internal variables can be used to store and load intermediary
       results. They can be accessed using the "ld" and "st" functions with an
       index argument varying from 0 to 9 to specify which internal variable
       to access.

       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(y, x)
	   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(idx)
	   Load the value of the internal variable with index idx, which was
	   previously stored with st(idx, 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.	Return the value of
	   the expression printed.

       random(idx)
	   Return a pseudo random value between 0.0 and 1.0. idx is the index
	   of the internal variable used to save the seed/state, which can be
	   previously stored with "st(idx)".

	   To initialize the seed, you need to store the seed value as a
	   64-bit unsigned integer in the internal variable with index idx.

	   For example, to store the seed with value 42 in the internal
	   variable with index 0 and print a few random values:

		   st(0,42); print(random(0)); print(random(0)); print(random(0))

       randomi(idx, min, max)
	   Return a pseudo random value in the interval between min and max.
	   idx is the index of the internal variable which will be used to
	   save the seed/state, which can be previously stored with "st(idx)".

	   To initialize the seed, you need to store the seed value as a
	   64-bit unsigned integer in the internal variable with index idx.

       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".

       sgn(x)
	   Compute sign of x.

       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(idx, expr)
	   Store the value of the expression expr in an internal variable. idx
	   specifies the index 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.

	   The stored value can be retrieved with "ld(var)".

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

	   When the series does not converge the result is undefined.

	   ld(idx) 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(idx)". If idx 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 / 2^50

       E   10^18 / 2^60

       Z   10^21 / 2^70

       Y   10^24 / 2^80

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.

	   pass1
	       Use internal 2pass ratecontrol in first pass mode.

	   pass2
	       Use internal 2pass ratecontrol in second pass mode.

	   gray
	       Only decode/encode grayscale.

	   psnr
	       Set error[?] variables during encoding.

	   truncated
	       Input bitstream might be randomly truncated.

	   drop_changed
	       Don't output frames whose parameters differ from first decoded
	       frame in stream.  Error AVERROR_INPUT_CHANGED is returned when
	       a frame is dropped.

	   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.

	   ilme
	       Apply interlaced motion estimation.

	   cgop
	       Use closed gop.

	   output_corrupt
	       Output even potentially corrupted frames.

       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.

       codec_tag integer
       bug flags (decoding,video)
	   Workaround not auto detected encoder bugs.

	   Possible values:

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

	   ump4
	       (autodetected if fourcc==UMP4)

	   no_padding
	       padding bug (autodetected)

	   amv
	   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

       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
       rc_override_count integer
       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).

       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.

       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
	   xvid
	   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
       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)

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

	   skip
	   startcode
	   er  error recognition

	   mmco
	       memory management control operations (H.264)

	   bugs
	   buffers
	       picture buffer allocations

	   thread_ops
	       threading operations

	   nomc
	       skip motion compensation

       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.

	   (1024, INT_MAX)
	       full motion estimation(slowest)

	   (768, 1024]
	       umh motion estimation

	   (512, 768]
	       hex motion estimation

	   (256, 512]
	       l2s diamond motion estimation

	   [2,256]
	       var diamond motion estimation

	   (-1,  2)
	       small diamond motion estimation

	   -1  funny diamond motion estimation

	   (INT_MIN, -1)
	       sab diamond motion estimation

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

       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.

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

       global_quality integer (encoding,audio,video)
       slice_flags 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

       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,subtitles)
	   Possible values:

	   fast
	       Allow non spec compliant speedup tricks.

	   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.

	   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.

	   skip_manual
	       Do not skip samples and export skip information as frame side
	       data.

	   ass_ro_flush_noop
	       Do not reset ASS ReadOrder field on flush.

	   icc_profiles
	       Generate/parse embedded ICC profiles from/to colorimetry tags.

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

	   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.

	   prft
	       Export encoder Producer Reference Time into packet side-data
	       (see "AV_PKT_DATA_PRFT") for codecs that support it.

	   venc_params
	       Export video encoding parameters through frame side data (see
	       "AV_FRAME_DATA_VIDEO_ENC_PARAMS") for codecs that support it.
	       At present, those are H.264 and VP9.

	   film_grain
	       Export film grain parameters through frame side data (see
	       "AV_FRAME_DATA_FILM_GRAIN_PARAMS").  Supported at present by
	       AV1 decoders.

       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.

       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)
	   Set encoder codec profile. Default value is unknown. Encoder
	   specific profiles are documented in the relevant encoder
	   documentation.

       level integer (encoding,audio,video)
	   Set the encoder level. This level depends on the specific codec,
	   and might correspond to the profile level. It is set by default to
	   unknown.

	   Possible values:

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

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

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

       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.

	   nointra
	       Discard all frames except I frames.

	   all Discard all frames.

	   Default value is default.

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

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

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

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

       mv0_threshold integer (encoding,video)
       compression_level integer (encoding,audio,video)
       bits_per_raw_sample integer
       channel_layout integer (decoding/encoding,audio)
	   See the Channel Layout section in the ffmpeg-utils(1) manual for
	   the required syntax.

       rc_max_vbv_use float (encoding,video)
       rc_min_vbv_use float (encoding,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

	   chroma-derived-nc
	       Chroma-derived NCL

	   chroma-derived-c
	       Chroma-derived CL

	   ictcp
	       ICtCp

       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
	   limited
	       MPEG (219*2^(n-8))

	   pc
	   jpeg
	   full
	       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 indentation:

		   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.

   av1
       AOMedia Video 1 (AV1) decoder.

       Options

       operating_point
	   Select an operating point of a scalable AV1 bitstream (0 - 31).
	   Default is 0.

   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

   libdav1d
       dav1d AV1 decoder.

       libdav1d allows libavcodec to decode the AOMedia Video 1 (AV1) codec.
       Requires the presence of the libdav1d headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-libdav1d".

       Options

       The following options are supported by the libdav1d wrapper.

       framethreads
	   Set amount of frame threads to use during decoding. The default
	   value is 0 (autodetect).  This option is deprecated for libdav1d >=
	   1.0 and will be removed in the future. Use the option
	   "max_frame_delay" and the global option "threads" instead.

       tilethreads
	   Set amount of tile threads to use during decoding. The default
	   value is 0 (autodetect).  This option is deprecated for libdav1d >=
	   1.0 and will be removed in the future. Use the global option
	   "threads" instead.

       max_frame_delay
	   Set max amount of frames the decoder may buffer internally. The
	   default value is 0 (autodetect).

       filmgrain
	   Apply film grain to the decoded video if present in the bitstream.
	   Defaults to the internal default of the library.  This option is
	   deprecated and will be removed in the future. See the global option
	   "export_side_data" to export Film Grain parameters instead of
	   applying it.

       oppoint
	   Select an operating point of a scalable AV1 bitstream (0 - 31).
	   Defaults to the internal default of the library.

       alllayers
	   Output all spatial layers of a scalable AV1 bitstream. The default
	   value is false.

   libdavs2
       AVS2-P2/IEEE1857.4 video decoder wrapper.

       This decoder allows libavcodec to decode AVS2 streams with davs2
       library.

   libuavs3d
       AVS3-P2/IEEE1857.10 video decoder.

       libuavs3d allows libavcodec to decode AVS3 streams.  Requires the
       presence of the libuavs3d headers and library during configuration.
       You need to explicitly configure the build with "--enable-libuavs3d".

       Options

       The following option is supported by the libuavs3d wrapper.

       frame_threads
	   Set amount of frame threads to use during decoding. The default
	   value is 0 (autodetect).

   libxevd
       eXtra-fast Essential Video Decoder (XEVD) MPEG-5 EVC decoder wrapper.

       This decoder requires the presence of the libxevd headers and library
       during configuration. You need to explicitly configure the build with
       --enable-libxevd.

       The xevd project website is at <https://github.com/mpeg5/xevd>.

       Options

       The following options are supported by the libxevd wrapper.  The xevd-
       equivalent options or values are listed in parentheses for easy
       migration.

       To get a more accurate and extensive documentation of the libxevd
       options, invoke the command  "xevd_app --help" or consult the libxevd
       documentation.

       threads (threads)
	   Force to use a specific number of threads

   QSV Decoders
       The family of Intel QuickSync Video decoders (VC1, MPEG-2, H.264, HEVC,
       JPEG/MJPEG, VP8, VP9, AV1).

       Common Options

       The following options are supported by all qsv decoders.

       async_depth
	   Internal parallelization depth, the higher the value the higher the
	   latency.

       gpu_copy
	   A GPU-accelerated copy between video and system memory

	   default
	   on
	   off

       HEVC Options

       Extra options for hevc_qsv.

       load_plugin
	   A user plugin to load in an internal session

	   none
	   hevc_sw
	   hevc_hw
       load_plugins
	   A :-separate list of hexadecimal plugin UIDs to load in an internal
	   session

   v210
       Uncompressed 4:2:2 10-bit decoder.

       Options

       custom_stride
	   Set the line size of the v210 data in bytes. The default value is 0
	   (autodetect). You can use the special -1 value for a strideless
	   v210 as seen in BOXX files.

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.
	   The default value is 1.  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
   libaribb24
       ARIB STD-B24 caption decoder.

       Implements profiles A and C of the ARIB STD-B24 standard.

       libaribb24 Decoder Options

       -aribb24-base-path path
	   Sets the base path for the libaribb24 library. This is utilized for
	   reading of configuration files (for custom unicode conversions),
	   and for dumping of non-text symbols as images under that location.

	   Unset by default.

       -aribb24-skip-ruby-text boolean
	   Tells the decoder wrapper to skip text blocks that contain half-
	   height ruby text.

	   Enabled by default.

   libaribcaption
       Yet another ARIB STD-B24 caption decoder using external libaribcaption
       library.

       Implements profiles A and C of the Japanse ARIB STD-B24 standard,
       Brazilian ABNT NBR 15606-1, and Philippines version of ISDB-T.

       Requires the presence of the libaribcaption headers and library
       (<https://github.com/xqq/libaribcaption>) during configuration.	You
       need to explicitly configure the build with "--enable-libaribcaption".
       If both libaribb24 and libaribcaption are enabled, libaribcaption
       decoder precedes.

       libaribcaption Decoder Options

       -sub_type subtitle_type
	   Specifies the format of the decoded subtitles.

	   bitmap
	       Graphical image.

	   ass ASS formatted text.

	   text
	       Simple text based output without formatting.

	   The default is ass as same as libaribb24 decoder.  Some present
	   players (e.g., mpv) expect ASS format for ARIB caption.

       -caption_encoding encoding_scheme
	   Specifies the encoding scheme of input subtitle text.

	   auto
	       Automatically detect text encoding (default).

	   jis 8bit-char JIS encoding defined in ARIB STD B24.	This encoding
	       used in Japan for ISDB captions.

	   utf8
	       UTF-8 encoding defined in ARIB STD B24.	This encoding is used
	       in Philippines for ISDB-T captions.

	   latin
	       Latin character encoding defined in ABNT NBR 15606-1.  This
	       encoding is used in South America for SBTVD / ISDB-Tb captions.

       -font font_name[,font_name2,...]
	   Specify comma-separated list of font family names to be used for
	   bitmap or ass type subtitle rendering.  Only first font name is
	   used for ass type subtitle.

	   If not specified, use internaly defined default font family.

       -ass_single_rect boolean
	   ARIB STD-B24 specifies that some captions may be displayed at
	   different positions at a time (multi-rectangle subtitle).  Since
	   some players (e.g., old mpv) can't handle multiple ASS rectangles
	   in a single AVSubtitle, or multiple ASS rectangles of indeterminate
	   duration with the same start timestamp, this option can change the
	   behavior so that all the texts are displayed in a single ASS
	   rectangle.

	   The default is false.

	   If your player cannot handle AVSubtitles with multiple ASS
	   rectangles properly, set this option to true or define
	   ASS_SINGLE_RECT=1 to change default behavior at compilation.

       -force_outline_text boolean
	   Specify whether always render outline text for all characters
	   regardless of the indication by charactor style.

	   The default is false.

       -outline_width number (0.0 - 3.0)
	   Specify width for outline text, in dots (relative).

	   The default is 1.5.

       -ignore_background boolean
	   Specify whether to ignore background color rendering.

	   The default is false.

       -ignore_ruby boolean
	   Specify whether to ignore rendering for ruby-like (furigana)
	   characters.

	   The default is false.

       -replace_drcs boolean
	   Specify whether to render replaced DRCS characters as Unicode
	   characters.

	   The default is true.

       -replace_msz_ascii boolean
	   Specify whether to replace MSZ (Middle Size; half width) fullwidth
	   alphanumerics with halfwidth alphanumerics.

	   The default is true.

       -replace_msz_japanese boolean
	   Specify whether to replace some MSZ (Middle Size; half width)
	   fullwidth japanese special characters with halfwidth ones.

	   The default is true.

       -replace_msz_glyph boolean
	   Specify whether to replace MSZ (Middle Size; half width) characters
	   with halfwidth glyphs if the fonts supports it.  This option works
	   under FreeType or DirectWrite renderer with Adobe-Japan1 compliant
	   fonts.  e.g., IBM Plex Sans JP, Morisawa BIZ UDGothic, Morisawa BIZ
	   UDMincho, Yu Gothic, Yu Mincho, and Meiryo.

	   The default is true.

       -canvas_size image_size
	   Specify the resolution of the canvas to render subtitles to;
	   usually, this should be frame size of input video.  This only
	   applies when "-subtitle_type" is set to bitmap.

	   The libaribcaption decoder assumes input frame size for bitmap
	   rendering as below:

	   1.  PROFILE_A : 1440 x 1080 with SAR (PAR) 4:3

	   2.  PROFILE_C : 320 x 180 with SAR (PAR) 1:1

	   If actual frame size of input video does not match above
	   assumption, the rendered captions may be distorted.	To make the
	   captions undistorted, add "-canvas_size" option to specify actual
	   input video size.

	   Note that the "-canvas_size" option is not required for video with
	   different size but same aspect ratio.  In such cases, the caption
	   will be stretched or shrunk to actual video size if "-canvas_size"
	   option is not specified.  If "-canvas_size" option is specified
	   with different size, the caption will be stretched or shrunk as
	   specified size with calculated SAR.

       libaribcaption decoder usage examples

       Display MPEG-TS file with ARIB subtitle by "ffplay" tool:

	       ffplay -sub_type bitmap MPEG.TS

       Display MPEG-TS file with input frame size 1920x1080 by "ffplay" tool:

	       ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS

       Embed ARIB subtitle in transcoded video:

	       ffmpeg -sub_type bitmap -i src.m2t -filter_complex "[0:v][0:s]overlay" -vcodec h264 dest.mp4

   dvbsub
       Options

       compute_clut
	   -2  Compute clut once if no matching CLUT is in the stream.

	   -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 commas, 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. Pages that do not match
	   the specified list are dropped. You may use the special "*" string
	   to match all pages, or "subtitle" to match all subtitle pages.
	   Default value is *.

       txt_default_region
	   Set default character set used for decoding, a value between 0 and
	   87 (see ETS 300 706, Section 15, Table 32). Default value is -1,
	   which does not override the libzvbi default. This option is needed
	   for some legacy level 1.0 transmissions which cannot signal the
	   proper charset.

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

       txt_format
	   Specifies the format of the decoded subtitles.

	   bitmap
	       The default format, you should use this for teletext pages,
	       because certain graphics and colors cannot be expressed in
	       simple text or even ASS.

	   text
	       Simple text based output without formatting.

	   ass Formatted ASS output, subtitle pages and teletext pages are
	       returned in different styles, subtitle pages are stripped down
	       to text, but an effort is made to keep the text alignment and
	       the formatting.

       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 -1 which means infinity
	   or until the next subtitle event comes.

       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.

       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 is the default 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.

	   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.

	       Uses a cheaper version of twoloop algorithm that doesn't try to
	       do as many clever adjustments. Worse with low bitrates (less
	       than 64kbps), but is better and much faster at higher bitrates.

       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.

       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 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, it's 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 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.

       eld_v2
	   Enable ELDv2 (LD-MPS extension for ELD stereo signals) for ELDv2 if
	   set to 1, disabled if set to 0.

	   Note that option is available when fdk-aac version
	   (AACENCODER_LIB_VL0.AACENCODER_LIB_VL1.AACENCODER_LIB_VL2) >
	   (4.0.0).

	   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.

       frame_length
	   Set the audio frame length in samples. Default value is the
	   internal default of the library. Refer to the library's
	   documentation for information about supported values.

       Examples

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

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

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

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

   liblc3
       liblc3 LC3 (Low Complexity Communication Codec) encoder wrapper.

       Requires the presence of the liblc3 headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-liblc3".

       This encoder has support for the Bluetooth SIG LC3 codec for the LE
       Audio protocol, and the following features of LC3plus:

       o   Frame duration of 2.5 and 5ms.

       o   High-Resolution mode, 48 KHz, and 96 kHz sampling rates.

       For more information see the liblc3 project at
       <https://github.com/google/liblc3>.

       Options

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

       b bitrate
	   Set the bit rate in bits/s. This will determine the fixed size of
	   the encoded frames, for a selected frame duration.

       ar frequency
	   Set the audio sampling rate (in Hz).

       channels
	   Set the number of audio channels.

       frame_duration
	   Set the audio frame duration in milliseconds. Default value is
	   10ms.  Allowed frame durations are 2.5ms, 5ms, 7.5ms and 10ms.  LC3
	   (Bluetooth LE Audio), allows 7.5ms and 10ms; and LC3plus 2.5ms, 5ms
	   and 10ms.

	   The 10ms frame duration is available in LC3 and LC3 plus standard.
	   In this mode, the produced bitstream can be referenced either as
	   LC3 or LC3plus.

       high_resolution boolean
	   Enable the high-resolution mode if set to 1. The high-resolution
	   mode is available with all LC3plus frame durations and for a
	   sampling rate of 48 KHz, and 96 KHz.

	   The encoder automatically turns off this mode at lower sampling
	   rates and activates it at 96 KHz.

	   This mode should be preferred at high bitrates. In this mode, the
	   audio bandwidth is always up to the Nyquist frequency, compared to
	   LC3 at 48 KHz, which limits the bandwidth to 20 KHz.

   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.

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

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

   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.

       fec (n/a)
	   Enable inband forward error correction. packet_loss must be non-
	   zero to take advantage - frequency of FEC 'side-data' is
	   proportional to expected packet loss.  Default is disabled.

       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 by disabling voice-
	       optimized 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). Note that libopus
	   forces a wideband cutoff for bitrates < 15 kbps, unless CELT-only
	   (application set to lowdelay) mode is used.

       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.

       apply_phase_inv (N.A.) (requires libopus >= 1.2)
	   If set to 0, disables the use of phase inversion for intensity
	   stereo, improving the quality of mono downmixes, but slightly
	   reducing normal stereo quality. The default is 1 (phase inversion
	   enabled).

   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.

   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.

       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)

       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.

   a64_multi, a64_multi5
       A64 / Commodore 64 multicolor charset encoder. "a64_multi5" is extended
       with 5th color (colram).

   Cinepak
       Cinepak aka CVID encoder.  Compatible with Windows 3.1 and vintage
       MacOS.

       Options

       g integer
	   Keyframe interval.  A keyframe is inserted at least every "-g"
	   frames, sometimes sooner.

       q:v integer
	   Quality factor. Lower is better. Higher gives lower bitrate.  The
	   following table lists bitrates when encoding akiyo_cif.y4m for
	   various values of "-q:v" with "-g 100":

	   "-q:v 1" 1918 kb/s
	   "-q:v 2" 1735 kb/s
	   "-q:v 4" 1500 kb/s
	   "-q:v 10" 1041 kb/s
	   "-q:v 20" 826 kb/s
	   "-q:v 40" 553 kb/s
	   "-q:v 100" 394 kb/s
	   "-q:v 200" 312 kb/s
	   "-q:v 400" 266 kb/s
	   "-q:v 1000" 237 kb/s
       max_extra_cb_iterations integer
	   Max extra codebook recalculation passes, more is better and slower.

       skip_empty_cb boolean
	   Avoid wasting bytes, ignore vintage MacOS decoder.

       max_strips integer
       min_strips integer
	   The minimum and maximum number of strips to use.  Wider range
	   sometimes improves quality.	More strips is generally better
	   quality but costs more bits.  Fewer strips tend to yield more
	   keyframes.  Vintage compatible is 1..3.

       strip_number_adaptivity integer
	   How much number of strips is allowed to change between frames.
	   Higher is better but slower.

   GIF
       GIF image/animation encoder.

       Options

       gifflags integer
	   Sets the flags used for GIF encoding.

	   offsetting
	       Enables picture offsetting.

	       Default is enabled.

	   transdiff
	       Enables transparency detection between frames.

	       Default is enabled.

       gifimage integer
	   Enables encoding one full GIF image per frame, rather than an
	   animated GIF.

	   Default value is 0.

       global_palette integer
	   Writes a palette to the global GIF header where feasible.

	   If disabled, every frame will always have a palette written, even
	   if there is a global palette supplied.

	   Default value is 1.

   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 integer
	   Can be set to either "j2k" or "jp2" (the default) that makes it
	   possible to store non-rgb pix_fmts.

       tile_width integer
	   Sets tile width. Range is 1 to 1073741824. Default is 256.

       tile_height integer
	   Sets tile height. Range is 1 to 1073741824. Default is 256.

       pred integer
	   Allows setting the discrete wavelet transform (DWT) type

	   dwt97int (Lossy)
	   dwt53 (Lossless)

	   Default is "dwt97int"

       sop boolean
	   Enable this to add SOP marker at the start of each packet. Disabled
	   by default.

       eph boolean
	   Enable this to add EPH marker at the end of each packet header.
	   Disabled by default.

       prog integer
	   Sets the progression order to be used by the encoder.  Possible
	   values are:

	   lrcp
	   rlcp
	   rpcl
	   pcrl
	   cprl

	   Set to "lrcp" by default.

       layer_rates string
	   By default, when this option is not used, compression is done using
	   the quality metric.	This option allows for compression using
	   compression ratio. The compression ratio for each level could be
	   specified. The compression ratio of a layer "l" species the what
	   ratio of total file size is contained in the first "l" layers.

	   Example usage:

		   ffmpeg -i input.bmp -c:v jpeg2000 -layer_rates "100,10,1" output.j2k

	   This would compress the image to contain 3 layers, where the data
	   contained in the first layer would be compressed by 1000 times,
	   compressed by 100 in the first two layers, and shall contain all
	   data while using all 3 layers.

   librav1e
       rav1e AV1 encoder wrapper.

       Requires the presence of the rav1e headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-librav1e".

       Options

       qmax
	   Sets the maximum quantizer to use when using bitrate mode.

       qmin
	   Sets the minimum quantizer to use when using bitrate mode.

       qp  Uses quantizer mode to encode at the given quantizer (0-255).

       speed
	   Selects the speed preset (0-10) to encode with.

       tiles
	   Selects how many tiles to encode with.

       tile-rows
	   Selects how many rows of tiles to encode with.

       tile-columns
	   Selects how many columns of tiles to encode with.

       rav1e-params
	   Set rav1e options using a list of key=value pairs separated by ":".
	   See rav1e --help for a list of options.

	   For example to specify librav1e encoding options with
	   -rav1e-params:

		   ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4

   libaom-av1
       libaom AV1 encoder wrapper.

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

       Options

       The wrapper supports the following standard libavcodec options:

       b   Set bitrate target in bits/second.  By default this will use
	   variable-bitrate mode.  If maxrate and minrate are also set to the
	   same value then it will use constant-bitrate mode, otherwise if crf
	   is set as well then it will use constrained-quality mode.

       g keyint_min
	   Set key frame placement.  The GOP size sets the maximum distance
	   between key frames; if zero the output stream will be intra-only.
	   The minimum distance is ignored unless it is the same as the GOP
	   size, in which case key frames will always appear at a fixed
	   interval.  Not set by default, so without this option the library
	   has completely free choice about where to place key frames.

       qmin qmax
	   Set minimum/maximum quantisation values.  Valid range is from 0 to
	   63 (warning: this does not match the quantiser values actually used
	   by AV1 - divide by four to map real quantiser values to this
	   range).  Defaults to min/max (no constraint).

       minrate maxrate bufsize rc_init_occupancy
	   Set rate control buffering parameters.  Not used if not set -
	   defaults to unconstrained variable bitrate.

       threads
	   Set the number of threads to use while encoding.  This may require
	   the tiles or row-mt options to also be set to actually use the
	   specified number of threads fully. Defaults to the number of
	   hardware threads supported by the host machine.

       profile
	   Set the encoding profile.  Defaults to using the profile which
	   matches the bit depth and chroma subsampling of the input.

       The wrapper also has some specific options:

       cpu-used
	   Set the quality/encoding speed tradeoff.  Valid range is from 0 to
	   8, higher numbers indicating greater speed and lower quality.  The
	   default value is 1, which will be slow and high quality.

       auto-alt-ref
	   Enable use of alternate reference frames.  Defaults to the internal
	   default of the library.

       arnr-max-frames (frames)
	   Set altref noise reduction max frame count. Default is -1.

       arnr-strength (strength)
	   Set altref noise reduction filter strength. Range is -1 to 6.
	   Default is -1.

       aq-mode (aq-mode)
	   Set adaptive quantization mode. Possible values:

	   none (0)
	       Disabled.

	   variance (1)
	       Variance-based.

	   complexity (2)
	       Complexity-based.

	   cyclic (3)
	       Cyclic refresh.

       tune (tune)
	   Set the distortion metric the encoder is tuned with. Default is
	   "psnr".

	   psnr (0)
	   ssim (1)
       lag-in-frames
	   Set the maximum number of frames which the encoder may keep in
	   flight at any one time for lookahead purposes.  Defaults to the
	   internal default of the library.

       error-resilience
	   Enable error resilience features:

	   default
	       Improve resilience against losses of whole frames.

	   Not enabled by default.

       crf Set the quality/size tradeoff for constant-quality (no bitrate
	   target) and constrained-quality (with maximum bitrate target)
	   modes. Valid range is 0 to 63, higher numbers indicating lower
	   quality and smaller output size.  Only used if set; by default only
	   the bitrate target is used.

       static-thresh
	   Set a change threshold on blocks below which they will be skipped
	   by the encoder.  Defined in arbitrary units as a nonnegative
	   integer, defaulting to zero (no blocks are skipped).

       drop-threshold
	   Set a threshold for dropping frames when close to rate control
	   bounds.  Defined as a percentage of the target buffer - when the
	   rate control buffer falls below this percentage, frames will be
	   dropped until it has refilled above the threshold.  Defaults to
	   zero (no frames are dropped).

       denoise-noise-level (level)
	   Amount of noise to be removed for grain synthesis. Grain synthesis
	   is disabled if this option is not set or set to 0.

       denoise-block-size (pixels)
	   Block size used for denoising for grain synthesis. If not set, AV1
	   codec uses the default value of 32.

       undershoot-pct (pct)
	   Set datarate undershoot (min) percentage of the target bitrate.
	   Range is -1 to 100.	Default is -1.

       overshoot-pct (pct)
	   Set datarate overshoot (max) percentage of the target bitrate.
	   Range is -1 to 1000.  Default is -1.

       minsection-pct (pct)
	   Minimum percentage variation of the GOP bitrate from the target
	   bitrate. If minsection-pct is not set, the libaomenc wrapper
	   computes it as follows: "(minrate * 100 / bitrate)".  Range is -1
	   to 100. Default is -1 (unset).

       maxsection-pct (pct)
	   Maximum percentage variation of the GOP bitrate from the target
	   bitrate. If maxsection-pct is not set, the libaomenc wrapper
	   computes it as follows: "(maxrate * 100 / bitrate)".  Range is -1
	   to 5000. Default is -1 (unset).

       frame-parallel (boolean)
	   Enable frame parallel decodability features. Default is true.

       tiles
	   Set the number of tiles to encode the input video with, as columns
	   x rows.  Larger numbers allow greater parallelism in both encoding
	   and decoding, but may decrease coding efficiency.  Defaults to the
	   minimum number of tiles required by the size of the input video
	   (this is 1x1 (that is, a single tile) for sizes up to and including
	   4K).

       tile-columns tile-rows
	   Set the number of tiles as log2 of the number of tile rows and
	   columns.  Provided for compatibility with libvpx/VP9.

       row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
	   Enable row based multi-threading. Disabled by default.

       enable-cdef (boolean)
	   Enable Constrained Directional Enhancement Filter. The libaom-av1
	   encoder enables CDEF by default.

       enable-restoration (boolean)
	   Enable Loop Restoration Filter. Default is true for libaom-av1.

       enable-global-motion (boolean)
	   Enable the use of global motion for block prediction. Default is
	   true.

       enable-intrabc (boolean)
	   Enable block copy mode for intra block prediction. This mode is
	   useful for screen content. Default is true.

       enable-rect-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable rectangular partitions. Default is true.

       enable-1to4-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable 1:4/4:1 partitions. Default is true.

       enable-ab-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable AB shape partitions. Default is true.

       enable-angle-delta (boolean) (Requires libaom >= v2.0.0)
	   Enable angle delta intra prediction. Default is true.

       enable-cfl-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable chroma predicted from luma intra prediction. Default is
	   true.

       enable-filter-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable filter intra predictor. Default is true.

       enable-intra-edge-filter (boolean) (Requires libaom >= v2.0.0)
	   Enable intra edge filter. Default is true.

       enable-smooth-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable smooth intra prediction mode. Default is true.

       enable-paeth-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable paeth predictor in intra prediction. Default is true.

       enable-palette (boolean) (Requires libaom >= v2.0.0)
	   Enable palette prediction mode. Default is true.

       enable-flip-idtx (boolean) (Requires libaom >= v2.0.0)
	   Enable extended transform type, including FLIPADST_DCT,
	   DCT_FLIPADST, FLIPADST_FLIPADST, ADST_FLIPADST, FLIPADST_ADST,
	   IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default
	   is true.

       enable-tx64 (boolean) (Requires libaom >= v2.0.0)
	   Enable 64-pt transform. Default is true.

       reduced-tx-type-set (boolean) (Requires libaom >= v2.0.0)
	   Use reduced set of transform types. Default is false.

       use-intra-dct-only (boolean) (Requires libaom >= v2.0.0)
	   Use DCT only for INTRA modes. Default is false.

       use-inter-dct-only (boolean) (Requires libaom >= v2.0.0)
	   Use DCT only for INTER modes. Default is false.

       use-intra-default-tx-only (boolean) (Requires libaom >= v2.0.0)
	   Use Default-transform only for INTRA modes. Default is false.

       enable-ref-frame-mvs (boolean) (Requires libaom >= v2.0.0)
	   Enable temporal mv prediction. Default is true.

       enable-reduced-reference-set (boolean) (Requires libaom >= v2.0.0)
	   Use reduced set of single and compound references. Default is
	   false.

       enable-obmc (boolean) (Requires libaom >= v2.0.0)
	   Enable obmc. Default is true.

       enable-dual-filter (boolean) (Requires libaom >= v2.0.0)
	   Enable dual filter. Default is true.

       enable-diff-wtd-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable difference-weighted compound. Default is true.

       enable-dist-wtd-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable distance-weighted compound. Default is true.

       enable-onesided-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable one sided compound. Default is true.

       enable-interinter-wedge (boolean) (Requires libaom >= v2.0.0)
	   Enable interinter wedge compound. Default is true.

       enable-interintra-wedge (boolean) (Requires libaom >= v2.0.0)
	   Enable interintra wedge compound. Default is true.

       enable-masked-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable masked compound. Default is true.

       enable-interintra-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable interintra compound. Default is true.

       enable-smooth-interintra (boolean) (Requires libaom >= v2.0.0)
	   Enable smooth interintra mode. Default is true.

       aom-params
	   Set libaom options using a list of key=value pairs separated by
	   ":". For a list of supported options, see aomenc --help under the
	   section "AV1 Specific Options".

	   For example to specify libaom encoding options with -aom-params:

		   ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4

   libsvtav1
       SVT-AV1 encoder wrapper.

       Requires the presence of the SVT-AV1 headers and library during
       configuration.  You need to explicitly configure the build with
       "--enable-libsvtav1".

       Options

       profile
	   Set the encoding profile.

	   main
	   high
	   professional
       level
	   Set the operating point level. For example: '4.0'

       hielevel
	   Set the Hierarchical prediction levels.

	   3level
	   4level
	       This is the default.

       tier
	   Set the operating point tier.

	   main
	       This is the default.

	   high
       qmax
	   Set the maximum quantizer to use when using a bitrate mode.

       qmin
	   Set the minimum quantizer to use when using a bitrate mode.

       crf Constant rate factor value used in crf rate control mode (0-63).

       qp  Set the quantizer used in cqp rate control mode (0-63).

       sc_detection
	   Enable scene change detection.

       la_depth
	   Set number of frames to look ahead (0-120).

       preset
	   Set the quality-speed tradeoff, in the range 0 to 13.  Higher
	   values are faster but lower quality.

       tile_rows
	   Set log2 of the number of rows of tiles to use (0-6).

       tile_columns
	   Set log2 of the number of columns of tiles to use (0-4).

       svtav1-params
	   Set SVT-AV1 options using a list of key=value pairs separated by
	   ":". See the SVT-AV1 encoder user guide for a list of accepted
	   parameters.

   libjxl
       libjxl JPEG XL encoder wrapper.

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

       Options

       The libjxl wrapper supports the following options:

       distance
	   Set the target Butteraugli distance. This is a quality setting:
	   lower distance yields higher quality, with distance=1.0 roughly
	   comparable to libjpeg Quality 90 for photographic content. Setting
	   distance=0.0 yields true lossless encoding. Valid values range
	   between 0.0 and 15.0, and sane values rarely exceed 5.0. Setting
	   distance=0.1 usually attains transparency for most input. The
	   default is 1.0.

       effort
	   Set the encoding effort used. Higher effort values produce more
	   consistent quality and usually produces a better quality/bpp curve,
	   at the cost of more CPU time required. Valid values range from 1 to
	   9, and the default is 7.

       modular
	   Force the encoder to use Modular mode instead of choosing
	   automatically. The default is to use VarDCT for lossy encoding and
	   Modular for lossless. VarDCT is generally superior to Modular for
	   lossy encoding but does not support lossless encoding.

   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.

       loopfilter
	   Enable loop filter, if set to 1 (automatically enabled). To disable
	   set a value of 0.

       profile
	   Set profile restrictions. If set to the value of main enable CABAC
	   (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

       max_nal_size
	   Set maximum NAL size in bytes.

       allow_skip_frames
	   Allow skipping frames to hit the target bitrate if set to 1.

   libtheora
       libtheora Theora encoder wrapper.

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

       For more information about the libtheora project see
       <http://www.theora.org/>.

       Options

       The following global options are mapped to internal libtheora options
       which affect the quality and the bitrate of the encoded stream.

       b   Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.
	   In case VBR (Variable Bit Rate) mode is enabled this option is
	   ignored.

       flags
	   Used to enable constant quality mode (VBR) encoding through the
	   qscale flag, and to enable the "pass1" and "pass2" modes.

       g   Set the GOP size.

       global_quality
	   Set the global quality as an integer in lambda units.

	   Only relevant when VBR mode is enabled with "flags +qscale". The
	   value is converted to QP units by dividing it by "FF_QP2LAMBDA",
	   clipped in the [0 - 10] range, and then multiplied by 6.3 to get a
	   value in the native libtheora range [0-63]. A higher value
	   corresponds to a higher quality.

       q   Enable VBR mode when set to a non-negative value, and set constant
	   quality value as a double floating point value in QP units.

	   The value is clipped in the [0-10] range, and then multiplied by
	   6.3 to get a value in the native libtheora range [0-63].

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

       Examples

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

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

       o   Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

		   ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

   libvpx
       VP8/VP9 format supported through libvpx.

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

       Options

       The following options are supported by the libvpx wrapper. The
       vpxenc-equivalent options or values are listed in parentheses for easy
       migration.

       To reduce the duplication of documentation, only the private options
       and some others requiring special attention are documented here. For
       the documentation of the undocumented generic options, see the Codec
       Options chapter.

       To get more documentation of the libvpx options, invoke the command
       ffmpeg -h encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc
       --help. Further information is available in the libvpx API
       documentation.

       b (target-bitrate)
	   Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
	   bits/s, while vpxenc's target-bitrate is in kilobits/s.

       g (kf-max-dist)
       keyint_min (kf-min-dist)
       qmin (min-q)
	   Minimum (Best Quality) Quantizer.

       qmax (max-q)
	   Maximum (Worst Quality) Quantizer.  Can be changed per-frame.

       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).	Values
	       greater than 1 enable multi-layer alternate reference frames
	       (VP9 only).

	   arnr-maxframes
	       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.

	   min-gf-interval
	       Set minimum golden/alternate reference frame interval (VP9
	       only).

       error-resilient
	   Enable error resiliency features.

       sharpness integer
	   Increase sharpness at the expense of lower PSNR.  The valid range
	   is [0, 7].

       ts-parameters
	   Sets the temporal scalability configuration using a :-separated
	   list of key=value pairs. For example, to specify temporal
	   scalability parameters with "ffmpeg":

		   ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
		   ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
		   ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT

	   Below is a brief explanation of each of the parameters, please
	   refer to "struct vpx_codec_enc_cfg" in "vpx/vpx_encoder.h" for more
	   details.

	   ts_number_layers
	       Number of temporal coding layers.

	   ts_target_bitrate
	       Target bitrate for each temporal layer (in kbps).  (bitrate
	       should be inclusive of the lower temporal layer).

	   ts_rate_decimator
	       Frame rate decimation factor for each temporal layer.

	   ts_periodicity
	       Length of the sequence defining frame temporal layer
	       membership.

	   ts_layer_id
	       Template defining the membership of frames to temporal layers.

	   ts_layering_mode
	       (optional) Selecting the temporal structure from a set of pre-
	       defined temporal layering modes.  Currently supports the
	       following options.

	       0   No temporal layering flags are provided internally, relies
		   on flags being passed in using "metadata" field in
		   "AVFrame" with following keys.

		   vp8-flags
		       Sets the flags passed into the encoder to indicate the
		       referencing scheme for the current frame.  Refer to
		       function "vpx_codec_encode" in "vpx/vpx_encoder.h" for
		       more details.

		   temporal_id
		       Explicitly sets the temporal id of the current frame to
		       encode.

	       2   Two temporal layers. 0-1...

	       3   Three temporal layers. 0-2-1-2...; with single reference
		   frame.

	       4   Same as option "3", except there is a dependency between
		   the two temporal layer 2 frames within the temporal period.

       VP8-specific options
	   screen-content-mode
	       Screen content mode, one of: 0 (off), 1 (screen), 2 (screen
	       with more aggressive rate control).

       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.

	   tune-content
	       Set content type: default (0), screen (1), film (2).

	   corpus-complexity
	       Corpus VBR mode is a variant of standard VBR where the
	       complexity distribution midpoint is passed in rather than
	       calculated for a specific clip or chunk.

	       The valid range is [0, 10000]. 0 (default) uses standard VBR.

	   enable-tpl boolean
	       Enable temporal dependency model.

	   ref-frame-config
	       Using per-frame metadata, set members of the structure
	       "vpx_svc_ref_frame_config_t" in "vpx/vp8cx.h" to fine-control
	       referencing schemes and frame buffer management.  Use a
	       :-separated list of key=value pairs.  For example,

		       av_dict_set(&av_frame->metadata, "ref-frame-config", \
		       "rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0");

	       rfc_update_buffer_slot
		   Indicates the buffer slot number to update

	       rfc_update_last
		   Indicates whether to update the LAST frame

	       rfc_update_golden
		   Indicates whether to update GOLDEN frame

	       rfc_update_alt_ref
		   Indicates whether to update ALT_REF frame

	       rfc_lst_fb_idx
		   LAST frame buffer index

	       rfc_gld_fb_idx
		   GOLDEN frame buffer index

	       rfc_alt_fb_idx
		   ALT_REF frame buffer index

	       rfc_reference_last
		   Indicates whether to reference LAST frame

	       rfc_reference_golden
		   Indicates whether to reference GOLDEN frame

	       rfc_reference_alt_ref
		   Indicates whether to reference ALT_REF frame

	       rfc_reference_duration
		   Indicates frame duration

       For more information about libvpx see: <http://www.webmproject.org/>

   libvvenc
       VVenC H.266/VVC encoder wrapper.

       This encoder requires the presence of the libvvenc headers and library
       during configuration. You need to explicitly configure the build with
       --enable-libvvenc.

       The VVenC project website is at
       <https://github.com/fraunhoferhhi/vvenc>.

       Supported Pixel Formats

       VVenC supports only 10-bit color spaces as input. But the internal
       (encoded) bit depth can be set to 8-bit or 10-bit at runtime.

       Options

       b   Sets target video bitrate.

       g   Set the GOP size. Currently support for g=1 (Intra only) or
	   default.

       preset
	   Set the VVenC preset.

       levelidc
	   Set level idc.

       tier
	   Set vvc tier.

       qp  Set constant quantization parameter.

       subopt boolean
	   Set subjective (perceptually motivated) optimization. Default is 1
	   (on).

       bitdepth8 boolean
	   Set 8bit coding mode instead of using 10bit. Default is 0 (off).

       period
	   set (intra) refresh period in seconds.

       vvenc-params
	   Set vvenc options using a list of key=value couples separated by
	   ":". See vvencapp --fullhelp or vvencFFapp --fullhelp for a list of
	   options.

	   For example, the options might be provided as:

		   intraperiod=64:decodingrefreshtype=idr:poc0idr=1:internalbitdepth=8

	   For example the encoding options might be provided with
	   -vvenc-params:

		   ffmpeg -i input -c:v libvvenc -b 1M -vvenc-params intraperiod=64:decodingrefreshtype=idr:poc0idr=1:internalbitdepth=8 output.mp4

   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.

       -quality float
	   For lossy encoding, this controls image quality. For lossless
	   encoding, this controls the effort and time spent in compression.
	   Range is 0 to 100. Default is 75.

       -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.

       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.

       In the list below, note that the x264 option name is shown in
       parentheses after the libavcodec corresponding name, in case there is a
       direct mapping.

       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)
	   Number of B-frames between I and P-frames

       g (keyint)
	   Maximum GOP size

       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 0-16.

       level (level)
	   Set the "x264_param_t.i_level_idc" value in case the value is
	   positive, it is ignored otherwise.

	   This value can be set using the "AVCodecContext" API (e.g. by
	   setting the "AVCodecContext" value directly), and is specified as
	   an integer mapped on a corresponding level (e.g. the value 31 maps
	   to H.264 level IDC "3.1", as defined in the "x264_levels" table).
	   It is ignored when set to a non positive value.

	   Alternatively it can be set as a private option, overriding the
	   value set in "AVCodecContext", and in this case must be specified
	   as the level IDC identifier (e.g. "3.1"), as defined by H.264 Annex
	   A.

       sc_threshold (scenecut)
	   Sets the threshold for the scene change detection.

       trellis (trellis)
	   Performs Trellis quantization to increase efficiency. Enabled by
	   default.

       nr (nr)
	   Noise reduction

       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)
	   Initial VBV buffer occupancy

       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 values.
	   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 opts
       x264-params opts
	   Override the x264 configuration using a :-separated list of
	   key=value options.

	   The argument for both options is a list of key=value couples
	   separated by ":". With x264opts the value can be omitted, and the
	   value 1 is assumed in that case.

	   For filter and psy-rd options values 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, the options might be provided as:

		   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

	   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

	   To get the complete list of the libx264 options, invoke the command
	   x264 --fullhelp or consult the libx264 documentation.

       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).

       udu_sei boolean
	   Import user data unregistered SEI if available into output. Default
	   is 0 (off).

       mb_info boolean
	   Set mb_info data through AVFrameSideData, only useful when used
	   from the API. Default is 0 (off).

       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

       b   Sets target video bitrate.

       bf
       g   Set the GOP size.

       keyint_min
	   Minimum GOP size.

       refs
	   Number of reference frames each P-frame can use. The range is from
	   1-16.

       preset
	   Set the x265 preset.

       tune
	   Set the x265 tune parameter.

       profile
	   Set profile restrictions.

       crf Set the quality for constant quality mode.

       qp  Set constant quantization rate control method parameter.

       qmin
	   Minimum quantizer scale.

       qmax
	   Maximum quantizer scale.

       qdiff
	   Maximum difference between quantizer scales.

       qblur
	   Quantizer curve blur

       qcomp
	   Quantizer curve compression factor

       i_qfactor
       b_qfactor
       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.

       udu_sei boolean
	   Import user data unregistered SEI if available into output. Default
	   is 0 (off).

       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

   libxavs2
       xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

       This encoder requires the presence of the libxavs2 headers and library
       during configuration. You need to explicitly configure the build with
       --enable-libxavs2.

       The following standard libavcodec options are used:

       o   b / bit_rate

       o   g / gop_size

       o   bf / max_b_frames

       The encoder also has its own specific options:

       Options

       lcu_row_threads
	   Set the number of parallel threads for rows from 1 to 8 (default
	   5).

       initial_qp
	   Set the xavs2 quantization parameter from 1 to 63 (default 34).
	   This is used to set the initial qp for the first frame.

       qp  Set the xavs2 quantization parameter from 1 to 63 (default 34).
	   This is used to set the qp value under constant-QP mode.

       max_qp
	   Set the max qp for rate control from 1 to 63 (default 55).

       min_qp
	   Set the min qp for rate control from 1 to 63 (default 20).

       speed_level
	   Set the Speed level from 0 to 9 (default 0). Higher is better but
	   slower.

       log_level
	   Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1:
	   warning, 2: info, 3: debug.

       xavs2-params
	   Set xavs2 options using a list of key=value couples separated by
	   ":".

	   For example to specify libxavs2 encoding options with
	   -xavs2-params:

		   ffmpeg -i input -c:v libxavs2 -xavs2-params RdoqLevel=0 output.avs2

   libxeve
       eXtra-fast Essential Video Encoder (XEVE) MPEG-5 EVC encoder wrapper.
       The xeve-equivalent options or values are listed in parentheses for
       easy migration.

       This encoder requires the presence of the libxeve headers and library
       during configuration. You need to explicitly configure the build with
       --enable-libxeve.

	   Many libxeve encoder options are mapped to FFmpeg global codec
	   options, while unique encoder options are provided through private
	   options.  Additionally the xeve-params private options allows one
	   to pass a list of key=value tuples as accepted by the libxeve
	   "parse_xeve_params" function.

       The xeve project website is at <https://github.com/mpeg5/xeve>.

       Options

       The following options are supported by the libxeve wrapper.  The xeve-
       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 libxeve
	   options, invoke the command	"xeve_app --help" or consult the
	   libxeve documentation.

       b (bitrate)
	   Set target video bitrate in bits/s.	Note that FFmpeg's b option is
	   expressed in bits/s, while xeve's bitrate is in kilobits/s.

       bf (bframes)
	   Set the maximum number of B frames (1,3,7,15).

       g (keyint)
	   Set the GOP size (I-picture period).

       preset (preset)
	   Set the xeve preset.  Set the encoder preset value to determine
	   encoding speed [fast, medium, slow, placebo]

       tune (tune)
	   Set the encoder tune parameter [psnr, zerolatency]

       profile (profile)
	   Set the encoder profile [0: baseline; 1: main]

       crf (crf)
	   Set the quality for constant quality mode.  Constant rate factor
	   <10..49> [default: 32]

       qp (qp)
	   Set constant quantization rate control method parameter.
	   Quantization parameter qp <0..51> [default: 32]

       threads (threads)
	   Force to use a specific number of threads

   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.

	   qpel
	       Enable quarter-pixel motion compensation.

	   cgop
	       Enable closed GOP.

	   global_header
	       Place global headers in extradata instead of every keyframe.

       gmc Enable the use of global motion compensation (GMC).	Default is 0
	   (disabled).

       me_quality
	   Set motion estimation quality level. Possible values in decreasing
	   order of speed and increasing order of quality:

	   0   Use no motion estimation (default).

	   1, 2
	       Enable advanced diamond zonal search for 16x16 blocks and half-
	       pixel refinement for 16x16 blocks.

	   3, 4
	       Enable all of the things described above, plus advanced diamond
	       zonal search for 8x8 blocks and half-pixel refinement for 8x8
	       blocks, also enable motion estimation on chroma planes for P
	       and B-frames.

	   5, 6
	       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.

       trellis
	   Set rate-distortion optimal quantization.

       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.

   MediaFoundation
       This provides wrappers to encoders (both audio and video) in the
       MediaFoundation framework. It can access both SW and HW encoders.
       Video encoders can take input in either of nv12 or yuv420p form (some
       encoders support both, some support only either - in practice, nv12 is
       the safer choice, especially among HW encoders).

   Microsoft RLE
       Microsoft RLE aka MSRLE encoder.  Only 8-bit palette mode supported.
       Compatible with Windows 3.1 and Windows 95.

       Options

       g integer
	   Keyframe interval.  A keyframe is inserted at least every "-g"
	   frames, sometimes sooner.

   mpeg2
       MPEG-2 video encoder.

       Options

       profile
	   Select the mpeg2 profile to encode:

	   422
	   high
	   ss  Spatially Scalable

	   snr SNR Scalable

	   main
	   simple
       level
	   Select the mpeg2 level to encode:

	   high
	   high1440
	   main
	   low
       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.

       video_format integer
	   Specifies the video_format written into the sequence display
	   extension indicating the source of the video pictures. The default
	   is unspecified, can be component, pal, ntsc, secam or mac.  For
	   maximum compatibility, use component.

       a53cc boolean
	   Import closed captions (which must be ATSC compatible format) into
	   output.  Default is 1 (on).

   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, HEVC,
       JPEG/MJPEG, VP9, AV1)

       Ratecontrol Method

       The ratecontrol method is selected as follows:

       o   When global_quality is specified, a quality-based mode is used.
	   Specifically this means either

	   -   CQP - constant quantizer scale, when the qscale codec flag is
	       also set (the -qscale ffmpeg option).

	   -   LA_ICQ - intelligent constant quality with lookahead, when the
	       look_ahead option is also set.

	   -   ICQ -- intelligent constant quality otherwise. For the ICQ
	       modes, global quality range is 1 to 51, with 1 being the best
	       quality.

       o   Otherwise when the desired average bitrate is specified with the b
	   option, a bitrate-based mode is used.

	   -   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, both
	       avbr_accuracy and avbr_convergence are set to non-zero. This
	       mode is available for H264 and HEVC on Windows.

       o   Otherwise the default ratecontrol method CQP is used.

       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.

       Global Options -> MSDK Options

       Additional libavcodec global options are mapped to MSDK options as
       follows:

       o   g/gop_size -> GopPicSize

       o   bf/max_b_frames+1 -> GopRefDist

       o   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       o   slices -> NumSlice

       o   refs -> NumRefFrame

       o   b_strategy/b_frame_strategy -> BRefType

       o   cgop/CLOSED_GOP codec flag -> GopOptFlag

       o   For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset
	   set the difference between QPP and QPI, and QPP and QPB
	   respectively.

       o   Setting the coder option to the value vlc will make the H.264
	   encoder use CAVLC instead of CABAC.

       Common Options

       Following options are used by all qsv encoders.

       async_depth
	   Specifies how many asynchronous operations an application performs
	   before the application explicitly synchronizes the result. If zero,
	   the value is not specified.

       preset
	   This option itemizes a range of choices from veryfast (best speed)
	   to veryslow (best quality).

	   veryfast
	   faster
	   fast
	   medium
	   slow
	   slower
	   veryslow
       forced_idr
	   Forcing I frames as IDR frames.

       low_power
	   For encoders set this flag to ON to reduce power consumption and
	   GPU usage.

       Runtime Options

       Following options can be used durning qsv encoding.

       global_quality
       i_quant_factor
       i_quant_offset
       b_quant_factor
       b_quant_offset
	   Supported in h264_qsv and hevc_qsv.	Change these value to reset
	   qsv codec's qp configuration.

       max_frame_size
	   Supported in h264_qsv and hevc_qsv.	Change this value to reset qsv
	   codec's MaxFrameSize configuration.

       gop_size
	   Change this value to reset qsv codec's gop configuration.

       int_ref_type
       int_ref_cycle_size
       int_ref_qp_delta
       int_ref_cycle_dist
	   Supported in h264_qsv and hevc_qsv.	Change these value to reset
	   qsv codec's Intra Refresh configuration.

       qmax
       qmin
       max_qp_i
       min_qp_i
       max_qp_p
       min_qp_p
       max_qp_b
       min_qp_b
	   Supported in h264_qsv.  Change these value to reset qsv codec's
	   max/min qp configuration.

       low_delay_brc
	   Supported in h264_qsv, hevc_qsv and av1_qsv.  Change this value to
	   reset qsv codec's low_delay_brc configuration.

       framerate
	   Change this value to reset qsv codec's framerate configuration.

       bit_rate
       rc_buffer_size
       rc_initial_buffer_occupancy
       rc_max_rate
	   Change these value to reset qsv codec's bitrate control
	   configuration.

       pic_timing_sei
	   Supported in h264_qsv and hevc_qsv.	Change this value to reset qsv
	   codec's pic_timing_sei configuration.

       qsv_params
	   Set QSV encoder parameters as a colon-separated list of key-value
	   pairs.

	   The qsv_params should be formatted as
	   "key1=value1:key2=value2:...".

	   These parameters are passed directly to the underlying Intel Quick
	   Sync Video (QSV) encoder using the MFXSetParameter function.

	   Example:

		   ffmpeg -i input.mp4 -c:v h264_qsv -qsv_params "CodingOption1=1:CodingOption2=2" output.mp4

	   This option allows fine-grained control over various encoder-
	   specific settings provided by the QSV encoder.

       H264 options

       These options are used by h264_qsv

       extbrc
	   Extended bitrate control.

       recovery_point_sei
	   Set this flag to insert the recovery point SEI message at the
	   beginning of every intra refresh cycle.

       rdo Enable rate distortion optimization.

       max_frame_size
	   Maximum encoded frame size in bytes.

       max_frame_size_i
	   Maximum encoded frame size for I frames in bytes. If this value is
	   set as larger than zero, then for I frames the value set by
	   max_frame_size is ignored.

       max_frame_size_p
	   Maximum encoded frame size for P frames in bytes. If this value is
	   set as larger than zero, then for P frames the value set by
	   max_frame_size is ignored.

       max_slice_size
	   Maximum encoded slice size in bytes.

       bitrate_limit
	   Toggle bitrate limitations.	Modifies bitrate to be in the range
	   imposed by the QSV encoder. Setting this flag off may lead to
	   violation of HRD conformance. Mind that specifying bitrate below
	   the QSV encoder range might significantly affect quality. If on
	   this option takes effect in non CQP modes: if bitrate is not in the
	   range imposed by the QSV encoder, it will be changed to be in the
	   range.

       mbbrc
	   Setting this flag enables macroblock level bitrate control that
	   generally improves subjective visual quality. Enabling this flag
	   may have negative impact on performance and objective visual
	   quality metric.

       low_delay_brc
	   Setting this flag turns on or off LowDelayBRC feautre in qsv
	   plugin, which provides more accurate bitrate control to minimize
	   the variance of bitstream size frame by frame. Value: -1-default
	   0-off 1-on

       adaptive_i
	   This flag controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       p_strategy
	   Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to
	   0).

       b_strategy
	   This option controls usage of B frames as reference.

       dblk_idc
	   This option disable deblocking. It has value in range 0~2.

       cavlc
	   If set, CAVLC is used; if unset, CABAC is used for encoding.

       vcm Video conferencing mode, please see ratecontrol method.

       idr_interval
	   Distance (in I-frames) between IDR frames.

       pic_timing_sei
	   Insert picture timing SEI with pic_struct_syntax element.

       single_sei_nal_unit
	   Put all the SEI messages into one NALU.

       max_dec_frame_buffering
	   Maximum number of frames buffered in the DPB.

       look_ahead
	   Use VBR algorithm with look ahead.

       look_ahead_depth
	   Depth of look ahead in number frames.

       look_ahead_downsampling
	   Downscaling factor for the frames saved for the lookahead analysis.

	   unknown
	   auto
	   off
	   2x
	   4x
       int_ref_type
	   Specifies intra refresh type. The major goal of intra refresh is
	   improvement of error resilience without significant impact on
	   encoded bitstream size caused by I frames. The SDK encoder achieves
	   this by encoding part of each frame in refresh cycle using intra
	   MBs. none means no refresh. vertical means vertical refresh, by
	   column of MBs. horizontal means horizontal refresh, by rows of MBs.
	   slice means horizontal refresh by slices without overlapping. In
	   case of slice, in_ref_cycle_size is ignored. To enable intra
	   refresh, B frame should be set to 0.

       int_ref_cycle_size
	   Specifies number of pictures within refresh cycle starting from 2.
	   0 and 1 are invalid values.

       int_ref_qp_delta
	   Specifies QP difference for inserted intra MBs. This is signed
	   value in [-51, 51] range if target encoding bit-depth for luma
	   samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75,
	   75] for 12 bit-depth respectively.

       int_ref_cycle_dist
	   Distance between the beginnings of the intra-refresh cycles in
	   frames.

       profile
	   unknown
	   baseline
	   main
	   high
       a53cc
	   Use A53 Closed Captions (if available).

       aud Insert the Access Unit Delimiter NAL.

       mfmode
	   Multi-Frame Mode.

	   off
	   auto
       repeat_pps
	   Repeat pps for every frame.

       max_qp_i
	   Maximum video quantizer scale for I frame.

       min_qp_i
	   Minimum video quantizer scale for I frame.

       max_qp_p
	   Maximum video quantizer scale for P frame.

       min_qp_p
	   Minimum video quantizer scale for P frame.

       max_qp_b
	   Maximum video quantizer scale for B frame.

       min_qp_b
	   Minimum video quantizer scale for B frame.

       scenario
	   Provides a hint to encoder about the scenario for the encoding
	   session.

	   unknown
	   displayremoting
	   videoconference
	   archive
	   livestreaming
	   cameracapture
	   videosurveillance
	   gamestreaming
	   remotegaming
       avbr_accuracy
	   Accuracy of the AVBR ratecontrol (unit of tenth of percent).

       avbr_convergence
	   Convergence of the AVBR ratecontrol (unit of 100 frames)

	   The parameters avbr_accuracy and avbr_convergence are for the
	   average variable bitrate control (AVBR) algorithm.  The algorithm
	   focuses on overall encoding quality while meeting the specified
	   bitrate, target_bitrate, within the accuracy range avbr_accuracy,
	   after a avbr_Convergence period. This method does not follow HRD
	   and the instant bitrate is not capped or padded.

       skip_frame
	   Use per-frame metadata "qsv_skip_frame" to skip frame when
	   encoding. This option defines the usage of this metadata.

	   no_skip
	       Frame skipping is disabled.

	   insert_dummy
	       Encoder inserts into bitstream frame where all macroblocks are
	       encoded as skipped.

	   insert_nothing
	       Similar to insert_dummy, but encoder inserts nothing into
	       bitstream. The skipped frames are still used in brc. For
	       example, gop still include skipped frames, and the frames after
	       skipped frames will be larger in size.

	   brc_only
	       skip_frame metadata indicates the number of missed frames
	       before the current frame.

       HEVC Options

       These options are used by hevc_qsv

       extbrc
	   Extended bitrate control.

       recovery_point_sei
	   Set this flag to insert the recovery point SEI message at the
	   beginning of every intra refresh cycle.

       rdo Enable rate distortion optimization.

       max_frame_size
	   Maximum encoded frame size in bytes.

       max_frame_size_i
	   Maximum encoded frame size for I frames in bytes. If this value is
	   set as larger than zero, then for I frames the value set by
	   max_frame_size is ignored.

       max_frame_size_p
	   Maximum encoded frame size for P frames in bytes. If this value is
	   set as larger than zero, then for P frames the value set by
	   max_frame_size is ignored.

       max_slice_size
	   Maximum encoded slice size in bytes.

       mbbrc
	   Setting this flag enables macroblock level bitrate control that
	   generally improves subjective visual quality. Enabling this flag
	   may have negative impact on performance and objective visual
	   quality metric.

       low_delay_brc
	   Setting this flag turns on or off LowDelayBRC feautre in qsv
	   plugin, which provides more accurate bitrate control to minimize
	   the variance of bitstream size frame by frame. Value: -1-default
	   0-off 1-on

       adaptive_i
	   This flag controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       p_strategy
	   Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to
	   0).

       b_strategy
	   This option controls usage of B frames as reference.

       dblk_idc
	   This option disable deblocking. It has value in range 0~2.

       idr_interval
	   Distance (in I-frames) between IDR frames.

	   begin_only
	       Output an IDR-frame only at the beginning of the stream.

       load_plugin
	   A user plugin to load in an internal session.

	   none
	   hevc_sw
	   hevc_hw
       load_plugins
	   A :-separate list of hexadecimal plugin UIDs to load in an internal
	   session.

       look_ahead_depth
	   Depth of look ahead in number frames, available when extbrc option
	   is enabled.

       profile
	   Set the encoding profile (scc requires libmfx >= 1.32).

	   unknown
	   main
	   main10
	   mainsp
	   rext
	   scc
       tier
	   Set the encoding tier (only level >= 4 can support high tier).
	   This option only takes effect when the level option is specified.

	   main
	   high
       gpb 1: GPB (generalized P/B frame)

	   0: regular P frame.

       tile_cols
	   Number of columns for tiled encoding.

       tile_rows
	   Number of rows for tiled encoding.

       aud Insert the Access Unit Delimiter NAL.

       pic_timing_sei
	   Insert picture timing SEI with pic_struct_syntax element.

       transform_skip
	   Turn this option ON to enable transformskip. It is supported on
	   platform equal or newer than ICL.

       int_ref_type
	   Specifies intra refresh type. The major goal of intra refresh is
	   improvement of error resilience without significant impact on
	   encoded bitstream size caused by I frames. The SDK encoder achieves
	   this by encoding part of each frame in refresh cycle using intra
	   MBs. none means no refresh. vertical means vertical refresh, by
	   column of MBs. horizontal means horizontal refresh, by rows of MBs.
	   slice means horizontal refresh by slices without overlapping. In
	   case of slice, in_ref_cycle_size is ignored. To enable intra
	   refresh, B frame should be set to 0.

       int_ref_cycle_size
	   Specifies number of pictures within refresh cycle starting from 2.
	   0 and 1 are invalid values.

       int_ref_qp_delta
	   Specifies QP difference for inserted intra MBs. This is signed
	   value in [-51, 51] range if target encoding bit-depth for luma
	   samples is 8 and this range is [-63, 63] for 10 bit-depth or [-75,
	   75] for 12 bit-depth respectively.

       int_ref_cycle_dist
	   Distance between the beginnings of the intra-refresh cycles in
	   frames.

       max_qp_i
	   Maximum video quantizer scale for I frame.

       min_qp_i
	   Minimum video quantizer scale for I frame.

       max_qp_p
	   Maximum video quantizer scale for P frame.

       min_qp_p
	   Minimum video quantizer scale for P frame.

       max_qp_b
	   Maximum video quantizer scale for B frame.

       min_qp_b
	   Minimum video quantizer scale for B frame.

       scenario
	   Provides a hint to encoder about the scenario for the encoding
	   session.

	   unknown
	   displayremoting
	   videoconference
	   archive
	   livestreaming
	   cameracapture
	   videosurveillance
	   gamestreaming
	   remotegaming
       avbr_accuracy
	   Accuracy of the AVBR ratecontrol (unit of tenth of percent).

       avbr_convergence
	   Convergence of the AVBR ratecontrol (unit of 100 frames)

	   The parameters avbr_accuracy and avbr_convergence are for the
	   average variable bitrate control (AVBR) algorithm.  The algorithm
	   focuses on overall encoding quality while meeting the specified
	   bitrate, target_bitrate, within the accuracy range avbr_accuracy,
	   after a avbr_Convergence period. This method does not follow HRD
	   and the instant bitrate is not capped or padded.

       skip_frame
	   Use per-frame metadata "qsv_skip_frame" to skip frame when
	   encoding. This option defines the usage of this metadata.

	   no_skip
	       Frame skipping is disabled.

	   insert_dummy
	       Encoder inserts into bitstream frame where all macroblocks are
	       encoded as skipped.

	   insert_nothing
	       Similar to insert_dummy, but encoder inserts nothing into
	       bitstream. The skipped frames are still used in brc. For
	       example, gop still include skipped frames, and the frames after
	       skipped frames will be larger in size.

	   brc_only
	       skip_frame metadata indicates the number of missed frames
	       before the current frame.

       MPEG2 Options

       These options are used by mpeg2_qsv

       profile
	   unknown
	   simple
	   main
	   high

       VP9 Options

       These options are used by vp9_qsv

       profile
	   unknown
	   profile0
	   profile1
	   profile2
	   profile3
       tile_cols
	   Number of columns for tiled encoding (requires libmfx >= 1.29).

       tile_rows
	   Number of rows for tiled encoding (requires libmfx  >= 1.29).

       AV1 Options

       These options are used by av1_qsv (requires libvpl).

       profile
	   unknown
	   main
       tile_cols
	   Number of columns for tiled encoding.

       tile_rows
	   Number of rows for tiled encoding.

       adaptive_i
	   This flag controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       b_strategy
	   This option controls usage of B frames as reference.

       extbrc
	   Extended bitrate control.

       look_ahead_depth
	   Depth of look ahead in number frames, available when extbrc option
	   is enabled.

       low_delay_brc
	   Setting this flag turns on or off LowDelayBRC feautre in qsv
	   plugin, which provides more accurate bitrate control to minimize
	   the variance of bitstream size frame by frame. Value: -1-default
	   0-off 1-on

       max_frame_size
	   Set the allowed max size in bytes for each frame. If the frame size
	   exceeds the limitation, encoder will adjust the QP value to control
	   the frame size.  Invalid in CQP rate control mode.

   snow
       Options

       iterative_dia_size
	   dia size for the iterative motion estimation

   VAAPI encoders
       Wrappers for hardware encoders accessible via VAAPI.

       These encoders only accept input in VAAPI hardware surfaces.  If you
       have input in software frames, use the hwupload filter to upload them
       to the GPU.

       The following standard libavcodec options are used:

       o   g / gop_size

       o   bf / max_b_frames

       o   profile

	   If not set, this will be determined automatically from the format
	   of the input frames and the profiles supported by the driver.

       o   level

       o   b / bit_rate

       o   maxrate / rc_max_rate

       o   bufsize / rc_buffer_size

       o   rc_init_occupancy / rc_initial_buffer_occupancy

       o   compression_level

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

       o   q / global_quality

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

       o   qmin

       o   qmax

       o   i_qfactor / i_quant_factor

       o   i_qoffset / i_quant_offset

       o   b_qfactor / b_quant_factor

       o   b_qoffset / b_quant_offset

       o   slices

       All encoders support the following options:

       low_power
	   Some drivers/platforms offer a second encoder for some codecs
	   intended to use less power than the default encoder; setting this
	   option will attempt to use that encoder.  Note that it may support
	   a reduced feature set, so some other options may not be available
	   in this mode.

       idr_interval
	   Set the number of normal intra frames between full-refresh (IDR)
	   frames in open-GOP mode.  The intra frames are still IRAPs, but
	   will not include global headers and may have non-decodable leading
	   pictures.

       b_depth
	   Set the B-frame reference depth.  When set to one (the default),
	   all B-frames will refer only to P- or I-frames.  When set to
	   greater values multiple layers of B-frames will be present, frames
	   in each layer only referring to frames in higher layers.

       async_depth
	   Maximum processing parallelism. Increase this to improve single
	   channel performance. This option doesn't work if driver doesn't
	   implement vaSyncBuffer function. Please make sure there are enough
	   hw_frames allocated if a large number of async_depth is used.

       max_frame_size
	   Set the allowed max size in bytes for each frame. If the frame size
	   exceeds the limitation, encoder will adjust the QP value to control
	   the frame size.  Invalid in CQP rate control mode.

       rc_mode
	   Set the rate control mode to use.  A given driver may only support
	   a subset of modes.

	   Possible modes:

	   auto
	       Choose the mode automatically based on driver support and the
	       other options.  This is the default.

	   CQP Constant-quality.

	   CBR Constant-bitrate.

	   VBR Variable-bitrate.

	   ICQ Intelligent constant-quality.

	   QVBR
	       Quality-defined variable-bitrate.

	   AVBR
	       Average variable bitrate.

       blbrc
	   Enable block level rate control, which assigns different bitrate
	   block by block.  Invalid for CQP mode.

       Each encoder also has its own specific options:

       av1_vaapi
	   profile sets the value of seq_profile.  tier sets the value of
	   seq_tier.  level sets the value of seq_level_idx.

	   tiles
	       Set the number of tiles to encode the input video with, as
	       columns x rows.	(default is auto, which means use minimal tile
	       column/row number).

	   tile_groups
	       Set tile groups number. All the tiles will be distributed as
	       evenly as possible to each tile group. (default is 1).

       h264_vaapi
	   profile sets the value of profile_idc and the
	   constraint_set*_flags.  level sets the value of level_idc.

	   coder
	       Set entropy encoder (default is cabac).	Possible values:

	       ac
	       cabac
		   Use CABAC.

	       vlc
	       cavlc
		   Use CAVLC.

	   aud Include access unit delimiters in the stream (not included by
	       default).

	   sei Set SEI message types to include.  Some combination of the
	       following values:

	       identifier
		   Include a user_data_unregistered message containing
		   information about the encoder.

	       timing
		   Include picture timing parameters (buffering_period and
		   pic_timing messages).

	       recovery_point
		   Include recovery points where appropriate (recovery_point
		   messages).

       hevc_vaapi
	   profile and level set the values of general_profile_idc and
	   general_level_idc respectively.

	   aud Include access unit delimiters in the stream (not included by
	       default).

	   tier
	       Set general_tier_flag.  This may affect the level chosen for
	       the stream if it is not explicitly specified.

	   sei Set SEI message types to include.  Some combination of the
	       following values:

	       hdr Include HDR metadata if the input frames have it
		   (mastering_display_colour_volume and content_light_level
		   messages).

	   tiles
	       Set the number of tiles to encode the input video with, as
	       columns x rows.	Larger numbers allow greater parallelism in
	       both encoding and decoding, but may decrease coding efficiency.

       mjpeg_vaapi
	   Only baseline DCT encoding is supported.  The encoder always uses
	   the standard quantisation and huffman tables - global_quality
	   scales the standard quantisation table (range 1-100).

	   For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported.
	   RGB is also supported, and will create an RGB JPEG.

	   jfif
	       Include JFIF header in each frame (not included by default).

	   huffman
	       Include standard huffman tables (on by default).  Turning this
	       off will save a few hundred bytes in each output frame, but may
	       lose compatibility with some JPEG decoders which don't fully
	       handle MJPEG.

       mpeg2_vaapi
	   profile and level set the value of profile_and_level_indication.

       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.

   vbn
       Vizrt Binary Image encoder.

       This format is used by the broadcast vendor Vizrt for quick texture
       streaming.  Advanced features of the format such as LZW compression of
       texture data or generation of mipmaps are not supported.

       Options

       format string
	   Sets the texture compression used by the VBN file. Can be dxt1,
	   dxt5 or raw. Default is dxt5.

   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

       palette
	   Specify the global palette used by the bitmaps.

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

       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.

   av1_metadata
       Modify metadata embedded in an AV1 stream.

       td  Insert or remove temporal delimiter OBUs in all temporal units of
	   the stream.

	   insert
	       Insert a TD at the beginning of every TU which does not already
	       have one.

	   remove
	       Remove the TD from the beginning of every TU which has one.

       color_primaries
       transfer_characteristics
       matrix_coefficients
	   Set the color description fields in the stream (see AV1 section
	   6.4.2).

       color_range
	   Set the color range in the stream (see AV1 section 6.4.2; note that
	   this cannot be set for streams using BT.709 primaries, sRGB
	   transfer characteristic and identity (RGB) matrix coefficients).

	   tv  Limited range.

	   pc  Full range.

       chroma_sample_position
	   Set the chroma sample location in the stream (see AV1 section
	   6.4.2).  This can only be set for 4:2:0 streams.

	   vertical
	       Left position (matching the default in MPEG-2 and H.264).

	   colocated
	       Top-left position.

       tick_rate
	   Set the tick rate (time_scale / num_units_in_display_tick) in the
	   timing info in the sequence header.

       num_ticks_per_picture
	   Set the number of ticks in each picture, to indicate that the
	   stream has a fixed framerate.  Ignored if tick_rate is not also
	   set.

       delete_padding
	   Deletes Padding OBUs.

   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 except when said
       packets already exactly begin with the extradata that is intended to be
       added.

       freq
	   The additional argument specifies which packets should be filtered.
	   It accepts the values:

	   k
	   keyframe
	       add extradata to all key packets

	   e
	   all 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

   dv_error_marker
       Blocks in DV which are marked as damaged are replaced by blocks of the
       specified color.

       color
	   The color to replace damaged blocks by

       sta A 16 bit mask which specifies which of the 16 possible error status
	   values are to be replaced by colored blocks. 0xFFFE is the default
	   which replaces all non 0 error status values.

	   ok  No error, no concealment

	   err Error, No concealment

	   res Reserved

	   notok
	       Error or concealment

	   notres
	       Not reserved

	   Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
	       The specific error status code

	   see page 44-46 or section 5.5 of
	   <http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf>

   eac3_core
       Extract the core from a E-AC-3 stream, dropping extra channels.

   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.

   filter_units
       Remove units with types in or not in a given set from the stream.

       pass_types
	   List of unit types or ranges of unit types to pass through while
	   removing all others.  This is specified as a '|'-separated list of
	   unit type values or ranges of values with '-'.

       remove_types
	   Identical to pass_types, except the units in the given set removed
	   and all others passed through.

       The types used by pass_types and remove_types correspond to NAL unit
       types (nal_unit_type) in H.264, HEVC and H.266 (see Table 7-1 in the
       H.264 and HEVC specifications or Table 5 in the H.266 specification),
       to marker values for JPEG (without 0xFF prefix) and to start codes
       without start code prefix (i.e. the byte following the 0x000001) for
       MPEG-2.	For VP8 and VP9, every unit has type zero.

       Extradata is unchanged by this transformation, but note that if the
       stream contains inline parameter sets then the output may be unusable
       if they are removed.

       For example, to remove all non-VCL NAL units from an H.264 stream:

	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT

       To remove all AUDs, SEI and filler from an H.265 stream:

	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT

       To remove all user data from a MPEG-2 stream, including Closed
       Captions:

	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=178' OUTPUT

       To remove all SEI from a H264 stream, including Closed Captions:

	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=6' OUTPUT

       To remove all prefix and suffix SEI from a HEVC stream, including
       Closed Captions and dynamic HDR:

	       ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=39|40' OUTPUT

   hapqa_extract
       Extract Rgb or Alpha part of an HAPQA file, without recompression, in
       order to create an HAPQ or an HAPAlphaOnly file.

       texture
	   Specifies the texture to keep.

	   color
	   alpha

       Convert HAPQA to HAPQ

	       ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov

       Convert HAPQA to HAPAlphaOnly

	       ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov

   h264_metadata
       Modify metadata embedded in an H.264 stream.

       aud Insert or remove AUD NAL units in all access units of the stream.

	   pass
	   insert
	   remove

	   Default is pass.

       sample_aspect_ratio
	   Set the sample aspect ratio of the stream in the VUI parameters.
	   See H.264 table E-1.

       overscan_appropriate_flag
	   Set whether the stream is suitable for display using overscan or
	   not (see H.264 section E.2.1).

       video_format
       video_full_range_flag
	   Set the video format in the stream (see H.264 section E.2.1 and
	   table E-2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
	   Set the colour description in the stream (see H.264 section E.2.1
	   and tables E-3, E-4 and E-5).

       chroma_sample_loc_type
	   Set the chroma sample location in the stream (see H.264 section
	   E.2.1 and figure E-1).

       tick_rate
	   Set the tick rate (time_scale / num_units_in_tick) in the VUI
	   parameters.	This is the smallest time unit representable in the
	   stream, and in many cases represents the field rate of the stream
	   (double the frame rate).

       fixed_frame_rate_flag
	   Set whether the stream has fixed framerate - typically this
	   indicates that the framerate is exactly half the tick rate, but the
	   exact meaning is dependent on interlacing and the picture structure
	   (see H.264 section E.2.1 and table E-6).

       zero_new_constraint_set_flags
	   Zero constraint_set4_flag and constraint_set5_flag in the SPS.
	   These bits were reserved in a previous version of the H.264 spec,
	   and thus some hardware decoders require these to be zero. The
	   result of zeroing this is still a valid bitstream.

       crop_left
       crop_right
       crop_top
       crop_bottom
	   Set the frame cropping offsets in the SPS.  These values will
	   replace the current ones if the stream is already cropped.

	   These fields are set in pixels.  Note that some sizes may not be
	   representable if the chroma is subsampled or the stream is
	   interlaced (see H.264 section 7.4.2.1.1).

       sei_user_data
	   Insert a string as SEI unregistered user data.  The argument must
	   be of the form UUID+string, where the UUID is as hex digits
	   possibly separated by hyphens, and the string can be anything.

	   For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert
	   the string ``hello'' associated with the given UUID.

       delete_filler
	   Deletes both filler NAL units and filler SEI messages.

       display_orientation
	   Insert, extract or remove Display orientation SEI messages.	See
	   H.264 section D.1.27 and D.2.27 for syntax and semantics.

	   pass
	   insert
	   remove
	   extract

	   Default is pass.

	   Insert mode works in conjunction with "rotate" and "flip" options.
	   Any pre-existing Display orientation messages will be removed in
	   insert or remove mode.  Extract mode attaches the display matrix to
	   the packet as side data.

       rotate
	   Set rotation in display orientation SEI (anticlockwise angle in
	   degrees).  Range is -360 to +360. Default is NaN.

       flip
	   Set flip in display orientation SEI.

	   horizontal
	   vertical

	   Default is unset.

       level
	   Set the level in the SPS.  Refer to H.264 section A.3 and tables
	   A-1 to A-5.

	   The argument must be the name of a level (for example, 4.2), a
	   level_idc value (for example, 42), or the special name auto
	   indicating that the filter should attempt to guess the level from
	   the input stream properties.

   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.

   h264_redundant_pps
       This applies a specific fixup to some Blu-ray streams which contain
       redundant PPSs modifying irrelevant parameters of the stream which
       confuse other transformations which require correct extradata.

   hevc_metadata
       Modify metadata embedded in an HEVC stream.

       aud Insert or remove AUD NAL units in all access units of the stream.

	   insert
	   remove
       sample_aspect_ratio
	   Set the sample aspect ratio in the stream in the VUI parameters.

       video_format
       video_full_range_flag
	   Set the video format in the stream (see H.265 section E.3.1 and
	   table E.2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
	   Set the colour description in the stream (see H.265 section E.3.1
	   and tables E.3, E.4 and E.5).

       chroma_sample_loc_type
	   Set the chroma sample location in the stream (see H.265 section
	   E.3.1 and figure E.1).

       tick_rate
	   Set the tick rate in the VPS and VUI parameters (time_scale /
	   num_units_in_tick). Combined with num_ticks_poc_diff_one, this can
	   set a constant framerate in the stream.  Note that it is likely to
	   be overridden by container parameters when the stream is in a
	   container.

       num_ticks_poc_diff_one
	   Set poc_proportional_to_timing_flag in VPS and VUI and use this
	   value to set num_ticks_poc_diff_one_minus1 (see H.265 sections
	   7.4.3.1 and E.3.1).	Ignored if tick_rate is not also set.

       crop_left
       crop_right
       crop_top
       crop_bottom
	   Set the conformance window cropping offsets in the SPS.  These
	   values will replace the current ones if the stream is already
	   cropped.

	   These fields are set in pixels.  Note that some sizes may not be
	   representable if the chroma is subsampled (H.265 section
	   7.4.3.2.1).

       width
       height
	   Set width and height after crop.

       level
	   Set the level in the VPS and SPS.  See H.265 section A.4 and tables
	   A.6 and A.7.

	   The argument must be the name of a level (for example, 5.1), a
	   general_level_idc value (for example, 153 for level 5.1), or the
	   special name auto indicating that the filter should attempt to
	   guess the level from the input stream properties.

   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.

   mpeg2_metadata
       Modify metadata embedded in an MPEG-2 stream.

       display_aspect_ratio
	   Set the display aspect ratio in the stream.

	   The following fixed values are supported:

	   4/3
	   16/9
	   221/100

	   Any other value will result in square pixels being signalled
	   instead (see H.262 section 6.3.3 and table 6-3).

       frame_rate
	   Set the frame rate in the stream.  This is constructed from a table
	   of known values combined with a small multiplier and divisor - if
	   the supplied value is not exactly representable, the nearest
	   representable value will be used instead (see H.262 section 6.3.3
	   and table 6-4).

       video_format
	   Set the video format in the stream (see H.262 section 6.3.6 and
	   table 6-6).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
	   Set the colour description in the stream (see H.262 section 6.3.6
	   and tables 6-7, 6-8 and 6-9).

   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
	   Accepts an expression whose evaluation per-packet determines how
	   often bytes in that packet will be modified. A value below 0 will
	   result in a variable frequency.  Default is 0 which results in no
	   modification. However, if neither amount nor drop is specified,
	   amount will be set to -1. See below for accepted variables.

       drop
	   Accepts an expression evaluated per-packet whose value determines
	   whether that packet is dropped.  Evaluation to a positive value
	   results in the packet being dropped. Evaluation to a negative value
	   results in a variable chance of it being dropped, roughly inverse
	   in proportion to the magnitude of the value. Default is 0 which
	   results in no drops. See below for accepted variables.

       dropamount
	   Accepts a non-negative integer, which assigns a variable chance of
	   it being dropped, roughly inverse in proportion to the value.
	   Default is 0 which results in no drops. This option is kept for
	   backwards compatibility and is equivalent to setting drop to a
	   negative value with the same magnitude i.e. "dropamount=4" is the
	   same as "drop=-4". Ignored if drop is also specified.

       Both "amount" and "drop" accept expressions containing the following
       variables:

       n   The index of the packet, starting from zero.

       tb  The timebase for packet timestamps.

       pts Packet presentation timestamp.

       dts Packet decoding timestamp.

       nopts
	   Constant representing AV_NOPTS_VALUE.

       startpts
	   First non-AV_NOPTS_VALUE PTS seen in the stream.

       startdts
	   First non-AV_NOPTS_VALUE DTS seen in the stream.

       duration
       d   Packet duration, in timebase units.

       pos Packet position in input; may be -1 when unknown or not set.

       size
	   Packet size, in bytes.

       key Whether packet is marked as a keyframe.

       state
	   A pseudo random integer, primarily derived from the content of
	   packet payload.

       Examples

       Apply modification to every byte but don't drop any packets.

	       ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv

       Drop every video packet not marked as a keyframe after timestamp 30s
       but do not modify any of the remaining packets.

	       ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv

       Drop one second of audio every 10 seconds and add some random noise to
       the rest.

	       ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv

   null
       This bitstream filter passes the packets through unchanged.

   pcm_rechunk
       Repacketize PCM audio to a fixed number of samples per packet or a
       fixed packet rate per second. This is similar to the asetnsamples audio
       filter but works on audio packets instead of audio frames.

       nb_out_samples, n
	   Set the number of samples per each output audio packet. 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 packet with
	   silence, so that it will contain the same number of samples (or
	   roughly the same number of samples, see frame_rate) as the previous
	   ones. Default value is 1.

       frame_rate, r
	   This option makes the filter output a fixed number of packets per
	   second instead of a fixed number of samples per packet. If the
	   audio sample rate is not divisible by the frame rate then the
	   number of samples will not be constant but will vary slightly so
	   that each packet will start as close to the frame boundary as
	   possible. Using this option has precedence over nb_out_samples.

       You can generate the well known 1602-1601-1602-1601-1602 pattern of
       48kHz audio for NTSC frame rate using the frame_rate option.

	       ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -

   pgs_frame_merge
       Merge a sequence of PGS Subtitle segments ending with an "end of
       display set" segment into a single packet.

       This is required by some containers that support PGS subtitles (muxer
       "matroska").

   prores_metadata
       Modify color property metadata embedded in prores stream.

       color_primaries
	   Set the color primaries.  Available values are:

	   auto
	       Keep the same color primaries property (default).

	   unknown
	   bt709
	   bt470bg
	       BT601 625

	   smpte170m
	       BT601 525

	   bt2020
	   smpte431
	       DCI P3

	   smpte432
	       P3 D65

       transfer_characteristics
	   Set the color transfer.  Available values are:

	   auto
	       Keep the same transfer characteristics property (default).

	   unknown
	   bt709
	       BT 601, BT 709, BT 2020

	   smpte2084
	       SMPTE ST 2084

	   arib-std-b67
	       ARIB STD-B67

       matrix_coefficients
	   Set the matrix coefficient.	Available values are:

	   auto
	       Keep the same colorspace property (default).

	   unknown
	   bt709
	   smpte170m
	       BT 601

	   bt2020nc

       Set Rec709 colorspace for each frame of the file

	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov

       Set Hybrid Log-Gamma parameters for each frame of the file

	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov

   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.

   setts
       Set PTS and DTS in packets.

       It accepts the following parameters:

       ts
       pts
       dts Set expressions for PTS, DTS or both.

       duration
	   Set expression for duration.

       time_base
	   Set output time base.

       The expressions are evaluated through the eval API and can contain the
       following constants:

       N   The count of the input packet. Starting from 0.

       TS  The demux timestamp in input in case of "ts" or "dts" option or
	   presentation timestamp in case of "pts" option.

       POS The original position in the file of the packet, or undefined if
	   undefined for the current packet

       DTS The demux timestamp in input.

       PTS The presentation timestamp in input.

       DURATION
	   The duration in input.

       STARTDTS
	   The DTS of the first packet.

       STARTPTS
	   The PTS of the first packet.

       PREV_INDTS
	   The previous input DTS.

       PREV_INPTS
	   The previous input PTS.

       PREV_INDURATION
	   The previous input duration.

       PREV_OUTDTS
	   The previous output DTS.

       PREV_OUTPTS
	   The previous output PTS.

       PREV_OUTDURATION
	   The previous output duration.

       NEXT_DTS
	   The next input DTS.

       NEXT_PTS
	   The next input PTS.

       NEXT_DURATION
	   The next input duration.

       TB  The timebase of stream packet belongs.

       TB_OUT
	   The output timebase.

       SR  The sample rate of stream packet belongs.

       NOPTS
	   The AV_NOPTS_VALUE constant.

       For example, to set PTS equal to DTS (not recommended if B-frames are
       involved):

	       ffmpeg -i INPUT -c:a copy -bsf:a setts=pts=DTS out.mkv

   showinfo
       Log basic packet information. Mainly useful for testing, debugging, and
       development.

   text2movsub
       Convert text subtitles to MOV subtitles (as used by the "mov_text"
       codec) with metadata headers.

       See also the mov2textsub filter.

   trace_headers
       Log trace output containing all syntax elements in the coded stream
       headers (everything above the level of individual coded blocks).  This
       can be useful for debugging low-level stream issues.

       Supports AV1, H.264, H.265, (M)JPEG, MPEG-2 and VP9, but depending on
       the build only a subset of these may be available.

   truehd_core
       Extract the core from a TrueHD stream, dropping ATMOS data.

   vp9_metadata
       Modify metadata embedded in a VP9 stream.

       color_space
	   Set the color space value in the frame header.  Note that any frame
	   set to RGB will be implicitly set to PC range and that RGB is
	   incompatible with profiles 0 and 2.

	   unknown
	   bt601
	   bt709
	   smpte170
	   smpte240
	   bt2020
	   rgb
       color_range
	   Set the color range value in the frame header.  Note that any value
	   imposed by the color space will take precedence over this value.

	   tv
	   pc

   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.

       max_probe_packets integer (input)
	   Set the maximum number of buffered packets when probing a codec.
	   Default is 2500 packets.

       packetsize integer (output)
	   Set packet size.

       fflags flags
	   Set format flags. Some are implemented for a limited number of
	   formats.

	   Possible values for input files:

	   discardcorrupt
	       Discard corrupted packets.

	   fastseek
	       Enable fast, but inaccurate seeks for some formats.

	   genpts
	       Generate missing PTS if DTS is present.

	   igndts
	       Ignore DTS if PTS is also set. In case the PTS is set, the DTS
	       value is set to NOPTS. This is ignored when the "nofillin" flag
	       is set.

	   ignidx
	       Ignore index.

	   nobuffer
	       Reduce the latency introduced by buffering during initial input
	       streams analysis.

	   nofillin
	       Do not fill in missing values in packet fields that can be
	       exactly calculated.

	   noparse
	       Disable AVParsers, this needs "+nofillin" too.

	   sortdts
	       Try to interleave output packets by DTS. At present, available
	       only for AVIs with an index.

	   Possible values for output files:

	   autobsf
	       Automatically apply bitstream filters as required by the output
	       format. Enabled by default.

	   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.

	   flush_packets
	       Write out packets immediately.

	   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 10000000 (10 seconds).

	   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 indentation:

		   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.

       skip_estimate_duration_from_pts bool (input)
	   Skip estimation of input duration if it requires an additional
	   probing for PTS at end of file.  At present, applicable for MPEG-PS
	   and MPEG-TS.

       duration_probesize integer (input)
	   Set probing size, in bytes, for input duration estimation when it
	   actually requires an additional probing for PTS at end of file (at
	   present: MPEG-PS and MPEG-TS).  It is aimed at users interested in
	   better durations probing for itself, or indirectly because using
	   the concat demuxer, for example.  The typical use case is an MPEG-
	   TS CBR with a high bitrate, high video buffering and ending
	   cleaning with similar PTS for video and audio: in such a scenario,
	   the large physical gap between the last video packet and the last
	   audio packet makes it necessary to read many bytes in order to get
	   the video stream duration.  Another use case is where the default
	   probing behaviour only reaches a single video frame which is not
	   the last one of the stream due to frame reordering, so the duration
	   is not accurate.  Setting this option has a performance impact even
	   for small files because the probing size is fixed.  Default
	   behaviour is a general purpose trade-off, largely adaptive, but the
	   probing size will not be extended to get streams durations at all
	   costs.  Must be an integer not lesser than 1, or 0 for default
	   behaviour.

       strict, f_strict integer (input/output)
	   Specify how strictly to follow the standards. "f_strict" is
	   deprecated and should be used only via the ffmpeg tool.

	   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.

   Format stream specifiers
       Format stream specifiers allow selection of one or more streams that
       match specific properties.

       The exact semantics of stream specifiers is defined by the
       "avformat_match_stream_specifier()" function declared in the
       libavformat/avformat.h header and documented in the Stream specifiers
       section in the ffmpeg(1) manual.

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.

   aac
       Raw Audio Data Transport Stream AAC demuxer.

       This demuxer is used to demux an ADTS input containing a single AAC
       stream alongwith any ID3v1/2 or APE tags in it.

   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. Default is enabled.

       -max_fps int
	   Maximum framerate in frames per second. Default of 0 imposes no
	   limit.

       -default_fps int
	   Default framerate in frames per second when none is specified in
	   the file (0 meaning as fast as possible). Default is 15.

   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.

	   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.  This directive is
	   deprecated, use "file_packet_meta" instead.

       "file_packet_meta 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.

       "option key value"
	   Option to access, open and probe the file.  Can be present multiple
	   times.

       "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.

       "stream_meta key value"
	   Metadata for the stream.  Can be present multiple times.

       "stream_codec value"
	   Codec for the stream.

       "stream_extradata hex_string"
	   Extradata for the string, encoded in hexadecimal.

       "chapter id start end"
	   Add a chapter. id is an unique identifier, possibly small and
	   consecutive.

       Options

       This demuxer accepts the following option:

       safe
	   If set to 1, reject unsafe file paths and directives.  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.

       auto_convert
	   If set to 1, try to perform automatic conversions on packet data to
	   make the streams concatenable.  The default is 1.

	   Currently, the only conversion is adding the h264_mp4toannexb
	   bitstream filter to H.264 streams in MP4 format. This is necessary
	   in particular if there are resolution changes.

       segment_time_metadata
	   If set to 1, every packet will contain the lavf.concat.start_time
	   and the lavf.concat.duration packet metadata values which are the
	   start_time and the duration of the respective file segments in the
	   concatenated output expressed in microseconds. The duration
	   metadata is only set if it is known based on the concat file.  The
	   default is 0.

       Examples

       o   Use absolute filenames and include some comments:

		   # my first filename
		   file /mnt/share/file-1.wav
		   # my second filename including whitespace
		   file '/mnt/share/file 2.wav'
		   # my third filename including whitespace plus single quote
		   file '/mnt/share/file 3'\''.wav'

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

		   ffconcat version 1.0

		   file file-1.wav
		   duration 20.0

		   file subdir/file-2.wav

   dash
       Dynamic Adaptive Streaming over HTTP demuxer.

       This demuxer presents all AVStreams found in the manifest.  By setting
       the discard flags on AVStreams the caller can decide which streams to
       actually receive.  Each stream mirrors the "id" and "bandwidth"
       properties from the "<Representation>" as metadata keys named "id" and
       "variant_bitrate" respectively.

       Options

       This demuxer accepts the following option:

       cenc_decryption_key
	   16-byte key, in hex, to decrypt files encrypted using ISO Common
	   Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

   dvdvideo
       DVD-Video demuxer, powered by libdvdnav and libdvdread.

       Can directly ingest DVD titles, specifically sequential PGCs, into a
       conversion pipeline. Menu assets, such as background video or audio,
       can also be demuxed given the menu's coordinates (at best effort).
       Seeking is not supported at this time.

       Block devices (DVD drives), ISO files, and directory structures are
       accepted.  Activate with "-f dvdvideo" in front of one of these inputs.

       This demuxer does NOT have decryption code of any kind. You are on your
       own working with encrypted DVDs, and should not expect support on the
       matter.

       Underlying playback is handled by libdvdnav, and structure parsing by
       libdvdread.  FFmpeg must be built with GPL library support available as
       well as the configure switches "--enable-libdvdnav" and
       "--enable-libdvdread".

       You will need to provide either the desired "title number" or exact
       PGC/PG coordinates.  Many open-source DVD players and tools can aid in
       providing this information.  If not specified, the demuxer will default
       to title 1 which works for many discs.  However, due to the flexibility
       of the format, it is recommended to check manually.  There are many
       discs that are authored strangely or with invalid headers.

       If the input is a real DVD drive, please note that there are some
       drives which may silently fail on reading bad sectors from the disc,
       returning random bits instead which is effectively corrupt data. This
       is especially prominent on aging or rotting discs.  A second pass and
       integrity checks would be needed to detect the corruption.  This is not
       an FFmpeg issue.

       Background

       DVD-Video is not a directly accessible, linear container format in the
       traditional sense. Instead, it allows for complex and programmatic
       playback of carefully muxed MPEG-PS streams that are stored in
       headerless VOB files.  To the end-user, these streams are known simply
       as "titles", but the actual logical playback sequence is defined by one
       or more "PGCs", or Program Group Chains, within the title. The PGC is
       in turn comprised of multiple "PGs", or Programs", which are the actual
       video segments (and for a typical video feature, sequentially ordered).
       The PGC structure, along with stream layout and metadata, are stored in
       IFO files that need to be parsed. PGCs can be thought of as playlists
       in easier terms.

       An actual DVD player relies on user GUI interaction via menus and an
       internal VM to drive the direction of demuxing. Generally, the user
       would either navigate (via menus) or automatically be redirected to the
       PGC of their choice. During this process and the subsequent playback,
       the DVD player's internal VM also maintains a state and executes
       instructions that can create jumps to different sectors during
       playback.  This is why libdvdnav is involved, as a linear read of the
       MPEG-PS blobs on the disc (VOBs) is not enough to produce the right
       sequence in many cases.

       There are many other DVD structures (a long subject) that will not be
       discussed here.	NAV packets, in particular, are handled by this
       demuxer to build accurate timing but not emitted as a stream. For a
       good high-level understanding, refer to:
       <https://code.videolan.org/videolan/libdvdnav/-/blob/master/doc/dvd_structures>

       Options

       This demuxer accepts the following options:

       title int
	   The title number to play. Must be set if pgc and pg are not set.
	   Not applicable to menus.  Default is 0 (auto), which currently only
	   selects the first available title (title 1) and notifies the user
	   about the implications.

       chapter_start int
	   The chapter, or PTT (part-of-title), number to start at. Not
	   applicable to menus.  Default is 1.

       chapter_end int
	   The chapter, or PTT (part-of-title), number to end at. Not
	   applicable to menus.  Default is 0, which is a special value to
	   signal end at the last possible chapter.

       angle int
	   The video angle number, referring to what is essentially an
	   additional video stream that is composed from alternate frames
	   interleaved in the VOBs.  Not applicable to menus.  Default is 1.

       region int
	   The region code to use for playback. Some discs may use this to
	   default playback at a particular angle in different regions. This
	   option will not affect the region code of a real DVD drive, if used
	   as an input. Not applicable to menus.  Default is 0, "world".

       menu bool
	   Demux menu assets instead of navigating a title. Requires exact
	   coordinates of the menu (menu_lu, menu_vts, pgc, pg).  Default is
	   false.

       menu_lu int
	   The menu language to demux. In DVD, menus are grouped by language.
	   Default is 0, the first language unit.

       menu_vts int
	   The VTS where the menu lives, or 0 if it is a VMG menu (root-
	   level).  Default is 0, VMG menu.

       pgc int
	   The entry PGC to start playback, in conjunction with pg.
	   Alternative to setting title.  Chapter markers are not supported at
	   this time.  Must be explicitly set for menus.  Default is 0,
	   automatically resolve from value of title.

       pg int
	   The entry PG to start playback, in conjunction with pgc.
	   Alternative to setting title.  Chapter markers are not supported at
	   this time.  Default is 0, automatically resolve from value of
	   title, or start from the beginning (PG 1) of the menu.

       preindex bool
	   Enable this to have accurate chapter (PTT) markers and duration
	   measurement, which requires a slow second pass read in order to
	   index the chapter marker timestamps from NAV packets. This is non-
	   ideal extra work for real optical drives.  It is recommended and
	   faster to use this option with a backup of the DVD structure stored
	   on a hard drive. Not compatible with pgc and pg.  Not applicable to
	   menus.  Default is 0, false.

       trim bool
	   Skip padding cells (i.e. cells shorter than 1 second) from the
	   beginning.  There exist many discs with filler segments at the
	   beginning of the PGC, often with junk data intended for controlling
	   a real DVD player's buffering speed and with no other material data
	   value.  Not applicable to menus.  Default is 1, true.

       Examples

       o   Open title 3 from a given DVD structure:

		   ffmpeg -f dvdvideo -title 3 -i <path to DVD> ...

       o   Open chapters 3-6 from title 1 from a given DVD structure:

		   ffmpeg -f dvdvideo -chapter_start 3 -chapter_end 6 -title 1 -i <path to DVD> ...

       o   Open only chapter 5 from title 1 from a given DVD structure:

		   ffmpeg -f dvdvideo -chapter_start 5 -chapter_end 5 -title 1 -i <path to DVD> ...

       o   Demux menu with language 1 from VTS 1, PGC 1, starting at PG 1:

		   ffmpeg -f dvdvideo -menu 1 -menu_lu 1 -menu_vts 1 -pgc 1 -pg 1 -i <path to DVD> ...

   ea
       Electronic Arts Multimedia format demuxer.

       This format is used by various Electronic Arts games.

       Options

       merge_alpha bool
	   Normally the VP6 alpha channel (if exists) is returned as a
	   secondary video stream, by setting this option you can make the
	   demuxer return a single video stream which contains the alpha
	   channel in addition to the ordinary video.

   imf
       Interoperable Master Format demuxer.

       This demuxer presents audio and video streams found in an IMF
       Composition, as specified in
       <https://doi.org/10.5594/SMPTE.ST2067-2.2020>.

	       ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...

       If "-assetmaps" is not specified, the demuxer looks for a file called
       ASSETMAP.xml in the same directory as the CPL.

   flv, live_flv, kux
       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.  KUX is a
       flv variant used on the Youku platform.

	       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.

       -flv_ignore_prevtag bool
	   Ignore the size of previous tag value.

       -flv_full_metadata bool
	   Output all context of the onMetadata.

   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

       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".

       It accepts the following options:

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

       prefer_x_start
	   prefer to use #EXT-X-START if it's in playlist instead of
	   live_start_index.

       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.

       m3u8_hold_counters
	   The maximum number of times to load m3u8 when it refreshes without
	   new segments.  Default value is 1000.

       http_persistent
	   Use persistent HTTP connections. Applicable only for HTTP streams.
	   Enabled by default.

       http_multiple
	   Use multiple HTTP connections for downloading HTTP segments.
	   Enabled by default for HTTP/1.1 servers.

       http_seekable
	   Use HTTP partial requests for downloading HTTP segments.  0 =
	   disable, 1 = enable, -1 = auto, Default is auto.

       seg_format_options
	   Set options for the demuxer of media segments using a list of
	   key=value pairs separated by ":".

       seg_max_retry
	   Maximum number of times to reload a segment on error, useful when
	   segment skip on network error is not desired.  Default value is 0.

   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.

       export_path_metadata
	   If set to 1, will add two extra fields to the metadata found in
	   input, making them also available for other filters (see drawtext
	   filter for examples). Default value is 0. The extra fields are
	   described below:

	   lavf.image2dec.source_path
	       Corresponds to the full path to the input file being read.

	   lavf.image2dec.source_basename
	       Corresponds to the name of the file being read.

       Examples

       o   Use ffmpeg for creating a video from the images in the file
	   sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame
	   rate of 10 frames per second:

		   ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

       o   As above, but start by reading from a file with index 100 in the
	   sequence:

		   ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

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

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

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

       See <https://bitbucket.org/mpyne/game-music-emu/overview> for more
       information.

       It accepts the following options:

       track_index
	   Set the index of which track to demux. The demuxer can only export
	   one track.  Track indexes start at 0. Default is to pick the first
	   track. Number of tracks is exported as tracks metadata entry.

       sample_rate
	   Set the sampling rate of the exported track. Range is 1000 to
	   999999. Default is 44100.

       max_size (bytes)
	   The demuxer buffers the entire file into memory. Adjust this value
	   to set the maximum buffer size, which in turn, acts as a ceiling
	   for the size of files that can be read.  Default is 50 MiB.

   libmodplug
       ModPlug based module demuxer

       See <https://github.com/Konstanty/libmodplug>

       It will export one 2-channel 16-bit 44.1 kHz audio stream.  Optionally,
       a "pal8" 16-color video stream can be exported with or without printed
       metadata.

       It accepts the following options:

       noise_reduction
	   Apply a simple low-pass filter. Can be 1 (on) or 0 (off). Default
	   is 0.

       reverb_depth
	   Set amount of reverb. Range 0-100. Default is 0.

       reverb_delay
	   Set delay in ms, clamped to 40-250 ms. Default is 0.

       bass_amount
	   Apply bass expansion a.k.a. XBass or megabass. Range is 0 (quiet)
	   to 100 (loud). Default is 0.

       bass_range
	   Set cutoff i.e. upper-bound for bass frequencies. Range is 10-100
	   Hz. Default is 0.

       surround_depth
	   Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to 100
	   (heavy). Default is 0.

       surround_delay
	   Set surround delay in ms, clamped to 5-40 ms. Default is 0.

       max_size
	   The demuxer buffers the entire file into memory. Adjust this value
	   to set the maximum buffer size, which in turn, acts as a ceiling
	   for the size of files that can be read. Range is 0 to 100 MiB.  0
	   removes buffer size limit (not recommended). Default is 5 MiB.

       video_stream_expr
	   String which is evaluated using the eval API to assign colors to
	   the generated video stream.	Variables which can be used are "x",
	   "y", "w", "h", "t", "speed", "tempo", "order", "pattern" and "row".

       video_stream
	   Generate video stream. Can be 1 (on) or 0 (off). Default is 0.

       video_stream_w
	   Set video frame width in 'chars' where one char indicates 8 pixels.
	   Range is 20-512. Default is 30.

       video_stream_h
	   Set video frame height in 'chars' where one char indicates 8
	   pixels. Range is 20-512. Default is 30.

       video_stream_ptxt
	   Print metadata on video stream. Includes "speed", "tempo", "order",
	   "pattern", "row" and "ts" (time in ms). Can be 1 (on) or 0 (off).
	   Default is 1.

   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
       Demuxer for Quicktime File Format & ISO/IEC Base Media File Format
       (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part
       12).

       Registered extensions: mov, mp4, m4a, 3gp, 3g2, mj2, psp, m4b, ism,
       ismv, isma, f4v

       Options

       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.

       seek_streams_individually
	   When seeking, identify the closest point in each stream
	   individually and demux packets in that stream from identified
	   point. This can lead to a different sequence of packets compared to
	   demuxing linearly from the beginning. Default is true.

       ignore_editlist
	   Ignore any edit list atoms. The demuxer, by default, modifies the
	   stream index to reflect the timeline described by the edit list.
	   Default is false.

       advanced_editlist
	   Modify the stream index to reflect the timeline described by the
	   edit list. "ignore_editlist" must be set to false for this option
	   to be effective.  If both "ignore_editlist" and this option are set
	   to false, then only the start of the stream index is modified to
	   reflect initial dwell time or starting timestamp described by the
	   edit list. Default is true.

       ignore_chapters
	   Don't parse chapters. This includes GoPro 'HiLight' tags/moments.
	   Note that chapters are only parsed when input is seekable. Default
	   is false.

       use_mfra_for
	   For seekable fragmented input, set fragment's starting timestamp
	   from media fragment random access box, if present.

	   Following options are available:

	   auto
	       Auto-detect whether to set mfra timestamps as PTS or DTS
	       (default)

	   dts Set mfra timestamps as DTS

	   pts Set mfra timestamps as PTS

	   0   Don't use mfra box to set timestamps

       use_tfdt
	   For fragmented input, set fragment's starting timestamp to
	   "baseMediaDecodeTime" from the "tfdt" box.  Default is enabled,
	   which will prefer to use the "tfdt" box to set DTS. Disable to use
	   the "earliest_presentation_time" from the "sidx" box.  In either
	   case, the timestamp from the "mfra" box will be used if it's
	   available and "use_mfra_for" is set to pts or dts.

       export_all
	   Export unrecognized boxes within the udta box as metadata entries.
	   The first four characters of the box type are set as the key.
	   Default is false.

       export_xmp
	   Export entire contents of XMP_ box and uuid box as a string with
	   key "xmp". Note that if "export_all" is set and this option isn't,
	   the contents of XMP_ box are still exported but with key "XMP_".
	   Default is false.

       activation_bytes
	   4-byte key required to decrypt Audible AAX and AAX+ files. See
	   Audible AAX subsection below.

       audible_fixed_key
	   Fixed key used for handling Audible AAX/AAX+ files. It has been
	   pre-set so should not be necessary to specify.

       decryption_key
	   16-byte key, in hex, to decrypt files encrypted using ISO Common
	   Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

       max_stts_delta
	   Very high sample deltas written in a trak's stts box may
	   occasionally be intended but usually they are written in error or
	   used to store a negative value for dts correction when treated as
	   signed 32-bit integers. This option lets the user set an upper
	   limit, beyond which the delta is clamped to 1. Values greater than
	   the limit if negative when cast to int32 are used to adjust onward
	   dts.

	   Unit is the track time scale. Range is 0 to UINT_MAX. Default is
	   "UINT_MAX - 48000*10" which allows up to a 10 second dts correction
	   for 48 kHz audio streams while accommodating 99.9% of "uint32"
	   range.

       interleaved_read
	   Interleave packets from multiple tracks at demuxer level. For badly
	   interleaved files, this prevents playback issues caused by large
	   gaps between packets in different tracks, as MOV/MP4 do not have
	   packet placement requirements.  However, this can cause excessive
	   seeking on very badly interleaved files, due to seeking between
	   tracks, so disabling it may prevent I/O issues, at the expense of
	   playback.

       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

   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.

       skip_unknown_pmt
	   Skip PMTs for programs not defined in the PAT. Default value is 0.

       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.

       merge_pmt_versions
	   Re-use existing streams when a PMT's version is updated and
	   elementary streams move to different PIDs. Default value is 0.

       max_packet_size
	   Set maximum size, in bytes, of packet emitted by the demuxer.
	   Payloads above this size are split across multiple packets. Range
	   is 1 to INT_MAX/2. Default is 204800 bytes.

   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

   rcwt
       RCWT (Raw Captions With Time) is a format native to ccextractor, a
       commonly used open source tool for processing 608/708 Closed Captions
       (CC) sources.  For more information on the format, see .

       This demuxer implements the specification as of March 2024, which has
       been stable and unchanged since April 2014.

       Examples

       o   Render CC to ASS using the built-in decoder:

		   ffmpeg -i CC.rcwt.bin CC.ass

	   Note that if your output appears to be empty, you may have to
	   manually set the decoder's data_field option to pick the desired CC
	   substream.

       o   Convert an RCWT backup to Scenarist (SCC) format:

		   ffmpeg -i CC.rcwt.bin -c:s copy CC.scc

	   Note that the SCC format does not support all of the possible CC
	   extensions that can be stored in RCWT (such as EIA-708).

   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

   vapoursynth
       Vapoursynth wrapper.

       Due to security concerns, Vapoursynth scripts will not be autodetected
       so the input format has to be forced. For ff* CLI tools, add "-f
       vapoursynth" before the input "-i yourscript.vpy".

       This demuxer accepts the following option:

       max_script_size
	   The demuxer buffers the entire script into memory. Adjust this
	   value to set the maximum buffer size, which in turn, acts as a
	   ceiling for the size of scripts that can be read.  Default is 1
	   MiB.

   w64
       Sony Wave64 Audio demuxer.

       This demuxer accepts the following options:

       max_size
	   See the same option for the wav demuxer.

   wav
       RIFF Wave Audio demuxer.

       This demuxer accepts the following options:

       max_size
	   Specify the maximum packet size in bytes for the demuxed packets.
	   By default this is set to 0, which means that a sensible value is
	   chosen based on the input format.

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.

   Raw muxers
       This section covers raw muxers. They accept a single stream matching
       the designated codec. They do not store timestamps or metadata. The
       recognized extension is the same as the muxer name unless indicated
       otherwise.

       It comprises the following muxers. The media type and the eventual
       extensions used to automatically selects the muxer from the output
       extensions are also shown.

       ac3 audio
	   Dolby Digital, also known as AC-3.

       adx audio
	   CRI Middleware ADX audio.

	   This muxer will write out the total sample count near the start of
	   the first packet when the output is seekable and the count can be
	   stored in 32 bits.

       aptx audio
	   aptX (Audio Processing Technology for Bluetooth)

       aptx_hd audio (aptxdh)
	   aptX HD (Audio Processing Technology for Bluetooth) audio

       avs2 video (avs, avs2)
	   AVS2-P2 (Audio Video Standard - Second generation - Part 2) / IEEE
	   1857.4 video

       avs3 video (avs3)
	   AVS3-P2 (Audio Video Standard - Third generation - Part 2) / IEEE
	   1857.10 video

       cavsvideo video (cavs)
	   Chinese AVS (Audio Video Standard - First generation)

       codec2raw audio
	   Codec 2 audio.

	   No extension is registered so format name has to be supplied e.g.
	   with the ffmpeg CLI tool "-f codec2raw".

       data any
	   Generic data muxer.

	   This muxer accepts a single stream with any codec of any type. The
	   input stream has to be selected using the "-map" option with the
	   ffmpeg CLI tool.

	   No extension is registered so format name has to be supplied e.g.
	   with the ffmpeg CLI tool "-f data".

       dfpwm audio (dfpwm)
	   Raw DFPWM1a (Dynamic Filter Pulse With Modulation) audio muxer.

       dirac video (drc, vc2)
	   BBC Dirac video.

	   The Dirac Pro codec is a subset and is standardized as SMPTE VC-2.

       dnxhd video (dnxhd, dnxhr)
	   Avid DNxHD video.

	   It is standardized as SMPTE VC-3. Accepts DNxHR streams.

       dts audio
	   DTS Coherent Acoustics (DCA) audio

       eac3 audio
	   Dolby Digital Plus, also known as Enhanced AC-3

       evc video (evc)
	   MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video

       g722 audio
	   ITU-T G.722 audio

       g723_1 audio (tco, rco)
	   ITU-T G.723.1 audio

       g726 audio
	   ITU-T G.726 big-endian ("left-justified") audio.

	   No extension is registered so format name has to be supplied e.g.
	   with the ffmpeg CLI tool "-f g726".

       g726le audio
	   ITU-T G.726 little-endian ("right-justified") audio.

	   No extension is registered so format name has to be supplied e.g.
	   with the ffmpeg CLI tool "-f g726le".

       gsm audio
	   Global System for Mobile Communications audio

       h261 video
	   ITU-T H.261 video

       h263 video
	   ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2
	   video

       h264 video (h264, 264)
	   ITU-T H.264 / MPEG-4 Part 10 AVC video. Bitstream shall be
	   converted to Annex B syntax if it's in length-prefixed mode.

       hevc video (hevc, h265, 265)
	   ITU-T H.265 / MPEG-H Part 2 HEVC video. Bitstream shall be
	   converted to Annex B syntax if it's in length-prefixed mode.

       m4v video
	   MPEG-4 Part 2 video

       mjpeg video (mjpg, mjpeg)
	   Motion JPEG video

       mlp audio
	   Meridian Lossless Packing, also known as Packed PCM

       mp2 audio (mp2, m2a, mpa)
	   MPEG-1 Audio Layer II audio

       mpeg1video video (mpg, mpeg, m1v)
	   MPEG-1 Part 2 video.

       mpeg2video video (m2v)
	   ITU-T H.262 / MPEG-2 Part 2 video

       obu video
	   AV1 low overhead Open Bitstream Units muxer.

	   Temporal delimiter OBUs will be inserted in all temporal units of
	   the stream.

       rawvideo video (yuv, rgb)
	   Raw uncompressed video.

       sbc audio (sbc, msbc)
	   Bluetooth SIG low-complexity subband codec audio

       truehd audio (thd)
	   Dolby TrueHD audio

       vc1 video
	   SMPTE 421M / VC-1 video

       Examples

       o   Store raw video frames with the rawvideo muxer using ffmpeg:

		   ffmpeg -f lavfi -i testsrc -t 10 -s hd1080p testsrc.yuv

	   Since the rawvideo muxer do not store the information related to
	   size and format, this information must be provided when demuxing
	   the file:

		   ffplay -video_size 1920x1080 -pixel_format rgb24 -f rawvideo testsrc.rgb

   Raw PCM muxers
       This section covers raw PCM (Pulse-Code Modulation) audio muxers.

       They accept a single stream matching the designated codec. They do not
       store timestamps or metadata. The recognized extension is the same as
       the muxer name.

       It comprises the following muxers. The optional additional extension
       used to automatically select the muxer from the output extension is
       also shown in parentheses.

       alaw (al)
	   PCM A-law

       f32be
	   PCM 32-bit floating-point big-endian

       f32le
	   PCM 32-bit floating-point little-endian

       f64be
	   PCM 64-bit floating-point big-endian

       f64le
	   PCM 64-bit floating-point little-endian

       mulaw (ul)
	   PCM mu-law

       s16be
	   PCM signed 16-bit big-endian

       s16le
	   PCM signed 16-bit little-endian

       s24be
	   PCM signed 24-bit big-endian

       s24le
	   PCM signed 24-bit little-endian

       s32be
	   PCM signed 32-bit big-endian

       s32le
	   PCM signed 32-bit little-endian

       s8 (sb)
	   PCM signed 8-bit

       u16be
	   PCM unsigned 16-bit big-endian

       u16le
	   PCM unsigned 16-bit little-endian

       u24be
	   PCM unsigned 24-bit big-endian

       u24le
	   PCM unsigned 24-bit little-endian

       u32be
	   PCM unsigned 32-bit big-endian

       u32le
	   PCM unsigned 32-bit little-endian

       u8 (ub)
	   PCM unsigned 8-bit

       vidc
	   PCM Archimedes VIDC

   MPEG-1/MPEG-2 program stream muxers
       This section covers formats belonging to the MPEG-1 and MPEG-2 Systems
       family.

       The MPEG-1 Systems format (also known as ISO/IEEC 11172-1 or MPEG-1
       program stream) has been adopted for the format of media track stored
       in VCD (Video Compact Disc).

       The MPEG-2 Systems standard (also known as ISO/IEEC 13818-1) covers two
       containers formats, one known as transport stream and one known as
       program stream; only the latter is covered here.

       The MPEG-2 program stream format (also known as VOB due to the
       corresponding file extension) is an extension of MPEG-1 program stream:
       in addition to support different codecs for the audio and video
       streams, it also stores subtitles and navigation metadata.  MPEG-2
       program stream has been adopted for storing media streams in SVCD and
       DVD storage devices.

       This section comprises the following muxers.

       mpeg (mpg,mpeg)
	   MPEG-1 Systems / MPEG-1 program stream muxer.

       vcd MPEG-1 Systems / MPEG-1 program stream (VCD) muxer.

	   This muxer can be used to generate tracks in the format accepted by
	   the VCD (Video Compact Disc) storage devices.

	   It is the same as the mpeg muxer with a few differences.

       vob MPEG-2 program stream (VOB) muxer.

       dvd MPEG-2 program stream (DVD VOB) muxer.

	   This muxer can be used to generate tracks in the format accepted by
	   the DVD (Digital Versatile Disc) storage devices.

	   This is the same as the vob muxer with a few differences.

       svcd (vob)
	   MPEG-2 program stream (SVCD VOB) muxer.

	   This muxer can be used to generate tracks in the format accepted by
	   the SVCD (Super Video Compact Disc) storage devices.

	   This is the same as the vob muxer with a few differences.

       Options

       muxrate rate
	   Set user-defined mux rate expressed as a number of bits/s. If not
	   specied the automatically computed mux rate is employed. Default
	   value is 0.

       preload delay
	   Set initial demux-decode delay in microseconds. Default value is
	   500000.

   MOV/MPEG-4/ISOMBFF muxers
       This section covers formats belonging to the QuickTime / MOV family,
       including the MPEG-4 Part 14 format and ISO base media file format
       (ISOBMFF). These formats share a common structure based on the ISO base
       media file format (ISOBMFF).

       The MOV format was originally developed for use with Apple QuickTime.
       It was later used as the basis for the MPEG-4 Part 1 (later Part 14)
       format, also known as ISO/IEC 14496-1. That format was then generalized
       into ISOBMFF, also named MPEG-4 Part 12 format, ISO/IEC 14496-12, or
       ISO/IEC 15444-12.

       It comprises the following muxers.

       3gp Third Generation Partnership Project (3GPP) format for 3G UMTS
	   multimedia services

       3g2 Third Generation Partnership Project 2 (3GP2 or 3GPP2) format for
	   3G CDMA2000 multimedia services, similar to 3gp with extensions and
	   limitations

       f4v Adobe Flash Video format

       ipod
	   MPEG-4 audio file format, as MOV/MP4 but limited to contain only
	   audio streams, typically played with the Apple ipod device

       ismv
	   Microsoft IIS (Internet Information Services) Smooth Streaming
	   Audio/Video (ISMV or ISMA) format. This is based on MPEG-4 Part 14
	   format with a few incompatible variants, used to stream media files
	   for the Microsoft IIS server.

       mov QuickTime player format identified by the ".mov" extension

       mp4 MP4 or MPEG-4 Part 14 format

       psp PlayStation Portable MP4/MPEG-4 Part 14 format variant. This is
	   based on MPEG-4 Part 14 format with a few incompatible variants,
	   used to play files on PlayStation devices.

       Fragmentation

       The mov, mp4, and ismv muxers support fragmentation. Normally, a
       MOV/MP4 file has all the metadata about all packets stored in one
       location.

       This data is usually written at the end of the file, but it can be
       moved to the start for better playback by adding "+faststart" to the
       "-movflags", or using the qt-faststart tool).

       A fragmented file consists of a number of fragments, where packets and
       metadata about these packets are stored together. Writing a fragmented
       file has the advantage that the file is decodable even if the writing
       is interrupted (while a normal MOV/MP4 is undecodable if it is not
       properly finished), and it requires less memory when writing very long
       files (since writing normal MOV/MP4 files stores info about every
       single packet in memory until the file is closed). The downside is that
       it is less compatible with other applications.

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

       frag_duration
       frag_size
       min_frag_duration
       movflags +frag_keyframe
       movflags +frag_custom

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

       Options

       brand brand_string
	   Override major brand.

       empty_hdlr_name bool
	   Enable to skip writing the name inside a "hdlr" box.  Default is
	   "false".

       encryption_key key
	   set the media encryption key in hexadecimal format

       encryption_kid kid
	   set the media encryption key identifier in hexadecimal format

       encryption_scheme scheme
	   configure the encryption scheme, allowed values are none, and cenc-
	   aes-ctr

       frag_duration duration
	   Create fragments that are duration microseconds long.

       frag_interleave	number
	   Interleave samples within fragments (max number of consecutive
	   samples, lower is tighter interleaving, but with more overhead. It
	   is set to 0 by default.

       frag_size size
	   create fragments that contain up to size bytes of payload data

       iods_audio_profile profile
	   specify iods number for the audio profile atom (from -1 to 255),
	   default is "-1"

       iods_video_profile profile
	   specify iods number for the video profile atom (from -1 to 255),
	   default is "-1"

       ism_lookahead num_entries
	   specify number of lookahead entries for ISM files (from 0 to 255),
	   default is 0

       min_frag_duration duration
	   do not create fragments that are shorter than duration microseconds
	   long

       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.

       mov_gamma gamma
	   specify gamma value for gama atom (as a decimal number from 0 to
	   10), default is 0.0, must be set together with "+ movflags"

       movflags flags
	   Set various muxing switches. The following flags can be used:

	   cmaf
	       write CMAF (Common Media Application Format) compatible
	       fragmented MP4 output

	   dash
	       write DASH (Dynamic Adaptive Streaming over HTTP) compatible
	       fragmented MP4 output

	   default_base_moof
	       Similarly to the omit_tfhd_offset flag, 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).

	   delay_moov
	       delay writing the initial moov until the first fragment is cut,
	       or until the first fragment flush

	   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.

	   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.

	   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.)

	   frag_discont
	       signal that the next fragment is discontinuous from earlier
	       ones

	   frag_every_frame
	       fragment at every frame

	   frag_keyframe
	       start a new fragment at each video keyframe

	   global_sidx
	       write a global sidx index at the start of the file

	   isml
	       create a live smooth streaming feed (for pushing to a
	       publishing point)

	   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.

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

	   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.

	   prefer_icc
	       If writing colr atom prioritise usage of ICC profile if it
	       exists in stream packet side data.

	   rtphint
	       add RTP hinting tracks to the output file

	   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.

	   skip_sidx
	       Skip writing of sidx atom. When bitrate overhead due to sidx
	       atom is high, this option could be used for cases where sidx
	       atom is not mandatory. When the global_sidx flag is enabled,
	       this option is ignored.

	   skip_trailer
	       skip writing the mfra/tfra/mfro trailer for fragmented files

	   use_metadata_tags
	       use mdta atom for metadata

	   write_colr
	       write colr atom even if the color info is unspecified. This
	       flag is experimental, may be renamed or changed, do not use
	       from scripts.

	   write_gama
	       write deprecated gama atom

	   hybrid_fragmented
	       For recoverability - write the output file as a fragmented
	       file.  This allows the intermediate file to be read while being
	       written (in particular, if the writing process is aborted
	       uncleanly). When writing is finished, the file is converted to
	       a regular, non-fragmented file, which is more compatible and
	       allows easier and quicker seeking.

	       If writing is aborted, the intermediate file can manually be
	       remuxed to get a regular, non-fragmented file of what had been
	       written into the unfinished file.

       movie_timescale scale
	   Set the timescale written in the movie header box ("mvhd").	Range
	   is 1 to INT_MAX. Default is 1000.

       rtpflags flags
	   Add RTP hinting tracks to the output file.

	   The following flags can be used:

	   h264_mode0
	       use mode 0 for H.264 in RTP

	   latm
	       use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC

	   rfc2190
	       use RFC 2190 packetization instead of RFC 4629 for H.263

	   send_bye
	       send RTCP BYE packets when finishing

	   skip_rtcp
	       do not send RTCP sender reports

       skip_iods bool
	   skip writing iods atom (default value is "true")

       use_editlist bool
	   use edit list (default value is "auto")

       use_stream_ids_as_track_ids bool
	   use stream ids as track ids (default value is "false")

       video_track_timescale scale
	   Set the timescale used for video tracks. Range is 0 to INT_MAX. If
	   set to 0, the timescale is automatically set based on the native
	   stream time base. Default is 0.

       write_btrt bool
	   Force or disable writing bitrate box inside stsd box of a track.
	   The box contains decoding buffer size (in bytes), maximum bitrate
	   and average bitrate for the track. The box will be skipped if none
	   of these values can be computed.  Default is "-1" or "auto", which
	   will write the box only in MP4 mode.

       write_prft option
	   Write producer time reference box (PRFT) with a specified time
	   source for the NTP field in the PRFT box. Set value as wallclock to
	   specify timesource as wallclock time and pts to specify timesource
	   as input packets' PTS values.

       write_tmcd bool
	   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).

	   Setting value to pts is applicable only for a live encoding use
	   case, where PTS values are set as as wallclock time at the source.
	   For example, an encoding use case with decklink capture source
	   where video_pts and audio_pts are set to abs_wallclock.

       Examples

       o   Push Smooth Streaming content in real time to a publishing point on
	   IIS with the ismv muxer using ffmpeg:

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

   a64
       A64 Commodore 64 video muxer.

       This muxer accepts a single "a64_multi" or "a64_multi5" codec video
       stream.

   ac4
       Raw AC-4 audio muxer.

       This muxer accepts a single "ac4" audio stream.

       Options

       write_crc bool
	   when enabled, write a CRC checksum for each packet to the output,
	   default is "false"

   adts
       Audio Data Transport Stream muxer.

       It accepts a single AAC stream.

       Options

       write_id3v2 bool
	   Enable to write ID3v2.4 tags at the start of the stream. Default is
	   disabled.

       write_apetag bool
	   Enable to write APE tags at the end of the stream. Default is
	   disabled.

       write_mpeg2 bool
	   Enable to set MPEG version bit in the ADTS frame header to 1 which
	   indicates MPEG-2. Default is 0, which indicates MPEG-4.

   aea
       MD STUDIO audio muxer.

       This muxer accepts a single ATRAC1 audio stream with either one or two
       channels and a sample rate of 44100Hz.

       As AEA supports storing the track title, this muxer will also write the
       title from stream's metadata to the container.

   aiff
       Audio Interchange File Format muxer.

       Options

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

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

   alp
       High Voltage Software's Lego Racers game audio muxer.

       It accepts a single ADPCM_IMA_ALP stream with no more than 2 channels
       and a sample rate not greater than 44100 Hz.

       Extensions: "tun", "pcm"

       Options

       type type
	   Set file type.

	   type accepts the following values:

	   tun Set file type as music. Must have a sample rate of 22050 Hz.

	   pcm Set file type as sfx.

	   auto
	       Set file type as per output file extension. ".pcm" results in
	       type "pcm" else type "tun" is set. (default)

   amr
       3GPP AMR (Adaptive Multi-Rate) audio muxer.

       It accepts a single audio stream containing an AMR NB stream.

   amv
       AMV (Actions Media Video) format muxer.

   apm
       Ubisoft Rayman 2 APM audio muxer.

       It accepts a single ADPCM IMA APM audio stream.

   apng
       Animated Portable Network Graphics muxer.

       It accepts a single APNG video stream.

       Options

       final_delay delay
	   Force a delay expressed in seconds after the last frame of each
	   repetition. Default value is 0.0.

       plays repetitions
	   specify how many times to play the content, 0 causes an infinte
	   loop, with 1 there is no loop

       Examples

       o   Use ffmpeg to generate an APNG output with 2 repetitions, and with
	   a delay of half a second after the first repetition:

		   ffmpeg -i INPUT -final_delay 0.5 -plays 2 out.apng

   argo_asf
       Argonaut Games ASF audio muxer.

       It accepts a single ADPCM audio stream.

       Options

       version_major version
	   override file major version, specified as an integer, default value
	   is 2

       version_minor version
	   override file minor version, specified as an integer, default value
	   is 1

       name name
	   Embed file name into file, if not specified use the output file
	   name. The name is truncated to 8 characters.

   argo_cvg
       Argonaut Games CVG audio muxer.

       It accepts a single one-channel ADPCM 22050Hz audio stream.

       The loop and reverb options set the corresponding flags in the header
       which can be later retrieved to process the audio stream accordingly.

       Options

       skip_rate_check bool
	   skip sample rate check (default is "false")

       loop bool
	   set loop flag (default is "false")

       reverb boolean
	   set reverb flag (default is "true")

   asf, asf_stream
       Advanced / Active Systems (or Streaming) Format audio muxer.

       The asf_stream variant should be selected for streaming.

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

       Options

       packet_size size
	   Set the muxer packet size as a number of bytes. 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 "64Ki".

   ass
       ASS/SSA (SubStation Alpha) subtitles muxer.

       It accepts a single ASS subtitles stream.

       Options

       ignore_readorder bool
	   Write dialogue events immediately, even if they are out-of-order,
	   default is "false", otherwise they are cached until the expected
	   time event is found.

   ast
       AST (Audio Stream) muxer.

       This format is used to play audio on some Nintendo Wii games.

       It accepts a single audio stream.

       The loopstart and loopend options can be used to define a section of
       the file to loop for players honoring such options.

       Options

       loopstart start
	   Specify loop start position expressesd in milliseconds, from "-1"
	   to "INT_MAX", in case "-1" is set then no loop is specified
	   (default -1) and the loopend value is ignored.

       loopend end
	   Specify loop end position expressed in milliseconds, from 0 to
	   "INT_MAX", default is 0, in case 0 is set it assumes the total
	   stream duration.

   au
       SUN AU audio muxer.

       It accepts a single audio stream.

   avi
       Audio Video Interleaved muxer.

       AVI is a proprietary format developed by Microsoft, and later formally
       specified through the Open DML specification.

       Because of differences in players implementations, it might be required
       to set some options to make sure that the generated output can be
       correctly played by the target player.

       Options

       flipped_raw_rgb bool
	   If set to "true", store positive height for raw RGB bitmaps, which
	   indicates bitmap is stored bottom-up. Note that this option does
	   not flip the bitmap which has to be done manually beforehand, e.g.
	   by using the vflip filter. Default is "false" and indicates bitmap
	   is stored top down.

       reserve_index_space size
	   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.

	   Default value is 0.

       write_channel_mask bool
	   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).

   avif
       AV1 (Alliance for Open Media Video codec 1) image format muxer.

       This muxers stores images encoded using the AV1 codec.

       It accepts one or two video streams. In case two video streams are
       provided, the second one shall contain a single plane storing the alpha
       mask.

       In case more than one image is provided, the generated output is
       considered an animated AVIF and the number of loops can be specified
       with the loop option.

       This is based on the specification by Alliance for Open Media at url
       <https://aomediacodec.github.io/av1-avif>.

       Options

       loop count
	   number of times to loop an animated AVIF, 0 specify an infinite
	   loop, default is 0

       movie_timescale timescale
	   Set the timescale written in the movie header box ("mvhd").	Range
	   is 1 to INT_MAX. Default is 1000.

   avm2
       ShockWave Flash (SWF) / ActionScript Virtual Machine 2 (AVM2) format
       muxer.

       It accepts one audio stream, one video stream, or both.

   bit
       G.729 (.bit) file format muxer.

       It accepts a single G.729 audio stream.

   caf
       Apple CAF (Core Audio Format) muxer.

       It accepts a single audio stream.

   codec2
       Codec2 audio audio muxer.

       It accepts a single codec2 audio stream.

   chromaprint
       Chromaprint fingerprinter muxers.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-chromaprint".

       This muxer feeds audio data to the Chromaprint library, which generates
       a fingerprint for the provided audio data. See:
       <https://acoustid.org/chromaprint>

       It takes a single signed native-endian 16-bit raw audio stream of at
       most 2 channels.

       Options

       algorithm version
	   Select version of algorithm to fingerprint with. Range is 0 to 4.
	   Version 3 enables silence detection. Default is 1.

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

	   base64
	       Base64 compressed fingerprint (default)

	   compressed
	       Binary compressed fingerprint

	   raw Binary raw fingerprint

       silence_threshold threshold
	   Threshold for detecting silence. Range is from "-1" to 32767, where
	   "-1" disables silence detection. Silence detection can only be used
	   with version 3 of the algorithm.

	   Silence detection must be disabled for use with the AcoustID
	   service. Default is "-1".

   crc
       CRC (Cyclic Redundancy Check) muxer.

       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

       o   Use ffmpeg to compute the CRC of the input, and store it in the
	   file out.crc:

		   ffmpeg -i INPUT -f crc out.crc

       o   Use ffmpeg to print the CRC to stdout with the command:

		   ffmpeg -i INPUT -f crc -

       o   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 -

   dash
       Dynamic Adaptive Streaming over HTTP (DASH) muxer.

       This muxer creates segments and manifest files according to the MPEG-
       DASH standard ISO/IEC 23009-1:2014 and following standard updates.

       For more information see:

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

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

       This muxer creates an MPD (Media Presentation Description) manifest
       file and segment files for each stream. Segment files are placed in the
       same directory of the MPD manifest file.

       The segment filename might contain pre-defined identifiers used in the
       manifest "SegmentTemplate" section as defined in section 5.3.9.4.4 of
       the standard.

       Available identifiers are "$RepresentationID$", "$Number$",
       "$Bandwidth$", and "$Time$". In addition to the standard identifiers,
       an ffmpeg-specific "$ext$" identifier is also supported. When
       specified, ffmpeg will replace "$ext$" in the file name with muxing
       format's extensions such as "mp4", "webm" etc.

       Options

       adaptation_sets adaptation_sets
	   Assign streams to adaptation sets, specified in the MPD manifest
	   "AdaptationSets" section.

	   An adaptation set contains a set of one or more streams accessed as
	   a single subset, e.g. corresponding streams encoded at different
	   size selectable by the user depending on the available bandwidth,
	   or to different audio streams with a different language.

	   Each adaptation set is specified with the syntax:

		   id=<index>,streams=<streams>

	   where index must be a numerical index, and streams is a sequence of
	   ","-separated stream indices. Multiple adaptation sets can be
	   specified, separated by spaces.

	   To map all video (or audio) streams to an adaptation set, "v" (or
	   "a") can be used as stream identifier instead of IDs.

	   When no assignment is defined, this defaults to an adaptation set
	   for each stream.

	   The following optional fields can also be specified:

	   descriptor
	       Define the descriptor as defined by ISO/IEC
	       23009-1:2014/Amd.2:2015.

	       For example:

		       <SupplementalProperty schemeIdUri=\"urn:mpeg:dash:srd:2014\" value=\"0,0,0,1,1,2,2\"/>

	       The descriptor string should be a self-closing XML tag.

	   frag_duration
	       Override the global fragment duration specified with the
	       frag_duration option.

	   frag_type
	       Override the global fragment type specified with the frag_type
	       option.

	   seg_duration
	       Override the global segment duration specified with the
	       seg_duration option.

	   trick_id
	       Mark an adaptation set as containing streams meant to be used
	       for Trick Mode for the referenced adaptation set.

	   A few examples of possible values for the adaptation_sets option
	   follow:

		   id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v id=1,seg_duration=2,frag_type=none,streams=a

		   id=0,seg_duration=2,frag_type=none,streams=0 id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1

       dash_segment_type type
	   Set DASH segment files type.

	   Possible values:

	   auto
	       The dash segment files format will be selected based on the
	       stream codec. This is the default mode.

	   mp4 the dash segment files will be in ISOBMFF/MP4 format

	   webm
	       the dash segment files will be in WebM format

       extra_window_size size
	   Set the maximum number of segments kept outside of the manifest
	   before removing from disk.

       format_options options_list
	   Set container format (mp4/webm) options using a ":"-separated list
	   of key=value parameters. Values containing ":" special characters
	   must be escaped.

       frag_duration duration
	   Set the length in seconds of fragments within segments, fractional
	   value can also be set.

       frag_type type
	   Set the type of interval for fragmentation.

	   Possible values:

	   auto
	       set one fragment per segment

	   every_frame
	       fragment at every frame

	   duration
	       fragment at specific time intervals

	   pframes
	       fragment at keyframes and following P-Frame reordering (Video
	       only, experimental)

       global_sidx bool
	   Write global "SIDX" atom. Applicable only for single file, mp4
	   output, non-streaming mode.

       hls_master_name file_name
	   HLS master playlist name. Default is master.m3u8.

       hls_playlist bool
	   Generate HLS playlist files. The master playlist is generated with
	   filename specified by the hls_master_name option. One media
	   playlist file is generated for each stream with filenames
	   media_0.m3u8, media_1.m3u8, etc.

       http_opts http_opts
	   Specify a list of ":"-separated key=value options to pass to the
	   underlying HTTP protocol. Applicable only for HTTP output.

       http_persistent bool
	   Use persistent HTTP connections. Applicable only for HTTP output.

       http_user_agent user_agent
	   Override User-Agent field in HTTP header. Applicable only for HTTP
	   output.

       ignore_io_errors bool
	   Ignore IO errors during open and write. Useful for long-duration
	   runs with network output. This is disabled by default.

       index_correction bool
	   Enable or disable segment index correction logic. Applicable only
	   when use_template is enabled and use_timeline is disabled. This is
	   disabled by default.

	   When enabled, the logic monitors the flow of segment indexes. If a
	   streams's segment index value is not at the expected real time
	   position, then the logic corrects that index value.

	   Typically this logic is needed in live streaming use cases. The
	   network bandwidth fluctuations are common during long run
	   streaming. Each fluctuation can cause the segment indexes fall
	   behind the expected real time position.

       init_seg_name init_name
	   DASH-templated name to use for the initialization segment. Default
	   is "init-stream$RepresentationID$.$ext$". "$ext$" is replaced with
	   the file name extension specific for the segment format.

       ldash bool
	   Enable Low-latency Dash by constraining the presence and values of
	   some elements. This is disabled by default.

       lhls bool
	   Enable Low-latency HLS (LHLS). Add "#EXT-X-PREFETCH" tag with
	   current segment's URI. hls.js player folks are trying to
	   standardize an open LHLS spec. The draft spec is available at
	   <https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md>.

	   This option tries to comply with the above open spec. It enables
	   streaming and hls_playlist options automatically.  This is an
	   experimental feature.

	   Note: This is not Apple's version LHLS. See
	   <https://datatracker.ietf.org/doc/html/draft-pantos-hls-rfc8216bis>

       master_m3u8_publish_rate segment_intervals_count
	   Publish master playlist repeatedly every after specified number of
	   segment intervals.

       max_playback_rate rate
	   Set the maximum playback rate indicated as appropriate for the
	   purposes of automatically adjusting playback latency and buffer
	   occupancy during normal playback by clients.

       media_seg_name segment_name
	   DASH-templated name to use for the media segments. Default is
	   "chunk-stream$RepresentationID$-$Number%05d$.$ext$". "$ext$" is
	   replaced with the file name extension specific for the segment
	   format.

       method method
	   Use the given HTTP method to create output files. Generally set to
	   "PUT" or "POST".

       min_playback_rate rate
	   Set the minimum playback rate indicated as appropriate for the
	   purposes of automatically adjusting playback latency and buffer
	   occupancy during normal playback by clients.

       mpd_profile flags
	   Set one or more MPD manifest profiles.

	   Possible values:

	   dash
	       MPEG-DASH ISO Base media file format live profile

	   dvb_dash
	       DVB-DASH profile

	   Default value is "dash".

       remove_at_exit bool
	   Enable or disable removal of all segments when finished. This is
	   disabled by default.

       seg_duration duration
	   Set the segment length in seconds (fractional value can be set).
	   The value is treated as average segment duration when the
	   use_template option is enabled and the use_timeline option is
	   disabled and as minimum segment duration for all the other use
	   cases.

	   Default value is 5.

       single_file bool
	   Enable or disable storing all segments in one file, accessed using
	   byte ranges. This is disabled by default.

	   The name of the single file can be specified with the
	   single_file_name option, if not specified assume the basename of
	   the manifest file with the output format extension.

       single_file_name file_name
	   DASH-templated name to use for the manifest "baseURL" element.
	   Imply that the single_file option is set to true. In the template,
	   "$ext$" is replaced with the file name extension specific for the
	   segment format.

       streaming bool
	   Enable or disable chunk streaming mode of output. In chunk
	   streaming mode, each frame will be a "moof" fragment which forms a
	   chunk. This is disabled by default.

       target_latency target_latency
	   Set an intended target latency in seconds for serving (fractional
	   value can be set). Applicable only when the streaming and
	   write_prft options are enabled. This is an informative fields
	   clients can use to measure the latency of the service.

       timeout timeout
	   Set timeout for socket I/O operations expressed in seconds
	   (fractional value can be set). Applicable only for HTTP output.

       update_period period
	   Set the MPD update period, for dynamic content. The unit is second.
	   If set to 0, the period is automatically computed.

	   Default value is 0.

       use_template bool
	   Enable or disable use of "SegmentTemplate" instead of "SegmentList"
	   in the manifest. This is enabled by default.

       use_timeline bool
	   Enable or disable use of "SegmentTimeline" within the
	   "SegmentTemplate" manifest section. This is enabled by default.

       utc_timing_url url
	   URL of the page that will return the UTC timestamp in ISO format,
	   for example "https://time.akamai.com/?iso"

       window_size size
	   Set the maximum number of segments kept in the manifest, discard
	   the oldest one. This is useful for live streaming.

	   If the value is 0, all segments are kept in the manifest. Default
	   value is 0.

       write_prft write_prft
	   Write Producer Reference Time elements on supported streams. This
	   also enables writing prft boxes in the underlying muxer. Applicable
	   only when the utc_url option is enabled. It is set to auto by
	   default, in which case the muxer will attempt to enable it only in
	   modes that require it.

       Example

       Generate a DASH output reading from an input source in realtime using
       ffmpeg.

       Two multimedia streams are generated from the input file, both
       containing a video stream encoded through libx264, and an audio stream
       encoded with libfdk_aac. The first multimedia stream contains video
       with a bitrate of 800k and audio at the default rate, the second with
       video scaled to 320x170 pixels at 300k and audio resampled at 22005 Hz.

       The window_size option keeps only the latest 5 segments with the
       default duration of 5 seconds.

	       ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 \
	       -b:v:0 800k -profile:v:0 main \
	       -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline -ar:a:1 22050 \
	       -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 -b_strategy 0 \
	       -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

   daud
       D-Cinema audio muxer.

       It accepts a single 6-channels audio stream resampled at 96000 Hz
       encoded with the pcm_24daud codec.

       Example

       Use ffmpeg to mux input audio to a 5.1 channel layout resampled at
       96000Hz:

	       ffmpeg -i INPUT -af aresample=96000,pan=5.1 slow.302

       For ffmpeg versions before 7.0 you might have to use the asetnsamples
       filter to limit the muxed packet size, because this format does not
       support muxing packets larger than 65535 bytes (3640 samples). For
       newer ffmpeg versions audio is automatically packetized to 36000 byte
       (2000 sample) packets.

   dv
       DV (Digital Video) muxer.

       It accepts exactly one dvvideo video stream and at most two pcm_s16
       audio streams. More constraints are defined by the property of the
       video, which must correspond to a DV video supported profile, and on
       the framerate.

       Example

       Use ffmpeg to convert the input:

	       ffmpeg -i INPUT -s:v 720x480 -pix_fmt yuv411p -r 29.97 -ac 2 -ar 48000 -y out.dv

   ffmetadata
       FFmpeg metadata muxer.

       This muxer writes the streams metadata in the ffmetadata format.

       See the Metadata chapter for information about the format.

       Example

       Use ffmpeg to extract metadata from an input file to a metadata.ffmeta
       file in ffmetadata format:

	       ffmpeg -i INPUT -f ffmetadata metadata.ffmeta

   fifo
       FIFO (First-In First-Out) muxer.

       The fifo pseudo-muxer allows the separation of encoding and muxing by
       using a 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.

       The target muxer is either selected from the output name or specified
       through the fifo_format option.

       The behavior of the fifo muxer if the queue fills up or if the output
       fails (e.g. if a packet cannot be written to the output) is selectable:

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

       o   Encoding can be blocked during temporary failure, or continue
	   transparently dropping packets in case the FIFO queue fills up.

       API users should be aware that callback functions
       ("interrupt_callback", "io_open" and "io_close") used within its
       "AVFormatContext" must be thread-safe.

       Options

       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 "false".

       drop_pkts_on_overflow bool
	   If set to "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
	   "false", so in such cases the encoder will be blocked until the
	   muxer processes some of the packets and none of them is lost.

       fifo_format format_name
	   Specify the format name. Useful if it cannot be guessed from the
	   output name suffix.

       format_opts options
	   Specify format options for the underlying muxer. Muxer options can
	   be specified as a list of key=value pairs separated by ':'.

       max_recovery_attempts count
	   Set maximum number of successive unsuccessful recovery attempts
	   after which the output fails permanently. By default this option is
	   set to 0 (unlimited).

       queue_size size
	   Specify size of the queue as a number of packets. Default value is
	   60.

       recover_any_error bool
	   If set to "true", recovery will be attempted regardless of type of
	   the error causing the failure. By default this option is set to
	   "false" and in case of certain (usually permanent) errors the
	   recovery is not attempted even when the attempt_recovery option is
	   set to "true".

       recovery_wait_streamtime bool
	   If set to "false", the real time is used when waiting for the
	   recovery attempt (i.e. the recovery will be attempted after the
	   time specified by the recovery_wait_time option).

	   If set to "true", the time of the processed stream is taken into
	   account instead (i.e. the recovery will be attempted after
	   discarding the packets corresponding to the recovery_wait_time
	   option).

	   By default this option is set to "false".

       recovery_wait_time duration
	   Specify waiting time in seconds before the next recovery attempt
	   after previous unsuccessful recovery attempt. Default value is 5.

       restart_with_keyframe bool
	   Specify whether to wait for the keyframe after recovering from
	   queue overflow or failure. This option is set to "false" by
	   default.

       timeshift duration
	   Buffer the specified amount of packets and delay writing the
	   output. Note that the value of the queue_size option must be big
	   enough to store the packets for timeshift. At the end of the input
	   the fifo buffer is flushed at realtime speed.

       Example

       Use ffmpeg to stream to an 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 \
		 -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 \
		 -map 0:v -map 0:a rtmp://example.com/live/stream_name

   film_cpk
       Sega film (.cpk) muxer.

       This format was used as internal format for several Sega games.

       For more information regarding the Sega film file format, visit
       <http://wiki.multimedia.cx/index.php?title=Sega_FILM>.

       It accepts at maximum one cinepak or raw video stream, and at maximum
       one audio stream.

   filmstrip
       Adobe Filmstrip muxer.

       This format is used by several Adobe tools to store a generated
       filmstrip export. It accepts a single raw video stream.

   fits
       Flexible Image Transport System (FITS) muxer.

       This image format is used to store astronomical data.

       For more information regarding the format, visit
       <https://fits.gsfc.nasa.gov>.

   flac
       Raw FLAC audio muxer.

       This muxer accepts exactly one FLAC audio stream. Additionally, it is
       possible to add images with disposition attached_pic.

       Options

       write_header bool
	   write the file header if set to "true", default is "true"

       Example

       Use ffmpeg to store the audio stream from an input file, together with
       several pictures used with attached_pic disposition:

	       ffmpeg -i INPUT -i pic1.png -i pic2.jpg -map 0:a -map 1 -map 2 -disposition:v attached_pic OUTPUT

   flv
       Adobe Flash Video Format muxer.

       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.

   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.

       Note that the GIF format has a very large time base: the delay between
       two frames can therefore not be smaller than one centi second.

       Options

       loop bool
	   Set the number of times to loop the output. Use "-1" for no loop, 0
	   for looping indefinitely (default).

       final_delay 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.

       Example

       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"

   gxf
       General eXchange Format (GXF) muxer.

       GXF was developed by Grass Valley Group, then standardized by SMPTE as
       SMPTE 360M and was extended in SMPTE RDD 14-2007 to include high-
       definition video resolutions.

       It accepts at most one video stream with codec mjpeg, or mpeg1video, or
       mpeg2video, or dvvideo with resolution 512x480 or 608x576, and several
       audio streams with rate 48000Hz and codec pcm16_le.

   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.

   hds
       HTTP Dynamic Streaming (HDS) muxer.

       HTTP dynamic streaming, or HDS, is an adaptive bitrate streaming method
       developed by Adobe. HDS delivers MP4 video content over HTTP
       connections. HDS can be used for on-demand streaming or live streaming.

       This muxer creates an .f4m (Adobe Flash Media Manifest File) manifest,
       an .abst (Adobe Bootstrap File) for each stream, and segment files in a
       directory specified as the output.

       These needs to be accessed by an HDS player throuhg HTTPS for it to be
       able to perform playback on the generated stream.

       Options

       extra_window_size int
	   number of fragments kept outside of the manifest before removing
	   from disk

       min_frag_duration microseconds
	   minimum fragment duration (in microseconds), default value is 1
	   second (10000000)

       remove_at_exit bool
	   remove all fragments when finished when set to "true"

       window_size int
	   number of fragments kept in the manifest, if set to a value
	   different from 0. By default all segments are kept in the output
	   directory.

       Example

       Use ffmpeg to generate HDS files to the output.hds directory in real-
       time rate:

	       ffmpeg -re -i INPUT -f hds -b:v 200k output.hds

   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.

       Make sure to require a closed GOP when encoding and to set the GOP size
       to fit your segment time constraint.

       For example, to convert an input file with ffmpeg:

	       ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 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

       hls_init_time duration
	   Set the initial target segment length. Default value is 0.

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

	   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 duration
	   Set the target segment length. Default value is 2.

	   duration must be a time duration specification, see the Time
	   duration section in the ffmpeg-utils(1) manual.  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_delete_threshold size
	   Set the number of unreferenced segments to keep on disk before
	   "hls_flags delete_segments" deletes them. Increase this to allow
	   continue clients to download segments which were recently
	   referenced in the playlist. Default value is 1, meaning segments
	   older than hls_list_size+1 will be deleted.

       hls_start_number_source 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 start numbers according to the start_number option
	       value.

	   epoch
	       Set the start number as the seconds since epoch (1970-01-01
	       00:00:00).

	   epoch_us
	       Set the start number as the microseconds since epoch
	       (1970-01-01 00:00:00).

	   datetime
	       Set the start number 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 bool
	   Explicitly set whether the client MAY \fIs0(1) or MUST NOT \fIs0(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 the hls_flags option is set with
	   single_file, filename is used as a string format with the segment
	   number appended.

	   For example:

		   ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

	   will produce the playlist, out.m3u8, and segment files: file000.ts,
	   file001.ts, file002.ts, etc.

	   filename may contain a full path or relative path specification,
	   but only the file name part without any path 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 strftime_mkdir is set, the whole expanded
	   value of filename will be written into the m3u8 segment list.

	   When var_stream_map is set with two or more variant streams, the
	   filename pattern must contain the string "%v", and this string will
	   be expanded to the position of variant stream index in the
	   generated segment file names.

	   For example:

		   ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
		     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
		     -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8

	   will produce the playlists segment file sets: file_0_000.ts,
	   file_0_001.ts, file_0_002.ts, etc. and file_1_000.ts,
	   file_1_001.ts, file_1_002.ts, etc.

	   The string "%v" may be present in the filename or in the last
	   directory name containing the file, but only in one of them.
	   (Additionally, %v may appear multiple times in the last sub-
	   directory or filename.) If the string %v is present in the
	   directory name, then sub-directories are created after expanding
	   the directory name pattern. This enables creation of segments
	   corresponding to different variant streams in subdirectories.

	   For example:

		   ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
		     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
		     -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8

	   will produce the playlists segment file sets: vs0/file_000.ts,
	   vs0/file_001.ts, vs0/file_002.ts, etc. and vs1/file_000.ts,
	   vs1/file_001.ts, vs1/file_002.ts, etc.

       strftime bool
	   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 set second_level_segment_index in the
	   hls_flag and %%d will be the specifier.

	   For example:

		   ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

	   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.

	   For example:

		   ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

	   will produce the playlist, out.m3u8, and segment files:
	   file-20160215-0001.ts, file-20160215-0002.ts, etc.

       strftime_mkdir bool
	   Used together with strftime, it will create all subdirectories
	   which are present in the expanded values of option
	   hls_segment_filename.

	   For example:

		   ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

	   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.

	   For example:

		   ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

	   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_segment_options options_list
	   Set output format options using a :-separated list of key=value
	   parameters. Values containing ":" special characters must be
	   escaped.

       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 bool
	   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
	   Specify a 16-octet 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
	   Specify the 16-octet initialization vector for every segment
	   instead of the autogenerated ones.

       hls_segment_type flags
	   Possible values:

	   mpegts
	       Output segment files in MPEG-2 Transport Stream format. This is
	       compatible with all HLS versions.

	   fmp4
	       Output segment files in fragmented MP4 format, similar to MPEG-
	       DASH.  fmp4 files may be used in HLS version 7 and above.

       hls_fmp4_init_filename filename
	   Set filename for the fragment files header file, default filename
	   is init.mp4.

	   When strftime is enabled, filename is expanded to the segment
	   filename with localtime.

	   For example:

		   ffmpeg -i in.nut -hls_segment_type fmp4 -strftime 1 -hls_fmp4_init_filename "%s_init.mp4" out.m3u8

	   will produce init like this 1602678741_init.mp4.

       hls_fmp4_init_resend bool
	   Resend init file after m3u8 file refresh every time, default is 0.

	   When var_stream_map is set with two or more variant streams, the
	   filename pattern must contain the string "%v", this string
	   specifies the position of variant stream index in the generated
	   init file names.  The string "%v" may be present in the filename or
	   in the last directory name containing the file. If the string is
	   present in the directory name, then sub-directories are created
	   after expanding the directory name pattern. This enables creation
	   of init files corresponding to different variant streams in
	   subdirectories.

       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.  If there are
	       no other features requiring higher HLS versions be used, then
	       this will allow ffmpeg to output a HLS version 2 m3u8.

	   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.

	   independent_segments
	       Add the "#EXT-X-INDEPENDENT-SEGMENTS" tag to playlists that has
	       video segments and when all the segments of that playlist are
	       guaranteed to start with a key frame.

	   iframes_only
	       Add the "#EXT-X-I-FRAMES-ONLY" tag to playlists that has video
	       segments and can play only I-frames in the "#EXT-X-BYTERANGE"
	       mode.

	   split_by_time
	       Allow segments to start on frames other than key frames. This
	       improves behavior on some players when the time between key
	       frames 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
	       Make it possible to use segment indexes as %%d in the
	       hls_segment_filename option expression besides date/time values
	       when strftime option is on. To get fixed width numbers with
	       trailing zeroes, %%0xd format is available where x is the
	       required width.

	   second_level_segment_size
	       Make it possible to use segment sizes (counted in bytes) as %%s
	       in hls_segment_filename option expression besides date/time
	       values when strftime is on. To get fixed width numbers with
	       trailing zeroes, %%0xs format is available where x is the
	       required width.

	   second_level_segment_duration
	       Make it possible to use segment duration (calculated in
	       microseconds) as %%t in hls_segment_filename option expression
	       besides date/time values when strftime is on. To get fixed
	       width numbers with trailing zeroes, %%0xt format is available
	       where x is the required width.

	       For example:

		       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 \
			  -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8

	       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.

	       This flag also affects how m3u8 playlist files are created. If
	       this flag is set, all playlist files will be written into a
	       temporary file and renamed after they are complete, similarly
	       as segments are handled. But playlists with "file" protocol and
	       with hls_playlist_type type other than vod are always written
	       into a temporary file regardless of this flag.

	       Master playlist files specified with master_pl_name, if any,
	       with "file" protocol, are always written into temporary file
	       regardless of this flag if master_pl_publish_rate value is
	       other than zero.

       hls_playlist_type type
	   If type is event, emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8
	   header. This forces hls_list_size to 0; the playlist can only be
	   appended to.

	   If type is vod, emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header.
	   This forces hls_list_size to 0; the playlist must not change.

       method method
	   Use the given HTTP method to create the hls files.

	   For example:

		   ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

	   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 agent
	   Override User-Agent field in HTTP header. Applicable only for HTTP
	   output.

       var_stream_map stream_map
	   Specify a map string defining how to group the audio, video and
	   subtitle streams into different variant streams. The variant stream
	   groups are separated by space.

	   Expected string format is like this "a:0,v:0 a:1,v:1 ....". Here
	   a:, v:, s: are the keys to specify audio, video and subtitle
	   streams respectively.  Allowed values are 0 to 9 (limited just
	   based on practical usage).

	   When there are two or more variant streams, the output filename
	   pattern must contain the string "%v": this string specifies the
	   position of variant stream index in the output media playlist
	   filenames. The string "%v" may be present in the filename or in the
	   last directory name containing the file. If the string is present
	   in the directory name, then sub-directories are created after
	   expanding the directory name pattern. This enables creation of
	   variant streams in subdirectories.

	   A few examples follow.

	   o   Create two hls variant streams. The first variant stream will
	       contain video stream of bitrate 1000k and audio stream of
	       bitrate 64k and the second variant stream will contain video
	       stream of bitrate 256k and audio stream of bitrate 32k. Here,
	       two media playlist with file names out_0.m3u8 and out_1.m3u8
	       will be created.

		       ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
			 -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
			 http://example.com/live/out_%v.m3u8

	   o   If you want something meaningful text instead of indexes in
	       result names, you may specify names for each or some of the
	       variants. The following example will create two hls variant
	       streams as in the previous one. But here, the two media
	       playlist with file names out_my_hd.m3u8 and out_my_sd.m3u8 will
	       be created.

		       ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
			 -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0,name:my_hd v:1,a:1,name:my_sd" \
			 http://example.com/live/out_%v.m3u8

	   o   Create three hls variant streams. The first variant stream will
	       be a video only stream with video bitrate 1000k, the second
	       variant stream will be an audio only stream with bitrate 64k
	       and the third variant stream will be a video only stream with
	       bitrate 256k. Here, three media playlist with file names
	       out_0.m3u8, out_1.m3u8 and out_2.m3u8 will be created.

		       ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
			 -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
			 http://example.com/live/out_%v.m3u8

	   o   Create the variant streams in subdirectories. Here, the first
	       media playlist is created at
	       http://example.com/live/vs_0/out.m3u8 and the second one at
	       http://example.com/live/vs_1/out.m3u8.

		       ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
			 -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
			 http://example.com/live/vs_%v/out.m3u8

	   o   Create two audio only and two video only variant streams. In
	       addition to the "#EXT-X-STREAM-INF" tag for each variant stream
	       in the master playlist, the "#EXT-X-MEDIA" tag is also added
	       for the two audio only variant streams and they are mapped to
	       the two video only variant streams with audio group names
	       'aud_low' and 'aud_high'.  By default, a single hls variant
	       containing all the encoded streams is created.

		       ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k  \
			 -map 0:a -map 0:a -map 0:v -map 0:v -f hls \
			 -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
			 -master_pl_name master.m3u8 \
			 http://example.com/live/out_%v.m3u8

	   o   Create two audio only and one video only variant streams. In
	       addition to the "#EXT-X-STREAM-INF" tag for each variant stream
	       in the master playlist, the "#EXT-X-MEDIA" tag is also added
	       for the two audio only variant streams and they are mapped to
	       the one video only variant streams with audio group name
	       'aud_low', and the audio group have default stat is NO or YES.
	       By default, a single hls variant containing all the encoded
	       streams is created.

		       ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
			 -map 0:a -map 0:a -map 0:v -f hls \
			 -var_stream_map "a:0,agroup:aud_low,default:yes a:1,agroup:aud_low v:0,agroup:aud_low" \
			 -master_pl_name master.m3u8 \
			 http://example.com/live/out_%v.m3u8

	   o   Create two audio only and one video only variant streams. In
	       addition to the "#EXT-X-STREAM-INF" tag for each variant stream
	       in the master playlist, the "#EXT-X-MEDIA" tag is also added
	       for the two audio only variant streams and they are mapped to
	       the one video only variant streams with audio group name
	       'aud_low', and the audio group have default stat is NO or YES,
	       and one audio have and language is named ENG, the other audio
	       language is named CHN. By default, a single hls variant
	       containing all the encoded streams is created.

		       ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k \
			 -map 0:a -map 0:a -map 0:v -f hls \
			 -var_stream_map "a:0,agroup:aud_low,default:yes,language:ENG a:1,agroup:aud_low,language:CHN v:0,agroup:aud_low" \
			 -master_pl_name master.m3u8 \
			 http://example.com/live/out_%v.m3u8

	   o   Create a single variant stream. Add the "#EXT-X-MEDIA" tag with
	       "TYPE=SUBTITLES" in the master playlist with webvtt subtitle
	       group name 'subtitle'. Make sure the input file has one text
	       subtitle stream at least.

		       ffmpeg -y -i input_with_subtitle.mkv \
			-b:v:0 5250k -c:v h264 -pix_fmt yuv420p -profile:v main -level 4.1 \
			-b:a:0 256k \
			-c:s webvtt -c:a mp2 -ar 48000 -ac 2 -map 0:v -map 0:a:0 -map 0:s:0 \
			-f hls -var_stream_map "v:0,a:0,s:0,sgroup:subtitle" \
			-master_pl_name master.m3u8 -t 300 -hls_time 10 -hls_init_time 4 -hls_list_size \
			10 -master_pl_publish_rate 10 -hls_flags \
			delete_segments+discont_start+split_by_time ./tmp/video.m3u8

       cc_stream_map cc_stream_map
	   Map string which specifies different closed captions groups and
	   their attributes. The closed captions stream groups are separated
	   by space.

	   Expected string format is like this "ccgroup:<group
	   name>,instreamid:<INSTREAM-ID>,language:<language code> ....".
	   'ccgroup' and 'instreamid' are mandatory attributes. 'language' is
	   an optional attribute.

	   The closed captions groups configured using this option are mapped
	   to different variant streams by providing the same 'ccgroup' name
	   in the var_stream_map string.

	   For example:

		   ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
		     -a53cc:0 1 -a53cc:1 1 \
		     -map 0:v -map 0:a -map 0:v -map 0:a -f hls \
		     -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \
		     -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \
		     -master_pl_name master.m3u8 \
		     http://example.com/live/out_%v.m3u8

	   will add two "#EXT-X-MEDIA" tags with "TYPE=CLOSED-CAPTIONS" in the
	   master playlist for the INSTREAM-IDs 'CC1' and 'CC2'. Also, it will
	   add "CLOSED-CAPTIONS" attribute with group name 'cc' for the two
	   output variant streams.

	   If var_stream_map is not set, then the first available ccgroup in
	   cc_stream_map is mapped to the output variant stream.

	   For example:

		   ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
		     -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \
		     -master_pl_name master.m3u8 \
		     http://example.com/live/out.m3u8

	   this will add "#EXT-X-MEDIA" tag with "TYPE=CLOSED-CAPTIONS" in the
	   master playlist with group name 'cc', language 'en' (english) and
	   INSTREAM-ID 'CC1'. Also, it will add "CLOSED-CAPTIONS" attribute
	   with group name 'cc' for the output variant stream.

       master_pl_name name
	   Create HLS master playlist with the given name.

	   For example:

		   ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8

	   creates an HLS master playlist with name master.m3u8 which is
	   published at <http://example.com/live/>.

       master_pl_publish_rate count
	   Publish master play list repeatedly every after specified number of
	   segment intervals.

	   For example:

		   ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \
		   -hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8

	   creates an HLS master playlist with name master.m3u8 and keeps
	   publishing it repeatedly every after 30 segments i.e. every after
	   60s.

       http_persistent bool
	   Use persistent HTTP connections. Applicable only for HTTP output.

       timeout timeout
	   Set timeout for socket I/O operations. Applicable only for HTTP
	   output.

       ignore_io_errors bool
	   Ignore IO errors during open, write and delete. Useful for long-
	   duration runs with network output.

       headers headers
	   Set custom HTTP headers, can override built in default headers.
	   Applicable only for HTTP output.

   iamf
       Immersive Audio Model and Formats (IAMF) muxer.

       IAMF is used to provide immersive audio content for presentation on a
       wide range of devices in both streaming and offline applications. These
       applications include internet audio streaming,
       multicasting/broadcasting services, file download, gaming,
       communication, virtual and augmented reality, and others. In these
       applications, audio may be played back on a wide range of devices,
       e.g., headphones, mobile phones, tablets, TVs, sound bars, home theater
       systems, and big screens.

       This format was promoted and desgined by Alliance for Open Media.

       For more information about this format, see
       <https://aomedia.org/iamf/>.

   ico
       ICO file muxer.

       Microsoft's icon file format (ICO) has some strict limitations that
       should be noted:

       o   Size cannot exceed 256 pixels in any dimension

       o   Only BMP and PNG images can be stored

       o   If a BMP image is used, it must be one of the following pixel
	   formats:

		   BMP Bit Depth      FFmpeg Pixel Format
		   1bit 	      pal8
		   4bit 	      pal8
		   8bit 	      pal8
		   16bit	      rgb555le
		   24bit	      bgr24
		   32bit	      bgra

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

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

   ilbc
       Internet Low Bitrate Codec (iLBC) raw muxer.

       It accepts a single ilbc audio stream.

   image2, image2pipe
       Image file muxer.

       The image2 muxer writes video frames to image files.

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

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

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

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

       The image muxer supports the .Y.U.V image file format. This format is
       special in 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.

       The image2pipe muxer accepts the same options as the image2 muxer, but
       ignores the pattern verification and expansion, as it is supposed to
       write to the command output rather than to an actual stored file.

       Options

       frame_pts bool
	   If set to 1, expand the filename with the packet PTS (presentation
	   time stamp).  Default value is 0.

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

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

       atomic_writing bool
	   Write output to a temporary file, which is renamed to target
	   filename once writing is completed. Default is disabled.

       protocol_opts options_list
	   Set protocol options as a :-separated list of key=value parameters.
	   Values containing the ":" special character must be escaped.

       Examples

       o   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

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

	   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"

       o   Set the file name with current frame's PTS:

		   ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg

       o   Publish contents of your desktop directly to a WebDAV server every
	   second:

		   ffmpeg -f x11grab -framerate 1 -i :0.0 -q:v 6 -update 1 -protocol_opts method=PUT http://example.com/desktop.jpg

   ircam
       Berkeley / IRCAM / CARL Sound Filesystem (BICSF) format muxer.

       The Berkeley/IRCAM/CARL Sound Format, developed in the 1980s, is a
       result of the merging of several different earlier sound file formats
       and systems including the csound system developed by Dr Gareth Loy at
       the Computer Audio Research Lab (CARL) at UC San Diego, the IRCAM sound
       file system developed by Rob Gross and Dan Timis at the Institut de
       Recherche et Coordination Acoustique / Musique in Paris and the
       Berkeley Fast Filesystem.

       It was developed initially as part of the Berkeley/IRCAM/CARL Sound
       Filesystem, a suite of programs designed to implement a filesystem for
       audio applications running under Berkeley UNIX. It was particularly
       popular in academic music research centres, and was used a number of
       times in the creation of early computer-generated compositions.

       This muxer accepts a single audio stream containing PCM data.

   ivf
       On2 IVF muxer.

       IVF was developed by On2 Technologies (formerly known as Duck
       Corporation), to store internally developed codecs.

       This muxer accepts a single vp8, vp9, or av1 video stream.

   jacosub
       JACOsub subtitle format muxer.

       This muxer accepts a single jacosub subtitles stream.

       For more information about the format, see
       <http://unicorn.us.com/jacosub/jscripts.html>.

   kvag
       Simon & Schuster Interactive VAG muxer.

       This custom VAG container is used by some Simon & Schuster Interactive
       games such as "Real War", and "Real War: Rogue States".

       This muxer accepts a single adpcm_ima_ssi audio stream.

   lc3
       Bluetooth SIG Low Complexity Communication Codec audio (LC3), or ETSI
       TS 103 634 Low Complexity Communication Codec plus (LC3plus).

       This muxer accepts a single lc3 audio stream.

   lrc
       LRC lyrics file format muxer.

       LRC (short for LyRiCs) is a computer file format that synchronizes song
       lyrics with an audio file, such as MP3, Vorbis, or MIDI.

       This muxer accepts a single subrip or text subtitles stream.

       Metadata

       The following metadata tags are converted to the format corresponding
       metadata:

       title
       album
       artist
       author
       creator
       encoder
       encoder_version

       If encoder_version is not explicitly set, it is automatically set to
       the libavformat version.

   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. This gets mapped to the
	   FileDescription element for a stream written as attachment.

       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

       reserve_index_space size
	   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
	   size bytes of space in the file header and then try to write the
	   cues there when the muxing finishes. If the reserved space does not
	   suffice, no Cues will be written, the file will be finalized and
	   writing the trailer will return an error.  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.

       cues_to_front bool
	   If set, the muxer will write the index at the beginning of the file
	   by shifting the main data if necessary. This can be combined with
	   reserve_index_space in which case the data is only shifted if the
	   initially reserved space turns out to be insufficient.

	   This option is ignored if the output is unseekable.

       cluster_size_limit size
	   Store at most the provided amount of bytes in a cluster.

	   If not specified, the limit is set automatically to a sensible
	   hardcoded fixed value.

       cluster_time_limit duration
	   Store at most the provided number of milliseconds in a cluster.

	   If not specified, the limit is set automatically to a sensible
	   hardcoded fixed value.

       dash bool
	   Create a WebM file conforming to WebM DASH specification. By
	   default it is set to "false".

       dash_track_number index
	   Track number for the DASH stream. By default it is set to 1.

       live bool
	   Write files assuming it is a live stream. By default it is set to
	   "false".

       allow_raw_vfw bool
	   Allow raw VFW mode. By default it is set to "false".

       flipped_raw_rgb bool
	   If set to "true", store positive height for raw RGB bitmaps, which
	   indicates bitmap is stored bottom-up. Note that this option does
	   not flip the bitmap which has to be done manually beforehand, e.g.
	   by using the vflip filter.  Default is "false" and indicates bitmap
	   is stored top down.

       write_crc32 bool
	   Write a CRC32 element inside every Level 1 element. By default it
	   is set to "true". This option is ignored for WebM.

       default_mode mode
	   Control how the FlagDefault of the output tracks will be set.  It
	   influences which tracks players should play by default. The default
	   mode is passthrough.

	   infer
	       Every track with disposition default will have the FlagDefault
	       set.  Additionally, for each type of track (audio, video or
	       subtitle), if no track with disposition default of this type
	       exists, then the first track of this type will be marked as
	       default (if existing). This ensures that the default flag is
	       set in a sensible way even if the input originated from
	       containers that lack the concept of default tracks.

	   infer_no_subs
	       This mode is the same as infer except that if no subtitle track
	       with disposition default exists, no subtitle track will be
	       marked as default.

	   passthrough
	       In this mode the FlagDefault is set if and only if the
	       AV_DISPOSITION_DEFAULT flag is set in the disposition of the
	       corresponding stream.

   md5
       MD5 testing format.

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

       See also the hash and framemd5 muxers.

       Examples

       o   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

       o   To print the MD5 hash to stdout:

		   ffmpeg -i INPUT -f md5 -

   microdvd
       MicroDVD subtitle format muxer.

       This muxer accepts a single microdvd subtitles stream.

   mmf
       Synthetic music Mobile Application Format (SMAF) format muxer.

       SMAF is a music data format specified by Yamaha for portable electronic
       devices, such as mobile phones and personal digital assistants.

       This muxer accepts a single adpcm_yamaha audio stream.

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

       o   An ID3v2 metadata header at the beginning (enabled by default).
	   Versions 2.3 and 2.4 are supported, the "id3v2_version" private
	   option controls which one is used (3 or 4). Setting "id3v2_version"
	   to 0 disables the ID3v2 header completely.

	   The muxer supports writing attached pictures (APIC frames) to the
	   ID3v2 header.  The pictures are supplied to the muxer in form of a
	   video stream with a single packet. There can be any number of those
	   streams, each will correspond to a single APIC frame.  The stream
	   metadata tags title and comment map to APIC description and picture
	   type respectively. See <http://id3.org/id3v2.4.0-frames> for
	   allowed picture types.

	   Note that the APIC frames must be written at the beginning, so the
	   muxer will buffer the audio frames until it gets all the pictures.
	   It is therefore advised to provide the pictures as soon as possible
	   to avoid excessive buffering.

       o   A Xing/LAME frame right after the ID3v2 header (if present). It is
	   enabled by default, but will be written only if the output is
	   seekable. The "write_xing" private option can be used to disable
	   it.	The frame contains various information that may be useful to
	   the decoder, like the audio duration or encoder delay.

       o   A legacy ID3v1 tag at the end of the file (disabled by default). It
	   may be enabled with the "write_id3v1" private option, but as its
	   capabilities are very limited, its usage is not recommended.

       Examples:

       Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

	       ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

       To attach a picture to an mp3 file select both the audio and the
       picture stream with "map":

	       ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
	       -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

       Write a "clean" MP3 without any extra features:

	       ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

   mpegts
       MPEG transport stream muxer.

       This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

       The recognized metadata settings in mpegts muxer are "service_provider"
       and "service_name". If they are not set the default for
       "service_provider" is FFmpeg and the default for "service_name" is
       Service01.

       Options

       The muxer options are:

       mpegts_transport_stream_id integer
	   Set the transport_stream_id. This identifies a transponder in DVB.
	   Default is 0x0001.

       mpegts_original_network_id integer
	   Set the original_network_id. This is unique identifier of a network
	   in DVB. Its main use is in the unique identification of a service
	   through the path Original_Network_ID, Transport_Stream_ID. Default
	   is 0x0001.

       mpegts_service_id integer
	   Set the service_id, also known as program in DVB. Default is
	   0x0001.

       mpegts_service_type integer
	   Set the program service_type. Default is "digital_tv".  Accepts the
	   following options:

	   hex_value
	       Any hexadecimal value between 0x01 and 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 PMTs. Default is 0x1000, minimum is 0x0020,
	   maximum is 0x1ffa. This option has no effect in m2ts mode where the
	   PMT PID is fixed 0x0100.

       mpegts_start_pid integer
	   Set the first PID for elementary streams. Default is 0x0100,
	   minimum is 0x0020, maximum is 0x1ffa. This option has no effect in
	   m2ts mode where the elementary stream PIDs are fixed.

       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.

	   nit Emit NIT table.

	   omit_rai
	       Disable writing of random access indicator.

       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.
	   Default is "-1" which means that the PCR interval will be
	   determined automatically: 20 ms is used for CBR streams, the
	   highest multiple of the frame duration which is less than 100 ms is
	   used for VBR streams.

       pat_period duration
	   Maximum time in seconds between PAT/PMT tables. Default is 0.1.

       sdt_period duration
	   Maximum time in seconds between SDT tables. Default is 0.5.

       nit_period duration
	   Maximum time in seconds between NIT tables. Default is 0.5.

       tables_version integer
	   Set PAT, PMT, SDT and NIT 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_opatom
       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 and mxf_opatom 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.

   rcwt
       RCWT (Raw Captions With Time) is a format native to ccextractor, a
       commonly used open source tool for processing 608/708 Closed Captions
       (CC) sources.  It can be used to archive the original extracted CC
       bitstream and to produce a source file for later processing or
       conversion. The format allows for interoperability between ccextractor
       and FFmpeg, is simple to parse, and can be used to create a backup of
       the CC presentation.

       This muxer implements the specification as of March 2024, which has
       been stable and unchanged since April 2014.

       This muxer will have some nuances from the way that ccextractor muxes
       RCWT.  No compatibility issues when processing the output with
       ccextractor have been observed as a result of this so far, but mileage
       may vary and outputs will not be a bit-exact match.

       A free specification of RCWT can be found here:
       <https://github.com/CCExtractor/ccextractor/blob/master/docs/BINARY_FILE_FORMAT.TXT>

       Examples

       o   Extract Closed Captions to RCWT using lavfi:

		   ffmpeg -f lavfi -i "movie=INPUT.mkv[out+subcc]" -map 0:s:0 -c:s copy -f rcwt CC.rcwt.bin

   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.

       min_seg_duration time
	   Set minimum segment duration to time, the value must be a duration
	   specification. This prevents the muxer ending segments at a
	   duration below this value. Only effective with "segment_time".
	   Default value is "0".

       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.

       Make sure to require a closed GOP when encoding and to set the GOP size
       to fit your segment time constraint.

       Examples

       o   Remux the content of file in.mkv to a list of segments out-000.nut,
	   out-001.nut, etc., and write the list of generated segments to
	   out.list:

		   ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut

       o   Segment input and set output format options for the output
	   segments:

		   ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4

       o   Segment the input file according to the split points specified by
	   the segment_times option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut

       o   Use the ffmpeg force_key_frames option to force key frames in the
	   input at the specified location, together with the segment option
	   segment_time_delta to account for possible roundings operated when
	   setting key frame times.

		   ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
		   -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut

	   In order to force key frames on the input file, transcoding is
	   required.

       o   Segment the input file by splitting the input file according to the
	   frame numbers sequence specified with the segment_frames option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut

       o   Convert the in.mkv to TS segments using the "libx264" and "aac"
	   encoders:

		   ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts

       o   Segment the input file, and create an M3U8 live playlist (can be
	   used as live HLS source):

		   ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
		   -segment_list_flags +live -segment_time 10 out%03d.mkv

   smoothstreaming
       Smooth Streaming muxer generates a set of files (Manifest, chunks)
       suitable for serving with conventional web server.

       window_size
	   Specify the number of fragments kept in the manifest. Default 0
	   (keep all).

       extra_window_size
	   Specify the number of fragments kept outside of the manifest before
	   removing from disk. Default 5.

       lookahead_count
	   Specify the number of lookahead fragments. Default 2.

       min_frag_duration
	   Specify the minimum fragment duration (in microseconds). Default
	   5000000.

       remove_at_exit
	   Specify whether to remove all fragments when finished. Default 0
	   (do not remove).

   streamhash
       Per stream hash testing format.

       This muxer computes and prints a cryptographic hash of all the input
       frames, on a per-stream basis. 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 one line per stream of the form:
       streamindex,streamtype,algo=hash, where streamindex is the index of the
       mapped stream, streamtype is a single character indicating the type of
       stream, 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 streamhash out.sha256

       To print an MD5 hash to stdout use the command:

	       ffmpeg -i INPUT -f streamhash -hash md5 -

       See also the hash and framehash muxers.

   tee
       The tee muxer can be used to write the same data to several outputs,
       such as files or streams.  It can be used, for example, to stream a
       video over a network and save it to disk at the same time.

       It is different from specifying several outputs to the ffmpeg command-
       line tool. With the tee muxer, the audio and video data will be encoded
       only once.  With conventional multiple outputs, multiple encoding
       operations in parallel are initiated, which can be a very expensive
       process. The tee muxer is not useful when using the libavformat API
       directly because it is then possible to feed the same packets to
       several muxers directly.

       Since the tee muxer does not represent any particular output format,
       ffmpeg cannot auto-select output streams. So all streams intended for
       output must be specified using "-map". See the examples below.

       Some encoders may need different options depending on the output
       format; the auto-detection of this can not work with the tee muxer, so
       they need to be explicitly specified.  The main example is the
       global_header flag.

       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, those must be
       escaped (see the "Quoting and escaping" section in the ffmpeg-utils(1)
       manual).

       Options

       use_fifo bool
	   If set to 1, slave outputs will be processed in separate threads
	   using the 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.

       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. Required if it cannot be guessed from the
	   output URL.

       bsfs[/spec]
	   Specify a list of bitstream filters to apply to the specified
	   output.

	   It is possible to specify to which streams a given bitstream filter
	   applies, by appending a stream specifier to the option separated by
	   "/". 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. This will cause that
	   output operation to fail if the output contains streams to which
	   the bitstream filter cannot be applied e.g. "h264_mp4toannexb"
	   being applied to an output containing an audio stream.

	   Options for a bitstream filter must be specified in the form of
	   "opt=value".

	   Several bitstream filters can be specified, separated by ",".

       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.

       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 mapped streams. This will cause that output operation to
	   fail if the output format does not accept all mapped streams.

	   You may use multiple stream specifiers separated by commas (",")
	   e.g.: "a:0,v"

       onfail
	   Specify behaviour on output failure. This can be set to either
	   "abort" (which is default) or "ignore". "abort" will cause whole
	   process to fail in case of failure on this slave output. "ignore"
	   will ignore failure on this output, so other outputs will continue
	   without being affected.

       Examples

       o   Encode something and both archive it in a WebM file and stream it
	   as MPEG-TS over UDP:

		   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		     "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       o   As above, but continue streaming even if output to local file fails
	   (for example local drive fills up):

		   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		     "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       o   Use ffmpeg to encode the input, and send the output to three
	   different destinations. The "dump_extra" bitstream filter is used
	   to add extradata information to all the output video keyframes
	   packets, as requested by the MPEG-TS format. The select option is
	   applied to out.aac in order to make it contain only audio packets.

		   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
			  -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

       o   As above, 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=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

   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

   webm_dash_manifest
       WebM DASH Manifest muxer.

       This muxer implements the WebM DASH Manifest specification to generate
       the DASH manifest XML. It also supports manifest generation for DASH
       live streams.

       For more information see:

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

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

       Options

       This muxer supports the following options:

       adaptation_sets
	   This option has the following syntax: "id=x,streams=a,b,c
	   id=y,streams=d,e" where x and y are the unique identifiers of the
	   adaptation sets and a,b,c,d and e are the indices of the
	   corresponding audio and video streams. Any number of adaptation
	   sets can be added using this option.

       live
	   Set this to 1 to create a live stream DASH Manifest. Default: 0.

       chunk_start_index
	   Start index of the first chunk. This will go in the startNumber
	   attribute of the SegmentTemplate element in the manifest. Default:
	   0.

       chunk_duration_ms
	   Duration of each chunk in milliseconds. This will go in the
	   duration attribute of the SegmentTemplate element in the manifest.
	   Default: 1000.

       utc_timing_url
	   URL of the page that will return the UTC timestamp in ISO format.
	   This will go in the value attribute of the UTCTiming element in the
	   manifest.  Default: None.

       time_shift_buffer_depth
	   Smallest time (in seconds) shifting buffer for which any
	   Representation is guaranteed to be available. This will go in the
	   timeShiftBufferDepth attribute of the MPD element. Default: 60.

       minimum_update_period
	   Minimum update period (in seconds) of the manifest. This will go in
	   the minimumUpdatePeriod attribute of the MPD element. Default: 0.

       Example

	       ffmpeg -f webm_dash_manifest -i video1.webm \
		      -f webm_dash_manifest -i video2.webm \
		      -f webm_dash_manifest -i audio1.webm \
		      -f webm_dash_manifest -i audio2.webm \
		      -map 0 -map 1 -map 2 -map 3 \
		      -c copy \
		      -f webm_dash_manifest \
		      -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
		      manifest.xml

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 nanoseconds.

	   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.

   amqp
       Advanced Message Queueing Protocol (AMQP) version 0-9-1 is a broker
       based publish-subscribe communication protocol.

       FFmpeg must be compiled with --enable-librabbitmq to support AMQP. A
       separate AMQP broker must also be run. An example open-source AMQP
       broker is RabbitMQ.

       After starting the broker, an FFmpeg client may stream data to the
       broker using the command:

	       ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]

       Where hostname and port (default is 5672) is the address of the broker.
       The client may also set a user/password for authentication. The default
       for both fields is "guest". Name of virtual host on broker can be set
       with vhost. The default value is "/".

       Muliple subscribers may stream from the broker using the command:

	       ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]

       In RabbitMQ all data published to the broker flows through a specific
       exchange, and each subscribing client has an assigned queue/buffer.
       When a packet arrives at an exchange, it may be copied to a client's
       queue depending on the exchange and routing_key fields.

       The following options are supported:

       exchange
	   Sets the exchange to use on the broker. RabbitMQ has several
	   predefined exchanges: "amq.direct" is the default exchange, where
	   the publisher and subscriber must have a matching routing_key;
	   "amq.fanout" is the same as a broadcast operation (i.e. the data is
	   forwarded to all queues on the fanout exchange independent of the
	   routing_key); and "amq.topic" is similar to "amq.direct", but
	   allows for more complex pattern matching (refer to the RabbitMQ
	   documentation).

       routing_key
	   Sets the routing key. The default value is "amqp". The routing key
	   is used on the "amq.direct" and "amq.topic" exchanges to decide
	   whether packets are written to the queue of a subscriber.

       pkt_size
	   Maximum size of each packet sent/received to the broker. Default is
	   131072.  Minimum is 4096 and max is any large value (representable
	   by an int). When receiving packets, this sets an internal buffer
	   size in FFmpeg. It should be equal to or greater than the size of
	   the published packets to the broker. Otherwise the received message
	   may be truncated causing decoding errors.

       connection_timeout
	   The timeout in seconds during the initial connection to the broker.
	   The default value is rw_timeout, or 5 seconds if rw_timeout is not
	   set.

       delivery_mode mode
	   Sets the delivery mode of each message sent to broker.  The
	   following values are accepted:

	   persistent
	       Delivery mode set to "persistent" (2). This is the default
	       value.  Messages may be written to the broker's disk depending
	       on its setup.

	   non-persistent
	       Delivery mode set to "non-persistent" (1).  Messages will stay
	       in broker's memory unless the broker is under memory pressure.

   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.

       The accepted options are:

       read_ahead_limit
	   Amount in bytes that may be read ahead when seeking isn't
	   supported. Range is -1 to INT_MAX.  -1 for unlimited. Default is
	   65536.

       URL Syntax is

	       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.

   concatf
       Physical concatenation protocol using a line break delimited list of
       resources.

       Read and seek from many resources in sequence as if they were a unique
       resource.

       A URL accepted by this protocol has the syntax:

	       concatf:<URL>

       where URL is the url containing a line break delimited list of
       resources to be concatenated, each one possibly specifying a distinct
       protocol. Special characters must be escaped with backslash or single
       quotes. See the "Quoting and escaping" section in the ffmpeg-utils(1)
       manual.

       For example to read a sequence of files split1.mpeg, split2.mpeg,
       split3.mpeg listed in separate lines within a file split.txt with
       ffplay use the command:

	       ffplay concatf:split.txt

       Where split.txt contains the lines:

	       split1.mpeg
	       split2.mpeg
	       split3.mpeg

   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

   fd
       File descriptor access protocol.

       The accepted syntax is:

	       fd: -fd <file_descriptor>

       If fd is not specified, by default the stdout file descriptor will be
       used for writing, stdin for reading. Unlike the pipe protocol, fd
       protocol has seek support if it corresponding to a regular file. fd
       protocol doesn't support pass file descriptor via URL for security.

       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.

       fd  Set file descriptor.

   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.

       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).

       seekable
	   Controls if seekability is advertised on the file. 0 means non-
	   seekable, -1 means auto (seekable for normal files, non-seekable
	   for named pipes).

	   Many demuxers handle seekable and non-seekable resources
	   differently, overriding this might speed up opening certain files
	   at the cost of losing some features (e.g. accurate seeking).

   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-user
	   Set a user to be used for authenticating to the FTP server. This is
	   overridden by the user in the FTP URL.

       ftp-password
	   Set a password to be used for authenticating to the FTP server.
	   This is overridden by the password in the FTP URL, or by ftp-
	   anonymous-password if no user is set.

       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.

   gopher
       Gopher protocol.

   gophers
       Gophers protocol.

       The Gopher protocol with TLS encapsulation.

   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.

       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.

       content_type
	   Set a specific content type for the POST messages or for listen
	   mode.

       user_agent
	   Override the User-Agent header. If not specified the protocol will
	   use a string describing the libavformat build. ("Lavf/<version>")

       referer
	   Set the Referer header. Include 'Referer: URL' header in HTTP
	   request.

       multiple_requests
	   Use persistent connections if set to 1, default is 0.

       post_data
	   Set custom HTTP post data.

       mime_type
	   Export the MIME type.

       http_version
	   Exports the HTTP response version number. Usually "1.0" or "1.1".

       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.

       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.

       metadata
	   Set an exported dictionary containing Icecast metadata from the
	   bitstream, if present.  Only useful with the C API.

       auth_type
	   Set HTTP authentication type. No option for Digest, since this
	   method requires getting nonce parameters from the server first and
	   can't be used straight away like Basic.

	   none
	       Choose the HTTP authentication type automatically. This is the
	       default.

	   basic
	       Choose the HTTP basic authentication.

	       Basic authentication sends a Base64-encoded string that
	       contains a user name and password for the client. Base64 is not
	       a form of encryption and should be considered the same as
	       sending the user name and password in clear text (Base64 is a
	       reversible encoding).  If a resource needs to be protected,
	       strongly consider using an authentication scheme other than
	       basic authentication. HTTPS/TLS should be used with basic
	       authentication.	Without these additional security
	       enhancements, basic authentication should not be used to
	       protect sensitive or valuable information.

       send_expect_100
	   Send an Expect: 100-continue header for POST. If set to 1 it will
	   send, if set to 0 it won't, if set to -1 it will try to send if it
	   is applicable. Default value is -1.

       location
	   An exported dictionary containing the content location. Only useful
	   with the C API.

       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.

       reconnect
	   Reconnect automatically when disconnected before EOF is hit.

       reconnect_at_eof
	   If set then eof is treated like an error and causes reconnection,
	   this is useful for live / endless streams.

       reconnect_on_network_error
	   Reconnect automatically in case of TCP/TLS errors during connect.

       reconnect_on_http_error
	   A comma separated list of HTTP status codes to reconnect on. The
	   list can include specific status codes (e.g. '503') or the strings
	   '4xx' / '5xx'.

       reconnect_streamed
	   If set then even streamed/non seekable streams will be reconnected
	   on errors.

       reconnect_delay_max
	   Set the maximum delay in seconds after which to give up
	   reconnecting.

       reconnect_max_retries
	   Set the maximum number of times to retry a connection. Default
	   unset.

       reconnect_delay_total_max
	   Set the maximum total delay in seconds after which to give up
	   reconnecting.

       respect_retry_after
	   If enabled, and a Retry-After header is encountered, its requested
	   reconnection delay will be honored, rather than using exponential
	   backoff. Useful for 429 and 503 errors. Default enabled.

       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 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>

       resource
	   The resource requested by a client, when the experimental HTTP
	   server is in use.

       reply_code
	   The HTTP code returned to the client, when the experimental HTTP
	   server is in use.

       short_seek_size
	   Set the threshold, in bytes, for when a readahead should be
	   prefered over a seek and new HTTP request. This is useful, for
	   example, to make sure the same connection is used for reading large
	   video packets with small audio packets in between.

       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.

       tls Establish a TLS (HTTPS) connection to Icecast.

	       icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

   ipfs
       InterPlanetary File System (IPFS) protocol support. One can access
       files stored on the IPFS network through so-called gateways. These are
       http(s) endpoints.  This protocol wraps the IPFS native protocols
       (ipfs:// and ipns://) to be sent to such a gateway. Users can (and
       should) host their own node which means this protocol will use one's
       local gateway to access files on the IPFS network.

       This protocol accepts the following options:

       gateway
	   Defines the gateway to use. When not set, the protocol will first
	   try locating the local gateway by looking at $IPFS_GATEWAY,
	   $IPFS_PATH and "$HOME/.ipfs/", in that order.

       One can use this protocol in 2 ways. Using IPFS:

	       ffplay ipfs://<hash>

       Or the IPNS protocol (IPNS is mutable IPFS):

	       ffplay ipns://<hash>

   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>]

       If fd isn't specified, 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.

       fd  Set file descriptor.

       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>

   rist
       Reliable Internet Streaming Transport protocol

       The accepted options are:

       rist_profile
	   Supported values:

	   simple
	   main
	       This one is default.

	   advanced
       buffer_size
	   Set internal RIST buffer size in milliseconds for retransmission of
	   data.  Default value is 0 which means the librist default (1 sec).
	   Maximum value is 30 seconds.

       fifo_size
	   Size of the librist receiver output fifo in number of packets. This
	   must be a power of 2.  Defaults to 8192 (vs the librist default of
	   1024).

       overrun_nonfatal=1|0
	   Survive in case of librist fifo buffer overrun. Default value is 0.

       pkt_size
	   Set maximum packet size for sending data. 1316 by default.

       log_level
	   Set loglevel for RIST logging messages. You only need to set this
	   if you explicitly want to enable debug level messages or packet
	   loss simulation, otherwise the regular loglevel is respected.

       secret
	   Set override of encryption secret, by default is unset.

       encryption
	   Set encryption type, by default is disabled.  Acceptable values are
	   128 and 256.

   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_enhanced_codecs
	   Specify the list of codecs the client advertises to support in an
	   enhanced RTMP stream. This option should be set to a comma
	   separated list of fourcc values, like "hvc1,av01,vp09" for multiple
	   codecs or "hvc1" for only one codec. The specified list will be
	   presented in the "fourCcLive" property of the Connect Command
	   Message.

       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.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark: Writing to the socket is currently not optimized to
	   minimize system calls and reduces the efficiency / effect of
	   TCP_NODELAY.

       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.

       buffer_size=size
	   Set the maximum UDP socket buffer size in bytes.

       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.

       localaddr=addr
	   Local IP address of a network interface used for sending packets or
	   joining multicast groups.

       timeout=n
	   Set timeout (in microseconds) of socket I/O operations 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".

       Muxer

       The following options are supported.

       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.

	   Default value is 0.

       rtsp_flags
	   Set RTSP flags.

	   The following values are accepted:

	   latm
	       Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

	   rfc2190
	       Use RFC 2190 packetization instead of RFC 4629 for H.263.

	   skip_rtcp
	       Don't send RTCP sender reports.

	   h264_mode0
	       Use mode 0 for H.264 in RTP.

	   send_bye
	       Send RTCP BYE packets when finishing.

	   Default value is 0.

       min_port
	   Set minimum local UDP port. Default value is 5000.

       max_port
	   Set maximum local UDP port. Default value is 65000.

       buffer_size
	   Set the maximum socket buffer size in bytes.

       pkt_size
	   Set max send packet size (in bytes). Default value is 1472.

       Demuxer

       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.

	   https
	       Use HTTPs tunneling as lower transport protocol, which is
	       useful for passing proxies and widely used for security
	       consideration.

	   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.

	   satip_raw
	       Export raw MPEG-TS stream instead of demuxing. The flag will
	       simply write out the raw stream, with the original PAT/PMT/PIDs
	       intact.

	   Default value is none.

       allowed_media_types
	   Set media types to accept from the server.

	   The following flags are accepted:

	   video
	   audio
	   data
	   subtitle

	   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.

       listen_timeout
	   Set maximum timeout (in seconds) to establish an initial
	   connection. Setting listen_timeout > 0 sets rtsp_flags to listen.
	   Default is -1 which means an infinite timeout when listen mode is
	   set.

       reorder_queue_size
	   Set number of packets to buffer for handling of reordered packets.

       timeout
	   Set socket TCP I/O timeout in microseconds.

       user_agent
	   Override User-Agent header. If not specified, it defaults to the
	   libavformat identifier string.

       buffer_size
	   Set the maximum socket buffer size in bytes.

       When receiving data over UDP, the demuxer tries to reorder received
       packets (since they may arrive out of order, or packets may get lost
       totally). This can be disabled by setting the maximum demuxing delay to
       zero (via the "max_delay" field of AVFormatContext).

       When watching multi-bitrate Real-RTSP streams with ffplay, the streams
       to display can be chosen with "-vst" n and "-ast" n for video and audio
       respectively, and can be switched on the fly by pressing "v" and "a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       o   Watch a stream over UDP, with a max reordering delay of 0.5
	   seconds:

		   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

       o   Watch a stream tunneled over HTTP:

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

       o   Send a stream in realtime to a RTSP server, for others to watch:

		   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       o   Receive a stream in realtime:

		   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a
       protocol handler in libavformat, it is a muxer and demuxer.  It is used
       for signalling of RTP streams, by announcing the SDP for the streams
       regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

	       sap://<destination>[:<port>][?<options>]

       The RTP packets are sent to destination on port port, or to port 5004
       if no port is specified.  options is a "&"-separated list. The
       following options are supported:

       announce_addr=address
	   Specify the destination IP address for sending the announcements
	   to.	If omitted, the announcements are sent to the commonly used
	   SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
	   or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
	   Specify the port to send the announcements on, defaults to 9875 if
	   not specified.

       ttl=ttl
	   Specify the time to live value for the announcements and RTP
	   packets, defaults to 255.

       same_port=0|1
	   If set to 1, send all RTP streams on the same port pair. If zero
	   (the default), all streams are sent on unique ports, with each
	   stream on a port 2 numbers higher than the previous.  VLC/Live555
	   requires this to be set to 1, to be able to receive the stream.
	   The RTP stack in libavformat for receiving requires all streams to
	   be sent on unique ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

	       ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

	       sap://[<address>][:<port>]

       address is the multicast address to listen for announcements on, if
       omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
       port that is listened on, 9875 if omitted.

       The demuxers listens for announcements on the given address and port.
       Once an announcement is received, it tries to receive that particular
       stream.

       Example command lines follow.

       To play back the first stream announced on the normal SAP multicast
       address:

	       ffplay sap://

       To play back the first stream announced on one the default IPv6 SAP
       multicast address:

	       ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

	       sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

       listen
	   If set to any value, listen for an incoming connection. Outgoing
	   connection is done by default.

       max_streams
	   Set the maximum number of streams. By default no limit is set.

   srt
       Haivision Secure Reliable Transport Protocol via libsrt.

       The supported syntax for a SRT URL is:

	       srt://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       or

	       <options> srt://<hostname>:<port>

       options contains a list of '-key val' options.

       This protocol accepts the following options.

       connect_timeout=milliseconds
	   Connection timeout; SRT cannot connect for RTT > 1500 msec (2
	   handshake exchanges) with the default connect timeout of 3 seconds.
	   This option applies to the caller and rendezvous connection modes.
	   The connect timeout is 10 times the value set for the rendezvous
	   mode (which can be used as a workaround for this connection problem
	   with earlier versions).

       ffs=bytes
	   Flight Flag Size (Window Size), in bytes. FFS is actually an
	   internal parameter and you should set it to not less than
	   recv_buffer_size and mss. The default value is relatively large,
	   therefore unless you set a very large receiver buffer, you do not
	   need to change this option. Default value is 25600.

       inputbw=bytes/seconds
	   Sender nominal input rate, in bytes per seconds. Used along with
	   oheadbw, when maxbw is set to relative (0), to calculate maximum
	   sending rate when recovery packets are sent along with the main
	   media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
	   while maxbw is set to relative (0), the actual input rate is
	   evaluated inside the library. Default value is 0.

       iptos=tos
	   IP Type of Service. Applies to sender only. Default value is 0xB8.

       ipttl=ttl
	   IP Time To Live. Applies to sender only. Default value is 64.

       latency=microseconds
	   Timestamp-based Packet Delivery Delay.  Used to absorb bursts of
	   missed packet retransmissions.  This flag sets both rcvlatency and
	   peerlatency to the same value. Note that prior to version 1.3.0
	   this is the only flag to set the latency, however this is
	   effectively equivalent to setting peerlatency, when side is sender
	   and rcvlatency when side is receiver, and the bidirectional stream
	   sending is not supported.

       listen_timeout=microseconds
	   Set socket listen timeout.

       maxbw=bytes/seconds
	   Maximum sending bandwidth, in bytes per seconds.  -1 infinite
	   (CSRTCC limit is 30mbps) 0 relative to input rate (see inputbw) >0
	   absolute limit value Default value is 0 (relative)

       mode=caller|listener|rendezvous
	   Connection mode.  caller opens client connection.  listener starts
	   server to listen for incoming connections.  rendezvous use Rendez-
	   Vous connection mode.  Default value is caller.

       mss=bytes
	   Maximum Segment Size, in bytes. Used for buffer allocation and rate
	   calculation using a packet counter assuming fully filled packets.
	   The smallest MSS between the peers is used. This is 1500 by default
	   in the overall internet.  This is the maximum size of the UDP
	   packet and can be only decreased, unless you have some unusual
	   dedicated network settings. Default value is 1500.

       nakreport=1|0
	   If set to 1, Receiver will send `UMSG_LOSSREPORT` messages
	   periodically until a lost packet is retransmitted or intentionally
	   dropped. Default value is 1.

       oheadbw=percents
	   Recovery bandwidth overhead above input rate, in percents.  See
	   inputbw. Default value is 25%.

       passphrase=string
	   HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
	   79 characters. The passphrase is the shared secret between the
	   sender and the receiver. It is used to generate the Key Encrypting
	   Key using PBKDF2 (Password-Based Key Derivation Function). It is
	   used only if pbkeylen is non-zero. It is used on the receiver only
	   if the received data is encrypted.  The configured passphrase
	   cannot be recovered (write-only).

       enforced_encryption=1|0
	   If true, both connection parties must have the same password set
	   (including empty, that is, with no encryption). If the password
	   doesn't match or only one side is unencrypted, the connection is
	   rejected. Default is true.

       kmrefreshrate=packets
	   The number of packets to be transmitted after which the encryption
	   key is switched to a new key. Default is -1.  -1 means auto
	   (0x1000000 in srt library). The range for this option is integers
	   in the 0 - "INT_MAX".

       kmpreannounce=packets
	   The interval between when a new encryption key is sent and when
	   switchover occurs. This value also applies to the subsequent
	   interval between when switchover occurs and when the old encryption
	   key is decommissioned. Default is -1.  -1 means auto (0x1000 in srt
	   library). The range for this option is integers in the 0 -
	   "INT_MAX".

       snddropdelay=microseconds
	   The sender's extra delay before dropping packets. This delay is
	   added to the default drop delay time interval value.

	   Special value -1: Do not drop packets on the sender at all.

       payload_size=bytes
	   Sets the maximum declared size of a packet transferred during the
	   single call to the sending function in Live mode. Use 0 if this
	   value isn't used (which is default in file mode).  Default is -1
	   (automatic), which typically means MPEG-TS; if you are going to use
	   SRT to send any different kind of payload, such as, for example,
	   wrapping a live stream in very small frames, then you can use a
	   bigger maximum frame size, though not greater than 1456 bytes.

       pkt_size=bytes
	   Alias for payload_size.

       peerlatency=microseconds
	   The latency value (as described in rcvlatency) that is set by the
	   sender side as a minimum value for the receiver.

       pbkeylen=bytes
	   Sender encryption key length, in bytes.  Only can be set to 0, 16,
	   24 and 32.  Enable sender encryption if not 0.  Not required on
	   receiver (set to 0), key size obtained from sender in HaiCrypt
	   handshake.  Default value is 0.

       rcvlatency=microseconds
	   The time that should elapse since the moment when the packet was
	   sent and the moment when it's delivered to the receiver application
	   in the receiving function.  This time should be a buffer time large
	   enough to cover the time spent for sending, unexpectedly extended
	   RTT time, and the time needed to retransmit the lost UDP packet.
	   The effective latency value will be the maximum of this options'
	   value and the value of peerlatency set by the peer side. Before
	   version 1.3.0 this option is only available as latency.

       recv_buffer_size=bytes
	   Set UDP receive buffer size, expressed in bytes.

       send_buffer_size=bytes
	   Set UDP send buffer size, expressed in bytes.

       timeout=microseconds
	   Set raise error timeouts for read, write and connect operations.
	   Note that the SRT library has internal timeouts which can be
	   controlled separately, the value set here is only a cap on those.

       tlpktdrop=1|0
	   Too-late Packet Drop. When enabled on receiver, it skips missing
	   packets that have not been delivered in time and delivers the
	   following packets to the application when their time-to-play has
	   come. It also sends a fake ACK to the sender. When enabled on
	   sender and enabled on the receiving peer, the sender drops the
	   older packets that have no chance of being delivered in time. It
	   was automatically enabled in the sender if the receiver supports
	   it.

       sndbuf=bytes
	   Set send buffer size, expressed in bytes.

       rcvbuf=bytes
	   Set receive buffer size, expressed in bytes.

	   Receive buffer must not be greater than ffs.

       lossmaxttl=packets
	   The value up to which the Reorder Tolerance may grow. When Reorder
	   Tolerance is > 0, then packet loss report is delayed until that
	   number of packets come in. Reorder Tolerance increases every time a
	   "belated" packet has come, but it wasn't due to retransmission
	   (that is, when UDP packets tend to come out of order), with the
	   difference between the latest sequence and this packet's sequence,
	   and not more than the value of this option. By default it's 0,
	   which means that this mechanism is turned off, and the loss report
	   is always sent immediately upon experiencing a "gap" in sequences.

       minversion
	   The minimum SRT version that is required from the peer. A
	   connection to a peer that does not satisfy the minimum version
	   requirement will be rejected.

	   The version format in hex is 0xXXYYZZ for x.y.z in human readable
	   form.

       streamid=string
	   A string limited to 512 characters that can be set on the socket
	   prior to connecting. This stream ID will be able to be retrieved by
	   the listener side from the socket that is returned from srt_accept
	   and was connected by a socket with that set stream ID. SRT does not
	   enforce any special interpretation of the contents of this string.
	   This option doesnXt make sense in Rendezvous connection; the result
	   might be that simply one side will override the value from the
	   other side and itXs the matter of luck which one would win

       srt_streamid=string
	   Alias for streamid to avoid conflict with ffmpeg command line
	   option.

       smoother=live|file
	   The type of Smoother used for the transmission for that socket,
	   which is responsible for the transmission and congestion control.
	   The Smoother type must be exactly the same on both connecting
	   parties, otherwise the connection is rejected.

       messageapi=1|0
	   When set, this socket uses the Message API, otherwise it uses
	   Buffer API. Note that in live mode (see transtype) thereXs only
	   message API available. In File mode you can chose to use one of two
	   modes:

	   Stream API (default, when this option is false). In this mode you
	   may send as many data as you wish with one sending instruction, or
	   even use dedicated functions that read directly from a file. The
	   internal facility will take care of any speed and congestion
	   control. When receiving, you can also receive as many data as
	   desired, the data not extracted will be waiting for the next call.
	   There is no boundary between data portions in the Stream mode.

	   Message API. In this mode your single sending instruction passes
	   exactly one piece of data that has boundaries (a message). Contrary
	   to Live mode, this message may span across multiple UDP packets and
	   the only size limitation is that it shall fit as a whole in the
	   sending buffer. The receiver shall use as large buffer as necessary
	   to receive the message, otherwise the message will not be given up.
	   When the message is not complete (not all packets received or there
	   was a packet loss) it will not be given up.

       transtype=live|file
	   Sets the transmission type for the socket, in particular, setting
	   this option sets multiple other parameters to their default values
	   as required for a particular transmission type.

	   live: Set options as for live transmission. In this mode, you
	   should send by one sending instruction only so many data that fit
	   in one UDP packet, and limited to the value defined first in
	   payload_size (1316 is default in this mode). There is no speed
	   control in this mode, only the bandwidth control, if configured, in
	   order to not exceed the bandwidth with the overhead transmission
	   (retransmitted and control packets).

	   file: Set options as for non-live transmission. See messageapi for
	   further explanations

       linger=seconds
	   The number of seconds that the socket waits for unsent data when
	   closing.  Default is -1. -1 means auto (off with 0 seconds in live
	   mode, on with 180 seconds in file mode). The range for this option
	   is integers in the 0 - "INT_MAX".

       tsbpd=1|0
	   When true, use Timestamp-based Packet Delivery mode. The default
	   behavior depends on the transmission type: enabled in live mode,
	   disabled in file mode.

       For more information see: <https://github.com/Haivision/srt>.

   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.  If set to 0,
	   extract till end of file.

       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

       Play a MPEG-TS file from start offset till end:

	       subfile,,start,32815239,end,0,,:video.ts

   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=2|1|0
	   Listen for an incoming connection. 0 disables listen, 1 enables
	   listen in single client mode, 2 enables listen in multi-client
	   mode. Default value is 0.

       local_addr=addr
	   Local IP address of a network interface used for tcp socket
	   connect.

       local_port=port
	   Local port used for tcp socket connect.

       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.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark: Writing to the socket is currently not optimized to
	   minimize system calls and reduces the efficiency / effect of
	   TCP_NODELAY.

       tcp_mss=bytes
	   Set maximum segment size for outgoing TCP packets, expressed in
	   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 other
	   backends, 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.

       http_proxy
	   The HTTP proxy to tunnel through, e.g. "http://example.com:1234".
	   The proxy must support the CONNECT method.

       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 32 KB for output, 384 KB for input.
	   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
	   Local IP address of a network interface used for sending packets or
	   joining multicast groups.

       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 from the specified addresses. In case of
	   multicast, also subscribe to multicast traffic coming from these
	   addresses only.

       block=address[,address]
	   Ignore packets sent from the specified addresses. In case of
	   multicast, also exclude the source addresses in the multicast
	   subscription.

       fifo_size=units
	   Set the UDP receiving circular buffer size, expressed as a number
	   of packets with size of 188 bytes. If not specified defaults to
	   7*4096.

       overrun_nonfatal=1|0
	   Survive in case of UDP receiving circular buffer overrun. Default
	   value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This option is only relevant in read mode: if no data arrived in
	   more than this time interval, raise error.

       broadcast=1|0
	   Explicitly allow or disallow UDP broadcasting.

	   Note that broadcasting may not work properly on networks having a
	   broadcast storm protection.

       Examples

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

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

       o   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP
	   packets, using a large input buffer:

		   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

       o   Use ffmpeg to receive over UDP from a remote endpoint:

		   ffmpeg -i udp://[<multicast-address>]:<port> ...

   unix
       Unix local socket

       The required syntax for a Unix socket URL is:

	       unix://<filepath>

       The following parameters can be set via command line options (or in
       code via "AVOption"s):

       timeout
	   Timeout in ms.

       listen
	   Create the Unix socket in listening mode.

   zmq
       ZeroMQ asynchronous messaging using the libzmq library.

       This library supports unicast streaming to multiple clients without
       relying on an external server.

       The required syntax for streaming or connecting to a stream is:

	       zmq:tcp://ip-address:port

       Example: Create a localhost stream on port 5555:

	       ffmpeg -re -i input -f mpegts zmq:tcp://127.0.0.1:5555

       Multiple clients may connect to the stream using:

	       ffplay zmq:tcp://127.0.0.1:5555

       Streaming to multiple clients is implemented using a ZeroMQ Pub-Sub
       pattern.  The server side binds to a port and publishes data. Clients
       connect to the server (via IP address/port) and subscribe to the
       stream. The order in which the server and client start generally does
       not matter.

       ffmpeg must be compiled with the --enable-libzmq option to support this
       protocol.

       Options can be set on the ffmpeg/ffplay command line. The following
       options are supported:

       pkt_size
	   Forces the maximum packet size for sending/receiving data. The
	   default value is 131,072 bytes. On the server side, this sets the
	   maximum size of sent packets via ZeroMQ. On the clients, it sets an
	   internal buffer size for receiving packets. Note that pkt_size on
	   the clients should be equal to or greater than pkt_size on the
	   server. Otherwise the received message may be truncated causing
	   decoding errors.

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.

   android_camera
       Android camera input device.

       This input devices uses the Android Camera2 NDK API which is available
       on devices with API level 24+. The availability of android_camera is
       autodetected during configuration.

       This device allows capturing from all cameras on an Android device,
       which are integrated into the Camera2 NDK API.

       The available cameras are enumerated internally and can be selected
       with the camera_index parameter. The input file string is discarded.

       Generally the back facing camera has index 0 while the front facing
       camera has index 1.

       Options

       video_size
	   Set the video size given as a string such as 640x480 or hd720.
	   Falls back to the first available configuration reported by Android
	   if requested video size is not available or by default.

       framerate
	   Set the video framerate.  Falls back to the first available
	   configuration reported by Android if requested framerate is not
	   available or by default (-1).

       camera_index
	   Set the index of the camera to use. Default is 0.

       input_queue_size
	   Set the maximum number of frames to buffer. Default is 5.

   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.

       -capture_raw_data
	   Capture the raw device data. Default is 0.  Using this option may
	   result in receiving the underlying data delivered to the
	   AVFoundation framework. E.g. for muxed devices that sends raw DV
	   data to the framework (like tape-based camcorders), setting this
	   option to false results in extracted video frames captured in the
	   designated pixel format only. Setting this option to true results
	   in receiving the raw DV stream untouched.

       Examples

       o   Print the list of AVFoundation supported devices and exit:

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

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

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

       o   Record video from video device 2 and audio from audio device 1 into
	   out.avi:

		   $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

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

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

       o   Record raw DV data from a suitable input device and write the
	   output into out.dv:

		   $ ffmpeg -f avfoundation -capture_raw_data true -i "zr100:none" out.dv

   bktr
       BSD video input device. Deprecated and will be removed - please contact
       the developers if you are interested in maintaining it.

       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. This option is deprecated, please use the "-sources" option
	   of ffmpeg to list the available input devices.

       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).  Default behavior is autodetection of the input video
	   format, if the hardware supports it.

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

	   auto
	       This is the default which means 8-bit YUV 422 or 8-bit ARGB if
	       format autodetection is used, 8-bit YUV 422 otherwise.

	   uyvy422
	       8-bit YUV 422.

	   yuv422p10
	       10-bit YUV 422.

	   argb
	       8-bit RGB.

	   bgra
	       8-bit RGB.

	   rgb10
	       10-bit RGB.

       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/profile mode. Must be unset, half,
	   full, one_sub_device_full, one_sub_device_half,
	   two_sub_device_full, four_sub_device_half Defaults to unset.

	   Note: DeckLink SDK 11.0 have replaced the duplex property by a
	   profile property.  For the DeckLink Duo 2 and DeckLink Quad 2, a
	   profile is shared between any 2 sub-devices that utilize the same
	   connectors. For the DeckLink 8K Pro, a profile is shared between
	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.

	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
	   one_sub_device_full, one_sub_device_half, two_sub_device_full,
	   four_sub_device_half

	   Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
	   full

       timecode_format
	   Timecode type to include in the frame and video stream metadata.
	   Must be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any,
	   vitc, vitc2, or serial.  Defaults to none (not included).

	   In order to properly support 50/60 fps timecodes, the ordering of
	   the queried timecode types for rp188any is HFR, VITC1, VITC2 and
	   LTC for >30 fps content. Note that this is slightly different to
	   the ordering used by the DeckLink API, which is HFR, VITC1, LTC,
	   VITC2.

       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, wallclock or abs_wallclock.  Defaults to video.

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

       audio_depth
	   Sets the audio sample bit depth. Must be 16 or 32.  Defaults to 16.

       decklink_copyts
	   If set to true, timestamps are forwarded as they are without
	   removing the initial offset.  Defaults to false.

       timestamp_align
	   Capture start time alignment in seconds. If set to nonzero, input
	   frames are dropped till the system timestamp aligns with configured
	   value.  Alignment difference of up to one frame duration is
	   tolerated.  This is useful for maintaining input synchronization
	   across N different hardware devices deployed for 'N-way'
	   redundancy. The system time of different hardware devices should be
	   synchronized with protocols such as NTP or PTP, before using this
	   option.  Note that this method is not foolproof. In some border
	   cases input synchronization may not happen due to thread scheduling
	   jitters in the OS.  Either sync could go wrong by 1 frame or in a
	   rarer case timestamp_align seconds.	Defaults to 0.

       wait_for_tc (bool)
	   Drop frames till a frame with timecode is received. Sometimes
	   serial timecode isn't received with the first input frame. If that
	   happens, the stored stream timecode will be inaccurate. If this
	   option is set to true, input frames are dropped till a frame with
	   timecode is received.  Option timecode_format must be specified.
	   Defaults to false.

       enable_klv(bool)
	   If set to true, extracts KLV data from VANC and outputs KLV
	   packets.  KLV VANC packets are joined based on MID and PSC fields
	   and aggregated into one KLV packet.	Defaults to false.

       Examples

       o   List input devices:

		   ffmpeg -sources decklink

       o   List supported formats:

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

       o   Capture video clip at 1080i50:

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

       o   Capture video clip at 1080i50 10 bit:

		   ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

       o   Capture video clip at 1080i50 with 16 audio channels:

		   ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

   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.

       use_video_device_timestamps
	   If set to false, the timestamp for video frames will be derived
	   from the wallclock instead of the timestamp provided by the capture
	   device. This allows working around devices that provide unreliable
	   timestamps.

       Examples

       o   Print the list of DirectShow supported devices and exit:

		   $ ffmpeg -list_devices true -f dshow -i dummy

       o   Open video device Camera:

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

       o   Open second video device with name Camera:

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

       o   Open video device Camera and audio device Microphone:

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

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

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

       o   Specify pin names to capture by name or alternative name, specify
	   alternative device name:

		   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

       o   Configure a crossbar device, specifying crossbar pins, allow user
	   to adjust video capture properties at startup:

		   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
			-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

   fbdev
       Linux framebuffer input device.

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

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

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

       To record from the framebuffer device /dev/fb0 with ffmpeg:

	       ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

       You can take a single screenshot image with the command:

	       ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

       Options

       framerate
	   Set the frame rate. Default is 25.

   gdigrab
       Win32 GDI-based screen capture device.

       This device allows you to capture a region of the display on Windows.

       Amongst options for the imput filenames are such elements as:

	       desktop

       or

	       title=<window_title>

       or

	       hwnd=<window_hwnd>

       The first option will capture the entire desktop, or a fixed region of
       the desktop. The second and third options will instead capture the
       contents of a single window, regardless of its position on the screen.

       For example, to grab the entire desktop using ffmpeg:

	       ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

       Grab a 640x480 region at position "10,20":

	       ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

       Grab the contents of the window named "Calculator"

	       ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. Use the value 0 to not
	   draw the pointer. Default value is 1.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

       show_region
	   Show grabbed region on screen.

	   If show_region is specified with 1, then the grabbing region will
	   be indicated on screen. With this option, it is easy to know what
	   is being grabbed if only a portion of the screen is grabbed.

	   Note that show_region is incompatible with grabbing the contents of
	   a single window.

	   For example:

		   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

       video_size
	   Set the video frame size. The default is to capture the full screen
	   if desktop is selected, or the full window size if
	   title=window_title is selected.

       offset_x
	   When capturing a region with video_size, set the distance from the
	   left edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary monitor on Windows. If you have a monitor positioned to the
	   left of your primary monitor, you will need to use a negative
	   offset_x value to move the region to that monitor.

       offset_y
	   When capturing a region with video_size, set the distance from the
	   top edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary monitor on Windows. If you have a monitor positioned above
	   your primary monitor, you will need to use a negative offset_y
	   value to move the region to that monitor.

   iec61883
       FireWire DV/HDV input device using libiec61883.

       To enable this input device, you need libiec61883, libraw1394 and
       libavc1394 installed on your system. Use the configure option
       "--enable-libiec61883" to compile with the device enabled.

       The iec61883 capture device supports capturing from a video device
       connected via IEEE1394 (FireWire), using libiec61883 and the new Linux
       FireWire stack (juju). This is the default DV/HDV input method in Linux
       Kernel 2.6.37 and later, since the old FireWire stack was removed.

       Specify the FireWire port to be used as input file, or "auto" to choose
       the first port connected.

       Options

       dvtype
	   Override autodetection of DV/HDV. This should only be used if auto
	   detection does not work, or if usage of a different device type
	   should be prohibited. Treating a DV device as HDV (or vice versa)
	   will not work and result in undefined behavior.  The values auto,
	   dv and hdv are supported.

       dvbuffer
	   Set maximum size of buffer for incoming data, in frames. For DV,
	   this is an exact value. For HDV, it is not frame exact, since HDV
	   does not have a fixed frame size.

       dvguid
	   Select the capture device by specifying its GUID. Capturing will
	   only be performed from the specified device and fails if no device
	   with the given GUID is found. This is useful to select the input if
	   multiple devices are connected at the same time.  Look at
	   /sys/bus/firewire/devices to find out the GUIDs.

       Examples

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

		   ffplay -f iec61883 -i auto

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

		   ffmpeg -f iec61883 -i auto -dvbuffer 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.

   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.  This can be autodetected if you
	   are running Linux 5.7 or later, but needs to be provided for
	   earlier versions.  Defaults to bgr0, which is the most common
	   format used by the Linux console and Xorg X server.

       format_modifier
	   Format modifier to signal on output frames.	This is necessary to
	   import correctly into some APIs.  It can be autodetected if you are
	   running Linux 5.7 or later, but will need to be provided explicitly
	   when needed in earlier versions.  See the libdrm documentation for
	   possible values.

       crtc_id
	   KMS CRTC ID to define the capture source.  The first active plane
	   on the given CRTC will be used.

       plane_id
	   KMS plane ID to define the capture source.  Defaults to the first
	   active plane found if neither crtc_id nor plane_id are specified.

       framerate
	   Framerate to capture at.  This is not synchronised to any page
	   flipping or framebuffer changes - it just defines the interval at
	   which the framebuffer is sampled.  Sampling faster than the
	   framebuffer update rate will generate independent frames with the
	   same content.  Defaults to 30.

       Examples

       o   Capture from the first active plane, download the result to normal
	   frames and encode.  This will only work if the framebuffer is both
	   linear and mappable - if not, the result may be scrambled or fail
	   to download.

		   ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4

       o   Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert
	   to NV12 and encode as H.264.

		   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

       o   To capture only part of a plane the output can be cropped - this
	   can be used to capture a single window, as long as it has a known
	   absolute position and size.	For example, to capture and encode the
	   middle quarter of a 1920x1080 plane:

		   ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4

   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.  The
       filtergraph is specified through the option graph.

       Options

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

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

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

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

       dumpgraph
	   Dump graph to stderr.

       Examples

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

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

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

		   ffplay -f lavfi color=c=pink

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

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

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

		   ffplay -f lavfi "amovie=test.wav"

       o   Read an audio stream and a video stream and play it back with
	   ffplay:

		   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

       o   Dump decoded frames to images and Closed Captions to an RCWT
	   backup:

		   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rcwt 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".

       Options

       framerate
	   Set the frame rate. Default is "ntsc", corresponding to a frame
	   rate of "30000/1001".

       pixel_format
	   Select the pixel format. Default is "uyvy422".

       video_size
	   Set the video size given as a string such as "640x480" or "hd720".
	   Default is "qvga".

   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
	   This option does nothing and is deprecated.

       fragment_size
	   Specify the size in bytes of the minimal buffering fragment in
	   PulseAudio, it will affect the audio latency. By default it is set
	   to 50 ms amount of data.

       wallclock
	   Set the initial PTS using the current time. Default is 1.

       Examples

       Record a stream from default device:

	       ffmpeg -f pulse -i default /tmp/pulse.wav

   sndio
       sndio input device.

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

       The filename to provide to the input device is the device node
       representing the sndio input device, and is usually set to /dev/audio0.

       For example to grab from /dev/audio0 using ffmpeg use the command:

	       ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   video4linux2, v4l2
       Video4Linux2 input video device.

       "v4l2" can be used as alias for "video4linux2".

       If FFmpeg is built with v4l-utils support (by using the
       "--enable-libv4l2" configure option), it is possible to use it with the
       "-use_libv4l2" input device option.

       The name of the device to grab is a file device node, usually Linux
       systems tend to automatically create such nodes when the device (e.g.
       an USB webcam) is plugged into the system, and has a name of the kind
       /dev/videoN, where N is a number associated to the device.

       Video4Linux2 devices usually support a limited set of widthxheight
       sizes and frame rates. You can check which are supported using
       -list_formats all for Video4Linux2 devices.  Some devices, like TV
       cards, support one or more standards. It is possible to list all the
       supported standards using -list_standards all.

       The time base for the timestamps is 1 microsecond. Depending on the
       kernel version and configuration, the timestamps may be derived from
       the real time clock (origin at the Unix Epoch) or the monotonic clock
       (origin usually at boot time, unaffected by NTP or manual changes to
       the clock). The -timestamps abs or -ts abs option can be used to force
       conversion into the real time clock.

       Some usage examples of the video4linux2 device with ffmpeg and ffplay:

       o   List supported formats for a video4linux2 device:

		   ffplay -f video4linux2 -list_formats all /dev/video0

       o   Grab and show the input of a video4linux2 device:

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

       o   Grab and record the input of a video4linux2 device, leave the frame
	   rate and size as previously set:

		   ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

       For more information about Video4Linux, check <http://linuxtv.org/>.

       Options

       standard
	   Set the standard. Must be the name of a supported standard. To get
	   a list of the supported standards, use the list_standards option.

       channel
	   Set the input channel number. Default to -1, which means using the
	   previously selected channel.

       video_size
	   Set the video frame size. The argument must be a string in the form
	   WIDTHxHEIGHT or a valid size abbreviation.

       pixel_format
	   Select the pixel format (only valid for raw video input).

       input_format
	   Set the preferred pixel format (for raw video) or a codec name.
	   This option allows one to select the input format, when several are
	   available.

       framerate
	   Set the preferred video frame rate.

       list_formats
	   List available formats (supported pixel formats, codecs, and frame
	   sizes) and exit.

	   Available values are:

	   all Show all available (compressed and non-compressed) formats.

	   raw Show only raw video (non-compressed) formats.

	   compressed
	       Show only compressed formats.

       list_standards
	   List supported standards and exit.

	   Available values are:

	   all Show all supported standards.

       timestamps, ts
	   Set type of timestamps for grabbed frames.

	   Available values are:

	   default
	       Use timestamps from the kernel.

	   abs Use absolute timestamps (wall clock).

	   mono2abs
	       Force conversion from monotonic to absolute timestamps.

	   Default value is "default".

       use_libv4l2
	   Use libv4l2 (v4l-utils) conversion functions. Default is 0.

   vfwcap
       VfW (Video for Windows) capture input device.

       The filename passed as input is the capture driver number, ranging from
       0 to 9. You may use "list" as filename to print a list of drivers. Any
       other filename will be interpreted as device number 0.

       Options

       video_size
	   Set the video frame size.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

   x11grab
       X11 video input device.

       To enable this input device during configuration you need libxcb
       installed on your system. It will be automatically detected during
       configuration.

       This device allows one to capture a region of an X11 display.

       The filename passed as input has the syntax:

	       [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

       hostname:display_number.screen_number specifies the X11 display name of
       the screen to grab from. hostname can be omitted, and defaults to
       "localhost". The environment variable DISPLAY contains the default
       display name.

       x_offset and y_offset specify the offsets of the grabbed area with
       respect to the top-left border of the X11 screen. They default to 0.

       Check the X11 documentation (e.g. man X) for more detailed information.

       Use the xdpyinfo program for getting basic information about the
       properties of your X11 display (e.g. grep for "name" or "dimensions").

       For example to grab from :0.0 using ffmpeg:

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

       Grab at position "10,20":

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

       Options

       select_region
	   Specify whether to select the grabbing area graphically using the
	   pointer.  A value of 1 prompts the user to select the grabbing area
	   graphically by clicking and dragging. A single click with no
	   dragging will select the whole screen. A region with zero width or
	   height will also select the whole screen. This option overwrites
	   the video_size, grab_x, and grab_y options. Default value is 0.

       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

       window_id
	   Grab this window, instead of the whole screen. Default value is 0,
	   which maps to the whole screen (root window).

	   The id of a window can be found using the xwininfo program,
	   possibly with options -tree and -root.

	   If the window is later enlarged, the new area is not recorded.
	   Video ends when the window is closed, unmapped (i.e., iconified) or
	   shrunk beyond the video size (which defaults to the initial window
	   size).

	   This option disables options follow_mouse and select_region.

       video_size
	   Set the video frame size. Default is the full desktop or window.

       grab_x
       grab_y
	   Set the grabbing region coordinates. They are expressed as offset
	   from the top left corner of the X11 window and correspond to the
	   x_offset and y_offset parameters in the device name. The default
	   value for both options is 0.

OUTPUT DEVICES
       Output devices are configured elements in FFmpeg that can write
       multimedia data to an output device attached to your system.

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

       You can disable all the output devices using the configure option
       "--disable-outdevs", and selectively enable an output device using the
       option "--enable-outdev=OUTDEV", or you can disable a particular input
       device using the option "--disable-outdev=OUTDEV".

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

       A description of the currently available output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) output device.

       Examples

       o   Play a file on default ALSA device:

		   ffmpeg -i INPUT -f alsa default

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

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

   AudioToolbox
       AudioToolbox output device.

       Allows native output to CoreAudio devices on OSX.

       The output filename can be empty (or "-") to refer to the default
       system output device or a number that refers to the device index as
       shown using: "-list_devices true".

       Alternatively, the audio input device can be chosen by index using the

	   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, UIDs and corresponding indices.

       Options

       AudioToolbox supports the following options:

       -audio_device_index <INDEX>
	   Specify the audio device by its index. Overrides anything given in
	   the output filename.

       Examples

       o   Print the list of supported devices and output a sine wave to the
	   default device:

		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -

       o   Output a sine wave to the device with the index 2, overriding any
	   output filename:

		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -

   caca
       CACA output device.

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

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

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

       Options

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

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

       driver
	   Set display driver.

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

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

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

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

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

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

       Examples

       o   The following command shows the ffmpeg output is an CACA window,
	   forcing its size to 80x25:

		   ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

       o   Show the list of available drivers and exit:

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

       o   Show the list of available dither colors and exit:

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

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

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

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

       Options

       list_devices
	   If set to true, print a list of devices and exit.  Defaults to
	   false. This option is deprecated, please use the "-sinks" option of
	   ffmpeg to list the available output devices.

       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.

       duplex_mode
	   Sets the decklink device duplex/profile mode. Must be unset, half,
	   full, one_sub_device_full, one_sub_device_half,
	   two_sub_device_full, four_sub_device_half Defaults to unset.

	   Note: DeckLink SDK 11.0 have replaced the duplex property by a
	   profile property.  For the DeckLink Duo 2 and DeckLink Quad 2, a
	   profile is shared between any 2 sub-devices that utilize the same
	   connectors. For the DeckLink 8K Pro, a profile is shared between
	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.

	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
	   one_sub_device_full, one_sub_device_half, two_sub_device_full,
	   four_sub_device_half

	   Valid profile modes for DeckLink Quad 2 and DeckLink Duo 2: half,
	   full

       timing_offset
	   Sets the genlock timing pixel offset on the used output.  Defaults
	   to unset.

       link
	   Sets the SDI video link configuration on the used output. Must be
	   unset, single link SDI, dual link SDI or quad link SDI.  Defaults
	   to unset.

       sqd Enable Square Division Quad Split mode for Quad-link SDI output.
	   Must be unset, true or false.  Defaults to unset.

       level_a
	   Enable SMPTE Level A mode on the used output.  Must be unset, true
	   or false.  Defaults to unset.

       vanc_queue_size
	   Sets maximum output buffer size in bytes for VANC data. If the
	   buffering reaches this value, outgoing VANC data will be dropped.
	   Defaults to 1048576.

       Examples

       o   List output devices:

		   ffmpeg -sinks decklink

       o   List supported formats:

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

       o   Play video clip:

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

       o   Play video clip with non-standard framerate or video size:

		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

   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. Deprecated and will be removed.

       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. Deprecated and will be
       removed.

       For monitoring purposes in FFmpeg, pipes and a video player such as
       ffplay can be used:

	       ffmpeg -i INPUT -f nut -c:v rawvideo - | ffplay -

       "sdl2" can be used as alias for "sdl".

       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_borderless
	   Set SDL window border off.  Default value is 0 (enable window
	   border).

       window_enable_quit
	   Enable quit action (using window button or keyboard key) when non-
	   zero value is provided.  Default value is 1 (enable quit action).

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

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

       window_x
       window_y
	   Set the position of the window on the screen.

       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.

   v4l2
       Video4Linux2 output device.

   xv
       XV (XVideo) output device.

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

       Options

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

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

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

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

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

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

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

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

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

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

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

       Examples

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

		   ffmpeg -i INPUT OUTPUT -f xv display

       o   Decode and display the input video to multiple X11 windows:

		   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
       The audio resampler supports the following named options.

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

       uchl, used_chlayout
	   Set used input channel layout. Default is unset. 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.

       ichl, in_chlayout
       ochl, out_chlayout
	   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,
       with a few API-only exceptions noted below.  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. The default width (alpha)
	       is 3 and can be changed by setting "param0".

	   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 (API only)
	   Set source width.

       srch (API only)
	   Set source height.

       dstw (API only)
	   Set destination width.

       dsth (API only)
	   Set destination height.

       src_format (API only)
	   Set source pixel format (must be expressed as an integer).

       dst_format (API only)
	   Set destination pixel format (must be expressed as an integer).

       src_range (boolean)
	   If value is set to 1, indicates source is full range. Default value
	   is 0, which indicates source is limited range.

       dst_range (boolean)
	   If value is set to 1, enable full range for destination. Default
	   value is 0, which enables limited range.

       param0, param1
	   Set scaling algorithm parameters. The specified values are specific
	   of some scaling algorithms and ignored by others. The specified
	   values are floating point number values.

       sws_dither
	   Set the dithering algorithm. Accepts one of the following values.
	   Default value is auto.

	   auto
	       automatic choice

	   none
	       no dithering

	   bayer
	       bayer dither

	   ed  error diffusion dither

	   a_dither
	       arithmetic dither, based using addition

	   x_dither
	       arithmetic dither, based using xor (more random/less apparent
	       patterning that a_dither).

       alphablend
	   Set the alpha blending to use when the input has alpha but the
	   output does not.  Default value is none.

	   uniform_color
	       Blend onto a uniform background color

	   checkerboard
	       Blend onto a checkerboard

	   none
	       No blending

FILTERING INTRODUCTION
       Filtering in FFmpeg is enabled through the libavfilter library.

       In libavfilter, a filter can have multiple inputs and multiple outputs.
       To illustrate the sorts of things that are possible, we consider the
       following filtergraph.

			       [main]
	       input --> split ---------------------> overlay --> output
			   |				 ^
			   |[tmp]		   [flip]|
			   +-----> crop --> vflip -------+

       This filtergraph splits the input stream in two streams, then sends one
       stream through the crop filter and the vflip filter, before merging it
       back with the other stream by overlaying it on top. You can use the
       following command to achieve this:

	       ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

       The result will be that the top half of the video is mirrored onto the
       bottom half of the output video.

       Filters in the same linear chain are separated by commas, and distinct
       linear chains of filters are separated by semicolons. In our example,
       crop,vflip are in one linear chain, split and overlay are separately in
       another. The points where the linear chains join are labelled by names
       enclosed in square brackets. In the example, the split filter generates
       two outputs that are associated to the labels [main] and [tmp].

       The stream sent to the second output of split, labelled as [tmp], is
       processed through the crop filter, which crops away the lower half part
       of the video, and then vertically flipped. The overlay filter takes in
       input the first unchanged output of the split filter (which was
       labelled as [main]), and overlay on its lower half the output generated
       by the crop,vflip filterchain.

       Some filters take in input a list of parameters: they are specified
       after the filter name and an equal sign, and are separated from each
       other by a colon.

       There exist so-called source filters that do not have an audio/video
       input, and sink filters that will not have audio/video output.

GRAPH
       The graph2dot program included in the FFmpeg tools directory can be
       used to parse a filtergraph description and issue a corresponding
       textual representation in the dot language.

       Invoke the command:

	       graph2dot -h

       to see how to use graph2dot.

       You can then pass the dot description to the dot program (from the
       graphviz suite of programs) and obtain a graphical representation of
       the filtergraph.

       For example the sequence of commands:

	       echo <GRAPH_DESCRIPTION> | \
	       tools/graph2dot -o graph.tmp && \
	       dot -Tpng graph.tmp -o graph.png && \
	       display graph.png

       can be used to create and display an image representing the graph
       described by the GRAPH_DESCRIPTION string. Note that this string must
       be a complete self-contained graph, with its inputs and outputs
       explicitly defined.  For example if your command line is of the form:

	       ffmpeg -i infile -vf scale=640:360 outfile

       your GRAPH_DESCRIPTION string will need to be of the form:

	       nullsrc,scale=640:360,nullsink

       you may also need to set the nullsrc parameters and add a format filter
       in order to simulate a specific input file.

FILTERGRAPH DESCRIPTION
       A filtergraph is a directed graph of connected filters. It can contain
       cycles, and there can be multiple links between a pair of filters. Each
       link has one input pad on one side connecting it to one filter from
       which it takes its input, and one output pad on the other side
       connecting it to one filter accepting its output.

       Each filter in a filtergraph is an instance of a filter class
       registered in the application, which defines the features and the
       number of input and output pads of the filter.

       A filter with no input pads is called a "source", and a filter with no
       output pads is called a "sink".

   Filtergraph syntax
       A filtergraph has a textual representation, which is recognized by the
       -filter/-vf/-af and -filter_complex options in ffmpeg and -vf/-af in
       ffplay, and by the "avfilter_graph_parse_ptr()" function defined in
       libavfilter/avfilter.h.

       A filterchain consists of a sequence of connected filters, each one
       connected to the previous one in the sequence. A filterchain is
       represented by a list of ","-separated filter descriptions.

       A filtergraph consists of a sequence of filterchains. A sequence of
       filterchains is represented by a list of ";"-separated filterchain
       descriptions.

       A filter is represented by a string of the form:
       [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]

       filter_name is the name of the filter class of which the described
       filter is an instance of, and has to be the name of one of the filter
       classes registered in the program optionally followed by "@id".	The
       name of the filter class is optionally followed by a string
       "=arguments".

       arguments is a string which contains the parameters used to initialize
       the filter instance. It may have one of two forms:

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

       o   A ':'-separated list of value. In this case, the keys are assumed
	   to be the option names in the order they are declared. E.g. the
	   "fade" filter declares three options in this order -- type,
	   start_frame and nb_frames. Then the parameter list in:0:30 means
	   that the value in is assigned to the option type, 0 to start_frame
	   and 30 to nb_frames.

       o   A ':'-separated list of mixed direct value and long key=value
	   pairs. The direct value must precede the key=value pairs, and
	   follow the same constraints order of the previous point. The
	   following key=value pairs can be set in any preferred order.

       If the option value itself is a list of items (e.g. the "format" filter
       takes a list of pixel formats), the items in the list are usually
       separated by |.

       The list of arguments can be quoted using the character ' as initial
       and ending mark, and the character \ for escaping the characters within
       the quoted text; otherwise the argument string is considered terminated
       when the next special character (belonging to the set []=;,) is
       encountered.

       A special syntax implemented in the ffmpeg CLI tool allows loading
       option values from files. This is done be prepending a slash '/' to the
       option name, then the supplied value is interpreted as a path from
       which the actual value is loaded. E.g.

	       ffmpeg -i <INPUT> -vf drawtext=/text=/tmp/some_text <OUTPUT>

       will load the text to be drawn from /tmp/some_text. API users wishing
       to implement a similar feature should use the
       "avfilter_graph_segment_*()" functions together with custom IO code.

       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.

       Leading and trailing whitespaces (space, tabs, or line feeds)
       separating tokens in the filtergraph specification are ignored. This
       means that the filtergraph can be expressed using empty lines and
       spaces to improve redability.

       For example, the filtergraph:

	       testsrc,split[L1],hflip[L2];[L1][L2] hstack

       can be represented as:

	       testsrc,
	       split [L1], hflip [L2];

	       [L1][L2] hstack

       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"

       In order to avoid cumbersome escaping when using a commandline tool
       accepting a filter specification as input, it is advisable to avoid
       direct inclusion of the filter or options specification in the shell.

       For example, in case of the drawtext filter, you might prefer to use
       the textfile option in place of text to specify the text to render.

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;
	   deprecated, do not use

       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.

CHANGING OPTIONS AT RUNTIME WITH A COMMAND
       Some options can be changed during the operation of the filter using a
       command. These options are marked 'T' on the output of ffmpeg -h
       filter=<name of filter>.  The name of the command is the name of the
       option and the argument is the new value.

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.

       ts_sync_mode
	   How strictly to sync streams based on secondary input timestamps;
	   it accepts one of the following values:

	   default
	       Frame from secondary input with the nearest lower or equal
	       timestamp to the primary input frame.

	   nearest
	       Frame from secondary input with the absolute nearest timestamp
	       to the primary input frame.

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.

   aap
       Apply Affine Projection algorithm to the first audio stream using the
       second audio stream.

       This adaptive filter is used to estimate unknown audio based on
       multiple input audio samples.  Affine projection algorithm can make
       trade-offs between computation complexity with convergence speed.

       A description of the accepted options follows.

       order
	   Set the filter order.

       projection
	   Set the projection order.

       mu  Set the filter mu.

       delta
	   Set the coefficient to initialize internal covariance matrix.

       out_mode
	   Set the filter output samples. It accepts the following values:

	   i   Pass the 1st input.

	   d   Pass the 2nd input.

	   o   Pass difference between desired, 2nd input and error signal
	       estimate.

	   n   Pass difference between input, 1st input and error signal
	       estimate.

	   e   Pass error signal estimated samples.

	       Default value is o.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

   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.

       mode
	   Set mode of compressor operation. Can be "upward" or "downward".
	   Default is "downward".

       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.

       Commands

       This filter supports the all above options as commands.

   acontrast
       Simple audio dynamic range compression/expansion filter.

       The filter accepts the following options:

       contrast
	   Set contrast. Default is 33. Allowed range is between 0 and 100.

   acopy
       Copy the input audio source unchanged to the output. This is mainly
       useful for testing purposes.

   acrossfade
       Apply cross fade from one input audio stream to another input audio
       stream.	The cross fade is applied for specified duration near the end
       of first stream.

       The filter accepts the following options:

       nb_samples, ns
	   Specify the number of samples for which the cross fade effect has
	   to last.  At the end of the cross fade effect the first input audio
	   will be completely silent. Default is 44100.

       duration, d
	   Specify the duration of the cross fade effect. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  By default the duration is determined by nb_samples.  If
	   set this option is used instead of nb_samples.

       overlap, o
	   Should first stream end overlap with second stream start. Default
	   is enabled.

       curve1
	   Set curve for cross fade transition for first stream.

       curve2
	   Set curve for cross fade transition for second stream.

	   For description of available curve types see afade filter
	   description.

       Examples

       o   Cross fade from one input to another:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

       o   Cross fade from one input to another but without overlapping:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

   acrossover
       Split audio stream into several bands.

       This filter splits audio stream into two or more frequency ranges.
       Summing all streams back will give flat output.

       The filter accepts the following options:

       split
	   Set split frequencies. Those must be positive and increasing.

       order
	   Set filter order for each band split. This controls filter roll-off
	   or steepness of filter transfer function.  Available values are:

	   2nd 12 dB per octave.

	   4th 24 dB per octave.

	   6th 36 dB per octave.

	   8th 48 dB per octave.

	   10th
	       60 dB per octave.

	   12th
	       72 dB per octave.

	   14th
	       84 dB per octave.

	   16th
	       96 dB per octave.

	   18th
	       108 dB per octave.

	   20th
	       120 dB per octave.

	   Default is 4th.

       level
	   Set input gain level. Allowed range is from 0 to 1. Default value
	   is 1.

       gains
	   Set output gain for each band. Default value is 1 for all bands.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

	   Default value is "auto".

       Examples

       o   Split input audio stream into two bands (low and high) with split
	   frequency of 1500 Hz, each band will be in separate stream:

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

       o   Same as above, but with higher filter order:

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

       o   Same as above, but also with additional middle band (frequencies
	   between 1500 and 8000):

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav

   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, 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.

       Commands

       This filter supports the all above options as commands.

   acue
       Delay audio filtering until a given wallclock timestamp. See the cue
       filter.

   adeclick
       Remove impulsive noise from input audio.

       Samples detected as impulsive noise are replaced by interpolated
       samples using autoregressive modelling.

       window, w
	   Set window size, in milliseconds. Allowed range is from 10 to 100.
	   Default value is 55 milliseconds.  This sets size of window which
	   will be processed at once.

       overlap, o
	   Set window overlap, in percentage of window size. Allowed range is
	   from 50 to 95. Default value is 75 percent.	Setting this to a very
	   high value increases impulsive noise removal but makes whole
	   process much slower.

       arorder, a
	   Set autoregression order, in percentage of window size. Allowed
	   range is from 0 to 25. Default value is 2 percent. This option also
	   controls quality of interpolated samples using neighbour good
	   samples.

       threshold, t
	   Set threshold value. Allowed range is from 1 to 100.  Default value
	   is 2.  This controls the strength of impulsive noise which is going
	   to be removed.  The lower value, the more samples will be detected
	   as impulsive noise.

       burst, b
	   Set burst fusion, in percentage of window size. Allowed range is 0
	   to 10. Default value is 2.  If any two samples detected as noise
	   are spaced less than this value then any sample between those two
	   samples will be also detected as noise.

       method, m
	   Set overlap method.

	   It accepts the following values:

	   add, a
	       Select overlap-add method. Even not interpolated samples are
	       slightly changed with this method.

	   save, s
	       Select overlap-save method. Not interpolated samples remain
	       unchanged.

	   Default value is "a".

   adeclip
       Remove clipped samples from input audio.

       Samples detected as clipped are replaced by interpolated samples using
       autoregressive modelling.

       window, w
	   Set window size, in milliseconds. Allowed range is from 10 to 100.
	   Default value is 55 milliseconds.  This sets size of window which
	   will be processed at once.

       overlap, o
	   Set window overlap, in percentage of window size. Allowed range is
	   from 50 to 95. Default value is 75 percent.

       arorder, a
	   Set autoregression order, in percentage of window size. Allowed
	   range is from 0 to 25. Default value is 8 percent. This option also
	   controls quality of interpolated samples using neighbour good
	   samples.

       threshold, t
	   Set threshold value. Allowed range is from 1 to 100.  Default value
	   is 10. Higher values make clip detection less aggressive.

       hsize, n
	   Set size of histogram used to detect clips. Allowed range is from
	   100 to 9999.  Default value is 1000. Higher values make clip
	   detection less aggressive.

       method, m
	   Set overlap method.

	   It accepts the following values:

	   add, a
	       Select overlap-add method. Even not interpolated samples are
	       slightly changed with this method.

	   save, s
	       Select overlap-save method. Not interpolated samples remain
	       unchanged.

	   Default value is "a".

   adecorrelate
       Apply decorrelation to input audio stream.

       The filter accepts the following options:

       stages
	   Set decorrelation stages of filtering. Allowed range is from 1 to
	   16. Default value is 6.

       seed
	   Set random seed used for setting delay in samples across channels.

   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.  If you want instead to delay in seconds,
	   append 's' to number.

       all Use last set delay for all remaining channels. By default is
	   disabled.  This option if enabled changes how option "delays" is
	   interpreted.

       Examples

       o   Delay first channel by 1.5 seconds, the third channel by 0.5
	   seconds and leave the second channel (and any other channels that
	   may be present) unchanged.

		   adelay=1500|0|500

       o   Delay second channel by 500 samples, the third channel by 700
	   samples and leave the first channel (and any other channels that
	   may be present) unchanged.

		   adelay=0|500S|700S

       o   Delay all channels by same number of samples:

		   adelay=delays=64S:all=1

   adenorm
       Remedy denormals in audio by adding extremely low-level noise.

       This filter shall be placed before any filter that can produce
       denormals.

       A description of the accepted parameters follows.

       level
	   Set level of added noise in dB. Default is "-351".  Allowed range
	   is from -451 to -90.

       type
	   Set type of added noise.

	   dc  Add DC signal.

	   ac  Add AC signal.

	   square
	       Add square signal.

	   pulse
	       Add pulse signal.

	   Default is "dc".

       Commands

       This filter supports the all above options as commands.

   aderivative, aintegral
       Compute derivative/integral of audio stream.

       Applying both filters one after another produces original audio.

   adrc
       Apply spectral dynamic range controller filter to input audio stream.

       A description of the accepted options follows.

       transfer
	   Set the transfer expression.

	   The expression can contain the following constants:

	   ch  current channel number

	   sn  current sample number

	   nb_channels
	       number of channels

	   t   timestamp expressed in seconds

	   sr  sample rate

	   p   current frequency power value, in dB

	   f   current frequency in Hz

	   Default value is "p".

       attack
	   Set the attack in milliseconds. Default is 50 milliseconds.
	   Allowed range is from 1 to 1000 milliseconds.

       release
	   Set the release in milliseconds. Default is 100 milliseconds.
	   Allowed range is from 5 to 2000 milliseconds.

       channels
	   Set which channels to filter, by default "all" channels in audio
	   stream are filtered.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Apply spectral compression to all frequencies with threshold of -50
	   dB and 1:6 ratio:

		   adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100

       o   Similar to above but with 1:2 ratio and filtering only front center
	   channel:

		   adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC

       o   Apply spectral noise gate to all frequencies with threshold of -85
	   dB and with short attack time and short release time:

		   adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5

       o   Apply spectral expansion to all frequencies with threshold of -10
	   dB and 1:2 ratio:

		   adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100

       o   Apply limiter to max -60 dB to all frequencies, with attack of 2 ms
	   and release of 10 ms:

		   adrc=transfer='min(p,-60)':attack=2:release=10

   adynamicequalizer
       Apply dynamic equalization to input audio stream.

       A description of the accepted options follows.

       threshold
	   Set the detection threshold used to trigger equalization.
	   Threshold detection is using detection filter.  Default value is 0.
	   Allowed range is from 0 to 100.

       dfrequency
	   Set the detection frequency in Hz used for detection filter used to
	   trigger equalization.  Default value is 1000 Hz. Allowed range is
	   between 2 and 1000000 Hz.

       dqfactor
	   Set the detection resonance factor for detection filter used to
	   trigger equalization.  Default value is 1. Allowed range is from
	   0.001 to 1000.

       tfrequency
	   Set the target frequency of equalization filter.  Default value is
	   1000 Hz. Allowed range is between 2 and 1000000 Hz.

       tqfactor
	   Set the target resonance factor for target equalization filter.
	   Default value is 1. Allowed range is from 0.001 to 1000.

       attack
	   Set the amount of milliseconds the signal from detection has to
	   rise above the detection threshold before equalization starts.
	   Default is 20. Allowed range is between 1 and 2000.

       release
	   Set the amount of milliseconds the signal from detection has to
	   fall below the detection threshold before equalization ends.
	   Default is 200. Allowed range is between 1 and 2000.

       ratio
	   Set the ratio by which the equalization gain is raised.  Default is
	   1. Allowed range is between 0 and 30.

       makeup
	   Set the makeup offset by which the equalization gain is raised.
	   Default is 0. Allowed range is between 0 and 100.

       range
	   Set the max allowed cut/boost amount. Default is 50.  Allowed range
	   is from 1 to 200.

       mode
	   Set the mode of filter operation, can be one of the following:

	   listen
	       Output only isolated detection signal.

	   cutbelow
	       Cut frequencies below detection threshold.

	   cutabove
	       Cut frequencies above detection threshold.

	   boostbelow
	       Boost frequencies below detection threshold.

	   boostabove
	       Boost frequencies above detection threshold.

	   Default mode is cutbelow.

       dftype
	   Set the type of detection filter, can be one of the following:

	   bandpass
	   lowpass
	   highpass
	   peak

	   Default type is bandpass.

       tftype
	   Set the type of target filter, can be one of the following:

	   bell
	   lowshelf
	   highshelf

	   Default type is bell.

       auto
	   Automatically gather threshold from detection filter. By default is
	   disabled.  This option is useful to detect threshold in certain
	   time frame of input audio stream, in such case option value is
	   changed at runtime.

	   Available values are:

	   disabled
	       Disable using automatically gathered threshold value.

	   off Stop picking threshold value.

	   on  Start picking threshold value.

	   adaptive
	       Adaptively pick threshold value, by calculating sliding window
	       entropy.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

       Commands

       This filter supports the all above options as commands.

   adynamicsmooth
       Apply dynamic smoothing to input audio stream.

       A description of the accepted options follows.

       sensitivity
	   Set an amount of sensitivity to frequency fluctations. Default is
	   2.  Allowed range is from 0 to 1e+06.

       basefreq
	   Set a base frequency for smoothing. Default value is 22050.
	   Allowed range is from 2 to 1e+06.

       Commands

       This filter supports the all above options as commands.

   aecho
       Apply echoing to the input audio.

       Echoes are reflected sound and can occur naturally amongst mountains
       (and sometimes large buildings) when talking or shouting; digital echo
       effects emulate this behaviour and are often used to help fill out the
       sound of a single instrument or vocal. The time difference between the
       original signal and the reflection is the "delay", and the loudness of
       the reflected signal is the "decay".  Multiple echoes can have
       different delays and decays.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain of reflected signal. Default is 0.6.

       out_gain
	   Set output gain of reflected signal. Default is 0.3.

       delays
	   Set list of time intervals in milliseconds between original signal
	   and reflections separated by '|'. Allowed range for each "delay" is
	   "(0 - 90000.0]".  Default is 1000.

       decays
	   Set list of loudness of reflected signals separated by '|'.
	   Allowed range for each "decay" is "(0 - 1.0]".  Default is 0.5.

       Examples

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

		   aecho=0.8:0.88:60:0.4

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

		   aecho=0.8:0.88:6:0.4

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

		   aecho=0.8:0.9:1000:0.3

       o   Same as above but with one more mountain:

		   aecho=0.8:0.9:1000|1800:0.3|0.25

   aemphasis
       Audio emphasis filter creates or restores material directly taken from
       LPs or emphased CDs with different filter curves. E.g. to store music
       on vinyl the signal has to be altered by a filter first to even out the
       disadvantages of this recording medium.	Once the material is played
       back the inverse filter has to be applied to restore the distortion of
       the frequency response.

       The filter accepts the following options:

       level_in
	   Set input gain.

       level_out
	   Set output gain.

       mode
	   Set filter mode. For restoring material use "reproduction" mode,
	   otherwise use "production" mode. Default is "reproduction" mode.

       type
	   Set filter type. Selects medium. Can be one of the following:

	   col select Columbia.

	   emi select EMI.

	   bsi select BSI (78RPM).

	   riaa
	       select RIAA.

	   cd  select Compact Disc (CD).

	   50fm
	       select 50Xs (FM).

	   75fm
	       select 75Xs (FM).

	   50kf
	       select 50Xs (FM-KF).

	   75kf
	       select 75Xs (FM-KF).

       Commands

       This filter supports the all above options as commands.

   aeval
       Modify an audio signal according to the specified expressions.

       This filter accepts one or more expressions (one for each channel),
       which are evaluated and used to modify a corresponding audio signal.

       It accepts the following parameters:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   If the number of input channels is greater than the number of
	   expressions, the last specified expression is used for the
	   remaining output channels.

       channel_layout, c
	   Set output channel layout. If not specified, the channel layout is
	   specified by the number of expressions. If set to same, it will use
	   by default the same input channel layout.

       Each expression in exprs can contain the following constants and
       functions:

       ch  channel number of the current expression

       n   number of the evaluated sample, starting from 0

       s   sample rate

       t   time of the evaluated sample expressed in seconds

       nb_in_channels
       nb_out_channels
	   input and output number of channels

       val(CH)
	   the value of input channel with number CH

       Note: this filter is slow. For faster processing you should use a
       dedicated filter.

       Examples

       o   Half volume:

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

       o   Invert phase of the second channel:

		   aeval=val(0)|-val(1)

   aexciter
       An exciter is used to produce high sound that is not present in the
       original signal. This is done by creating harmonic distortions of the
       signal which are restricted in range and added to the original signal.
       An Exciter raises the upper end of an audio signal without simply
       raising the higher frequencies like an equalizer would do to create a
       more "crisp" or "brilliant" sound.

       The filter accepts the following options:

       level_in
	   Set input level prior processing of signal.	Allowed range is from
	   0 to 64.  Default value is 1.

       level_out
	   Set output level after processing of signal.  Allowed range is from
	   0 to 64.  Default value is 1.

       amount
	   Set the amount of harmonics added to original signal.  Allowed
	   range is from 0 to 64.  Default value is 1.

       drive
	   Set the amount of newly created harmonics.  Allowed range is from
	   0.1 to 10.  Default value is 8.5.

       blend
	   Set the octave of newly created harmonics.  Allowed range is from
	   -10 to 10.  Default value is 0.

       freq
	   Set the lower frequency limit of producing harmonics in Hz.
	   Allowed range is from 2000 to 12000 Hz.  Default is 7500 Hz.

       ceil
	   Set the upper frequency limit of producing harmonics.  Allowed
	   range is from 9999 to 20000 Hz.  If value is lower than 10000 Hz no
	   limit is applied.

       listen
	   Mute the original signal and output only added harmonics.  By
	   default is disabled.

       Commands

       This filter supports the all above options as commands.

   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

	   losi
	       select logistic sigmoid

	   sinc
	       select sine cardinal function

	   isinc
	       select inverted sine cardinal function

	   quat
	       select quartic

	   quatr
	       select quartic root

	   qsin2
	       select squared quarter of sine wave

	   hsin2
	       select squared half of sine wave

	   nofade
	       no fade applied

       silence
	   Set the initial gain for fade-in or final gain for fade-out.
	   Default value is 0.0.

       unity
	   Set the initial gain for fade-out or final gain for fade-in.
	   Default value is 1.0.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Fade in first 15 seconds of audio:

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

       o   Fade out last 25 seconds of a 900 seconds audio:

		   afade=t=out:st=875:d=25

   afftdn
       Denoise audio samples with FFT.

       A description of the accepted parameters follows.

       noise_reduction, nr
	   Set the noise reduction in dB, allowed range is 0.01 to 97.
	   Default value is 12 dB.

       noise_floor, nf
	   Set the noise floor in dB, allowed range is -80 to -20.  Default
	   value is -50 dB.

       noise_type, nt
	   Set the noise type.

	   It accepts the following values:

	   white, w
	       Select white noise.

	   vinyl, v
	       Select vinyl noise.

	   shellac, s
	       Select shellac noise.

	   custom, c
	       Select custom noise, defined in "bn" option.

	       Default value is white noise.

       band_noise, bn
	   Set custom band noise profile for every one of 15 bands.  Bands are
	   separated by ' ' or '|'.

       residual_floor, rf
	   Set the residual floor in dB, allowed range is -80 to -20.  Default
	   value is -38 dB.

       track_noise, tn
	   Enable noise floor tracking. By default is disabled.  With this
	   enabled, noise floor is automatically adjusted.

       track_residual, tr
	   Enable residual tracking. By default is disabled.

       output_mode, om
	   Set the output mode.

	   It accepts the following values:

	   input, i
	       Pass input unchanged.

	   output, o
	       Pass noise filtered out.

	   noise, n
	       Pass only noise.

	       Default value is output.

       adaptivity, ad
	   Set the adaptivity factor, used how fast to adapt gains adjustments
	   per each frequency bin. Value 0 enables instant adaptation, while
	   higher values react much slower.  Allowed range is from 0 to 1.
	   Default value is 0.5.

       floor_offset, fo
	   Set the noise floor offset factor. This option is used to adjust
	   offset applied to measured noise floor. It is only effective when
	   noise floor tracking is enabled.  Allowed range is from -2.0 to
	   2.0. Default value is 1.0.

       noise_link, nl
	   Set the noise link used for multichannel audio.

	   It accepts the following values:

	   none
	       Use unchanged channel's noise floor.

	   min Use measured min noise floor of all channels.

	   max Use measured max noise floor of all channels.

	   average
	       Use measured average noise floor of all channels.

	       Default value is min.

       band_multiplier, bm
	   Set the band multiplier factor, used how much to spread bands
	   across frequency bins.  Allowed range is from 0.2 to 5. Default
	   value is 1.25.

       sample_noise, sn
	   Toggle capturing and measurement of noise profile from input audio.

	   It accepts the following values:

	   start, begin
	       Start sample noise capture.

	   stop, end
	       Stop sample noise capture and measure new noise band profile.

	       Default value is "none".

       gain_smooth, gs
	   Set gain smooth spatial radius, used to smooth gains applied to
	   each frequency bin.	Useful to reduce random music noise artefacts.
	   Higher values increases smoothing of gains.	Allowed range is from
	   0 to 50.  Default value is 0.

       Commands

       This filter supports the some above mentioned options as commands.

       Examples

       o   Reduce white noise by 10dB, and use previously measured noise floor
	   of -40dB:

		   afftdn=nr=10:nf=-40

       o   Reduce white noise by 10dB, also set initial noise floor to -80dB
	   and enable automatic tracking of noise floor so noise floor will
	   gradually change during processing:

		   afftdn=nr=10:nf=-80:tn=1

       o   Reduce noise by 20dB, using noise floor of -40dB and using commands
	   to take noise profile of first 0.4 seconds of input audio:

		   asendcmd=0.0 afftdn sn start,asendcmd=0.4 afftdn sn stop,afftdn=nr=20:nf=-40

   afftfilt
       Apply arbitrary expressions to samples in frequency domain.

       real
	   Set frequency domain real expression for each separate channel
	   separated by '|'. Default is "re".  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 '|'. Default is "im".

	   Each expression in real and imag can contain the following
	   constants and functions:

	   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

	   re  current real part of frequency bin of current channel

	   im  current imaginary part of frequency bin of current channel

	   real(b, ch)
	       Return the value of real part of frequency bin at location
	       (bin,channel)

	   imag(b, ch)
	       Return the value of imaginary part of frequency bin at location
	       (bin,channel)

       win_size
	   Set window size. Allowed range is from 16 to 131072.  Default is
	   4096

       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
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set window overlap. If set to 1, the recommended overlap for
	   selected window function will be picked. Default is 0.75.

       Examples

       o   Leave almost only low frequencies in audio:

		   afftfilt="'real=re * (1-clip((b/nb)*b,0,1))':imag='im * (1-clip((b/nb)*b,0,1))'"

       o   Apply robotize effect:

		   afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"

       o   Apply whisper effect:

		   afftfilt="real='hypot(re,im)*cos((random(0)*2-1)*2*3.14)':imag='hypot(re,im)*sin((random(1)*2-1)*2*3.14)':win_size=128:overlap=0.8"

       o   Apply phase shift:

		   afftfilt="real=re*cos(1)-im*sin(1):imag=re*sin(1)+im*cos(1)"

   afir
       Apply an arbitrary Finite Impulse Response filter.

       This filter is designed for applying long FIR filters, up to 60 seconds
       long.

       It can be used as component for digital crossover filters, room
       equalization, cross talk cancellation, wavefield synthesis,
       auralization, ambiophonics, ambisonics and spatialization.

       This filter uses the streams higher than first one as FIR coefficients.
       If the non-first stream holds a single channel, it will be used for all
       input channels in the first stream, otherwise the number of channels in
       the non-first stream must be same as the number of channels in the
       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.

       gtype
	   This option is deprecated, and does nothing.

       irnorm
	   Set norm to be applied to IR coefficients before filtering.
	   Allowed range is from -1 to 2.  IR coefficients are normalized with
	   calculated vector norm set by this option.  For negative values, no
	   norm is calculated, and IR coefficients are not modified at all.
	   Default is 1.

       irlink
	   For multichannel IR if this option is set to true, all IR channels
	   will be normalized with maximal measured gain of all IR channels
	   coefficients as set by "irnorm" option.  When disabled, all IR
	   coefficients in each IR channel will be normalized independently.
	   Default is true.

       irgain
	   Set gain to be applied to IR coefficients before filtering.
	   Allowed range is 0 to 1. This gain is applied after any gain
	   applied with irnorm option.

       irfmt
	   Set format of IR stream. Can be "mono" or "input".  Default is
	   "input".

       maxir
	   Set max allowed Impulse Response filter duration in seconds.
	   Default is 30 seconds.  Allowed range is 0.1 to 60 seconds.

       response
	   This option is deprecated, and does nothing.

       channel
	   This option is deprecated, and does nothing.

       size
	   This option is deprecated, and does nothing.

       rate
	   This option is deprecated, and does nothing.

       minp
	   Set minimal partition size used for convolution. Default is 8192.
	   Allowed range is from 1 to 65536.  Lower values decreases latency
	   at cost of higher CPU usage.

       maxp
	   Set maximal partition size used for convolution. Default is 8192.
	   Allowed range is from 8 to 65536.  Lower values may increase CPU
	   usage.

       nbirs
	   Set number of input impulse responses streams which will be
	   switchable at runtime.  Allowed range is from 1 to 32. Default is
	   1.

       ir  Set IR stream which will be used for convolution, starting from 0,
	   should always be lower than supplied value by "nbirs" option.
	   Default is 0.  This option can be changed at runtime via commands.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

	   Default value is auto.

       irload
	   Set when to load IR stream. Can be "init" or "access".  First one
	   load and prepares all IRs on initialization, second one once on
	   first access of specific IR.  Default is "init".

       Examples

       o   Apply reverb to stream using mono IR file as second input, complete
	   command using ffmpeg:

		   ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

       o   Apply true stereo processing given input stereo stream, and two
	   stereo impulse responses for left and right channel, the impulse
	   response files are files with names l_ir.wav and r_ir.wav, and
	   setting irnorm option value:

		   "pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irnorm=1.2,pan=stereo|FL<c0+c2|FR<c1+c3"

       o   Similar to above example, but with "irgain" explicitly set to
	   estimated value and with "irnorm" disabled:

		   "pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:irgain=-5dB:irnom=-1,pan=stereo|FL<c0+c2|FR<c1+c3"

   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, f
	   A '|'-separated list of requested sample formats.

       sample_rates, r
	   A '|'-separated list of requested sample rates.

       channel_layouts, cl
	   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

   afreqshift
       Apply frequency shift to input audio samples.

       The filter accepts the following options:

       shift
	   Specify frequency shift. Allowed range is -INT_MAX to INT_MAX.
	   Default value is 0.0.

       level
	   Set output gain applied to final output. Allowed range is from 0.0
	   to 1.0.  Default value is 1.0.

       order
	   Set filter order used for filtering. Allowed range is from 1 to 16.
	   Default value is 8.

       Commands

       This filter supports the all above options as commands.

   afwtdn
       Reduce broadband noise from input samples using Wavelets.

       A description of the accepted options follows.

       sigma
	   Set the noise sigma, allowed range is from 0 to 1.  Default value
	   is 0.  This option controls strength of denoising applied to input
	   samples.  Most useful way to set this option is via decibels, eg.
	   -45dB.

       levels
	   Set the number of wavelet levels of decomposition.  Allowed range
	   is from 1 to 12.  Default value is 10.  Setting this too low make
	   denoising performance very poor.

       wavet
	   Set wavelet type for decomposition of input frame.  They are sorted
	   by number of coefficients, from lowest to highest.  More
	   coefficients means worse filtering speed, but overall better
	   quality.  Available wavelets are:

	   sym2
	   sym4
	   rbior68
	   deb10
	   sym10
	   coif5
	   bl3
       percent
	   Set percent of full denoising. Allowed range is from 0 to 100
	   percent.  Default value is 85 percent or partial denoising.

       profile
	   If enabled, first input frame will be used as noise profile.  If
	   first frame samples contain non-noise performance will be very
	   poor.

       adaptive
	   If enabled, input frames are analyzed for presence of noise.  If
	   noise is detected with high possibility then input frame profile
	   will be used for processing following frames, until new noise frame
	   is detected.

       samples
	   Set size of single frame in number of samples. Allowed range is
	   from 512 to 65536. Default frame size is 8192 samples.

       softness
	   Set softness applied inside thresholding function. Allowed range is
	   from 0 to 10. Default softness is 1.

       Commands

       This filter supports the all above options as commands.

   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.

       mode
	   Set the mode of operation. Can be "upward" or "downward".  Default
	   is "downward". If set to "upward" mode, higher parts of signal will
	   be amplified, expanding dynamic range in upward direction.
	   Otherwise, in case of "downward" lower parts of signal will be
	   reduced.

       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.
	   Setting this to 0 disables reduction and then filter behaves like
	   expander.

       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".

       Commands

       This filter supports the all above options as commands.

   aiir
       Apply an arbitrary Infinite Impulse Response filter.

       It accepts the following parameters:

       zeros, z
	   Set B/numerator/zeros/reflection coefficients.

       poles, p
	   Set A/denominator/poles/ladder coefficients.

       gains, k
	   Set channels gains.

       dry_gain
	   Set input gain.

       wet_gain
	   Set output gain.

       format, f
	   Set coefficients format.

	   ll  lattice-ladder function

	   sf  analog transfer function

	   tf  digital transfer function

	   zp  Z-plane zeros/poles, cartesian (default)

	   pr  Z-plane zeros/poles, polar radians

	   pd  Z-plane zeros/poles, polar degrees

	   sp  S-plane zeros/poles

       process, r
	   Set type of processing.

	   d   direct processing

	   s   serial processing

	   p   parallel processing

       precision, e
	   Set filtering precision.

	   dbl double-precision floating-point (default)

	   flt single-precision floating-point

	   i32 32-bit integers

	   i16 16-bit integers

       normalize, n
	   Normalize filter coefficients, by default is enabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       mix How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       response
	   Show IR frequency response, magnitude(magenta), phase(green) and
	   group delay(yellow) in additional video stream.  By default it is
	   disabled.

       channel
	   Set for which IR channel to display frequency response. By default
	   is first channel displayed. This option is used only when response
	   is enabled.

       size
	   Set video stream size. This option is used only when response is
	   enabled.

       Coefficients in "tf" and "sf" format are separated by spaces and are in
       ascending order.

       Coefficients in "zp" format are separated by spaces and order of
       coefficients doesn't matter. Coefficients in "zp" format are complex
       numbers with i imaginary unit.

       Different coefficients and gains can be provided for every channel, in
       such case use '|' to separate coefficients or gains. Last provided
       coefficients will be used for all remaining channels.

       Examples

       o   Apply 2 pole elliptic notch at around 5000Hz for 48000 Hz sample
	   rate:

		   aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d

       o   Same as above but in "zp" format:

		   aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s

       o   Apply 3-rd order analog normalized Butterworth low-pass filter,
	   using analog transfer function format:

		   aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d

   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.

       latency
	   Compensate the delay introduced by using the lookahead buffer set
	   with attack parameter. Also flush the valid audio data in the
	   lookahead buffer when the stream hits EOF.

       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

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       order, o
	   Set the filter order, can be 1 or 2. Default is 2.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change allpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change allpass width_type.  Syntax for the command is :
	   "width_type"

       width, w
	   Change allpass width.  Syntax for the command is : "width"

       mix, m
	   Change allpass mix.	Syntax for the command is : "mix"

   aloop
       Loop audio samples.

       The filter accepts the following options:

       loop
	   Set the number of loops. Setting this value to -1 will result in
	   infinite loops.  Default is 0.

       size
	   Set maximal number of samples. Default is 0.

       start
	   Set first sample of loop. Default is 0.

       time
	   Set the time of loop start in seconds.  Only used if option named
	   start is set to "-1".

   amerge
       Merge two or more audio streams into a single multi-channel stream.

       The filter accepts the following options:

       inputs
	   Set the number of inputs. Default is 2.

       If the channel layouts of the inputs are disjoint, and therefore
       compatible, the channel layout of the output will be set accordingly
       and the channels will be reordered as necessary. If the channel layouts
       of the inputs are not disjoint, the output will have all the channels
       of the first input then all the channels of the second input, in that
       order, and the channel layout of the output will be the default value
       corresponding to the total number of channels.

       For example, if the first input is in 2.1 (FL+FR+LF) and the second
       input is FC+BL+BR, then the output will be in 5.1, with the channels in
       the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of
       the first input, b1 is the first channel of the second input).

       On the other hand, if both input are in stereo, the output channels
       will be in the default order: a1, a2, b1, b2, and the channel layout
       will be arbitrarily set to 4.0, which may or may not be the expected
       value.

       All inputs must have the same sample rate, and format.

       If inputs do not have the same duration, the output will stop with the
       shortest.

       Examples

       o   Merge two mono files into a stereo stream:

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

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

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

   amix
       Mixes multiple audio inputs into a single output.

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

       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.

       weights
	   Specify weight of each input audio stream as a sequence of numbers
	   separated by a space. If fewer weights are specified compared to
	   number of inputs, the last weight is assigned to the remaining
	   inputs.  Default weight for each input is 1.

       normalize
	   Always scale inputs instead of only doing summation of samples.
	   Beware of heavy clipping if inputs are not normalized prior or
	   after filtering by this filter if this option is disabled. By
	   default is enabled.

       Examples

       o   This 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:

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

       o   This will mix one vocal and one music input audio stream to a
	   single output with the same duration as the longest input. The
	   music will have quarter the weight as the vocals, and the inputs
	   are not normalized:

		   ffmpeg -i VOCALS -i MUSIC -filter_complex amix=inputs=2:duration=longest:dropout_transition=0:weights="1 0.25":normalize=0 OUTPUT

       Commands

       This filter supports the following commands:

       weights
       normalize
	   Syntax is same as option with same name.

   amultiply
       Multiply first audio stream with second audio stream and store result
       in output audio stream. Multiplication is done by multiplying each
       sample from first stream with sample at same position from second
       stream.

       With this element-wise multiplication one can create amplitude fades
       and amplitude modulations.

   anequalizer
       High-order parametric multiband equalizer for each channel.

       It accepts the following parameters:

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

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

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

	   w   Set band width in Hertz.

	   g   Set band gain in dB.

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

	       0   Butterworth, this is default.

	       1   Chebyshev type 1.

	       2   Chebyshev type 2.

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

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

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

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

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

       Examples

       o   Lower gain by 10 of central frequency 200Hz and width 100 Hz for
	   first 2 channels using Chebyshev type 1 filter:

		   anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

       Commands

       This filter supports the following commands:

       change
	   Alter existing filter parameters.  Syntax for the commands is :
	   "fN|f=freq|w=width|g=gain"

	   fN is existing filter number, starting from 0, if no such filter is
	   available error is returned.  freq set new frequency parameter.
	   width set new width parameter in Hertz.  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=...

   anlmdn
       Reduce broadband noise in audio samples using Non-Local Means
       algorithm.

       Each sample is adjusted by looking for other samples with similar
       contexts. This context similarity is defined by comparing their
       surrounding patches of size p. Patches are searched in an area of r
       around the sample.

       The filter accepts the following options:

       strength, s
	   Set denoising strength. Allowed range is from 0.00001 to 10000.
	   Default value is 0.00001.

       patch, p
	   Set patch radius duration. Allowed range is from 1 to 100
	   milliseconds.  Default value is 2 milliseconds.

       research, r
	   Set research radius duration. Allowed range is from 2 to 300
	   milliseconds.  Default value is 6 milliseconds.

       output, o
	   Set the output mode.

	   It accepts the following values:

	   i   Pass input unchanged.

	   o   Pass noise filtered out.

	   n   Pass only noise.

	       Default value is o.

       smooth, m
	   Set smooth factor. Default value is 11. Allowed range is from 1 to
	   1000.

       Commands

       This filter supports the all above options as commands.

   anlmf, anlms
       Apply Normalized Least-Mean-(Squares|Fourth) algorithm to the first
       audio stream using the second audio stream.

       This adaptive filter is used to mimic a desired filter by finding the
       filter coefficients that relate to producing the least mean square of
       the error signal (difference between the desired, 2nd input audio
       stream and the actual signal, the 1st input audio stream).

       A description of the accepted options follows.

       order
	   Set filter order.

       mu  Set filter mu.

       eps Set the filter eps.

       leakage
	   Set the filter leakage.

       out_mode
	   It accepts the following values:

	   i   Pass the 1st input.

	   d   Pass the 2nd input.

	   o   Pass difference between desired, 2nd input and error signal
	       estimate.

	   n   Pass difference between input, 1st input and error signal
	       estimate.

	   e   Pass error signal estimated samples.

	       Default value is o.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

       Examples

       o   One of many usages of this filter is noise reduction, input audio
	   is filtered with same samples that are delayed by fixed amount, one
	   such example for stereo audio is:

		   asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o

       Commands

       This filter supports the same commands as options, excluding option
       "order".

   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.

       pad_dur
	   Specify the duration of samples of silence to add. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax. Used only if set to non-negative value.

       whole_dur
	   Specify the minimum total duration in the output audio stream. See
	   the Time duration section in the ffmpeg-utils(1) manual for the
	   accepted syntax. Used only if set to non-negative value. 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_dur

       If neither the pad_len nor the whole_len nor pad_dur nor whole_dur
       option is set, the filter will add silence to the end of the input
       stream indefinitely.

       Note that for ffmpeg 4.4 and earlier a zero pad_dur or whole_dur also
       caused the filter to add silence indefinitely.

       Examples

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

		   apad=pad_len=1024

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

		   apad=whole_len=10000

       o   Use ffmpeg to pad the audio input with silence, so that the video
	   stream will always result the shortest and will be converted until
	   the end in the output file when using the shortest option:

		   ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

   aphaser
       Add a phasing effect to the input audio.

       A phaser filter creates series of peaks and troughs in the frequency
       spectrum.  The position of the peaks and troughs are modulated so that
       they vary over time, creating a sweeping effect.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.74

       delay
	   Set delay in milliseconds. Default is 3.0.

       decay
	   Set decay. Default is 0.4.

       speed
	   Set modulation speed in Hz. Default is 0.5.

       type
	   Set modulation type. Default is triangular.

	   It accepts the following values:

	   triangular, t
	   sinusoidal, s

   aphaseshift
       Apply phase shift to input audio samples.

       The filter accepts the following options:

       shift
	   Specify phase shift. Allowed range is from -1.0 to 1.0.  Default
	   value is 0.0.

       level
	   Set output gain applied to final output. Allowed range is from 0.0
	   to 1.0.  Default value is 1.0.

       order
	   Set filter order used for filtering. Allowed range is from 1 to 16.
	   Default value is 8.

       Commands

       This filter supports the all above options as commands.

   apsnr
       Measure Audio Peak Signal-to-Noise Ratio.

       This filter takes two audio streams for input, and outputs first audio
       stream.	Results are in dB per channel at end of either input.

   apsyclip
       Apply Psychoacoustic clipper to input audio stream.

       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].

       clip
	   Set the clipping start value. Default value is 0dBFS or 1.

       diff
	   Output only difference samples, useful to hear introduced
	   distortions.  By default is disabled.

       adaptive
	   Set strength of adaptive distortion applied. Default value is 0.5.
	   Allowed range is from 0 to 1.

       iterations
	   Set number of iterations of psychoacoustic clipper.	Allowed range
	   is from 1 to 20. Default value is 10.

       level
	   Auto level output signal. Default is disabled.  This normalizes
	   audio back to 0dBFS if enabled.

       Commands

       This filter supports the all above options as commands.

   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 "Resampler Options" section
       in the ffmpeg-resampler(1) manual for the complete list of supported
       options.

       Examples

       o   Resample the input audio to 44100Hz:

		   aresample=44100

       o   Stretch/squeeze samples to the given timestamps, with a maximum of
	   1000 samples per second compensation:

		   aresample=async=1000

   areverse
       Reverse an audio clip.

       Warning: This filter requires memory to buffer the entire clip, so
       trimming is suggested.

       Examples

       o   Take the first 5 seconds of a clip, and reverse it.

		   atrim=end=5,areverse

   arls
       Apply Recursive Least Squares algorithm to the first audio stream using
       the second audio stream.

       This adaptive filter is used to mimic a desired filter by recursively
       finding the filter coefficients that relate to producing the minimal
       weighted linear least squares cost function of the error signal
       (difference between the desired, 2nd input audio stream and the actual
       signal, the 1st input audio stream).

       A description of the accepted options follows.

       order
	   Set the filter order.

       lambda
	   Set the forgetting factor.

       delta
	   Set the coefficient to initialize internal covariance matrix.

       out_mode
	   Set the filter output samples. It accepts the following values:

	   i   Pass the 1st input.

	   d   Pass the 2nd input.

	   o   Pass difference between desired, 2nd input and error signal
	       estimate.

	   n   Pass difference between input, 1st input and error signal
	       estimate.

	   e   Pass error signal estimated samples.

	       Default value is o.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

   arnndn
       Reduce noise from speech using Recurrent Neural Networks.

       This filter accepts the following options:

       model, m
	   Set train model file to load. This option is always required.

       mix Set how much to mix filtered samples into final output.  Allowed
	   range is from -1 to 1. Default value is 1.  Negative values are
	   special, they set how much to keep filtered noise in the final
	   filter output. Set this option to -1 to hear actual noise removed
	   from input signal.

       Commands

       This filter supports the all above options as commands.

   asdr
       Measure Audio Signal-to-Distortion Ratio.

       This filter takes two audio streams for input, and outputs first audio
       stream.	Results are in dB per channel at end of either input.

   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.

       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.

   asisdr
       Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.

       This filter takes two audio streams for input, and outputs first audio
       stream.	Results are in dB per channel at end of either input.

   asoftclip
       Apply audio soft clipping.

       Soft clipping is a type of distortion effect where the amplitude of a
       signal is saturated along a smooth curve, rather than the abrupt shape
       of hard-clipping.

       This filter accepts the following options:

       type
	   Set type of soft-clipping.

	   It accepts the following values:

	   hard
	   tanh
	   atan
	   cubic
	   exp
	   alg
	   quintic
	   sin
	   erf
       threshold
	   Set threshold from where to start clipping. Default value is 0dB or
	   1.

       output
	   Set gain applied to output. Default value is 0dB or 1.

       param
	   Set additional parameter which controls sigmoid function.

       oversample
	   Set oversampling factor.

       Commands

       This filter supports the all above options as commands.

   aspectralstats
       Display frequency domain statistical information about the audio
       channels.  Statistics are calculated and stored as metadata for each
       audio channel and for each audio frame.

       It accepts the following option:

       win_size
	   Set the window length in samples. Default value is 2048.  Allowed
	   range is from 32 to 65536.

       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
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set window overlap. Allowed range is from 0 to 1. Default value is
	   0.5.

       measure
	   Select the parameters which are measured. The metadata keys can be
	   used as flags, default is all which measures everything.  none
	   disables all measurement.

       A list of each metadata key follows:

       mean
       variance
       centroid
       spread
       skewness
       kurtosis
       entropy
       flatness
       crest
       flux
       slope
       decrease
       rolloff

   asr
       Automatic Speech Recognition

       This filter uses PocketSphinx for speech recognition. To enable
       compilation of this filter, you need to configure FFmpeg with
       "--enable-pocketsphinx".

       It accepts the following options:

       rate
	   Set sampling rate of input audio. Defaults is 16000.  This need to
	   match speech models, otherwise one will get poor results.

       hmm Set dictionary containing acoustic model files.

       dict
	   Set pronunciation dictionary.

       lm  Set language model file.

       lmctl
	   Set language model set.

       lmname
	   Set which language model to use.

       logfn
	   Set output for log messages.

       The filter exports recognized speech as the frame metadata
       "lavfi.asr.text".

   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 - 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: Bit_depth Crest_factor
	   DC_offset Dynamic_range Entropy Flat_factor Max_difference
	   Max_level Mean_difference Min_difference Min_level Noise_floor
	   Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals
	   Peak_count Abs_Peak_count Peak_level RMS_difference RMS_peak
	   RMS_trough Zero_crossings Zero_crossings_rate

	   and for "Overall": Bit_depth DC_offset Entropy Flat_factor
	   Max_difference Max_level Mean_difference Min_difference Min_level
	   Noise_floor Noise_floor_count Number_of_Infs Number_of_NaNs
	   Number_of_denormals Number_of_samples Peak_count Abs_Peak_count
	   Peak_level RMS_difference RMS_level RMS_peak RMS_trough

	   For example, a full key looks like "lavfi.astats.1.DC_offset" or
	   "lavfi.astats.Overall.Peak_count".

	   Read below for the description of the keys.

       reset
	   Set the number of frames over which cumulative stats are calculated
	   before being reset. Default is disabled.

       measure_perchannel
	   Select the parameters which are measured per channel. The metadata
	   keys can be used as flags, default is all which measures
	   everything.	none disables all per channel measurement.

       measure_overall
	   Select the parameters which are measured overall. The metadata keys
	   can be used as flags, default is all which measures everything.
	   none disables all overall measurement.

       A description of the measure keys follow:

       none
	   no measures

       all all measures

       Bit_depth
	   overall bit depth of audio, i.e. number of bits used for each
	   sample

       Crest_factor
	   standard ratio of peak to RMS level (note: not in dB)

       DC_offset
	   mean amplitude displacement from zero

       Dynamic_range
	   measured dynamic range of audio in dB

       Entropy
	   entropy measured across whole audio, entropy of value near 1.0 is
	   typically measured for white noise

       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)

       Max_difference
	   maximal difference between two consecutive samples

       Max_level
	   maximal sample level

       Mean_difference
	   mean difference between two consecutive samples, i.e. the average
	   of each difference between two consecutive samples

       Min_difference
	   minimal difference between two consecutive samples

       Min_level
	   minimal sample level

       Noise_floor
	   minimum local peak measured in dBFS over a short window

       Noise_floor_count
	   number of occasions (not the number of samples) that the signal
	   attained Noise floor

       Number_of_Infs
	   number of samples with an infinite value

       Number_of_NaNs
	   number of samples with a NaN (not a number) value

       Number_of_denormals
	   number of samples with a subnormal value

       Number_of_samples
	   number of samples

       Peak_count
	   number of occasions (not the number of samples) that the signal
	   attained either Min_level or Max_level

       Abs_Peak_count
	   number of occasions that the absolute samples taken from the signal
	   attained max absolute value of Min_level and Max_level

       Peak_level
	   standard peak level measured in dBFS

       RMS_difference
	   Root Mean Square difference between two consecutive samples

       RMS_level
	   standard RMS level measured in dBFS

       RMS_peak
       RMS_trough
	   peak and trough values for RMS level measured over a short window,
	   measured in dBFS.

       Zero crossings
	   number of points where the waveform crosses the zero level axis

       Zero crossings rate
	   rate of Zero crossings and number of audio samples

   asubboost
       Boost subwoofer frequencies.

       The filter accepts the following options:

       dry Set dry gain, how much of original signal is kept. Allowed range is
	   from 0 to 1.  Default value is 1.0.

       wet Set wet gain, how much of filtered signal is kept. Allowed range is
	   from 0 to 1.  Default value is 1.0.

       boost
	   Set max boost factor. Allowed range is from 1 to 12. Default value
	   is 2.

       decay
	   Set delay line decay gain value. Allowed range is from 0 to 1.
	   Default value is 0.0.

       feedback
	   Set delay line feedback gain value. Allowed range is from 0 to 1.
	   Default value is 0.9.

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is 50 to 900.  Default
	   value is 100.

       slope
	   Set slope amount for cutoff frequency. Allowed range is 0.0001 to
	   1.  Default value is 0.5.

       delay
	   Set delay. Allowed range is from 1 to 100.  Default value is 20.

       channels
	   Set the channels to process. Default value is all available.

       Commands

       This filter supports the all above options as commands.

   asubcut
       Cut subwoofer frequencies.

       This filter allows to set custom, steeper roll off than highpass
       filter, and thus is able to more attenuate frequency content in stop-
       band.

       The filter accepts the following options:

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is 2 to 200.  Default
	   value is 20.

       order
	   Set filter order. Available values are from 3 to 20.  Default value
	   is 10.

       level
	   Set input gain level. Allowed range is from 0 to 1. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asupercut
       Cut super frequencies.

       The filter accepts the following options:

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is 20000 to 192000.
	   Default value is 20000.

       order
	   Set filter order. Available values are from 3 to 20.  Default value
	   is 10.

       level
	   Set input gain level. Allowed range is from 0 to 1. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asuperpass
       Apply high order Butterworth band-pass filter.

       The filter accepts the following options:

       centerf
	   Set center frequency in Hertz. Allowed range is 2 to 999999.
	   Default value is 1000.

       order
	   Set filter order. Available values are from 4 to 20.  Default value
	   is 4.

       qfactor
	   Set Q-factor. Allowed range is from 0.01 to 100. Default value is
	   1.

       level
	   Set input gain level. Allowed range is from 0 to 2. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asuperstop
       Apply high order Butterworth band-stop filter.

       The filter accepts the following options:

       centerf
	   Set center frequency in Hertz. Allowed range is 2 to 999999.
	   Default value is 1000.

       order
	   Set filter order. Available values are from 4 to 20.  Default value
	   is 4.

       qfactor
	   Set Q-factor. Allowed range is from 0.01 to 100. Default value is
	   1.

       level
	   Set input gain level. Allowed range is from 0 to 2. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   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, 100.0] range.

       Note that tempo greater than 2 will skip some samples rather than blend
       them in.  If for any reason this is a concern it is always possible to
       daisy-chain several instances of atempo to achieve the desired product
       tempo.

       Examples

       o   Slow down audio to 80% tempo:

		   atempo=0.8

       o   To speed up audio to 300% tempo:

		   atempo=3

       o   To speed up audio to 300% tempo by daisy-chaining two atempo
	   instances:

		   atempo=sqrt(3),atempo=sqrt(3)

       Commands

       This filter supports the following commands:

       tempo
	   Change filter tempo scale factor.  Syntax for the command is :
	   "tempo"

   atilt
       Apply spectral tilt filter to audio stream.

       This filter apply any spectral roll-off slope over any specified
       frequency band.

       The filter accepts the following options:

       freq
	   Set central frequency of tilt in Hz. Default is 10000 Hz.

       slope
	   Set slope direction of tilt. Default is 0. Allowed range is from -1
	   to 1.

       width
	   Set width of tilt. Default is 1000. Allowed range is from 100 to
	   10000.

       order
	   Set order of tilt filter.

       level
	   Set input volume level. Allowed range is from 0 to 4.  Default is
	   1.

       Commands

       This filter supports the all above options as commands.

   atrim
       Trim the input so that the output contains one continuous subpart of
       the input.

       It accepts the following parameters:

       start
	   Timestamp (in seconds) of the start of the section to keep. I.e.
	   the audio sample with the timestamp start will be the first sample
	   in the output.

       end Specify time of the first audio sample that will be dropped, i.e.
	   the audio sample immediately preceding the one with the timestamp
	   end will be the last sample in the output.

       start_pts
	   Same as start, except this option sets the start timestamp in
	   samples instead of seconds.

       end_pts
	   Same as end, except this option sets the end timestamp in samples
	   instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_sample
	   The number of the first sample that should be output.

       end_sample
	   The number of the first sample that should be dropped.

       start, end, and duration are expressed as time duration specifications;
       see the Time duration section in the ffmpeg-utils(1) manual.

       Note that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the _sample options simply
       count the samples that pass through the filter. So start/end_pts and
       start/end_sample will give different results when the timestamps are
       wrong, inexact or do not start at zero. Also note that this filter does
       not modify the timestamps. If you wish to have the output timestamps
       start at zero, insert the asetpts filter after the atrim filter.

       If multiple start or end options are set, this filter tries to be
       greedy and keep all samples that match at least one of the specified
       constraints. To keep only the part that matches all the constraints at
       once, chain multiple atrim filters.

       The defaults are such that all the input is kept. So it is possible to
       set e.g.  just the end values to keep everything before the specified
       time.

       Examples:

       o   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -af atrim=60:120

       o   Keep only the first 1000 samples:

		   ffmpeg -i INPUT -af atrim=end_sample=1000

   axcorrelate
       Calculate normalized windowed cross-correlation between two input audio
       streams.

       Resulted samples are always between -1 and 1 inclusive.	If result is 1
       it means two input samples are highly correlated in that selected
       segment.  Result 0 means they are not correlated at all.  If result is
       -1 it means two input samples are out of phase, which means they cancel
       each other.

       The filter accepts the following options:

       size
	   Set size of segment over which cross-correlation is calculated.
	   Default is 256. Allowed range is from 2 to 131072.

       algo
	   Set algorithm for cross-correlation. Can be "slow" or "fast" or
	   "best".  Default is "best". Fast algorithm assumes mean values over
	   any given segment are always zero and thus need much less
	   calculations to make.  This is generally not true, but is valid for
	   typical audio streams.

       Examples

       o   Calculate correlation between channels in stereo audio stream:

		   ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav

   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

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change bandpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change bandpass width_type.	Syntax for the command is :
	   "width_type"

       width, w
	   Change bandpass width.  Syntax for the command is : "width"

       mix, m
	   Change bandpass mix.  Syntax for the command is : "mix"

   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

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change bandreject frequency.  Syntax for the command is :
	   "frequency"

       width_type, t
	   Change bandreject width_type.  Syntax for the command is :
	   "width_type"

       width, w
	   Change bandreject width.  Syntax for the command is : "width"

       mix, m
	   Change bandreject mix.  Syntax for the command is : "mix"

   bass, lowshelf
       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

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change bass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change bass width_type.  Syntax for the command is : "width_type"

       width, w
	   Change bass width.  Syntax for the command is : "width"

       gain, g
	   Change bass gain.  Syntax for the command is : "gain"

       mix, m
	   Change bass mix.  Syntax for the command is : "mix"

   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.

       Commands

       This filter supports the following commands:

       a0
       a1
       a2
       b0
       b1
       b2  Change biquad parameter.  Syntax for the command is : "value"

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

   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. Mixing different types of mappings is not allowed
	   and will result in a parse error.

       channel_layout
	   The channel layout of the output stream. If not specified, then
	   filter will guess it based on the out_channel names or the number
	   of mappings.  Guessed layouts will not necessarily contain channels
	   in the order of the mappings.

       If no mapping is present, the filter will implicitly map input channels
       to output channels, preserving indices.

       Examples

       o   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.

       o   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".

       channels
	   A channel layout describing the channels to be extracted as
	   separate output streams or "all" to extract each input channel as a
	   separate stream. The default is "all".

	   Choosing channels not present in channel layout in the input will
	   result in an error.

       Examples

       o   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.

       o   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

       o   Extract only LFE from a 5.1 WAV file:

		   ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
		   -map '[LFE]' lfe.wav

   chorus
       Add a chorus effect to the audio.

       Can make a single vocal sound like a chorus, but can also be applied to
       instrumentation.

       Chorus resembles an echo effect with a short delay, but whereas with
       echo the delay is constant, with chorus, it is varied using using
       sinusoidal or triangular modulation.  The modulation depth defines the
       range the modulated delay is played before or after the delay. Hence
       the delayed sound will sound slower or faster, that is the delayed
       sound tuned around the original one, like in a chorus where some vocals
       are slightly off key.

       It accepts the following parameters:

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.4.

       delays
	   Set delays. A typical delay is around 40ms to 60ms.

       decays
	   Set decays.

       speeds
	   Set speeds.

       depths
	   Set depths.

       Examples

       o   A single delay:

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

       o   Two delays:

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

       o   Fuller sounding chorus with three delays:

		   chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

   compand
       Compress or expand the audio's dynamic range.

       It accepts the following parameters:

       attacks
       decays
	   A list of times in seconds for each channel over which the
	   instantaneous level of the input signal is averaged to determine
	   its volume. attacks refers to increase of volume and decays refers
	   to decrease of volume. For most situations, the attack time
	   (response to the audio getting louder) should be shorter than the
	   decay time, because the human ear is more sensitive to sudden loud
	   audio than sudden soft audio. A typical value for attack is 0.3
	   seconds and a typical value for decay is 0.8 seconds.  If specified
	   number of attacks & decays is lower than number of channels, the
	   last set attack/decay will be used for all remaining channels.

       points
	   A list of points for the transfer function, specified in dB
	   relative to the maximum possible signal amplitude. Each key points
	   list must be defined using the following syntax:
	   "x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

	   The input values must be in strictly increasing order but the
	   transfer function does not have to be monotonically rising. The
	   point "0/0" is assumed but may be overridden (by "0/out-dBn").
	   Typical values for the transfer function are "-70/-70|-60/-20|1/0".

       soft-knee
	   Set the curve radius in dB for all joints. It defaults to 0.01.

       gain
	   Set the additional gain in dB to be applied at all points on the
	   transfer function. This allows for easy adjustment of the overall
	   gain.  It defaults to 0.

       volume
	   Set an initial volume, in dB, to be assumed for each channel when
	   filtering starts. This permits the user to supply a nominal level
	   initially, so that, for example, a very large gain is not applied
	   to initial signal levels before the companding has begun to
	   operate. A typical value for audio which is initially quiet is -90
	   dB. It defaults to 0.

       delay
	   Set a delay, in seconds. The input audio is analyzed immediately,
	   but audio is delayed before being fed to the volume adjuster.
	   Specifying a delay approximately equal to the attack/decay times
	   allows the filter to effectively operate in predictive rather than
	   reactive mode. It defaults to 0.

       Examples

       o   Make music with both quiet and loud passages suitable for listening
	   to in a noisy environment:

		   compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

	   Another example for audio with whisper and explosion parts:

		   compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

       o   A noise gate for when the noise is at a lower level than the
	   signal:

		   compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

       o   Here is another noise gate, this time for when the noise is at a
	   higher level than the signal (making it, in some ways, similar to
	   squelch):

		   compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

       o   2:1 compression starting at -6dB:

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

       o   2:1 compression starting at -9dB:

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

       o   2:1 compression starting at -12dB:

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

       o   2:1 compression starting at -18dB:

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

       o   3:1 compression starting at -15dB:

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

       o   Compressor/Gate:

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

       o   Expander:

		   compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3

       o   Hard limiter at -6dB:

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

       o   Hard limiter at -12dB:

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

       o   Hard noise gate at -35 dB:

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

       o   Soft limiter:

		   compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8

   compensationdelay
       Compensation Delay Line is a metric based delay to compensate differing
       positions of microphones or speakers.

       For example, you have recorded guitar with two microphones placed in
       different locations. 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 a
       distance of ~30 cm between microphones makes one microphone capture the
       signal in antiphase to the other microphone. That makes the final mix
       sound 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.

       The filter 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 in degrees Celsius. This is the temperature of the
	   environment.  Default is 20.

       Commands

       This filter supports the all above options as commands.

   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.

       slope
	   Set curve slope of low shelf filter. Default is 0.5.  Allowed range
	   is from 0.01 to 1.

       level_in
	   Set input gain. Default is 0.9.

       level_out
	   Set output gain. Default is 1.

       block_size
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the all above options as commands.

   crystalizer
       Simple algorithm for audio noise sharpening.

       This filter linearly increases differences between each audio sample.

       The filter accepts the following options:

       i   Sets the intensity of effect (default: 2.0). Must be in range
	   between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect).  To
	   inverse filtering use negative value.

       c   Enable clipping. By default is enabled.

       Commands

       This filter supports the all above options as commands.

   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.

   deesser
       Apply de-essing to the audio samples.

       i   Set intensity for triggering de-essing. Allowed range is from 0 to
	   1.  Default is 0.

       m   Set amount of ducking on treble part of sound. Allowed range is
	   from 0 to 1.  Default is 0.5.

       f   How much of original frequency content to keep when de-essing.
	   Allowed range is from 0 to 1.  Default is 0.5.

       s   Set the output mode.

	   It accepts the following values:

	   i   Pass input unchanged.

	   o   Pass ess filtered out.

	   e   Pass only ess.

	       Default value is o.

   dialoguenhance
       Enhance dialogue in stereo audio.

       This filter accepts stereo input and produce surround (3.0) channels
       output.	The newly produced front center channel have enhanced speech
       dialogue originally available in both stereo channels.  This filter
       outputs front left and front right channels same as available in stereo
       input.

       The filter accepts the following options:

       original
	   Set the original center factor to keep in front center channel
	   output.  Allowed range is from 0 to 1. Default value is 1.

       enhance
	   Set the dialogue enhance factor to put in front center channel
	   output.  Allowed range is from 0 to 3. Default value is 1.

       voice
	   Set the voice detection factor.  Allowed range is from 2 to 32.
	   Default value is 2.

       Commands

       This filter supports the all above options as commands.

   drmeter
       Measure audio dynamic range.

       DR values of 14 and higher is found in very dynamic material. DR of 8
       to 13 is found in transition material. And anything less that 8 have
       very poor dynamics and is very compressed.

       The filter accepts the following options:

       length
	   Set window length in seconds used to split audio into segments of
	   equal length.  Default is 3 seconds.

   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.

       framelen, 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.

       gausssize, 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.

       peak, 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.

       maxgain, 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.

       targetrms, 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.

       coupling, 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.

       correctdc, 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.

       altboundary, 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.

       compress, 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.

       threshold, t
	   Set the target threshold value. This specifies the lowest
	   permissible magnitude level for the audio input which will be
	   normalized.	If input frame volume is above this value frame will
	   be normalized.  Otherwise frame may not be normalized at all. The
	   default value is set to 0, which means all input frames will be
	   normalized.	This option is mostly useful if digital noise is not
	   wanted to be amplified.

       channels, h
	   Specify which channels to filter, by default all available channels
	   are filtered.

       overlap, o
	   Specify overlap for frames. If set to 0 (default) no frame
	   overlapping is done.  Using >0 and <1 values will make less
	   conservative gain adjustments, like when framelen option is set to
	   smaller value, if framelen option value is compensated for non-zero
	   overlap then gain adjustments will be smoother across time compared
	   to zero overlap case.

       curve, v
	   Specify the peak mapping curve expression which is going to be used
	   when calculating gain applied to frames. The max output frame gain
	   will still be limited by other options mentioned previously for
	   this filter.

	   The expression can contain the following constants:

	   ch  current channel number

	   sn  current sample number

	   nb_channels
	       number of channels

	   t   timestamp expressed in seconds

	   sr  sample rate

	   p   current frame peak value

       Commands

       This filter supports the all above options as commands.

   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

	   k   kHz

       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.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Examples

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

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

       o   Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz
	   with Q 2:

		   equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

       Commands

       This filter supports the following commands:

       frequency, f
	   Change equalizer frequency.	Syntax for the command is :
	   "frequency"

       width_type, t
	   Change equalizer width_type.  Syntax for the command is :
	   "width_type"

       width, w
	   Change equalizer width.  Syntax for the command is : "width"

       gain, g
	   Change equalizer gain.  Syntax for the command is : "gain"

       mix, m
	   Change equalizer mix.  Syntax for the command is : "mix"

   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.

       Commands

       This filter supports the all above options as commands.

   firequalizer
       Apply FIR Equalization using arbitrary frequency response.

       The filter accepts the following option:

       gain
	   Set gain curve equation (in dB). The expression can contain
	   variables:

	   f   the evaluated frequency

	   sr  sample rate

	   ch  channel number, set to 0 when multichannels evaluation is
	       disabled

	   chid
	       channel id, see libavutil/channel_layout.h, set to the first
	       channel id when multichannels evaluation is disabled

	   chs number of channels

	   chlayout
	       channel_layout, see libavutil/channel_layout.h

	   and functions:

	   gain_interpolate(f)
	       interpolate gain on frequency f based on gain_entry

	   cubic_interpolate(f)
	       same as gain_interpolate, but smoother

	   This option is also available as command. Default is
	   gain_interpolate(f).

       gain_entry
	   Set gain entry for gain_interpolate function. The expression can
	   contain functions:

	   entry(f, g)
	       store gain entry at frequency f with value g

	   This option is also available as command.

       delay
	   Set filter delay in seconds. Higher value means more accurate.
	   Default is 0.01.

       accuracy
	   Set filter accuracy in Hz. Lower value means more accurate.
	   Default is 5.

       wfunc
	   Set window function. Acceptable values are:

	   rectangular
	       rectangular window, useful when gain curve is already smooth

	   hann
	       hann window (default)

	   hamming
	       hamming window

	   blackman
	       blackman window

	   nuttall3
	       3-terms continuous 1st derivative nuttall window

	   mnuttall3
	       minimum 3-terms discontinuous nuttall window

	   nuttall
	       4-terms continuous 1st derivative nuttall window

	   bnuttall
	       minimum 4-terms discontinuous nuttall (blackman-nuttall) window

	   bharris
	       blackman-harris window

	   tukey
	       tukey window

       fixed
	   If enabled, use fixed number of audio samples. This improves speed
	   when filtering with large delay. Default is disabled.

       multi
	   Enable multichannels evaluation on gain. Default is disabled.

       zero_phase
	   Enable zero phase mode by subtracting timestamp to compensate
	   delay.  Default is disabled.

       scale
	   Set scale used by gain. Acceptable values are:

	   linlin
	       linear frequency, linear gain

	   linlog
	       linear frequency, logarithmic (in dB) gain (default)

	   loglin
	       logarithmic (in octave scale where 20 Hz is 0) frequency,
	       linear gain

	   loglog
	       logarithmic frequency, logarithmic gain

       dumpfile
	   Set file for dumping, suitable for gnuplot.

       dumpscale
	   Set scale for dumpfile. Acceptable values are same with scale
	   option.  Default is linlog.

       fft2
	   Enable 2-channel convolution using complex FFT. This improves speed
	   significantly.  Default is disabled.

       min_phase
	   Enable minimum phase impulse response. Default is disabled.

       Examples

       o   lowpass at 1000 Hz:

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

       o   lowpass at 1000 Hz with gain_entry:

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

       o   custom equalization:

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

       o   higher delay with zero phase to compensate delay:

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

       o   lowpass on left channel, highpass on right channel:

		   firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
		   :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on

   flanger
       Apply a flanging effect to the audio.

       The filter accepts the following options:

       delay
	   Set base delay in milliseconds. Range from 0 to 30. Default value
	   is 0.

       depth
	   Set added sweep delay in milliseconds. Range from 0 to 10. Default
	   value is 2.

       regen
	   Set percentage regeneration (delayed signal feedback). Range from
	   -95 to 95.  Default value is 0.

       width
	   Set percentage of delayed signal mixed with original. Range from 0
	   to 100.  Default value is 71.

       speed
	   Set sweeps per second (Hz). Range from 0.1 to 10. Default value is
	   0.5.

       shape
	   Set swept wave shape, can be triangular or sinusoidal.  Default
	   value is sinusoidal.

       phase
	   Set swept wave percentage-shift for multi channel. Range from 0 to
	   100.  Default value is 25.

       interp
	   Set delay-line interpolation, linear or quadratic.  Default is
	   linear.

   haas
       Apply Haas effect to audio.

       Note that this makes most sense to apply on mono signals.  With this
       filter applied to mono signals it give some directionality and
       stretches its stereo image.

       The filter accepts the following options:

       level_in
	   Set input level. By default is 1, or 0dB

       level_out
	   Set output level. By default is 1, or 0dB.

       side_gain
	   Set gain applied to side part of signal. By default is 1.

       middle_source
	   Set kind of middle source. Can be one of the following:

	   left
	       Pick left channel.

	   right
	       Pick right channel.

	   mid Pick middle part signal of stereo image.

	   side
	       Pick side part signal of stereo image.

       middle_phase
	   Change middle phase. By default is disabled.

       left_delay
	   Set left channel delay. By default is 2.05 milliseconds.

       left_balance
	   Set left channel balance. By default is -1.

       left_gain
	   Set left channel gain. By default is 1.

       left_phase
	   Change left phase. By default is disabled.

       right_delay
	   Set right channel delay. By defaults is 2.12 milliseconds.

       right_balance
	   Set right channel balance. By default is 1.

       right_gain
	   Set right channel gain. By default is 1.

       right_phase
	   Change right phase. By default is enabled.

   hdcd
       Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM
       stream with embedded HDCD codes is expanded into a 20-bit PCM stream.

       The filter supports the Peak Extend and Low-level Gain Adjustment
       features of HDCD, and detects the Transient Filter flag.

	       ffmpeg -i HDCD16.flac -af hdcd OUT24.flac

       When using the filter with wav, note the default encoding for wav is
       16-bit, so the resulting 20-bit stream will be truncated back to
       16-bit. Use something like -acodec pcm_s24le after the filter to get
       24-bit PCM output.

	       ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
	       ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav

       The filter accepts the following options:

       disable_autoconvert
	   Disable any automatic format conversion or resampling in the filter
	   graph.

       process_stereo
	   Process the stereo channels together. If target_gain does not match
	   between channels, consider it invalid and use the last valid
	   target_gain.

       cdt_ms
	   Set the code detect timer period in ms.

       force_pe
	   Always extend peaks above -3dBFS even if PE isn't signaled.

       analyze_mode
	   Replace audio with a solid tone and adjust the amplitude to signal
	   some specific aspect of the decoding process. The output file can
	   be loaded in an audio editor alongside the original to aid
	   analysis.

	   "analyze_mode=pe:force_pe=true" can be used to see all samples
	   above the PE level.

	   Modes are:

	   0, off
	       Disabled

	   1, lle
	       Gain adjustment level at each sample

	   2, pe
	       Samples where peak extend occurs

	   3, cdt
	       Samples where the code detect timer is active

	   4, tgm
	       Samples where the target gain does not match between channels

   headphone
       Apply head-related transfer functions (HRTFs) to create virtual
       loudspeakers around the user for binaural listening via headphones.
       The HRIRs are provided via additional streams, for each channel one
       stereo input stream is needed.

       The filter accepts the following options:

       map Set mapping of input streams for convolution.  The argument is a
	   '|'-separated list of channel names in order as they are given as
	   additional stream inputs for filter.  This also specify number of
	   input streams. Number of input streams must be not less than number
	   of channels in first stream plus one.

       gain
	   Set gain applied to audio. Value is in dB. Default is 0.

       type
	   Set processing type. Can be time or freq. time is processing audio
	   in time domain which is slow.  freq is processing audio in
	   frequency domain which is fast.  Default is freq.

       lfe Set custom gain for LFE channels. Value is in dB. Default is 0.

       size
	   Set size of frame in number of samples which will be processed at
	   once.  Default value is 1024. Allowed range is from 1024 to 96000.

       hrir
	   Set format of hrir stream.  Default value is stereo. Alternative
	   value is multich.  If value is set to stereo, number of additional
	   streams should be greater or equal to number of input channels in
	   first input stream.	Also each additional stream should have stereo
	   number of channels.	If value is set to multich, number of
	   additional streams should be exactly one. Also number of input
	   channels of additional stream should be equal or greater than twice
	   number of channels of first input stream.

       Examples

       o   Full example using wav files as coefficients with amovie filters
	   for 7.1 downmix, each amovie filter use stereo file with IR
	   coefficients as input.  The files give coefficients for each
	   position of virtual loudspeaker:

		   ffmpeg -i input.wav
		   -filter_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];[0:a][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
		   output.wav

       o   Full example using wav files as coefficients with amovie filters
	   for 7.1 downmix, but now in multich hrir format.

		   ffmpeg -i input.wav -filter_complex "amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
		   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

	   k   kHz

       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.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change highpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change highpass width_type.	Syntax for the command is :
	   "width_type"

       width, w
	   Change highpass width.  Syntax for the command is : "width"

       mix, m
	   Change highpass mix.  Syntax for the command is : "mix"

   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.

       latency, l
	   Enable latency compensation, by default is disabled.  Only used if
	   plugin have inputs.

       Examples

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

		   ladspa=file=amp

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

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

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

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

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

		   ladspa=file=tap_reverb:tap_reverb

       o   Generate white noise, with 0.2 amplitude:

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

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

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

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

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

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

		   ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

       o   Attenuate low frequencies using Multiband EQ from Steve Harris "SWH
	   Plugins" collection:

		   ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0

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

		   ladspa=caps:Narrower

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

		   ladspa=caps:White:.2

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

		   ladspa=caps:Fractal:c=c1=1

       o   Dynamic volume normalization using "VLevel" plugin:

		   ladspa=vlevel-ladspa:vlevel_mono

       Commands

       This filter supports the following commands:

       cN  Modify the N-th control value.

	   If the specified value is not valid, it is ignored and prior one is
	   kept.

   loudnorm
       EBU R128 loudness normalization. Includes both dynamic and linear
       normalization modes.  Support for both single pass (livestreams, files)
       and double pass (files) modes.  This algorithm can target IL, LRA, and
       maximum true peak. In dynamic mode, to accurately detect true peaks,
       the audio stream will be upsampled to 192 kHz.  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 - 50.0. 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 by linearly scaling the source audio.  "measured_I",
	   "measured_LRA", "measured_TP", and "measured_thresh" must all be
	   specified. Target LRA shouldn't be lower than source LRA and the
	   change in integrated loudness shouldn't result in a true peak which
	   exceeds the target TP. If any of these conditions aren't met,
	   normalization mode will revert to dynamic.  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

	   k   kHz

       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.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Examples

       o   Lowpass only LFE channel, it LFE is not present it does nothing:

		   lowpass=c=LFE

       Commands

       This filter supports the following commands:

       frequency, f
	   Change lowpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change lowpass width_type.  Syntax for the command is :
	   "width_type"

       width, w
	   Change lowpass width.  Syntax for the command is : "width"

       mix, m
	   Change lowpass mix.	Syntax for the command is : "mix"

   lv2
       Load a LV2 (LADSPA Version 2) plugin.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-lv2".

       plugin, p
	   Specifies the plugin URI. You may need to escape ':'.

       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).  If controls is set
	   to "help", all available controls and their valid ranges are
	   printed.

       sample_rate, s
	   Specify the sample rate, default to 44100. Only used if plugin have
	   zero inputs.

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default is 1024. Only used if plugin have zero inputs.

       duration, d
	   Set the minimum duration of the sourced audio. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater than the
	   specified duration, as the generated audio is always cut at the end
	   of a complete frame.  If not specified, or the expressed duration
	   is negative, the audio is supposed to be generated forever.	Only
	   used if plugin have zero inputs.

       Examples

       o   Apply bass enhancer plugin from Calf:

		   lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2

       o   Apply vinyl plugin from Calf:

		   lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5

       o   Apply bit crusher plugin from ArtyFX:

		   lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3

       Commands

       This filter supports all options that are exported by plugin as
       commands.

   mcompand
       Multiband Compress or expand the audio's dynamic range.

       The input audio is divided into bands using 4th order Linkwitz-Riley
       IIRs.  This is akin to the crossover of a loudspeaker, and results in
       flat frequency response when absent compander action.

       It accepts the following parameters:

       args
	   This option syntax is: attack,decay,[attack,decay..] soft-knee
	   points crossover_frequency [delay [initial_volume [gain]]] |
	   attack,decay ...  For explanation of each item refer to compand
	   filter documentation.

   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".

       The filter accepts the following exported read-only options:

       track_gain
	   Exported track gain in dB at end of stream.

       track_peak
	   Exported track peak at end of stream.

   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.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-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

       Commands

       This filter supports the following commands:

       tempo
	   Change filter tempo scale factor.  Syntax for the command is :
	   "tempo"

       pitch
	   Change filter pitch scale factor.  Syntax for the command is :
	   "pitch"

   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.

       mode
	   Set mode of compressor operation. Can be "upward" or "downward".
	   Default is "downward".

       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.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Full ffmpeg example taking 2 audio inputs, 1st input to be
	   compressed depending on the signal of 2nd input and later
	   compressed signal to be merged with 2nd input:

		   ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

   sidechaingate
       A sidechain gate acts like a normal (wideband) gate but has the ability
       to filter the detected signal before sending it to the gain reduction
       stage.  Normally a gate uses the full range signal to detect a level
       above the threshold.  For example: If you cut all lower frequencies
       from your sidechain signal the gate will decrease the volume of your
       track only if not enough highs appear. With this technique you are able
       to reduce the resonation of a natural drum or remove "rumbling" of
       muted strokes from a heavily distorted guitar.  It needs two input
       streams and returns one output stream.  First input stream will be
       processed depending on second stream signal.

       The filter accepts the following options:

       level_in
	   Set input level before filtering.  Default is 1. Allowed range is
	   from 0.015625 to 64.

       mode
	   Set the mode of operation. Can be "upward" or "downward".  Default
	   is "downward". If set to "upward" mode, higher parts of signal will
	   be amplified, expanding dynamic range in upward direction.
	   Otherwise, in case of "downward" lower parts of signal will be
	   reduced.

       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.
	   Setting this to 0 disables reduction and then filter behaves like
	   expander.

       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.

       Commands

       This filter supports the all above options as commands.

   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
       "lavfi.silence_start" or "lavfi.silence_start.X" metadata key is set on
       the first frame whose timestamp equals or exceeds the detection
       duration and it contains the timestamp of the first frame of the
       silence.

       The "lavfi.silence_duration" or "lavfi.silence_duration.X" and
       "lavfi.silence_end" or "lavfi.silence_end.X" metadata keys are set on
       the first frame after the silence. If mono is enabled, and each channel
       is evaluated separately, the ".X" suffixed keys are used, and "X"
       corresponds to the channel number.

       The filter accepts the following options:

       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.

       duration, d
	   Set silence duration until notification (default is 2 seconds). See
	   the Time duration section in the ffmpeg-utils(1) manual for the
	   accepted syntax.

       mono, m
	   Process each channel separately, instead of combined. By default is
	   disabled.

       Examples

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

		   silencedetect=n=-50dB:d=5

       o   Complete example with ffmpeg to detect silence with 0.0001 noise
	   tolerance in silence.mp3:

		   ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

   silenceremove
       Remove silence from the beginning, middle or end of the audio.

       The filter accepts the following options:

       start_periods
	   This value is used to indicate if audio should be trimmed at
	   beginning of the audio. A value of zero indicates no silence should
	   be trimmed from the beginning. When specifying a non-zero value, it
	   trims audio up until it finds non-silence. Normally, when trimming
	   silence from beginning of audio the start_periods will be 1 but it
	   can be increased to higher values to trim all audio up to specific
	   count of non-silence periods.  Default value is 0.

       start_duration
	   Specify the amount of time that non-silence must be detected before
	   it stops trimming audio. By increasing the duration, bursts of
	   noises can be treated as silence and trimmed off. Default value is
	   0.

       start_threshold
	   This indicates what sample value should be treated as silence. For
	   digital audio, a value of 0 may be fine but for audio recorded from
	   analog, you may wish to increase the value to account for
	   background noise.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude ratio. Default value is 0.

       start_silence
	   Specify max duration of silence at beginning that will be kept
	   after trimming. Default is 0, which is equal to trimming all
	   samples detected as silence.

       start_mode
	   Specify mode of detection of silence end at start of multi-channel
	   audio.  Can be any or all. Default is any.  With any, any sample
	   from any channel that is detected as non-silence will trigger end
	   of silence trimming at start of audio stream.  With all, only if
	   every sample from every channel is detected as non-silence will
	   trigger end of silence trimming at start of audio stream, limited
	   usage.

       stop_periods
	   Set the count for trimming silence from the end of audio. When
	   specifying a positive value, it trims audio after it finds
	   specified silence period.  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 stop_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.

       stop_silence
	   Specify max duration of silence at end that will be kept after
	   trimming. Default is 0, which is equal to trimming all samples
	   detected as silence.

       stop_mode
	   Specify mode of detection of silence start after start of multi-
	   channel audio.  Can be any or all. Default is all.  With any, any
	   sample from any channel that is detected as silence will trigger
	   start of silence trimming after start of audio stream, limited
	   usage.  With all, only if every sample from every channel is
	   detected as silence will trigger start of silence trimming after
	   start of audio stream.

       detection
	   Set how is silence detected.

	   avg Mean of absolute values of samples in moving window.

	   rms Root squared mean of absolute values of samples in moving
	       window.

	   peak
	       Maximum of absolute values of samples in moving window.

	   median
	       Median of absolute values of samples in moving window.

	   ptp Absolute of max peak to min peak difference of samples in
	       moving window.

	   dev Standard deviation of values of samples in moving window.

	   Default value is "rms".

       window
	   Set duration in number of seconds used to calculate size of window
	   in number of samples for detecting silence. Using 0 will
	   effectively disable any windowing and use only single sample per
	   channel for silence detection.  In that case it may be needed to
	   also set start_silence and/or stop_silence to nonzero values with
	   also start_duration and/or stop_duration to nonzero values.
	   Default value is 0.02. Allowed range is from 0 to 10.

       timestamp
	   Set processing mode of every audio frame output timestamp.

	   write
	       Full timestamps rewrite, keep only the start time for the first
	       output frame.

	   copy
	       Non-dropped frames are left with same timestamp as input audio
	       frame.

	   Defaults value is "write".

       Examples

       o   The following example shows how this filter can be used to start a
	   recording that does not contain the delay at the start which
	   usually occurs between pressing the record button and the start of
	   the performance:

		   silenceremove=start_periods=1:start_duration=5:start_threshold=0.02

       o   Trim all silence encountered from beginning to end where there is
	   more than 1 second of silence in audio:

		   silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB

       o   Trim all digital silence samples, using peak detection, from
	   beginning to end where there is more than 0 samples of digital
	   silence in audio and digital silence is detected in all channels at
	   same positions in stream:

		   silenceremove=window=0:detection=peak:stop_mode=all:start_mode=all:stop_periods=-1:stop_threshold=0

       o   Trim every 2nd encountered silence period from beginning to end
	   where there is more than 1 second of silence per silence period in
	   audio:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB

       o   Similar as above, but keep maximum of 0.5 seconds of silence from
	   each trimmed period:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5

       o   Similar as above, but keep maximum of 1.5 seconds of silence from
	   start of audio:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5:start_periods=1:start_duration=1:start_silence=1.5:stop_threshold=-90dB

       Commands

       This filter supports some above options as commands.

   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.

       framesize
	   Set custom frame size in number of samples. Default is 1024.
	   Allowed range is from 1024 to 96000. Only used if option type is
	   set to freq.

       normalize
	   Should all IRs be normalized upon importing SOFA file.  By default
	   is enabled.

       interpolate
	   Should nearest IRs be interpolated with neighbor IRs if exact
	   position does not match. By default is disabled.

       minphase
	   Minphase all IRs upon loading of SOFA file. By default is disabled.

       anglestep
	   Set neighbor search angle step. Only used if option interpolate is
	   enabled.

       radstep
	   Set neighbor search radius step. Only used if option interpolate is
	   enabled.

       Examples

       o   Using ClubFritz6 sofa file:

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

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

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

       o   Similar as above but with custom speaker positions for front left,
	   front right, back left and back right and also with custom gain:

		   "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"

   speechnorm
       Speech Normalizer.

       This filter expands or compresses each half-cycle of audio samples
       (local set of samples all above or all below zero and between two
       nearest zero crossings) depending on threshold value, so audio reaches
       target peak value under conditions controlled by below options.

       The filter accepts the following options:

       peak, p
	   Set the expansion target peak value. This specifies the highest
	   allowed absolute amplitude level for the normalized audio input.
	   Default value is 0.95. Allowed range is from 0.0 to 1.0.

       expansion, e
	   Set the maximum expansion factor. Allowed range is from 1.0 to
	   50.0. Default value is 2.0.	This option controls maximum local
	   half-cycle of samples expansion. The maximum expansion would be
	   such that local peak value reaches target peak value but never to
	   surpass it and that ratio between new and previous peak value does
	   not surpass this option value.

       compression, c
	   Set the maximum compression factor. Allowed range is from 1.0 to
	   50.0. Default value is 2.0.	This option controls maximum local
	   half-cycle of samples compression. This option is used only if
	   threshold option is set to value greater than 0.0, then in such
	   cases when local peak is lower or same as value set by threshold
	   all samples belonging to that peak's half-cycle will be compressed
	   by current compression factor.

       threshold, t
	   Set the threshold value. Default value is 0.0. Allowed range is
	   from 0.0 to 1.0.  This option specifies which half-cycles of
	   samples will be compressed and which will be expanded.  Any half-
	   cycle samples with their local peak value below or same as this
	   option value will be compressed by current compression factor,
	   otherwise, if greater than threshold value they will be expanded
	   with expansion factor so that it could reach peak target value but
	   never surpass it.

       raise, r
	   Set the expansion raising amount per each half-cycle of samples.
	   Default value is 0.001.  Allowed range is from 0.0 to 1.0. This
	   controls how fast expansion factor is raised per each new half-
	   cycle until it reaches expansion value.  Setting this options too
	   high may lead to distortions.

       fall, f
	   Set the compression raising amount per each half-cycle of samples.
	   Default value is 0.001.  Allowed range is from 0.0 to 1.0. This
	   controls how fast compression factor is raised per each new half-
	   cycle until it reaches compression value.

       channels, h
	   Specify which channels to filter, by default all available channels
	   are filtered.

       invert, i
	   Enable inverted filtering, by default is disabled. This inverts
	   interpretation of threshold option. When enabled any half-cycle of
	   samples with their local peak value below or same as threshold
	   option will be expanded otherwise it will be compressed.

       link, l
	   Link channels when calculating gain applied to each filtered
	   channel sample, by default is disabled.  When disabled each
	   filtered channel gain calculation is independent, otherwise when
	   this option is enabled the minimum of all possible gains for each
	   filtered channel is used.

       rms, m
	   Set the expansion target RMS value. This specifies the highest
	   allowed RMS level for the normalized audio input. Default value is
	   0.0, thus disabled. Allowed range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Weak and slow amplification:

		   speechnorm=e=3:r=0.00001:l=1

       o   Moderate and slow amplification:

		   speechnorm=e=6.25:r=0.00001:l=1

       o   Strong and fast amplification:

		   speechnorm=e=12.5:r=0.0001:l=1

       o   Very strong and fast amplification:

		   speechnorm=e=25:r=0.0001:l=1

       o   Extreme and fast amplification:

		   speechnorm=e=50:r=0.0001:l=1

   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.

	   ms>rl
	       Mid/Side to Right/Left.

	   lr>l-r
	       Left/Right to Left - 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.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Apply karaoke like effect:

		   stereotools=mlev=0.015625

       o   Convert M/S signal to L/R:

		   "stereotools=mode=ms>lr"

   stereowiden
       This filter enhance the stereo effect by suppressing signal common to
       both channels and by delaying the signal of left into right and vice
       versa, thereby widening the stereo effect.

       The filter accepts the following options:

       delay
	   Time in milliseconds of the delay of left signal into right and
	   vice versa.	Default is 20 milliseconds.

       feedback
	   Amount of gain in delayed signal into right and vice versa. Gives a
	   delay effect of left signal in right output and vice versa which
	   gives widening effect. Default is 0.3.

       crossfeed
	   Cross feed of left into right with inverted phase. This helps in
	   suppressing the mono. If the value is 1 it will cancel all the
	   signal common to both channels. Default is 0.3.

       drymix
	   Set level of input signal of original channel. Default is 0.8.

       Commands

       This filter supports the all above options except "delay" as commands.

   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.

       lfe_mode
	   Set LFE mode, can be add or sub. Default is add.  In add mode, LFE
	   channel is created from input audio and added to output.  In sub
	   mode, LFE channel is created from input audio and added to output
	   but also all non-LFE output channels are subtracted with output LFE
	   channel.

       smooth
	   Set temporal smoothness strength, used to gradually change factors
	   when transforming stereo sound in time. Allowed range is from 0.0
	   to 1.0.  Useful to improve output quality with focus option values
	   greater than 0.0.  Default is 0.0. Only values inside this range
	   and without edges are effective.

       angle
	   Set angle of stereo surround transform, Allowed range is from 0 to
	   360.  Default is 90.

       focus
	   Set focus of stereo surround transform, Allowed range is from -1 to
	   1.  Default is 0.

       fc_in
	   Set front center input volume. By default, this is 1.

       fc_out
	   Set front center output volume. By default, this is 1.

       fl_in
	   Set front left input volume. By default, this is 1.

       fl_out
	   Set front left output volume. By default, this is 1.

       fr_in
	   Set front right input volume. By default, this is 1.

       fr_out
	   Set front right output volume. By default, this is 1.

       sl_in
	   Set side left input volume. By default, this is 1.

       sl_out
	   Set side left output volume. By default, this is 1.

       sr_in
	   Set side right input volume. By default, this is 1.

       sr_out
	   Set side right output volume. By default, this is 1.

       bl_in
	   Set back left input volume. By default, this is 1.

       bl_out
	   Set back left output volume. By default, this is 1.

       br_in
	   Set back right input volume. By default, this is 1.

       br_out
	   Set back right output volume. By default, this is 1.

       bc_in
	   Set back center input volume. By default, this is 1.

       bc_out
	   Set back 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.

       allx
	   Set spread usage of stereo image across X axis for all channels.
	   Allowed range is from -1 to 15.  By default this value is negative
	   -1, and thus unused.

       ally
	   Set spread usage of stereo image across Y axis for all channels.
	   Allowed range is from -1 to 15.  By default this value is negative
	   -1, and thus unused.

       fcx, flx, frx, blx, brx, slx, srx, bcx
	   Set spread usage of stereo image across X axis for each channel.
	   Allowed range is from 0.06 to 15.  By default this value is 0.5.

       fcy, fly, fry, bly, bry, sly, sry, bcy
	   Set spread usage of stereo image across Y axis for each channel.
	   Allowed range is from 0.06 to 15.  By default this value is 0.5.

       win_size
	   Set window size. Allowed range is from 1024 to 65536. Default size
	   is 4096.

       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
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set window overlap. If set to 1, the recommended overlap for
	   selected window function will be picked. Default is 0.5.

   tiltshelf
       Boost or cut the lower frequencies and cut or boost higher 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 3000 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports some options as commands.

   treble, highshelf
       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

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter, by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precision of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If this value is
	   set to high enough value (higher than impulse response length
	   truncated when reaches near zero values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change treble frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change treble width_type.  Syntax for the command is : "width_type"

       width, w
	   Change treble width.  Syntax for the command is : "width"

       gain, g
	   Change treble gain.	Syntax for the command is : "gain"

       mix, m
	   Change treble mix.  Syntax for the command is : "mix"

   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.

   virtualbass
       Apply audio Virtual Bass filter.

       This filter accepts stereo input and produce stereo with LFE (2.1)
       channels output.  The newly produced LFE channel have enhanced virtual
       bass originally obtained from both stereo channels.  This filter
       outputs front left and front right channels unchanged as available in
       stereo input.

       The filter accepts the following options:

       cutoff
	   Set the virtual bass cutoff frequency. Default value is 250 Hz.
	   Allowed range is from 100 to 500 Hz.

       strength
	   Set the virtual bass strength. Allowed range is from 0.5 to 3.
	   Default value is 3.

   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.

       replaygain_noclip
	   Prevent clipping by limiting the gain applied.

	   Default value for replaygain_noclip is 1.

       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; deprecated, do not use

       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.

       Examples

       o   Halve the input audio volume:

		   volume=volume=0.5
		   volume=volume=1/2
		   volume=volume=-6.0206dB

	   In all the above example the named key for volume can be omitted,
	   for example like in:

		   volume=0.5

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

		   volume=volume=6dB:precision=fixed

       o   Fade volume after time 10 with an annihilation period of 5 seconds:

		   volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

   volumedetect
       Detect the volume of the input video.

       The filter has no parameters. It supports only 16-bit signed integer
       samples, so the input will be converted when needed. Statistics about
       the volume will be printed in the log when the input stream end is
       reached.

       In particular it will show the mean volume (root mean square), maximum
       volume (on a per-sample basis), and the beginning of a histogram of the
       registered volume values (from the maximum value to a cumulated 1/1000
       of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

	       [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
	       [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
	       [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
	       [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
	       [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
	       [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
	       [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
	       [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
	       [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means that:

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

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

       o   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

       In other words, raising the volume by +4 dB does not cause any
       clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
       Below is a description of the currently available audio sources.

   abuffer
       Buffer audio frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in libavfilter/buffersrc.h.

       It accepts the following parameters:

       time_base
	   The timebase which will be used for timestamps of submitted frames.
	   It must be either a floating-point number or in
	   numerator/denominator form.

       sample_rate
	   The sample rate of the incoming audio buffers.

       sample_fmt
	   The sample format of the incoming audio buffers.  Either a sample
	   format name or its corresponding integer representation from the
	   enum AVSampleFormat in libavutil/samplefmt.h

       channel_layout
	   The channel layout of the incoming audio buffers.  Either a channel
	   layout name from channel_layout_map in libavutil/channel_layout.c
	   or its corresponding integer representation from the AV_CH_LAYOUT_*
	   macros in libavutil/channel_layout.h

       channels
	   The number of channels of the incoming audio buffers.  If both
	   channels and channel_layout are specified, then they must be
	   consistent.

       Examples

	       abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

       will instruct the source to accept planar 16bit signed stereo at
       44100Hz.  Since the sample format with name "s16p" corresponds to the
       number 6 and the "stereo" channel layout corresponds to the value 0x3,
       this is equivalent to:

	       abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

   aevalsrc
       Generate an audio signal specified by an expression.

       This source accepts in input one or more expressions (one for each
       channel), which are evaluated and used to generate a corresponding
       audio signal.

       This source accepts the following options:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   In case the channel_layout option is not specified, the selected
	   channel layout depends on the number of provided expressions.
	   Otherwise the last specified expression is applied to the remaining
	   output channels.

       channel_layout, c
	   Set the channel layout. The number of channels in the specified
	   layout must be equal to the number of specified expressions.

       duration, d
	   Set the minimum duration of the sourced audio. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater than the
	   specified duration, as the generated audio is always cut at the end
	   of a complete frame.

	   If not specified, or the expressed duration is negative, the audio
	   is supposed to be generated forever.

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default to 1024.

       sample_rate, s
	   Specify the sample rate, default to 44100.

       Each expression in exprs can contain the following constants:

       n   number of the evaluated sample, starting from 0

       t   time of the evaluated sample expressed in seconds, starting from 0

       s   sample rate

       Examples

       o   Generate silence:

		   aevalsrc=0

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

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

       o   Generate a two channels signal, specify the channel layout (Front
	   Center + Back Center) explicitly:

		   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       o   Generate white noise:

		   aevalsrc="-2+random(0)"

       o   Generate an amplitude modulated signal:

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

       o   Generate 2.5 Hz binaural beats on a 360 Hz carrier:

		   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

   afdelaysrc
       Generate a fractional delay FIR coefficients.

       The resulting stream can be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       delay, d
	   Set the fractional delay. Default is 0.

       sample_rate, r
	   Set the sample rate, default is 44100.

       nb_samples, n
	   Set the number of samples per each frame. Default is 1024.

       taps, t
	   Set the number of filter coefficients in output audio stream.
	   Default value is 0.

       channel_layout, c
	   Specifies the channel layout, and can be a string representing a
	   channel layout.  The default value of channel_layout is "stereo".

   afireqsrc
       Generate a FIR equalizer coefficients.

       The resulting stream can be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       preset, p
	   Set equalizer preset.  Default preset is "flat".

	   Available presets are:

	   custom
	   flat
	   acoustic
	   bass
	   beats
	   classic
	   clear
	   deep bass
	   dubstep
	   electronic
	   hard-style
	   hip-hop
	   jazz
	   metal
	   movie
	   pop
	   r&b
	   rock
	   vocal booster
       gains, g
	   Set custom gains for each band. Only used if the preset option is
	   set to "custom".  Gains are separated by white spaces and each gain
	   is set in dBFS.  Default is "0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0".

       bands, b
	   Set the custom bands from where custon equalizer gains are set.
	   This must be in strictly increasing order. Only used if the preset
	   option is set to "custom".  Bands are separated by white spaces and
	   each band represent frequency in Hz.  Default is "25 40 63 100 160
	   250 400 630 1000 1600 2500 4000 6300 10000 16000 24000".

       taps, t
	   Set number of filter coefficients in output audio stream.  Default
	   value is 4096.

       sample_rate, r
	   Set sample rate of output audio stream, default is 44100.

       nb_samples, n
	   Set number of samples per each frame in output audio stream.
	   Default is 1024.

       interp, i
	   Set interpolation method for FIR equalizer coefficients. Can be
	   "linear" or "cubic".

       phase, h
	   Set phase type of FIR filter. Can be "linear" or "min": minimum-
	   phase.  Default is minimum-phase filter.

   afirsrc
       Generate a FIR coefficients using frequency sampling method.

       The resulting stream can be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       taps, t
	   Set number of filter coefficients in output audio stream.  Default
	   value is 1025.

       frequency, f
	   Set frequency points from where magnitude and phase are set.  This
	   must be in non decreasing order, and first element must be 0, while
	   last element must be 1. Elements are separated by white spaces.

       magnitude, m
	   Set magnitude value for every frequency point set by frequency.
	   Number of values must be same as number of frequency points.
	   Values are separated by white spaces.

       phase, p
	   Set phase value for every frequency point set by frequency.	Number
	   of values must be same as number of frequency points.  Values are
	   separated by white spaces.

       sample_rate, r
	   Set sample rate, default is 44100.

       nb_samples, n
	   Set number of samples per each frame. Default is 1024.

       win_func, w
	   Set window function. Default is blackman.

   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.

       duration, d
	   Set the duration of the sourced audio. 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 audio
	   is supposed to be generated forever.

       Examples

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

		   anullsrc=r=48000:cl=4

       o   Do the same operation with a more obvious syntax:

		   anullsrc=r=48000:cl=mono

       All the parameters need to be explicitly defined.

   flite
       Synthesize a voice utterance using the libflite library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libflite".

       Note that versions of the flite library prior to 2.0 are not thread-
       safe.

       The filter accepts the following options:

       list_voices
	   If set to 1, list the names of the available voices and exit
	   immediately. Default value is 0.

       nb_samples, n
	   Set the maximum number of samples per frame. Default value is 512.

       textfile
	   Set the filename containing the text to speak.

       text
	   Set the text to speak.

       voice, v
	   Set the voice to use for the speech synthesis. Default value is
	   "kal". See also the list_voices option.

       Examples

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

		   flite=textfile=speech.txt

       o   Read the specified text selecting the "slt" voice:

		   flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       o   Input text to ffmpeg:

		   ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       o   Make ffplay speak the specified text, using "flite" and the "lavfi"
	   device:

		   ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

       For more information about libflite, check:
       <http://www.festvox.org/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, violet and velvet. 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.

       density
	   Set the density (0.0 - 1.0) for the velvet noise generator, default
	   is 0.05.

       Examples

       o   Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate
	   and an amplitude of 0.5:

		   anoisesrc=d=60:c=pink:r=44100:a=0.5

   hilbert
       Generate odd-tap Hilbert transform FIR coefficients.

       The resulting stream can be used with afir filter for phase-shifting
       the signal by 90 degrees.

       This is used in many matrix coding schemes and for analytic signal
       generation.  The process is often written as a multiplication by i (or
       j), the imaginary unit.

       The filter accepts the following options:

       sample_rate, s
	   Set sample rate, default is 44100.

       taps, t
	   Set length of FIR filter, default is 22051.

       nb_samples, n
	   Set number of samples per each frame.

       win_func, w
	   Set window function to be used when generating FIR coefficients.

   sinc
       Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or
       band-reject FIR coefficients.

       The resulting stream can be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       sample_rate, r
	   Set sample rate, default is 44100.

       nb_samples, n
	   Set number of samples per each frame. Default is 1024.

       hp  Set high-pass frequency. Default is 0.

       lp  Set low-pass frequency. Default is 0.  If high-pass frequency is
	   lower than low-pass frequency and low-pass frequency is higher than
	   0 then filter will create band-pass filter coefficients, otherwise
	   band-reject filter coefficients.

       phase
	   Set filter phase response. Default is 50. Allowed range is from 0
	   to 100.

       beta
	   Set Kaiser window beta.

       att Set stop-band attenuation. Default is 120dB, allowed range is from
	   40 to 180 dB.

       round
	   Enable rounding, by default is disabled.

       hptaps
	   Set number of taps for high-pass filter.

       lptaps
	   Set number of taps for low-pass filter.

   sine
       Generate an audio signal made of a sine wave with amplitude 1/8.

       The audio signal is bit-exact.

       The filter accepts the following options:

       frequency, f
	   Set the carrier frequency. Default is 440 Hz.

       beep_factor, b
	   Enable a periodic beep every second with frequency beep_factor
	   times the carrier frequency. Default is 0, meaning the beep is
	   disabled.

       sample_rate, r
	   Specify the sample rate, default is 44100.

       duration, d
	   Specify the duration of the generated audio stream.

       samples_per_frame
	   Set the number of samples per output frame.

	   The expression can contain the following constants:

	   n   The (sequential) number of the output audio frame, starting
	       from 0.

	   pts The PTS (Presentation TimeStamp) of the output audio frame,
	       expressed in TB units.

	   t   The PTS of the output audio frame, expressed in seconds.

	   TB  The timebase of the output audio frames.

	   Default is 1024.

       Examples

       o   Generate a simple 440 Hz sine wave:

		   sine

       o   Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5
	   seconds:

		   sine=220:4:d=5
		   sine=f=220:b=4:d=5
		   sine=frequency=220:beep_factor=4:duration=5

       o   Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602"
	   NTSC pattern:

		   sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'

AUDIO SINKS
       Below is a description of the currently available audio sinks.

   abuffersink
       Buffer audio frames, and make them available to the end of filter
       chain.

       This sink is mainly intended for programmatic use, in particular
       through the interface defined in libavfilter/buffersink.h or the
       options system.

       It accepts a pointer to an AVABufferSinkContext structure, which
       defines the incoming buffers' formats, to be passed as the opaque
       parameter to "avfilter_init_filter" for initialization.

   anullsink
       Null audio sink; do absolutely nothing with the input audio. It is
       mainly useful as a template and for use in analysis / debugging tools.

VIDEO FILTERS
       When you configure your FFmpeg build, you can disable any of the
       existing filters using "--disable-filters".  The configure output will
       show the video filters included in your build.

       Below is a description of the currently available video filters.

   addroi
       Mark a region of interest in a video frame.

       The frame data is passed through unchanged, but metadata is attached to
       the frame indicating regions of interest which can affect the behaviour
       of later encoding.  Multiple regions can be marked by applying the
       filter multiple times.

       x   Region distance in pixels from the left edge of the frame.

       y   Region distance in pixels from the top edge of the frame.

       w   Region width in pixels.

       h   Region height in pixels.

	   The parameters x, y, w and h are expressions, and may contain the
	   following variables:

	   iw  Width of the input frame.

	   ih  Height of the input frame.

       qoffset
	   Quantisation offset to apply within the region.

	   This must be a real value in the range -1 to +1.  A value of zero
	   indicates no quality change.  A negative value asks for better
	   quality (less quantisation), while a positive value asks for worse
	   quality (greater quantisation).

	   The range is calibrated so that the extreme values indicate the
	   largest possible offset - if the rest of the frame is encoded with
	   the worst possible quality, an offset of -1 indicates that this
	   region should be encoded with the best possible quality anyway.
	   Intermediate values are then interpolated in some codec-dependent
	   way.

	   For example, in 10-bit H.264 the quantisation parameter varies
	   between -12 and 51.	A typical qoffset value of -1/10 therefore
	   indicates that this region should be encoded with a QP around one-
	   tenth of the full range better than the rest of the frame.  So, if
	   most of the frame were to be encoded with a QP of around 30, this
	   region would get a QP of around 24 (an offset of approximately
	   -1/10 * (51 - -12) = -6.3).	An extreme value of -1 would indicate
	   that this region should be encoded with the best possible quality
	   regardless of the treatment of the rest of the frame - that is,
	   should be encoded at a QP of -12.

       clear
	   If set to true, remove any existing regions of interest marked on
	   the frame before adding the new one.

       Examples

       o   Mark the centre quarter of the frame as interesting.

		   addroi=iw/4:ih/4:iw/2:ih/2:-1/10

       o   Mark the 100-pixel-wide region on the left edge of the frame as
	   very uninteresting (to be encoded at much lower quality than the
	   rest of the frame).

		   addroi=0:0:100:ih:+1/5

   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]

   amplify
       Amplify differences between current pixel and pixels of adjacent frames
       in same pixel location.

       This filter accepts the following options:

       radius
	   Set frame radius. Default is 2. Allowed range is from 1 to 63.  For
	   example radius of 3 will instruct filter to calculate average of 7
	   frames.

       factor
	   Set factor to amplify difference. Default is 2. Allowed range is
	   from 0 to 65535.

       threshold
	   Set threshold for difference amplification. Any difference greater
	   or equal to this value will not alter source pixel. Default is 10.
	   Allowed range is from 0 to 65535.

       tolerance
	   Set tolerance for difference amplification. Any difference lower to
	   this value will not alter source pixel. Default is 0.  Allowed
	   range is from 0 to 65535.

       low Set lower limit for changing source pixel. Default is 65535.
	   Allowed range is from 0 to 65535.  This option controls maximum
	   possible value that will decrease source pixel value.

       high
	   Set high limit for changing source pixel. Default is 65535. Allowed
	   range is from 0 to 65535.  This option controls maximum possible
	   value that will increase source pixel value.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Commands

       This filter supports the following commands that corresponds to option
       of same name:

       factor
       threshold
       tolerance
       low
       high
       planes

   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 9.
	   Must be odd number in range [5, 129].

       p   Set what planes of frame filter will use for averaging. Default is
	   all.

       a   Set what variant of algorithm filter will use for averaging.
	   Default is "p" parallel.  Alternatively can be set to "s" serial.

	   Parallel can be faster then serial, while other way around is never
	   true.  Parallel will abort early on first change being greater then
	   thresholds, while serial will continue processing other side of
	   frames if they are equal or below thresholds.

       0s
       1s
       2s  Set sigma for 1st plane, 2nd plane or 3rd plane. Default is 32767.
	   Valid range is from 0 to 32767.  This options controls weight for
	   each pixel in radius defined by size.  Default value means every
	   pixel have same weight.  Setting this option to 0 effectively
	   disables filtering.

       Commands

       This filter supports same commands as options except option "s".  The
       command accepts the same syntax of the corresponding option.

   avgblur
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.

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

       sizeY
	   Set vertical radius size, if zero it will be same as "sizeX".
	   Default is 0.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   backgroundkey
       Turns a static background into transparency.

       The filter accepts the following option:

       threshold
	   Threshold for scene change detection.

       similarity
	   Similarity percentage with the background.

       blend
	   Set the blend amount for pixels that are not similar.

       Commands

       This filter supports the all above options as commands.

   bbox
       Compute the bounding box for the non-black pixels in the input frame
       luma plane.

       This filter computes the bounding box containing all the pixels with a
       luma 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 luma value. Default is 16.

       Commands

       This filter supports the all above options as commands.

   bilateral
       Apply bilateral filter, spatial smoothing while preserving edges.

       The filter accepts the following options:

       sigmaS
	   Set sigma of gaussian function to calculate spatial weight.
	   Allowed range is 0 to 512. Default is 0.1.

       sigmaR
	   Set sigma of gaussian function to calculate range weight.  Allowed
	   range is 0 to 1. Default is 0.1.

       planes
	   Set planes to filter. Default is first only.

       Commands

       This filter supports the all above options as commands.

   bilateral_cuda
       CUDA accelerated bilateral filter, an edge preserving filter.  This
       filter is mathematically accurate thanks to the use of GPU
       acceleration.  For best output quality, use one to one chroma
       subsampling, i.e. yuv444p format.

       The filter accepts the following options:

       sigmaS
	   Set sigma of gaussian function to calculate spatial weight, also
	   called sigma space.	Allowed range is 0.1 to 512. Default is 0.1.

       sigmaR
	   Set sigma of gaussian function to calculate color range weight,
	   also called sigma color.  Allowed range is 0.1 to 512. Default is
	   0.1.

       window_size
	   Set window size of the bilateral function to determine the number
	   of neighbours to loop on.  If the number entered is even, one will
	   be added automatically.  Allowed range is 1 to 255. Default is 1.

       Examples

       o   Apply the bilateral filter on a video.

		   ./ffmpeg -v verbose \
		   -hwaccel cuda -hwaccel_output_format cuda -i input.mp4  \
		   -init_hw_device cuda \
		   -filter_complex \
		   " \
		   [0:v]scale_cuda=format=yuv444p[scaled_video];
		   [scaled_video]bilateral_cuda=window_size=9:sigmaS=3.0:sigmaR=50.0" \
		   -an -sn -c:v h264_nvenc -cq 20 out.mp4

   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.

       The filter outputs its detection analysis to both the log as well as
       frame metadata. If a black segment of at least the specified minimum
       duration is found, a line with the start and end timestamps as well as
       duration is printed to the log with level "info". In addition, a log
       line with level "debug" is printed per frame showing the black amount
       detected for that frame.

       The filter also attaches metadata to the first frame of a black segment
       with key "lavfi.black_start" and to the first frame after the black
       segment ends with key "lavfi.black_end". The value is the frame's
       timestamp. This metadata is added regardless of the minimum duration
       specified.

       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 luma value for which a
	   pixel is considered "black". The provided value is scaled according
	   to the following equation:

		   <absolute_threshold> = <luma_minimum_value> + <pixel_black_th> * <luma_range_size>

	   luma_range_size and luma_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
       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
	   and
	   average
	   bleach
	   burn
	   darken
	   difference
	   divide
	   dodge
	   exclusion
	   extremity
	   freeze
	   geometric
	   glow
	   grainextract
	   grainmerge
	   hardlight
	   hardmix
	   hardoverlay
	   harmonic
	   heat
	   interpolate
	   lighten
	   linearlight
	   multiply
	   multiply128
	   negation
	   normal
	   or
	   overlay
	   phoenix
	   pinlight
	   reflect
	   screen
	   softdifference
	   softlight
	   stain
	   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 for the plane being filtered. It is the
	       ratio between the dimensions of the current plane to the luma
	       plane, e.g. for a "yuv420p" frame, the values are "1,1" for the
	       luma plane and "0.5,0.5" for the chroma planes.

	   T   Time of the current frame, expressed in seconds.

	   TOP, A
	       Value of pixel component at current location for first video
	       frame (top layer).

	   BOTTOM, B
	       Value of pixel component at current location for second video
	       frame (bottom layer).

       The "blend" filter also supports the framesync options.

       Examples

       o   Apply transition from bottom layer to top layer in first 10
	   seconds:

		   blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

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

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

       o   Apply 1x1 checkerboard effect:

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

       o   Apply uncover left effect:

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

       o   Apply uncover down effect:

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

       o   Apply uncover up-left effect:

		   blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

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

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

       o   Display differences between the current and the previous frame:

		   tblend=all_mode=grainextract

       Commands

       This filter supports same commands as options.

   blockdetect
       Determines blockiness of frames without altering the input frames.

       Based on Remco Muijs and Ihor Kirenko: "A no-reference blocking
       artifact measure for adaptive video processing." 2005 13th European
       signal processing conference.

       The filter accepts the following options:

       period_min
       period_max
	   Set minimum and maximum values for determining pixel grids
	   (periods).  Default values are [3,24].

       planes
	   Set planes to filter. Default is first only.

       Examples

       o   Determine blockiness for the first plane and search for periods
	   within [8,32]:

		   blockdetect=period_min=8:period_max=32:planes=1

   blurdetect
       Determines blurriness of frames without altering the input frames.

       Based on Marziliano, Pina, et al. "A no-reference perceptual blur
       metric."  Allows for a block-based abbreviation.

       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".

       radius
	   Define the radius to search around an edge pixel for local maxima.

       block_pct
	   Determine blurriness only for the most significant blocks, given in
	   percentage.

       block_width
	   Determine blurriness for blocks of width block_width. If set to any
	   value smaller 1, no blocks are used and the whole image is
	   processed as one no matter of block_height.

       block_height
	   Determine blurriness for blocks of height block_height. If set to
	   any value smaller 1, no blocks are used and the whole image is
	   processed as one no matter of block_width.

       planes
	   Set planes to filter. Default is first only.

       Examples

       o   Determine blur for 80% of most significant 32x32 blocks:

		   blurdetect=block_width=32:block_height=32:block_pct=80

   bm3d
       Denoise frames using Block-Matching 3D algorithm.

       The filter accepts the following options.

       sigma
	   Set denoising strength. Default value is 1.	Allowed range is from
	   0 to 999.9.	The denoising algorithm is very sensitive to sigma, so
	   adjust it according to the source.

       block
	   Set local patch size. This sets dimensions in 2D.

       bstep
	   Set sliding step for processing blocks. Default value is 4.
	   Allowed range is from 1 to 64.  Smaller values allows processing
	   more reference blocks and is slower.

       group
	   Set maximal number of similar blocks for 3rd dimension. Default
	   value is 1.	When set to 1, no block matching is done. Larger
	   values allows more blocks in single group.  Allowed range is from 1
	   to 256.

       range
	   Set radius for search block matching. Default is 9.	Allowed range
	   is from 1 to INT32_MAX.

       mstep
	   Set step between two search locations for block matching. Default
	   is 1.  Allowed range is from 1 to 64. Smaller is slower.

       thmse
	   Set threshold of mean square error for block matching. Valid range
	   is 0 to INT32_MAX.

       hdthr
	   Set thresholding parameter for hard thresholding in 3D transformed
	   domain.  Larger values results in stronger hard-thresholding
	   filtering in frequency domain.

       estim
	   Set filtering estimation mode. Can be "basic" or "final".  Default
	   is "basic".

       ref If enabled, filter will use 2nd stream for block matching.  Default
	   is disabled for "basic" value of estim option, and always enabled
	   if value of estim is "final".

       planes
	   Set planes to filter. Default is all available except alpha.

       Examples

       o   Basic filtering with bm3d:

		   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic

       o   Same as above, but filtering only luma:

		   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1

       o   Same as above, but with both estimation modes:

		   split[a][b],[a]bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

       o   Same as above, but prefilter with nlmeans filter instead:

		   split[a][b],[a]nlmeans=s=3:r=7:p=3[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

   boxblur
       Apply a boxblur algorithm to the input video.

       It accepts the following parameters:

       luma_radius, lr
       luma_power, lp
       chroma_radius, cr
       chroma_power, cp
       alpha_radius, ar
       alpha_power, ap

       A description of the accepted options follows.

       luma_radius, lr
       chroma_radius, cr
       alpha_radius, ar
	   Set an expression for the box radius in pixels used for blurring
	   the corresponding input plane.

	   The radius value must be a non-negative number, and must not be
	   greater than the value of the expression "min(w,h)/2" for the luma
	   and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

	   Default value for luma_radius is "2". If not specified,
	   chroma_radius and alpha_radius default to the corresponding value
	   set for luma_radius.

	   The expressions can contain the following constants:

	   w
	   h   The input width and height in pixels.

	   cw
	   ch  The input chroma image width and height in pixels.

	   hsub
	   vsub
	       The horizontal and vertical chroma subsample values. For
	       example, for the pixel format "yuv422p", hsub is 2 and vsub is
	       1.

       luma_power, lp
       chroma_power, cp
       alpha_power, ap
	   Specify how many times the boxblur filter is applied to the
	   corresponding plane.

	   Default value for luma_power is 2. If not specified, chroma_power
	   and alpha_power default to the corresponding value set for
	   luma_power.

	   A value of 0 will disable the effect.

       Examples

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

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

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

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

       o   Set the luma and chroma radii to a fraction of the video dimension:

		   boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

   bwdif
       Deinterlace the input video ("bwdif" stands for "Bob Weaver
       Deinterlacing Filter").

       Motion adaptive deinterlacing based on yadif with the use of w3fdif and
       cubic interpolation algorithms.	It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It accepts one of the following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   The default value is "send_field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first will be
	   assumed.

       deint
	   Specify which frames to deinterlace. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   bwdif_cuda
       Deinterlace the input video using the bwdif algorithm, but implemented
       in CUDA so that it can work as part of a GPU accelerated pipeline with
       nvdec and/or nvenc.

       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. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   ccrepack
       Repack CEA-708 closed captioning side data

       This filter fixes various issues seen with commerical encoders related
       to upstream malformed CEA-708 payloads, specifically incorrect number
       of tuples (wrong cc_count for the target FPS), and incorrect ordering
       of tuples (i.e. the CEA-608 tuples are not at the first entries in the
       payload).

   cas
       Apply Contrast Adaptive Sharpen filter to video stream.

       The filter accepts the following options:

       strength
	   Set the sharpening strength. Default value is 0.

       planes
	   Set planes to filter. Default value is to filter all planes except
	   alpha plane.

       Commands

       This filter supports same commands as options.

   chromahold
       Remove all color information for all colors except for certain one.

       The filter accepts the following options:

       color
	   The color which will not be replaced with neutral chroma.

       similarity
	   Similarity percentage with the above color.	0.01 matches only the
	   exact key color, while 1.0 matches everything.

       blend
	   Blend percentage.  0.0 makes pixels either fully gray, or not gray
	   at all.  Higher values result in more preserved 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.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

       Examples

       o   Make every green pixel in the input image transparent:

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

       o   Overlay a greenscreen-video on top of a static black background.

		   ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv

   chromakey_cuda
       CUDA accelerated YUV colorspace color/chroma keying.

       This filter works like normal chromakey filter but operates on CUDA
       frames.	for more details and parameters see chromakey.

       Examples

       o   Make all the green pixels in the input video transparent and use it
	   as an overlay for another video:

		   ./ffmpeg \
		       -hwaccel cuda -hwaccel_output_format cuda -i input_green.mp4  \
		       -hwaccel cuda -hwaccel_output_format cuda -i base_video.mp4 \
		       -init_hw_device cuda \
		       -filter_complex \
		       " \
			   [0:v]chromakey_cuda=0x25302D:0.1:0.12:1[overlay_video]; \
			   [1:v]scale_cuda=format=yuv420p[base]; \
			   [base][overlay_video]overlay_cuda" \
		       -an -sn -c:v h264_nvenc -cq 20 output.mp4

       o   Process two software sources, explicitly uploading the frames:

		   ./ffmpeg -init_hw_device cuda=cuda -filter_hw_device cuda \
		       -f lavfi -i color=size=800x600:color=white,format=yuv420p \
		       -f lavfi -i yuvtestsrc=size=200x200,format=yuv420p \
		       -filter_complex \
		       " \
			   [0]hwupload[under]; \
			   [1]hwupload,chromakey_cuda=green:0.1:0.12[over]; \
			   [under][over]overlay_cuda" \
		       -c:v hevc_nvenc -cq 18 -preset slow output.mp4

   chromanr
       Reduce chrominance noise.

       The filter accepts the following options:

       thres
	   Set threshold for averaging chrominance values.  Sum of absolute
	   difference of Y, U and V pixel components of current pixel and
	   neighbour pixels lower than this threshold will be used in
	   averaging. Luma component is left unchanged and is copied to
	   output.  Default value is 30. Allowed range is from 1 to 200.

       sizew
	   Set horizontal radius of rectangle used for averaging.  Allowed
	   range is from 1 to 100. Default value is 5.

       sizeh
	   Set vertical radius of rectangle used for averaging.  Allowed range
	   is from 1 to 100. Default value is 5.

       stepw
	   Set horizontal step when averaging. Default value is 1.  Allowed
	   range is from 1 to 50.  Mostly useful to speed-up filtering.

       steph
	   Set vertical step when averaging. Default value is 1.  Allowed
	   range is from 1 to 50.  Mostly useful to speed-up filtering.

       threy
	   Set Y threshold for averaging chrominance values.  Set finer
	   control for max allowed difference between Y components of current
	   pixel and neigbour pixels.  Default value is 200. Allowed range is
	   from 1 to 200.

       threu
	   Set U threshold for averaging chrominance values.  Set finer
	   control for max allowed difference between U components of current
	   pixel and neigbour pixels.  Default value is 200. Allowed range is
	   from 1 to 200.

       threv
	   Set V threshold for averaging chrominance values.  Set finer
	   control for max allowed difference between V components of current
	   pixel and neigbour pixels.  Default value is 200. Allowed range is
	   from 1 to 200.

       distance
	   Set distance type used in calculations.

	   manhattan
	       Absolute difference.

	   euclidean
	       Difference squared.

	   Default distance type is manhattan.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

   chromashift
       Shift chroma pixels horizontally and/or vertically.

       The filter accepts the following options:

       cbh Set amount to shift chroma-blue horizontally.

       cbv Set amount to shift chroma-blue vertically.

       crh Set amount to shift chroma-red horizontally.

       crv Set amount to shift chroma-red vertically.

       edge
	   Set edge mode, can be smear, default, or warp.

       Commands

       This filter supports the all above options as commands.

   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
	   dcip3
       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.

       fill
	   Fill with CIE colors. By default is enabled.

   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:

       block
	   Display block partition structure using the luma plane.

       mv  Set motion vectors to visualize.

	   Available flags for mv are:

	   pf  forward predicted MVs of P-frames

	   bf  forward predicted MVs of B-frames

	   bb  backward predicted MVs of B-frames

       qp  Display quantization parameters using the chroma planes.

       mv_type, mvt
	   Set motion vectors type to visualize. Includes MVs from all frames
	   unless specified by frame_type option.

	   Available flags for mv_type are:

	   fp  forward predicted MVs

	   bp  backward predicted MVs

       frame_type, ft
	   Set frame type to visualize motion vectors of.

	   Available flags for frame_type are:

	   if  intra-coded frames (I-frames)

	   pf  predicted frames (P-frames)

	   bf  bi-directionally predicted frames (B-frames)

       Examples

       o   Visualize forward predicted MVs of all frames using ffplay:

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

       o   Visualize multi-directionals MVs of P and B-Frames using ffplay:

		   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb

   colorbalance
       Modify intensity of primary colors (red, green and blue) of input
       frames.

       The filter allows an input frame to be adjusted in the shadows,
       midtones or highlights regions for the red-cyan, green-magenta or blue-
       yellow balance.

       A positive adjustment value shifts the balance towards the primary
       color, a negative value towards the complementary color.

       The filter accepts the following options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

	   Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

       pl  Preserve lightness when changing color balance. Default is
	   disabled.

       Examples

       o   Add red color cast to shadows:

		   colorbalance=rs=.3

       Commands

       This filter supports the all above options as commands.

   colorcontrast
       Adjust color contrast between RGB components.

       The filter accepts the following options:

       rc  Set the red-cyan contrast. Defaults is 0.0. Allowed range is from
	   -1.0 to 1.0.

       gm  Set the green-magenta contrast. Defaults is 0.0. Allowed range is
	   from -1.0 to 1.0.

       by  Set the blue-yellow contrast. Defaults is 0.0. Allowed range is
	   from -1.0 to 1.0.

       rcw
       gmw
       byw Set the weight of each "rc", "gm", "by" option value. Default value
	   is 0.0.  Allowed range is from 0.0 to 1.0. If all weights are 0.0
	   filtering is disabled.

       pl  Set the amount of preserving lightness. Default value is 0.0.
	   Allowed range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

   colorcorrect
       Adjust color white balance selectively for blacks and whites.  This
       filter operates in YUV colorspace.

       The filter accepts the following options:

       rl  Set the red shadow spot. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       bl  Set the blue shadow spot. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       rh  Set the red highlight spot. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       bh  Set the blue highlight spot. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       saturation
	   Set the amount of saturation. Allowed range is from -3.0 to 3.0.
	   Default value is 1.

       analyze
	   If set to anything other than "manual" it will analyze every frame
	   and use derived parameters for filtering output frame.

	   Possible values are:

	   manual
	   average
	   minmax
	   median

	   Default value is "manual".

       Commands

       This filter supports the all above options as commands.

   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]".

       pc  Set preserve color mode. The accepted values are:

	   none
	       Disable color preserving, this is default.

	   lum Preserve luminance.

	   max Preserve max value of RGB triplet.

	   avg Preserve average value of RGB triplet.

	   sum Preserve sum value of RGB triplet.

	   nrm Preserve normalized value of RGB triplet.

	   pwr Preserve power value of RGB triplet.

       pa  Set the preserve color amount when changing colors. Allowed range
	   is from "[0.0, 1.0]".  Default is 0.0, thus disabled.

       Examples

       o   Convert source to grayscale:

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

       o   Simulate sepia tones:

		   colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

       Commands

       This filter supports the all above options as commands.

   colorize
       Overlay a solid color on the video stream.

       The filter accepts the following options:

       hue Set the color hue. Allowed range is from 0 to 360.  Default value
	   is 0.

       saturation
	   Set the color saturation. Allowed range is from 0 to 1.  Default
	   value is 0.5.

       lightness
	   Set the color lightness. Allowed range is from 0 to 1.  Default
	   value is 0.5.

       mix Set the mix of source lightness. By default is set to 1.0.  Allowed
	   range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

   colorkey
       RGB colorspace color keying.  This filter operates on 8-bit RGB format
       frames by setting the alpha component of each pixel which falls within
       the similarity radius of the key color to 0. The alpha value for pixels
       outside the similarity radius depends on the value of the blend option.

       The filter accepts the following options:

       color
	   Set the color for which alpha will be set to 0 (full transparency).
	   See "Color" section in the ffmpeg-utils manual.  Default is
	   "black".

       similarity
	   Set the radius from the key color within which other colors also
	   have full transparency.  The computed distance is related to the
	   unit fractional distance in 3D space between the RGB values of the
	   key color and the pixel's color. Range is 0.01 to 1.0. 0.01 matches
	   within a very small radius around the exact key color, while 1.0
	   matches everything.	Default is 0.01.

       blend
	   Set how the alpha value for pixels that fall outside the similarity
	   radius is computed.	0.0 makes pixels either fully transparent or
	   fully opaque.  Higher values result in semi-transparent pixels,
	   with greater transparency the more similar the pixel color is to
	   the key color.  Range is 0.0 to 1.0. Default is 0.0.

       Examples

       o   Make every green pixel in the input image transparent:

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

       o   Overlay a greenscreen-video on top of a static background image.

		   ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   colorhold
       Remove all color information for all RGB colors except for certain one.

       The filter accepts the following options:

       color
	   The color which will not be replaced with neutral gray.

       similarity
	   Similarity percentage with the above color.	0.01 matches only the
	   exact key color, while 1.0 matches everything.

       blend
	   Blend percentage. 0.0 makes pixels fully gray.  Higher values
	   result in more preserved color.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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.

       preserve
	   Set preserve color mode. The accepted values are:

	   none
	       Disable color preserving, this is default.

	   lum Preserve luminance.

	   max Preserve max value of RGB triplet.

	   avg Preserve average value of RGB triplet.

	   sum Preserve sum value of RGB triplet.

	   nrm Preserve normalized value of RGB triplet.

	   pwr Preserve power value of RGB triplet.

       Examples

       o   Make video output darker:

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

       o   Increase contrast:

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

       o   Make video output lighter:

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

       o   Increase brightness:

		   colorlevels=romin=0.5:gomin=0.5:bomin=0.5

       Commands

       This filter supports the all above options as commands.

   colormap
       Apply custom color maps to video stream.

       This filter needs three input video streams.  First stream is video
       stream that is going to be filtered out.  Second and third video stream
       specify color patches for source color to target color mapping.

       The filter accepts the following options:

       patch_size
	   Set the source and target video stream patch size in pixels.

       nb_patches
	   Set the max number of used patches from source and target video
	   stream.  Default value is number of patches available in additional
	   video streams.  Max allowed number of patches is 64.

       type
	   Set the adjustments used for target colors. Can be "relative" or
	   "absolute".	Defaults is "absolute".

       kernel
	   Set the kernel used to measure color differences between mapped
	   colors.

	   The accepted values are:

	   euclidean
	   weuclidean

	   Default is "euclidean".

   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

   colorspace_cuda
       CUDA accelerated implementation of the colorspace filter.

       It is by no means feature complete compared to the software colorspace
       filter, and at the current time only supports color range conversion
       between jpeg/full and mpeg/limited range.

       The filter accepts the following options:

       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

   colortemperature
       Adjust color temperature in video to simulate variations in ambient
       color temperature.

       The filter accepts the following options:

       temperature
	   Set the temperature in Kelvin. Allowed range is from 1000 to 40000.
	   Default value is 6500 K.

       mix Set mixing with filtered output. Allowed range is from 0 to 1.
	   Default value is 1.

       pl  Set the amount of preserving lightness. Allowed range is from 0 to
	   1.  Default value is 0.

       Commands

       This filter supports same commands as options.

   convolution
       Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up to 49
       elements.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set matrix for each plane.  Matrix is sequence of 9, 25 or 49
	   signed integers in square mode, and from 1 to 49 odd number of
	   signed integers in row mode.

       0rdiv
       1rdiv
       2rdiv
       3rdiv
	   Set multiplier for calculated value for each plane.	If unset or 0,
	   it will be 1/sum of all matrix elements.

       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.

       0mode
       1mode
       2mode
       3mode
	   Set matrix mode for each plane. Can be square, row or column.
	   Default is square.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Apply sharpen:

		   convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"

       o   Apply blur:

		   convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"

       o   Apply edge enhance:

		   convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"

       o   Apply edge detect:

		   convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"

       o   Apply laplacian edge detector which includes diagonals:

		   convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"

       o   Apply emboss:

		   convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"

   convolve
       Apply 2D convolution of video stream in frequency domain using second
       stream as impulse.

       The filter accepts the following options:

       planes
	   Set which planes to process.

       impulse
	   Set which impulse video frames will be processed, can be first or
	   all. Default is all.

       The "convolve" filter also supports the framesync options.

   copy
       Copy the input video source unchanged to the output. This is mainly
       useful for testing purposes.

   coreimage
       Video filtering on GPU using Apple's CoreImage API on OSX.

       Hardware acceleration is based on an OpenGL context. Usually, this
       means it is processed by video hardware. However, software-based OpenGL
       implementations exist which means there is no guarantee for hardware
       processing. It depends on the respective OSX.

       There are many filters and image generators provided by Apple that come
       with a large variety of options. The filter has to be referenced by its
       name along with its options.

       The coreimage filter accepts the following options:

       list_filters
	   List all available filters and generators along with all their
	   respective options as well as possible minimum and maximum values
	   along with the default values.

		   list_filters=true

       filter
	   Specify all filters by their respective name and options.  Use
	   list_filters to determine all valid filter names and options.
	   Numerical options are specified by a float value and are
	   automatically clamped to their respective value range.  Vector and
	   color options have to be specified by a list of space separated
	   float values. Character escaping has to be done.  A special option
	   name "default" is available to use default options for a filter.

	   It is required to specify either "default" or at least one of the
	   filter options.  All omitted options are used with their default
	   values.  The syntax of the filter string is as follows:

		   filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]

       output_rect
	   Specify a rectangle where the output of the filter chain is copied
	   into the input image. It is given by a list of space separated
	   float values:

		   output_rect=x\ y\ width\ height

	   If not given, the output rectangle equals the dimensions of the
	   input image.  The output rectangle is automatically cropped at the
	   borders of the input image. Negative values are valid for each
	   component.

		   output_rect=25\ 25\ 100\ 100

       Several filters can be chained for successive processing without GPU-
       HOST transfers allowing for fast processing of complex filter chains.
       Currently, only filters with zero (generators) or exactly one (filters)
       input image and one output image are supported. Also, transition
       filters are not yet usable as intended.

       Some filters generate output images with additional padding depending
       on the respective filter kernel. The padding is automatically removed
       to ensure the filter output has the same size as the input image.

       For image generators, the size of the output image is determined by the
       previous output image of the filter chain or the input image of the
       whole filterchain, respectively. The generators do not use the pixel
       information of this image to generate their output. However, the
       generated output is blended onto this image, resulting in partial or
       complete coverage of the output image.

       The coreimagesrc video source can be used for generating input images
       which are directly fed into the filter chain. By using it, providing
       input images by another video source or an input video is not required.

       Examples

       o   List all filters available:

		   coreimage=list_filters=true

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

		   coreimage=filter=CIBoxBlur@default

       o   Use a filter chain with CISepiaTone at default values and
	   CIVignetteEffect with its center at 100x100 and a radius of 50
	   pixels:

		   coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50

       o   Use nullsrc and CIQRCodeGenerator to create a QR code for the
	   FFmpeg homepage, given as complete and escaped command-line for
	   Apple's standard bash shell:

		   ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

   corr
       Obtain the correlation between two input videos.

       This filter takes two input videos.

       Both input videos 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 per component, average, min and max correlation is printed
       through the logging system.

       The filter stores the calculated correlation of each frame in frame
       metadata.

       This filter also supports the framesync options.

       In the below example the input file main.mpg being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi corr -f null -

   cover_rect
       Cover a rectangular object

       It accepts the following options:

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

       mode
	   Set covering mode.

	   It accepts the following values:

	   cover
	       cover it by the supplied image

	   blur
	       cover it by interpolating the surrounding pixels

	   Default value is blur.

       Examples

       o   Cover a rectangular object by the supplied image of a given video
	   using ffmpeg:

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

   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;
	   deprecated, do not use

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
	   is unknown.

       The expression for out_w may depend on the value of out_h, and the
       expression for out_h may depend on out_w, but they cannot depend on x
       and y, as x and y are evaluated after out_w and out_h.

       The x and y parameters specify the expressions for the position of the
       top-left corner of the output (non-cropped) area. They are evaluated
       for each frame. If the evaluated value is not valid, it is approximated
       to the nearest valid value.

       The expression for x may depend on y, and the expression for y may
       depend on x.

       Examples

       o   Crop area with size 100x100 at position (12,34).

		   crop=100:100:12:34

	   Using named options, the example above becomes:

		   crop=w=100:h=100:x=12:y=34

       o   Crop the central input area with size 100x100:

		   crop=100:100

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

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

       o   Crop the input video central square:

		   crop=out_w=in_h
		   crop=in_h

       o   Delimit the rectangle with the top-left corner placed at position
	   100:100 and the right-bottom corner corresponding to the right-
	   bottom corner of the input image.

		   crop=in_w-100:in_h-100:100:100

       o   Crop 10 pixels from the left and right borders, and 20 pixels from
	   the top and bottom borders

		   crop=in_w-2*10:in_h-2*20

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

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

       o   Crop height for getting Greek harmony:

		   crop=in_w:1/PHI*in_w

       o   Apply trembling effect:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

       o   Apply erratic camera effect depending on timestamp:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)

       o   Set x depending on the value of y:

		   crop=in_w/2:in_h/2:y:10+10*sin(n/10)

       Commands

       This filter supports the following commands:

       w, out_w
       h, out_h
       x
       y   Set width/height of the output video and the horizontal/vertical
	   position in the input video.  The command accepts the same syntax
	   of the corresponding option.

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

   cropdetect
       Auto-detect the crop size.

       It calculates the necessary cropping parameters and prints the
       recommended parameters via the logging system. The detected dimensions
       correspond to the non-black or video area of the input video according
       to mode.

       It accepts the following parameters:

       mode
	   Depending on mode crop detection is based on either the mere black
	   value of surrounding pixels or a combination of motion vectors and
	   edge pixels.

	   black
	       Detect black pixels surrounding the playing video. For fine
	       control use option limit.

	   mvedges
	       Detect the playing video by the motion vectors inside the video
	       and scanning for edge pixels typically forming the border of a
	       playing video.

       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.

       skip
	   Set the number of initial frames for which evaluation is skipped.
	   Default is 2. Range is 0 to INT_MAX.

       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.

       mv_threshold
	   Set motion in pixel units as threshold for motion detection. It
	   defaults to 8.

       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 "5/255", and default value for high is
	   "15/255".

       Examples

       o   Find video area surrounded by black borders:

		   ffmpeg -i file.mp4 -vf cropdetect,metadata=mode=print -f null -

       o   Find an embedded video area, generate motion vectors beforehand:

		   ffmpeg -i file.mp4 -vf mestimate,cropdetect=mode=mvedges,metadata=mode=print -f null -

       o   Find an embedded video area, use motion vectors from decoder:

		   ffmpeg -flags2 +export_mvs -i file.mp4 -vf cropdetect=mode=mvedges,metadata=mode=print -f null -

       Commands

       This filter supports the following commands:

       limit
	   The command accepts the same syntax of the corresponding option.
	   If the specified expression is not valid, it is kept at its current
	   value.

   cue
       Delay video filtering until a given wallclock timestamp. The filter
       first passes on preroll amount of frames, then it buffers at most
       buffer amount of frames and waits for the cue. After reaching the cue
       it forwards the buffered frames and also any subsequent frames coming
       in its input.

       The filter can be used synchronize the output of multiple ffmpeg
       processes for realtime output devices like decklink. By putting the
       delay in the filtering chain and pre-buffering frames the process can
       pass on data to output almost immediately after the target wallclock
       timestamp is reached.

       Perfect frame accuracy cannot be guaranteed, but the result is good
       enough for some use cases.

       cue The cue timestamp expressed in a UNIX timestamp in microseconds.
	   Default is 0.

       preroll
	   The duration of content to pass on as preroll expressed in seconds.
	   Default is 0.

       buffer
	   The maximum duration of content to buffer before waiting for the
	   cue expressed in seconds. Default is 0.

   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 will be defined to pass smoothly through all these new
       coordinates. The new defined points need to be strictly increasing over
       the x-axis, and their x and y values must be in the [0;1] interval. The
       curve is formed by using a natural or monotonic cubic spline
       interpolation, depending on the interp option (default: "natural"). The
       "natural" spline produces a smoother curve in general while the
       monotonic ("pchip") spline guarantees the transitions between the
       specified points to be monotonic. 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.

       interp
	   Specify the kind of interpolation. Available algorithms are:

	   natural
	       Natural cubic spline using a piece-wise cubic polynomial that
	       is twice continuously differentiable.

	   pchip
	       Monotonic cubic spline using a piecewise cubic Hermite
	       interpolating polynomial (PCHIP).

       To avoid some filtergraph syntax conflicts, each key points list need
       to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

       Commands

       This filter supports same commands as options.

       Examples

       o   Increase slightly the middle level of blue:

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

       o   Vintage effect:

		   curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'

	   Here we obtain the following coordinates for each components:

	   red "(0;0.11) (0.42;0.51) (1;0.95)"

	   green
	       "(0;0) (0.50;0.48) (1;1)"

	   blue
	       "(0;0.22) (0.49;0.44) (1;0.80)"

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

		   curves=preset=vintage

       o   Or simply:

		   curves=vintage

       o   Use a Photoshop preset and redefine the points of the green
	   component:

		   curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

       o   Check out the curves of the "cross_process" profile using ffmpeg
	   and gnuplot:

		   ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
		   gnuplot -p /tmp/curves.plt

   datascope
       Video data analysis filter.

       This filter shows hexadecimal pixel values of part of video.

       The filter accepts the following options:

       size, s
	   Set output video size.

       x   Set x offset from where to pick pixels.

       y   Set y offset from where to pick pixels.

       mode
	   Set scope mode, can be one of the following:

	   mono
	       Draw hexadecimal pixel values with white color on black
	       background.

	   color
	       Draw hexadecimal pixel values with input video pixel color on
	       black background.

	   color2
	       Draw hexadecimal pixel values on color background picked from
	       input video, the text color is picked in such way so its always
	       visible.

       axis
	   Draw rows and columns numbers on left and top of video.

       opacity
	   Set background opacity.

       format
	   Set display number format. Can be "hex", or "dec". Default is
	   "hex".

       components
	   Set pixel components to display. By default all pixel components
	   are displayed.

       Commands

       This filter supports same commands as options excluding "size" option.

   dblur
       Apply Directional blur filter.

       The filter accepts the following options:

       angle
	   Set angle of directional blur. Default is 45.

       radius
	   Set radius of directional blur. Default is 5.

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

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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.

       Commands

       This filter supports the all above options as commands.

   deblock
       Remove blocking artifacts from input video.

       The filter accepts the following options:

       filter
	   Set filter type, can be weak or strong. Default is strong.  This
	   controls what kind of deblocking is applied.

       block
	   Set size of block, allowed range is from 4 to 512. Default is 8.

       alpha
       beta
       gamma
       delta
	   Set blocking detection thresholds. Allowed range is 0 to 1.
	   Defaults are: 0.098 for alpha and 0.05 for the rest.  Using higher
	   threshold gives more deblocking strength.  Setting alpha controls
	   threshold detection at exact edge of block.	Remaining options
	   controls threshold detection near the edge. Each one for
	   below/above or left/right. Setting any of those to 0 disables
	   deblocking.

       planes
	   Set planes to filter. Default is to filter all available planes.

       Examples

       o   Deblock using weak filter and block size of 4 pixels.

		   deblock=filter=weak:block=4

       o   Deblock using strong filter, block size of 4 pixels and custom
	   thresholds for deblocking more edges.

		   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05

       o   Similar as above, but filter only first plane.

		   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=1

       o   Similar as above, but filter only second and third plane.

		   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6

       Commands

       This filter supports the all above options as commands.

   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.

       mixed
	   Set whether or not the input only partially contains content to be
	   decimated.  Default is "false".  If enabled video output stream
	   will be in variable frame rate.

   deconvolve
       Apply 2D deconvolution 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.

       noise
	   Set noise when doing divisions. Default is 0.0000001. Useful when
	   width and height are not same and not power of 2 or if stream prior
	   to convolving had noise.

       The "deconvolve" filter also supports the framesync options.

   dedot
       Reduce cross-luminance (dot-crawl) and cross-color (rainbows) from
       video.

       It accepts the following options:

       m   Set mode of operation. Can be combination of dotcrawl for cross-
	   luminance reduction and/or rainbows for cross-color reduction.

       lt  Set spatial luma threshold. Lower values increases reduction of
	   cross-luminance.

       tl  Set tolerance for temporal luma. Higher values increases reduction
	   of cross-luminance.

       tc  Set tolerance for chroma temporal variation. Higher values
	   increases reduction of cross-color.

       ct  Set temporal chroma threshold. Lower values increases reduction of
	   cross-color.

   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.

       Commands

       This filter supports the all above options as commands.

   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.

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

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

       Examples

       o   Set a rectangle covering the area with top left corner coordinates
	   0,0 and size 100x77:

		   delogo=x=0:y=0:w=100:h=77

   derain
       Remove the rain in the input image/video by applying the derain methods
       based on convolutional neural networks. Supported models:

       o   Recurrent Squeeze-and-Excitation Context Aggregation Net (RESCAN).
	   See
	   <http://openaccess.thecvf.com/content_ECCV_2018/papers/Xia_Li_Recurrent_Squeeze-and-Excitation_Context_ECCV_2018_paper.pdf>.

       Training as well as model generation scripts are provided in the
       repository at <https://github.com/XueweiMeng/derain_filter.git>.

       The filter accepts the following options:

       filter_type
	   Specify which filter to use. This option accepts the following
	   values:

	   derain
	       Derain filter. To conduct derain filter, you need to use a
	       derain model.

	   dehaze
	       Dehaze filter. To conduct dehaze filter, you need to use a
	       dehaze model.

	   Default value is derain.

       dnn_backend
	   Specify which DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to install
	       the TensorFlow for C library (see
	       <https://www.tensorflow.org/install/lang_c>) and configure
	       FFmpeg with "--enable-libtensorflow"

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats. TensorFlow can load files for only its format.

       To get full functionality (such as async execution), please use the
       dnn_processing filter.

   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.

   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.

       Commands

       This filter supports the all above options as commands.

   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

       Commands

       This filter supports the all above options as commands.

   displace
       Displace pixels as indicated by second and third input stream.

       It takes three input streams and outputs one stream, the first input is
       the source, and second and third input are displacement maps.

       The second input specifies how much to displace pixels along the
       x-axis, while the third input specifies how much to displace pixels
       along the y-axis.  If one of displacement map streams terminates, last
       frame from that displacement map will be used.

       Note that once generated, displacements maps can be reused over and
       over again.

       A description of the accepted options follows.

       edge
	   Set displace behavior for pixels that are out of range.

	   Available values are:

	   blank
	       Missing pixels are replaced by black pixels.

	   smear
	       Adjacent pixels will spread out to replace missing pixels.

	   wrap
	       Out of range pixels are wrapped so they point to pixels of
	       other side.

	   mirror
	       Out of range pixels will be replaced with mirrored pixels.

	   Default is smear.

       Examples

       o   Add ripple effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT

       o   Add wave effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT

   dnn_classify
       Do classification with deep neural networks based on bounding boxes.

       The filter accepts the following options:

       dnn_backend
	   Specify which DNN backend to use for model loading and execution.
	   This option accepts only openvino now, tensorflow backends will be
	   added.

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       confidence
	   Set the confidence threshold (default: 0.5).

       labels
	   Set path to label file specifying the mapping between label id and
	   name.  Each label name is written in one line, tailing spaces and
	   empty lines are skipped.  The first line is the name of label id 0,
	   and the second line is the name of label id 1, etc.	The label id
	   is considered as name if the label file is not provided.

       backend_configs
	   Set the configs to be passed into backend

	   For tensorflow backend, you can set its configs with sess_config
	   options, please use tools/python/tf_sess_config.py to get the
	   configs for your system.

   dnn_detect
       Do object detection with deep neural networks.

       The filter accepts the following options:

       dnn_backend
	   Specify which DNN backend to use for model loading and execution.
	   This option accepts only openvino now, tensorflow backends will be
	   added.

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       confidence
	   Set the confidence threshold (default: 0.5).

       labels
	   Set path to label file specifying the mapping between label id and
	   name.  Each label name is written in one line, tailing spaces and
	   empty lines are skipped.  The first line is the name of label id 0
	   (usually it is 'background'), and the second line is the name of
	   label id 1, etc.  The label id is considered as name if the label
	   file is not provided.

       backend_configs
	   Set the configs to be passed into backend. To use async execution,
	   set async (default: set).  Roll back to sync execution if the
	   backend does not support async.

   dnn_processing
       Do image processing with deep neural networks. It works together with
       another filter which converts the pixel format of the Frame to what the
       dnn network requires.

       The filter accepts the following options:

       dnn_backend
	   Specify which DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to install
	       the TensorFlow for C library (see
	       <https://www.tensorflow.org/install/lang_c>) and configure
	       FFmpeg with "--enable-libtensorflow"

	   openvino
	       OpenVINO backend. To enable this backend you need to build and
	       install the OpenVINO for C library (see
	       <https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md>)
	       and configure FFmpeg with "--enable-libopenvino"
	       (--extra-cflags=-I... --extra-ldflags=-L... might be needed if
	       the header files and libraries are not installed into system
	       path)

	   torch
	       Libtorch backend. To enable this backend you need to build and
	       install Libtroch for C++ library. Please download cxx11 ABI
	       version (see <https://pytorch.org/get-started/locally>) and
	       configure FFmpeg with "--enable-libtorch
	       --extra-cflags=-I/libtorch_root/libtorch/include
	       --extra-cflags=-I/libtorch_root/libtorch/include/torch/csrc/api/include
	       --extra-ldflags=-L/libtorch_root/libtorch/lib/"

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats. TensorFlow, OpenVINO and Libtorch backend can load files
	   for only its format.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       backend_configs
	   Set the configs to be passed into backend. To use async execution,
	   set async (default: set).  Roll back to sync execution if the
	   backend does not support async.

	   For tensorflow backend, you can set its configs with sess_config
	   options, please use tools/python/tf_sess_config.py to get the
	   configs of TensorFlow backend for your system.

       Examples

       o   Remove rain in rgb24 frame with can.pb (see derain filter):

		   ./ffmpeg -i rain.jpg -vf format=rgb24,dnn_processing=dnn_backend=tensorflow:model=can.pb:input=x:output=y derain.jpg

       o   Handle the Y channel with srcnn.pb (see sr filter) for frame with
	   yuv420p (planar YUV formats supported):

		   ./ffmpeg -i 480p.jpg -vf format=yuv420p,scale=w=iw*2:h=ih*2,dnn_processing=dnn_backend=tensorflow:model=srcnn.pb:input=x:output=y -y srcnn.jpg

       o   Handle the Y channel with espcn.pb (see sr filter), which changes
	   frame size, for format yuv420p (planar YUV formats supported),
	   please use tools/python/tf_sess_config.py to get the configs of
	   TensorFlow backend for your system.

		   ./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y:backend_configs=sess_config=0x10022805320e09cdccccccccccec3f20012a01303801 -y tmp.espcn.jpg

   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.  A value
	   of "fill" will create a filled box. Default value is 3.

	   See below for the list of accepted constants.

       replace
	   Applicable if the input has alpha. With value 1, the pixels of the
	   painted box will overwrite the video's color and alpha pixels.
	   Default is 0, which composites the box onto the input, leaving the
	   video's alpha intact.

       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.

       box_source
	   Box source can be set as side_data_detection_bboxes if you want to
	   use box data in detection bboxes of side data.

	   If box_source is set, the x, y, width and height will be ignored
	   and still use box data in detection bboxes of side data. So please
	   do not use this parameter if you were not sure about the box
	   source.

       t   The thickness of the drawn box.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

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

		   drawbox

       o   Draw a box with color red and an opacity of 50%:

		   drawbox=10:20:200:60:red@0.5

	   The previous example can be specified as:

		   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       o   Fill the box with pink color:

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

       o   Draw a 2-pixel red 2.40:1 mask:

		   drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   drawgraph
       Draw a graph using input video 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".

       rate, r
	   Set the output frame rate. Default value is 25.

	   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

   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.

       replace
	   Applicable if the input has alpha. With 1 the pixels of the painted
	   grid will overwrite the video's color and alpha pixels.  Default is
	   0, which composites the grid onto the input, leaving the video's
	   alpha intact.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input grid cell width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y coordinates of some point of grid intersection (meant
	   to configure offset).

       w
       h   The width and height of the drawn cell.

       t   The thickness of the drawn cell.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

       o   Draw a grid with cell 100x100 pixels, thickness 2 pixels, with
	   color red and an opacity of 50%:

		   drawgrid=width=100:height=100:thickness=2:color=red@0.5

       o   Draw a white 3x3 grid with an opacity of 50%:

		   drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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" and "--enable-libharfbuzz".  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 value must be specified using one of the following
	   formats:

	   *<"boxborderw=10" set the width of all the borders to 10>
	   *<"boxborderw=10|20" set the width of the top and bottom borders to
	   10>
		   and the width of the left and right borders to 20

	   *<"boxborderw=10|20|30" set the width of the top border to 10, the
	   width>
		   of the bottom border to 30 and the width of the left and right borders to 20

	   *<"boxborderw=10|20|30|40" set the borders width to 10 (top), 20
	   (right),>
		   30 (bottom), 40 (left)

	   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. The default value of line_spacing
	   is 0.

       text_align
	   Set the vertical and horizontal alignment of the text with respect
	   to the box boundaries.  The value is combination of flags, one for
	   the vertical alignment (T=top, M=middle, B=bottom) and one for the
	   horizontal alignment (L=left, C=center, R=right).  Please note that
	   tab characters are only supported with the left horizontal
	   alignment.

       y_align
	   Specify what the y value is referred to. Possible values are:

	   *<"text" the top of the highest glyph of the first text line is
	   placed at y>
	   *<"baseline" the baseline of the first text line is placed at y>
	   *<"font" the baseline of the first text line is placed at y plus
	   the>
		   ascent (in pixels) defined in the font metrics

	   The default value of y_align is "text" for backward compatibility.

       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".

       boxw
	   Set the width of the box to be drawn around text.  The default
	   value of boxw is computed automatically to match the text width

       boxh
	   Set the height of the box to be drawn around text.  The default
	   value of boxh is computed automatically to match the text height

       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). Value will be rounded
	   to nearest integer. Minimum value is "1".  Drop-frame timecode is
	   supported for frame rates 30 & 60.

       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.

       text_source
	   Text source should be set as side_data_detection_bboxes if you want
	   to use text data in detection bboxes of side data.

	   If text source is set, text and textfile will be ignored and still
	   use text data in detection bboxes of side data. So please do not
	   use this parameter if you are not sure about the text source.

       reload
	   The textfile will be reloaded at specified frame interval.  Be sure
	   to update textfile atomically, or it may be read partially, or even
	   fail.  Range is 0 to INT_MAX. Default is 0.

       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

       font_a
	   the ascent size defined in the font metrics

       font_d
	   the descent size defined in the font metrics

       top_a
	   the maximum ascender of the glyphs of the first text line

       bottom_d
	   the maximum descender of the glyphs of the last text line

       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 to each
	   other, so you can for example specify "y=x/dar".

       pict_type
	   A one character description of the current frame's picture type.

       pkt_pos
	   The current packet's position in the input file or stream (in
	   bytes, from the start of the input). A value of -1 indicates this
	   info is not available.

       duration
	   The current packet's duration, in seconds.

       pkt_size
	   The current packet's size (in bytes).

       Text expansion

       If expansion is set to "strftime", the filter recognizes sequences
       accepted by the "strftime" C function in the provided text and expands
       them accordingly. Check the documentation of "strftime". This feature
       is deprecated in favor of "normal" expansion with the "gmtime" or
       "localtime" expansion functions.

       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 with the textfile option
       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" C function format string.  The
	   format string is extended to support the variable %[1-6]N which
	   prints fractions of the second with optionally specified number of
	   digits.

       localtime
	   The time at which the filter is running, expressed in the local
	   time zone.  It can accept an argument: a "strftime" C function
	   format string.  The format string is extended to support the
	   variable %[1-6]N which prints fractions of the second with
	   optionally specified number of digits.

       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.

	   Available metadata can be identified by inspecting entries starting
	   with TAG included within each frame section printed by running
	   "ffprobe -show_frames".

	   String metadata generated in filters leading to the drawtext filter
	   are also available.

       n, frame_num
	   The frame number, starting from 0.

       pict_type
	   A one 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 "hms", a third argument "24HH" may be
	   supplied to present the hour part of the formatted timestamp in 24h
	   format (00-23).

	   If the format is set to "localtime" or "gmtime", a third argument
	   may be supplied: a "strftime" C function format string.  By
	   default, YYYY-MM-DD HH:MM:SS format will be used.

       Commands

       This filter supports altering parameters via commands:

       reinit
	   Alter existing filter parameters.

	   Syntax for the argument is the same as for filter invocation, e.g.

		   fontsize=56:fontcolor=green:text='Hello World'

	   Full filter invocation with sendcmd would look like this:

		   sendcmd=c='56.0 drawtext reinit fontsize=56\:fontcolor=green\:text=Hello\\ World'

	   If the entire argument can't be parsed or applied as valid values
	   then the filter will continue with its existing parameters.

       The following options are also supported as commands:

       *<x>
       *<y>
       *<alpha>
       *<fontsize>
       *<fontcolor>
       *<boxcolor>
       *<bordercolor>
       *<shadowcolor>
       *<box>
       *<boxw>
       *<boxh>
       *<boxborderw>
       *<line_spacing>
       *<text_align>
       *<shadowx>
       *<shadowy>
       *<borderw>

       Examples

       o   Draw "Test Text" with font FreeSerif, using the default values for
	   the optional parameters.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

       o   Draw 'Test Text' with font FreeSerif of size 24 at position x=100
	   and y=50 (counting from the top-left corner of the screen), text is
	   yellow with a red box around it. Both the text and the box have an
	   opacity of 20%.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
			     x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

	   Note that the double quotes are not necessary if spaces are not
	   used within the parameter list.

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

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

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

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

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

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

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

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

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

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

       o   Show text for 1 second every 3 seconds:

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

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

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

       o   Draw "Test Text" with font size dependent on height of the video.

		   drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"

       o   Print the date of a real-time encoding (see documentation for the
	   "strftime" C function):

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

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

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

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

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

       o   Plot special lavf.image2dec.source_basename metadata onto each
	   frame if such metadata exists. Otherwise, plot the string "NA".
	   Note that image2 demuxer must have option -export_path_metadata 1
	   for the special metadata fields to be available for filters.

		   drawtext="fontsize=20:fontcolor=white:fontfile=FreeSans.ttf:text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=10:y=10"

       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/>.

       For more information about libharfbuzz, check:
       <https://github.com/harfbuzz/harfbuzz>.

   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.

	   canny
	       Apply Canny edge detector on all selected planes.

	   Default value is wires.

       planes
	   Select planes for filtering. By default all available planes are
	   filtered.

       Examples

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

		   edgedetect=low=0.1:high=0.4

       o   Painting effect without thresholding:

		   edgedetect=mode=colormix:high=0

   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. Default is disabled.

       use_alpha
	   Include alpha values in the quantization calculation. Allows
	   creating palettized output images (e.g. PNG8) with multiple alpha
	   smooth blending.

   entropy
       Measure graylevel entropy in histogram of color channels of video
       frames.

       It accepts the following parameters:

       mode
	   Can be either normal or diff. Default is normal.

	   diff mode measures entropy of histogram delta values, absolute
	   differences between neighbour histogram values.

   epx
       Apply the EPX magnification filter which is designed for pixel art.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "2xEPX", 3 for "3xEPX".  Default
	   is 3.

   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 "-1000.0" to 1000.0. 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; deprecated, do not use

       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

       Commands

       This filter supports the all above options as commands.

   estdif
       Deinterlace the input video ("estdif" stands for "Edge Slope Tracing
       Deinterlacing Filter").

       Spatial only filter that uses edge slope tracing algorithm to
       interpolate missing lines.  It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It accepts one of the following
	   values:

	   frame
	       Output one frame for each frame.

	   field
	       Output one frame for each field.

	   The default value is "field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   tff Assume the top field is first.

	   bff Assume the bottom field is first.

	   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. Accepts one of the following
	   values:

	   all Deinterlace all frames.

	   interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

       rslope
	   Specify the search radius for edge slope tracing. Default value is
	   1.  Allowed range is from 1 to 15.

       redge
	   Specify the search radius for best edge matching. Default value is
	   2.  Allowed range is from 0 to 15.

       ecost
	   Specify the edge cost for edge matching. Default value is 2.
	   Allowed range is from 0 to 50.

       mcost
	   Specify the middle cost for edge matching. Default value is 1.
	   Allowed range is from 0 to 50.

       dcost
	   Specify the distance cost for edge matching. Default value is 1.
	   Allowed range is from 0 to 50.

       interp
	   Specify the interpolation used. Default is 4-point interpolation.
	   It accepts one of the following values:

	   2p  Two-point interpolation.

	   4p  Four-point interpolation.

	   6p  Six-point interpolation.

       Commands

       This filter supports same commands as options.

   exposure
       Adjust exposure of the video stream.

       The filter accepts the following options:

       exposure
	   Set the exposure correction in EV. Allowed range is from -3.0 to
	   3.0 EV Default value is 0 EV.

       black
	   Set the black level correction. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       Commands

       This filter supports same commands as options.

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

       The filter accepts the following option:

       planes
	   Set plane(s) to extract.

	   Available values for planes are:

	   y
	   u
	   v
	   a
	   r
	   g
	   b

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

       Examples

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

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

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

       It accepts the following parameters:

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

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

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

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

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

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

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

       Examples

       o   Fade in the first 30 frames of video:

		   fade=in:0:30

	   The command above is equivalent to:

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

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

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

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

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

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

		   fade=in:5:20:color=yellow

       o   Fade in alpha over first 25 frames of video:

		   fade=in:0:25:alpha=1

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

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

   feedback
       Apply feedback video filter.

       This filter pass cropped input frames to 2nd output.  From there it can
       be filtered with other video filters.  After filter receives frame from
       2nd input, that frame is combined on top of original frame from 1st
       input and passed to 1st output.

       The typical usage is filter only part of frame.

       The filter accepts the following options:

       x
       y   Set the top left crop position.

       w
       h   Set the crop size.

       Examples

       o   Blur only top left rectangular part of video frame size 100x100
	   with gblur filter.

		   [in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]gblur=8[blurin]

       o   Draw black box on top left part of video frame of size 100x100 with
	   drawbox filter.

		   [in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]drawbox=x=0:y=0:w=100:h=100:t=100[blurin]

       o   Pixelize rectangular part of video frame of size 100x100 with
	   pixelize filter.

		   [in][blurin]feedback=x=320:y=240:w=100:h=100[out][blurout];[blurout]pixelize[blurin]

   fftdnoiz
       Denoise frames using 3D FFT (frequency domain filtering).

       The filter accepts the following options:

       sigma
	   Set the noise sigma constant. This sets denoising strength.
	   Default value is 1. Allowed range is from 0 to 30.  Using very high
	   sigma with low overlap may give blocking artifacts.

       amount
	   Set amount of denoising. By default all detected noise is reduced.
	   Default value is 1. Allowed range is from 0 to 1.

       block
	   Set size of block in pixels, Default is 32, can be 8 to 256.

       overlap
	   Set block overlap. Default is 0.5. Allowed range is from 0.2 to
	   0.8.

       method
	   Set denoising method. Default is "wiener", can also be "hard".

       prev
	   Set number of previous frames to use for denoising. By default is
	   set to 0.

       next
	   Set number of next frames to to use for denoising. By default is
	   set to 0.

       planes
	   Set planes which will be filtered, by default are all available
	   filtered except alpha.

   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.

       WS
       HS  The size of FFT array for horizontal and vertical processing.

       Examples

       o   High-pass:

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

       o   Low-pass:

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

       o   Sharpen:

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

       o   Blur:

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

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

       The filter accepts the following options:

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

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

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

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

	   If optionally followed by "+" output frame will be marked as
	   interlaced, else if followed by "-" output frame will be marked as
	   progressive, else it will be marked same as input frame.  If
	   optionally followed by "t" output frame will use only top field, or
	   in case of "b" it will use only bottom field.  If line starts with
	   "#" or ";" that line is skipped.

       mode
	   Can be item "absolute" or "relative" or "pattern". Default is
	   "absolute".	The "pattern" mode is same as "relative" mode, except
	   at last entry of file if there are more frames to process than
	   "hint" file is seek back to start.

       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

   fillborders
       Fill borders of the input video, without changing video stream
       dimensions.  Sometimes video can have garbage at the four edges and you
       may not want to crop video input to keep size multiple of some number.

       This filter accepts the following options:

       left
	   Number of pixels to fill from left border.

       right
	   Number of pixels to fill from right border.

       top Number of pixels to fill from top border.

       bottom
	   Number of pixels to fill from bottom border.

       mode
	   Set fill mode.

	   It accepts the following values:

	   smear
	       fill pixels using outermost pixels

	   mirror
	       fill pixels using mirroring (half sample symmetric)

	   fixed
	       fill pixels with constant value

	   reflect
	       fill pixels using reflecting (whole sample symmetric)

	   wrap
	       fill pixels using wrapping

	   fade
	       fade pixels to constant value

	   margins
	       fill pixels at top and bottom with weighted averages pixels
	       near borders

	   Default is smear.

       color
	   Set color for pixels in fixed or fade mode. Default is black.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   find_rect
       Find a rectangular object in the input video.

       The object to search for must be specified as a gray8 image specified
       with the object option.

       For each possible match, a score is computed. If the score reaches the
       specified threshold, the object is considered found.

       If the input video contains multiple instances of the object, the
       filter will find only one of them.

       When an object is found, the following metadata entries are set in the
       matching frame:

       lavfi.rect.w
	   width of object

       lavfi.rect.h
	   height of object

       lavfi.rect.x
	   x position of object

       lavfi.rect.y
	   y position of object

       lavfi.rect.score
	   match score of the found object

       It accepts the following options:

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

       threshold
	   Detection threshold, expressed as a decimal number in the range
	   0-1.

	   A threshold value of 0.01 means only exact matches, a threshold of
	   0.99 means almost everything matches.

	   Default value is 0.5.

       mipmaps
	   Number of mipmaps, default is 3.

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

       discard
	   Discard frames where object is not detected. Default is disabled.

       Examples

       o   Cover a rectangular object by the supplied image of a given video
	   using ffmpeg:

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

       o   Find the position of an object in each frame using ffprobe and
	   write it to a log file:

		   ffprobe -f lavfi movie=test.mp4,find_rect=object=object.pgm:threshold=0.3 \
		     -show_entries frame=pkt_pts_time:frame_tags=lavfi.rect.x,lavfi.rect.y \
		     -of csv -o find_rect.csv

   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".

       color_spaces
	   A '|'-separated list of color space names, such as
	   "color_spaces=bt709|bt470bg|bt2020nc".

       color_ranges
	   A '|'-separated list of color range names, such as
	   "color_spaces=tv|pc".

       Examples

       o   Convert the input video to the yuv420p format

		   format=pix_fmts=yuv420p

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

		   format=pix_fmts=yuv420p|yuv444p|yuv410p

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

       It accepts the following parameters:

       fps The desired output frame rate. It accepts expressions containing
	   the following constants:

	   source_fps
	       The input's frame rate

	   ntsc
	       NTSC frame rate of "30000/1001"

	   pal PAL frame rate of 25.0

	   film
	       Film frame rate of 24.0

	   ntsc_film
	       NTSC-film frame rate of "24000/1001"

	   The default is 25.

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

       round
	   Timestamp (PTS) rounding method.

	   Possible values are:

	   zero
	       round towards 0

	   inf round away from 0

	   down
	       round towards -infinity

	   up  round towards +infinity

	   near
	       round to nearest

	   The default is "near".

       eof_action
	   Action performed when reading the last frame.

	   Possible values are:

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

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

	   The default is "round".

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

       See also the setpts filter.

       Examples

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

		   fps=fps=25

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

		   fps=fps=film:round=near

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

       It accepts the following parameters:

       format
	   The desired packing format. Supported values are:

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

	   tab The views are on top of each other.

	   lines
	       The views are packed by line.

	   columns
	       The views are packed by column.

	   frameseq
	       The views are temporally interleaved.

       Some examples:

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

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

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

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

       A description of the accepted options follows.

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

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

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

       scene
	   Specify the level at which a scene change is detected as a value
	   between 0 and 100 to indicate a new scene; a low value reflects a
	   low probability for the current frame to introduce a new scene,
	   while a higher value means the current frame is more likely to be
	   one.  The default is 8.2.

       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.

   freezedetect
       Detect frozen video.

       This filter logs a message and sets frame metadata when it detects that
       the input video has no significant change in content during a specified
       duration.  Video freeze detection calculates the mean average absolute
       difference of all the components of video frames and compares it to a
       noise floor.

       The printed times and duration are expressed in seconds. The
       "lavfi.freezedetect.freeze_start" metadata key is set on the first
       frame whose timestamp equals or exceeds the detection duration and it
       contains the timestamp of the first frame of the freeze. The
       "lavfi.freezedetect.freeze_duration" and
       "lavfi.freezedetect.freeze_end" metadata keys are set on the first
       frame after the freeze.

       The filter accepts the following options:

       noise, n
	   Set noise tolerance. Can be specified in dB (in case "dB" is
	   appended to the specified value) or as a difference ratio between 0
	   and 1. Default is -60dB, or 0.001.

       duration, d
	   Set freeze duration until notification (default is 2 seconds).

   freezeframes
       Freeze video frames.

       This filter freezes video frames using frame from 2nd input.

       The filter accepts the following options:

       first
	   Set number of first frame from which to start freeze.

       last
	   Set number of last frame from which to end freeze.

       replace
	   Set number of frame from 2nd input which will be used instead of
	   replaced frames.

   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 a color
       description as specified in the "Color" section in the ffmpeg-utils
       manual, a position (specified as X/Y, where X and Y are floating point
       numbers) and/or a string.

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

       Examples

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

		   frei0r=filter_name=distort0r:filter_params=0.5|0.01

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

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

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

		   frei0r=perspective:0.2/0.2|0.8/0.2

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

       Commands

       This filter supports the filter_params option as commands.

   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).

   fsync
       Synchronize video frames with an external mapping from a file.

       For each input PTS given in the map file it either drops or creates as
       many frames as necessary to recreate the sequence of output frames
       given in the map file.

       This filter is useful to recreate the output frames of a framerate
       conversion by the fps filter, recorded into a map file using the ffmpeg
       option "-stats_mux_pre", and do further processing to the corresponding
       frames e.g. quality comparison.

       Each line of the map file must contain three items per input frame, the
       input PTS (decimal), the output PTS (decimal) and the output TIMEBASE
       (decimal/decimal), seperated by a space.  This file format corresponds
       to the output of "-stats_mux_pre_fmt="{ptsi} {pts} {tb}"".

       The filter assumes the map file is sorted by increasing input PTS.

       The filter accepts the following options:

       file, f
	   The filename of the map file to be used.

       Example:

	       # Convert a video to 25 fps and record a MAP_FILE file with the default format of this filter
	       ffmpeg -i INPUT -vf fps=fps=25 -stats_mux_pre MAP_FILE -stats_mux_pre_fmt "{ptsi} {pts} {tb}" OUTPUT

	       # Sort MAP_FILE by increasing input PTS
	       sort -n MAP_FILE

	       # Use INPUT, OUTPUT and the MAP_FILE from above to compare the corresponding frames in INPUT and OUTPUT via SSIM
	       ffmpeg -i INPUT -i OUTPUT -filter_complex '[0:v]fsync=file=MAP_FILE[ref];[1:v][ref]ssim' -f null -

   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. Default 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".

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   geq
       Apply generic equation to each pixel.

       The filter accepts the following options:

       lum_expr, lum
	   Set the luma 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 luma 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 luma 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.

       psum(x,y), lumsum(x, y), cbsum(x,y), crsum(x,y), rsum(x,y), gsum(x,y),
       bsum(x,y), alphasum(x,y)
	   Sum of sample values in the rectangle from (0,0) to (x,y), this
	   allows obtaining sums of samples within a rectangle. See the
	   functions without the sum postfix.

       interpolation
	   Set one of interpolation methods:

	   nearest, n
	   bilinear, b

	   Default is bilinear.

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

       Please note that this filter can use multiple threads in which case
       each slice will have its own expression state. If you want to use only
       a single expression state because your expressions depend on previous
       state then you should limit the number of filter threads to 1.

       Examples

       o   Flip the image horizontally:

		   geq=p(W-X\,Y)

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

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

       o   Generate a fancy enigmatic moving light:

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

       o   Generate a quick emboss effect:

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

       o   Modify RGB components depending on pixel position:

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

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

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

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

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

       It accepts the following parameters:

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

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

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

       Examples

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

		   gradfun=3.5:8

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

		   gradfun=radius=8

   graphmonitor
       Show various filtergraph stats.

       With this filter one can debug complete filtergraph.  Especially issues
       with links filling with queued frames.

       The filter accepts the following options:

       size, s
	   Set video output size. Default is hd720.

       opacity, o
	   Set video opacity. Default is 0.9. Allowed range is from 0 to 1.

       mode, m
	   Set output mode flags.

	   Available values for flags are:

	   full
	       No any filtering. Default.

	   compact
	       Show only filters with queued frames.

	   nozero
	       Show only filters with non-zero stats.

	   noeof
	       Show only filters with non-eof stat.

	   nodisabled
	       Show only filters that are enabled in timeline.

       flags, f
	   Set flags which enable which stats are shown in video.

	   Available values for flags are:

	   none
	       All flags turned off.

	   all All flags turned on.

	   queue
	       Display number of queued frames in each link.

	   frame_count_in
	       Display number of frames taken from filter.

	   frame_count_out
	       Display number of frames given out from filter.

	   frame_count_delta
	       Display delta number of frames between above two values.

	   pts Display current filtered frame pts.

	   pts_delta
	       Display pts delta between current and previous frame.

	   time
	       Display current filtered frame time.

	   time_delta
	       Display time delta between current and previous frame.

	   timebase
	       Display time base for filter link.

	   format
	       Display used format for filter link.

	   size
	       Display video size or number of audio channels in case of audio
	       used by filter link.

	   rate
	       Display video frame rate or sample rate in case of audio used
	       by filter link.

	   eof Display link output status.

	   sample_count_in
	       Display number of samples taken from filter.

	   sample_count_out
	       Display number of samples given out from filter.

	   sample_count_delta
	       Display delta number of samples between above two values.

	   disabled
	       Show the timeline filter status.

       rate, r
	   Set upper limit for video rate of output stream, Default value is
	   25.	This guarantee that output video frame rate will not be higher
	   than this value.

   grayworld
       A color constancy filter that applies color correction based on the
       grayworld assumption

       See:
       <https://www.researchgate.net/publication/275213614_A_New_Color_Correction_Method_for_Underwater_Imaging>

       The algorithm  uses linear light, so input data should be linearized
       beforehand (and possibly correctly tagged).

	       ffmpeg -i INPUT -vf zscale=transfer=linear,grayworld,zscale=transfer=bt709,format=yuv420p OUTPUT

   greyedge
       A color constancy variation filter which estimates scene illumination
       via grey edge algorithm and corrects the scene colors accordingly.

       See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>

       The filter accepts the following options:

       difford
	   The order of differentiation to be applied on the scene. Must be
	   chosen in the range [0,2] and default value is 1.

       minknorm
	   The Minkowski parameter to be used for calculating the Minkowski
	   distance. Must be chosen in the range [0,20] and default value is
	   1. Set to 0 for getting max value instead of calculating Minkowski
	   distance.

       sigma
	   The standard deviation of Gaussian blur to be applied on the scene.
	   Must be chosen in the range [0,1024.0] and default value = 1.
	   floor( sigma * break_off_sigma(3) ) can't be equal to 0 if difford
	   is greater than 0.

       Examples

       o   Grey Edge:

		   greyedge=difford=1:minknorm=5:sigma=2

       o   Max Edge:

		   greyedge=difford=1:minknorm=0:sigma=2

   guided
       Apply guided filter for edge-preserving smoothing, dehazing and so on.

       The filter accepts the following options:

       radius
	   Set the box radius in pixels.  Allowed range is 1 to 20. Default is
	   3.

       eps Set regularization parameter (with square).	Allowed range is 0 to
	   1. Default is 0.01.

       mode
	   Set filter mode. Can be "basic" or "fast".  Default is "basic".

       sub Set subsampling ratio for "fast" mode.  Range is 2 to 64. Default
	   is 4.  No subsampling occurs in "basic" mode.

       guidance
	   Set guidance mode. Can be "off" or "on". Default is "off".  If
	   "off", single input is required.  If "on", two inputs of the same
	   resolution and pixel format are required.  The second input serves
	   as the guidance.

       planes
	   Set planes to filter. Default is first only.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Edge-preserving smoothing with guided filter:

		   ffmpeg -i in.png -vf guided out.png

       o   Dehazing, structure-transferring filtering, detail enhancement with
	   guided filter.  For the generation of guidance image, refer to
	   paper "Guided Image Filtering".  See:
	   <http://kaiminghe.com/publications/pami12guidedfilter.pdf>.

		   ffmpeg -i in.png -i guidance.png -filter_complex guided=guidance=on out.png

   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:

       clut
	   Set which CLUT video frames will be processed from second input
	   stream, can be first or all. Default is all.

       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).

       This filter also supports the framesync options.

       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>.

       Commands

       This filter supports the "interp" option as commands.

       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.

       colors_mode
	   Set colors mode.  It accepts the following values:

	   whiteonblack
	   blackonwhite
	   whiteongray
	   blackongray
	   coloronblack
	   coloronwhite
	   colorongray
	   blackoncolor
	   whiteoncolor
	   grayoncolor

	   Default is "whiteonblack".

       Examples

       o   Calculate and draw histogram:

		   ffplay -i input -vf histogram

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

       It accepts the following optional parameters:

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

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

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

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

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   hwdownload
       Download hardware frames to system memory.

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

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

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

       o   Hardware frame input, normal frame output

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

       o   Normal frame input, hardware frame output

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

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

       o   Hardware frame input and output

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

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

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

       The following additional parameters are accepted:

       mode
	   Set the frame mapping mode.	Some combination of:

	   read
	       The mapped frame should be readable.

	   write
	       The mapped frame should be writeable.

	   overwrite
	       The mapping will always overwrite the entire frame.

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

	   direct
	       The mapping must not involve any copying.

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

	   Defaults to read+write if not specified.

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

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

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

   hwupload
       Upload system memory frames to hardware surfaces.

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

       The following additional parameters are accepted:

       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.

   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 accepts 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.

   hsvhold
       Turns a certain HSV range into gray values.

       This filter measures color difference between set HSV color in options
       and ones measured in video stream. Depending on options, output colors
       can be changed to be gray or not.

       The filter accepts the following options:

       hue Set the hue value which will be used in color difference
	   calculation.  Allowed range is from -360 to 360. Default value is
	   0.

       sat Set the saturation value which will be used in color difference
	   calculation.  Allowed range is from -1 to 1. Default value is 0.

       val Set the value which will be used in color difference calculation.
	   Allowed range is from -1 to 1. Default value is 0.

       similarity
	   Set similarity percentage with the key color.  Allowed range is
	   from 0 to 1. Default value is 0.01.

	   0.00001 matches only the exact key color, while 1.0 matches
	   everything.

       blend
	   Blend percentage.  Allowed range is from 0 to 1. Default value is
	   0.

	   0.0 makes pixels either fully gray, or not gray at all.

	   Higher values result in more gray pixels, with a higher gray pixel
	   the more similar the pixels color is to the key color.

   hsvkey
       Turns a certain HSV range into transparency.

       This filter measures color difference between set HSV color in options
       and ones measured in video stream. Depending on options, output colors
       can be changed to transparent by adding alpha channel.

       The filter accepts the following options:

       hue Set the hue value which will be used in color difference
	   calculation.  Allowed range is from -360 to 360. Default value is
	   0.

       sat Set the saturation value which will be used in color difference
	   calculation.  Allowed range is from -1 to 1. Default value is 0.

       val Set the value which will be used in color difference calculation.
	   Allowed range is from -1 to 1. Default value is 0.

       similarity
	   Set similarity percentage with the key color.  Allowed range is
	   from 0 to 1. Default value is 0.01.

	   0.00001 matches only the exact key color, while 1.0 matches
	   everything.

       blend
	   Blend percentage.  Allowed range is from 0 to 1. Default value is
	   0.

	   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.

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

       It accepts the following parameters:

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

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

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

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

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

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

       n   frame count of the input frame starting from 0

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

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

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

       tb  time base of the input video

       Examples

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

		   hue=h=90:s=1

       o   Same command but expressing the hue in radians:

		   hue=H=PI/2:s=1

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

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

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

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

	   The general fade-in expression can be written as:

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

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

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

	   The general fade-out expression can be written as:

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

       Commands

       This filter supports the following commands:

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

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

   huesaturation
       Apply hue-saturation-intensity adjustments to input video stream.

       This filter operates in RGB colorspace.

       This filter accepts the following options:

       hue Set the hue shift in degrees to apply. Default is 0.  Allowed range
	   is from -180 to 180.

       saturation
	   Set the saturation shift. Default is 0.  Allowed range is from -1
	   to 1.

       intensity
	   Set the intensity shift. Default is 0.  Allowed range is from -1 to
	   1.

       colors
	   Set which primary and complementary colors are going to be
	   adjusted.  This options is set by providing one or multiple values.
	   This can select multiple colors at once. By default all colors are
	   selected.

	   r   Adjust reds.

	   y   Adjust yellows.

	   g   Adjust greens.

	   c   Adjust cyans.

	   b   Adjust blues.

	   m   Adjust magentas.

	   a   Adjust all colors.

       strength
	   Set strength of filtering. Allowed range is from 0 to 100.  Default
	   value is 1.

       rw, gw, bw
	   Set weight for each RGB component. Allowed range is from 0 to 1.
	   By default is set to 0.333, 0.334, 0.333.  Those options are used
	   in saturation and lightess processing.

       lightness
	   Set preserving lightness, by default is disabled.  Adjusting hues
	   can change lightness from original RGB triplet, with this option
	   enabled lightness is kept at same 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.

       The "hysteresis" filter also supports the framesync options.

   iccdetect
       Detect the colorspace  from an embedded ICC profile (if present), and
       update the frame's tags accordingly.

       This filter accepts the following options:

       force
	   If true, the frame's existing colorspace tags will always be
	   overridden by values detected from an ICC profile. Otherwise, they
	   will only be assigned if they contain "unknown". Enabled by
	   default.

   iccgen
       Generate ICC profiles and attach them to frames.

       This filter accepts the following options:

       color_primaries
       color_trc
	   Configure the colorspace that the ICC profile will be generated
	   for. The default value of "auto" infers the value from the input
	   frame's metadata, defaulting to BT.709/sRGB as appropriate.

	   See the setparams filter for a list of possible values, but note
	   that "unknown" are not valid values for this filter.

       force
	   If true, an ICC profile will be generated even if it would
	   overwrite an already existing ICC profile. Disabled by default.

   identity
       Obtain the identity score between two input videos.

       This filter takes two input videos.

       Both input videos 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 per component, average, min and max identity score is
       printed through the logging system.

       The filter stores the calculated identity scores of each frame in frame
       metadata.

       This filter also supports the framesync options.

       In the below example the input file main.mpg being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi identity -f null -

   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

       Examples

       Inspect the field order of the first 360 frames in a video, in verbose
       detail:

	       ffmpeg -i INPUT -filter:v idet,metadata=mode=print -frames:v 360 -an -f null -

       The idet filter will add analysis metadata to each frame, which will
       then be discarded. At the end, the filter will also print a final
       report with statistics.

   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.

       Commands

       This filter supports the all above options as commands.

   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.

       Commands

       This filter supports the all above options as commands.

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

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

       It accepts the following optional parameters:

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

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

	   0, off
	       Disable vertical lowpass filter

	   1, linear
	       Enable linear filter (default)

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

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

       The description of the accepted parameters follows.

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

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

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

       sharp
	   Enable additional sharpening if set to 1. Default is 0.

       twoway
	   Enable twoway sharpening if set to 1. Default is 0.

       Examples

       o   Apply default values:

		   kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

       o   Enable additional sharpening:

		   kerndeint=sharp=1

       o   Paint processed pixels in white:

		   kerndeint=map=1

   kirsch
       Apply kirsch 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.

       Commands

       This filter supports the all above options as commands.

   lagfun
       Slowly update darker pixels.

       This filter makes short flashes of light appear longer.	This filter
       accepts the following options:

       decay
	   Set factor for decaying. Default is .95. Allowed range is from 0 to
	   1.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Commands

       This filter supports the all above options as commands.

   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. Default is 0.5.

       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. Default is 0.5.

       k1  Coefficient of the quadratic correction term. This value has a
	   range [-1,1]. 0 means no correction. Default is 0.

       k2  Coefficient of the double quadratic correction term. This value has
	   a range [-1,1].  0 means no correction. Default is 0.

       i   Set interpolation type. Can be "nearest" or "bilinear".  Default is
	   "nearest".

       fc  Specify the color of the unmapped pixels. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.
	   Default color is "black@0".

       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.

       Commands

       This filter supports the all above options as commands.

   lensfun
       Apply lens correction via the lensfun library
       (<http://lensfun.sourceforge.net/>).

       The "lensfun" filter requires the camera make, camera model, and lens
       model to apply the lens correction. The filter will load the lensfun
       database and query it to find the corresponding camera and lens entries
       in the database. As long as these entries can be found with the given
       options, the filter can perform corrections on frames. Note that
       incomplete strings will result in the filter choosing the best match
       with the given options, and the filter will output the chosen camera
       and lens models (logged with level "info"). You must provide the make,
       camera model, and lens model as they are required.

       To obtain a list of available makes and models, leave out one or both
       of "make" and "model" options. The filter will send the full list to
       the log with level "INFO".  The first column is the make and the second
       column is the model.  To obtain a list of available lenses, set any
       values for make and model and leave out the "lens_model" option. The
       filter will send the full list of lenses in the log with level "INFO".
       The ffmpeg tool will exit after the list is printed.

       The filter accepts the following options:

       make
	   The make of the camera (for example, "Canon"). This option is
	   required.

       model
	   The model of the camera (for example, "Canon EOS 100D"). This
	   option is required.

       lens_model
	   The model of the lens (for example, "Canon EF-S 18-55mm f/3.5-5.6
	   IS STM"). This option is required.

       db_path
	   The full path to the lens database folder. If not set, the filter
	   will attempt to load the database from the install path when the
	   library was built. Default is unset.

       mode
	   The type of correction to apply. The following values are valid
	   options:

	   vignetting
	       Enables fixing lens vignetting.

	   geometry
	       Enables fixing lens geometry. This is the default.

	   subpixel
	       Enables fixing chromatic aberrations.

	   vig_geo
	       Enables fixing lens vignetting and lens geometry.

	   vig_subpixel
	       Enables fixing lens vignetting and chromatic aberrations.

	   distortion
	       Enables fixing both lens geometry and chromatic aberrations.

	   all Enables all possible corrections.

       focal_length
	   The focal length of the image/video (zoom; expected constant for
	   video). For example, a 18--55mm lens has focal length range of
	   [18--55], so a value in that range should be chosen when using that
	   lens. Default 18.

       aperture
	   The aperture of the image/video (expected constant for video). Note
	   that aperture is only used for vignetting correction. Default 3.5.

       focus_distance
	   The focus distance of the image/video (expected constant for
	   video). Note that focus distance is only used for vignetting and
	   only slightly affects the vignetting correction process. If
	   unknown, leave it at the default value (which is 1000).

       scale
	   The scale factor which is applied after transformation. After
	   correction the video is no longer necessarily rectangular. This
	   parameter controls how much of the resulting image is visible. The
	   value 0 means that a value will be chosen automatically such that
	   there is little or no unmapped area in the output image. 1.0 means
	   that no additional scaling is done. Lower values may result in more
	   of the corrected image being visible, while higher values may avoid
	   unmapped areas in the output.

       target_geometry
	   The target geometry of the output image/video. The following values
	   are valid options:

	   rectilinear (default)
	   fisheye
	   panoramic
	   equirectangular
	   fisheye_orthographic
	   fisheye_stereographic
	   fisheye_equisolid
	   fisheye_thoby
       reverse
	   Apply the reverse of image correction (instead of correcting
	   distortion, apply it).

       interpolation
	   The type of interpolation used when correcting distortion. The
	   following values are valid options:

	   nearest
	   linear (default)
	   lanczos

       Examples

       o   Apply lens correction with make "Canon", camera model "Canon EOS
	   100D", and lens model "Canon EF-S 18-55mm f/3.5-5.6 IS STM" with
	   focal length of "18" and aperture of "8.0".

		   ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8 -c:v h264 -b:v 8000k output.mov

       o   Apply the same as before, but only for the first 5 seconds of
	   video.

		   ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8:enable='lte(t\,5)' -c:v h264 -b:v 8000k output.mov

   libplacebo
       Flexible GPU-accelerated processing filter based on libplacebo
       (<https://code.videolan.org/videolan/libplacebo>).

       Options

       The options for this filter are divided into the following sections:

       Output mode

       These options control the overall output mode. By default, libplacebo
       will try to preserve the source colorimetry and size as best as it can,
       but it will apply any embedded film grain, dolby vision metadata or
       anamorphic SAR present in source frames.

       inputs
	   Set the number of inputs. This can be used, alongside the "idx"
	   variable, to allow placing/blending multiple inputs inside the
	   output frame. This effectively enables functionality similar to
	   hstack, overlay, etc.

       w
       h   Set the output video dimension expression. Default values are "iw"
	   and "ih".

	   Allows for the same expressions as the scale filter.

       crop_x
       crop_y
	   Set the input crop x/y expressions, default values are "(iw-cw)/2"
	   and "(ih-ch)/2".

       crop_w
       crop_h
	   Set the input crop width/height expressions, default values are
	   "iw" and "ih".

       pos_x
       pos_y
	   Set the output placement x/y expressions, default values are
	   "(ow-pw)/2" and "(oh-ph)/2".

       pos_w
       pos_h
	   Set the output placement width/height expressions, default values
	   are "ow" and "oh".

       fps Set the output frame rate. This can be rational, e.g. "60000/1001".
	   If set to the special string "none" (the default), input timestamps
	   will instead be passed through to the output unmodified. Otherwise,
	   the input video frames will be interpolated as necessary to rescale
	   the video to the specified target framerate, in a manner as
	   determined by the frame_mixer option.

       format
	   Set the output format override. If unset (the default), frames will
	   be output in the same format as the respective input frames.
	   Otherwise, format conversion will be performed.

       force_original_aspect_ratio
       force_divisible_by
	   Work the same as the identical scale filter options.

       normalize_sar
	   If enabled, output frames will always have a pixel aspect ratio of
	   1:1. This will introduce additional padding/cropping as necessary.
	   If disabled (the default), any aspect ratio mismatches, including
	   those from e.g. anamorphic video sources, are forwarded to the
	   output pixel aspect ratio.

       pad_crop_ratio
	   Specifies a ratio (between 0.0 and 1.0) between padding and
	   cropping when the input aspect ratio does not match the output
	   aspect ratio and normalize_sar is in effect. The default of 0.0
	   always pads the content with black borders, while a value of 1.0
	   always crops off parts of the content. Intermediate values are
	   possible, leading to a mix of the two approaches.

       fillcolor
	   Set the color used to fill the output area not covered by the
	   output image, for example as a result of normalize_sar. For the
	   general syntax of this option, check the "Color" section in the
	   ffmpeg-utils manual. Defaults to "black".

       corner_rounding
	   Render frames with rounded corners. The value, given as a float
	   ranging from 0.0 to 1.0, indicates the relative degree of rounding,
	   from fully square to fully circular. In other words, it gives the
	   radius divided by half the smaller side length. Defaults to 0.0.

       extra_opts
	   Pass extra libplacebo internal configuration options. These can be
	   specified as a list of key=value pairs separated by ':'. The
	   following example shows how to configure a custom filter kernel
	   ("EWA LanczosSharp") and use it to double the input image
	   resolution:

		   -vf "libplacebo=w=iw*2:h=ih*2:extra_opts='upscaler=custom\:upscaler_preset=ewa_lanczos\:upscaler_blur=0.9812505644269356'"

       colorspace
       color_primaries
       color_trc
       range
	   Configure the colorspace that output frames will be delivered in.
	   The default value of "auto" outputs frames in the same format as
	   the input frames, leading to no change. For any other value,
	   conversion will be performed.

	   See the setparams filter for a list of possible values.

       apply_filmgrain
	   Apply film grain (e.g. AV1 or H.274) if present in source frames,
	   and strip it from the output. Enabled by default.

       apply_dolbyvision
	   Apply Dolby Vision RPU metadata if present in source frames, and
	   strip it from the output. Enabled by default. Note that Dolby
	   Vision will always output BT.2020+PQ, overriding the usual input
	   frame metadata. These will also be picked as the values of "auto"
	   for the respective frame output options.

       In addition to the expression constants documented for the scale
       filter, the crop_w, crop_h, crop_x, crop_y, pos_w, pos_h, pos_x and
       pos_y options can also contain the following constants:

       in_idx, idx
	   The (0-based) numeric index of the currently active input stream.

       crop_w, cw
       crop_h, ch
	   The computed values of crop_w and crop_h.

       pos_w, pw
       pos_h, ph
	   The computed values of pos_w and pos_h.

       in_t, t
	   The input frame timestamp, in seconds. NAN if input timestamp is
	   unknown.

       out_t, ot
	   The input frame timestamp, in seconds. NAN if input timestamp is
	   unknown.

       n   The input frame number, starting with 0.

       Scaling

       The options in this section control how libplacebo performs upscaling
       and (if necessary) downscaling. Note that libplacebo will always
       internally operate on 4:4:4 content, so any sub-sampled chroma formats
       such as "yuv420p" will necessarily be upsampled and downsampled as part
       of the rendering process. That means scaling might be in effect even if
       the source and destination resolution are the same.

       upscaler
       downscaler
	   Configure the filter kernel used for upscaling and downscaling. The
	   respective defaults are "spline36" and "mitchell". For a full list
	   of possible values, pass "help" to these options. The most
	   important values are:

	   none
	       Forces the use of built-in GPU texture sampling (typically
	       bilinear). Extremely fast but poor quality, especially when
	       downscaling.

	   bilinear
	       Bilinear interpolation. Can generally be done for free on GPUs,
	       except when doing so would lead to aliasing. Fast and low
	       quality.

	   nearest
	       Nearest-neighbour interpolation. Sharp but highly aliasing.

	   oversample
	       Algorithm that looks visually similar to nearest-neighbour
	       interpolation but tries to preserve pixel aspect ratio. Good
	       for pixel art, since it results in minimal distortion of the
	       artistic appearance.

	   lanczos
	       Standard sinc-sinc interpolation kernel.

	   spline36
	       Cubic spline approximation of lanczos. No difference in
	       performance, but has very slightly less ringing.

	   ewa_lanczos
	       Elliptically weighted average version of lanczos, based on a
	       jinc-sinc kernel.  This is also popularly referred to as just
	       "Jinc scaling". Slow but very high quality.

	   gaussian
	       Gaussian kernel. Has certain ideal mathematical properties, but
	       subjectively very blurry.

	   mitchell
	       Cubic BC spline with parameters recommended by Mitchell and
	       Netravali. Very little ringing.

       frame_mixer
	   Controls the kernel used for mixing frames temporally. The default
	   value is "none", which disables frame mixing. For a full list of
	   possible values, pass "help" to this option. The most important
	   values are:

	   none
	       Disables frame mixing, giving a result equivalent to "nearest
	       neighbour" semantics.

	   oversample
	       Oversamples the input video to create a "Smooth Motion"-type
	       effect: if an output frame would exactly fall on the transition
	       between two video frames, it is blended according to the
	       relative overlap. This is the recommended option whenever
	       preserving the original subjective appearance is desired.

	   mitchell_clamp
	       Larger filter kernel that smoothly interpolates multiple frames
	       in a manner designed to eliminate ringing and other artefacts
	       as much as possible. This is the recommended option wherever
	       maximum visual smoothness is desired.

	   linear
	       Linear blend/fade between frames. Especially useful for
	       constructing e.g.  slideshows.

       lut_entries
	   Configures the size of scaler LUTs, ranging from 1 to 256. The
	   default of 0 will pick libplacebo's internal default, typically 64.

       antiringing
	   Enables anti-ringing (for non-EWA filters). The value (between 0.0
	   and 1.0) configures the strength of the anti-ringing algorithm. May
	   increase aliasing if set too high. Disabled by default.

       sigmoid
	   Enable sigmoidal compression during upscaling. Reduces ringing
	   slightly.  Enabled by default.

       Debanding

       Libplacebo comes with a built-in debanding filter that is good at
       counteracting many common sources of banding and blocking. Turning this
       on is highly recommended whenever quality is desired.

       deband
	   Enable (fast) debanding algorithm. Disabled by default.

       deband_iterations
	   Number of deband iterations of the debanding algorithm. Each
	   iteration is performed with progressively increased radius (and
	   diminished threshold).  Recommended values are in the range 1 to 4.
	   Defaults to 1.

       deband_threshold
	   Debanding filter strength. Higher numbers lead to more aggressive
	   debanding.  Defaults to 4.0.

       deband_radius
	   Debanding filter radius. A higher radius is better for slow
	   gradients, while a lower radius is better for steep gradients.
	   Defaults to 16.0.

       deband_grain
	   Amount of extra output grain to add. Helps hide imperfections.
	   Defaults to 6.0.

       Color adjustment

       A collection of subjective color controls. Not very rigorous, so the
       exact effect will vary somewhat depending on the input primaries and
       colorspace.

       brightness
	   Brightness boost, between "-1.0" and 1.0. Defaults to 0.0.

       contrast
	   Contrast gain, between 0.0 and 16.0. Defaults to 1.0.

       saturation
	   Saturation gain, between 0.0 and 16.0. Defaults to 1.0.

       hue Hue shift in radians, between "-3.14" and 3.14. Defaults to 0.0.
	   This will rotate the UV subvector, defaulting to BT.709
	   coefficients for RGB inputs.

       gamma
	   Gamma adjustment, between 0.0 and 16.0. Defaults to 1.0.

       cones
	   Cone model to use for color blindness simulation. Accepts any
	   combination of "l", "m" and "s". Here are some examples:

	   m   Deuteranomaly / deuteranopia (affecting 3%-4% of the
	       population)

	   l   Protanomaly / protanopia (affecting 1%-2% of the population)

	   l+m Monochromacy (very rare)

	   l+m+s
	       Achromatopsy (complete loss of daytime vision, extremely rare)

       cone-strength
	   Gain factor for the cones specified by "cones", between 0.0 and
	   10.0. A value of 1.0 results in no change to color vision. A value
	   of 0.0 (the default) simulates complete loss of those cones. Values
	   above 1.0 result in exaggerating the differences between cones,
	   which may help compensate for reduced color vision.

       Peak detection

       To help deal with sources that only have static HDR10 metadata (or no
       tagging whatsoever), libplacebo uses its own internal frame analysis
       compute shader to analyze source frames and adapt the tone mapping
       function in realtime. If this is too slow, or if exactly reproducible
       frame-perfect results are needed, it's recommended to turn this feature
       off.

       peak_detect
	   Enable HDR peak detection. Ignores static MaxCLL/MaxFALL values in
	   favor of dynamic detection from the input. Note that the detected
	   values do not get written back to the output frames, they merely
	   guide the internal tone mapping process. Enabled by default.

       smoothing_period
	   Peak detection smoothing period, between 0.0 and 1000.0. Higher
	   values result in peak detection becoming less responsive to changes
	   in the input. Defaults to 100.0.

       minimum_peak
	   Lower bound on the detected peak (relative to SDR white), between
	   0.0 and 100.0. Defaults to 1.0.

       scene_threshold_low
       scene_threshold_high
	   Lower and upper thresholds for scene change detection. Expressed in
	   a logarithmic scale between 0.0 and 100.0. Default to 5.5 and 10.0,
	   respectively. Setting either to a negative value disables this
	   functionality.

       percentile
	   Which percentile of the frame brightness histogram to use as the
	   source peak for tone-mapping. Defaults to 99.995, a fairly
	   conservative value.	Setting this to 100.0 disables frame histogram
	   measurement and instead uses the true peak brightness for tone-
	   mapping.

       Tone mapping

       The options in this section control how libplacebo performs tone-
       mapping and gamut-mapping when dealing with mismatches between wide-
       gamut or HDR content.  In general, libplacebo relies on accurate source
       tagging and mastering display gamut information to produce the best
       results.

       gamut_mode
	   How to handle out-of-gamut colors that can occur as a result of
	   colorimetric gamut mapping.

	   clip
	       Do nothing, simply clip out-of-range colors to the RGB volume.
	       Low quality but extremely fast.

	   perceptual
	       Perceptually soft-clip colors to the gamut volume. This is the
	       default.

	   relative
	       Relative colorimetric hard-clip. Similar to "perceptual" but
	       without the soft knee.

	   saturation
	       Saturation mapping, maps primaries directly to primaries in RGB
	       space.  Not recommended except for artificial computer graphics
	       for which a bright, saturated display is desired.

	   absolute
	       Absolute colorimetric hard-clip. Performs no adjustment of the
	       white point.

	   desaturate
	       Hard-desaturates out-of-gamut colors towards white, while
	       preserving the luminance. Has a tendency to distort the visual
	       appearance of bright objects.

	   darken
	       Linearly reduces content brightness to preserves saturated
	       details, followed by clipping the remaining out-of-gamut
	       colors.

	   warn
	       Highlight out-of-gamut pixels (by inverting/marking them).

	   linear
	       Linearly reduces chromaticity of the entire image to make it
	       fit within the target color volume. Be careful when using this
	       on BT.2020 sources without proper mastering metadata, as doing
	       so will lead to excessive desaturation.

       tonemapping
	   Tone-mapping algorithm to use. Available values are:

	   auto
	       Automatic selection based on internal heuristics. This is the
	       default.

	   clip
	       Performs no tone-mapping, just clips out-of-range colors.
	       Retains perfect color accuracy for in-range colors but
	       completely destroys out-of-range information.  Does not perform
	       any black point adaptation. Not configurable.

	   st2094-40
	       EETF from SMPTE ST 2094-40 Annex B, which applies the Bezier
	       curves from HDR10+ dynamic metadata based on Bezier curves to
	       perform tone-mapping. The OOTF used is adjusted based on the
	       ratio between the targeted and actual display peak luminances.

	   st2094-10
	       EETF from SMPTE ST 2094-10 Annex B.2, which takes into account
	       the input signal average luminance in addition to the
	       maximum/minimum. The configurable contrast parameter influences
	       the slope of the linear output segment, defaulting to 1.0 for
	       no increase/decrease in contrast. Note that this does not
	       currently include the subjective gain/offset/gamma controls
	       defined in Annex B.3.

	   bt.2390
	       EETF from the ITU-R Report BT.2390, a hermite spline roll-off
	       with linear segment. The knee point offset is configurable.
	       Note that this parameter defaults to 1.0, rather than the value
	       of 0.5 from the ITU-R spec.

	   bt.2446a
	       EETF from ITU-R Report BT.2446, method A. Designed for well-
	       mastered HDR sources. Can be used for both forward and inverse
	       tone mapping. Not configurable.

	   spline
	       Simple spline consisting of two polynomials, joined by a single
	       pivot point.  The parameter gives the pivot point (in PQ
	       space), defaulting to 0.30.  Can be used for both forward and
	       inverse tone mapping.

	   reinhard
	       Simple non-linear, global tone mapping algorithm. The parameter
	       specifies the local contrast coefficient at the display peak.
	       Essentially, a parameter of 0.5 implies that the reference
	       white will be about half as bright as when clipping. Defaults
	       to 0.5, which results in the simplest formulation of this
	       function.

	   mobius
	       Generalization of the reinhard tone mapping algorithm to
	       support an additional linear slope near black. The tone mapping
	       parameter indicates the trade-off between the linear section
	       and the non-linear section. Essentially, for a given parameter
	       x, every color value below x will be mapped linearly, while
	       higher values get non-linearly tone-mapped. Values near 1.0
	       make this curve behave like "clip", while values near 0.0 make
	       this curve behave like "reinhard". The default value is 0.3,
	       which provides a good balance between colorimetric accuracy and
	       preserving out-of-gamut details.

	   hable
	       Piece-wise, filmic tone-mapping algorithm developed by John
	       Hable for use in Uncharted 2, inspired by a similar tone-
	       mapping algorithm used by Kodak.  Popularized by its use in
	       video games with HDR rendering. Preserves both dark and bright
	       details very well, but comes with the drawback of changing the
	       average brightness quite significantly. This is sort of similar
	       to "reinhard" with parameter 0.24.

	   gamma
	       Fits a gamma (power) function to transfer between the source
	       and target color spaces, effectively resulting in a perceptual
	       hard-knee joining two roughly linear sections. This preserves
	       details at all scales fairly accurately, but can result in an
	       image with a muted or dull appearance. The parameter is used as
	       the cutoff point, defaulting to 0.5.

	   linear
	       Linearly stretches the input range to the output range, in PQ
	       space. This will preserve all details accurately, but results
	       in a significantly different average brightness. Can be used
	       for inverse tone-mapping in addition to regular tone-mapping.
	       The parameter can be used as an additional linear gain
	       coefficient (defaulting to 1.0).

       tonemapping_param
	   For tunable tone mapping functions, this parameter can be used to
	   fine-tune the curve behavior. Refer to the documentation of
	   "tonemapping". The default value of 0.0 is replaced by the curve's
	   preferred default setting.

       inverse_tonemapping
	   If enabled, this filter will also attempt stretching SDR signals to
	   fill HDR output color volumes. Disabled by default.

       tonemapping_lut_size
	   Size of the tone-mapping LUT, between 2 and 1024. Defaults to 256.
	   Note that this figure is squared when combined with "peak_detect".

       contrast_recovery
	   Contrast recovery strength. If set to a value above 0.0, the source
	   image will be divided into high-frequency and low-frequency
	   components, and a portion of the high-frequency image is added back
	   onto the tone-mapped output.  May cause excessive ringing artifacts
	   for some HDR sources, but can improve the subjective sharpness and
	   detail left over in the image after tone-mapping.  Defaults to
	   0.30.

       contrast_smoothness
	   Contrast recovery lowpass kernel size. Defaults to 3.5. Increasing
	   or decreasing this will affect the visual appearance substantially.
	   Has no effect when "contrast_recovery" is disabled.

       Dithering

       By default, libplacebo will dither whenever necessary, which includes
       rendering to any integer format below 16-bit precision. It's
       recommended to always leave this on, since not doing so may result in
       visible banding in the output, even if the "debanding" filter is
       enabled. If maximum performance is needed, use "ordered_fixed" instead
       of disabling dithering.

       dithering
	   Dithering method to use. Accepts the following values:

	   none
	       Disables dithering completely. May result in visible banding.

	   blue
	       Dither with pseudo-blue noise. This is the default.

	   ordered
	       Tunable ordered dither pattern.

	   ordered_fixed
	       Faster ordered dither with a fixed size of 6. Texture-less.

	   white
	       Dither with white noise. Texture-less.

       dither_lut_size
	   Dither LUT size, as log base2 between 1 and 8. Defaults to 6,
	   corresponding to a LUT size of "64x64".

       dither_temporal
	   Enables temporal dithering. Disabled by default.

       Custom shaders

       libplacebo supports a number of custom shaders based on the mpv .hook
       GLSL syntax. A collection of such shaders can be found here:
       <https://github.com/mpv-player/mpv/wiki/User-Scripts#user-shaders>

       A full description of the mpv shader format is beyond the scope of this
       section, but a summary can be found here:
       <https://mpv.io/manual/master/#options-glsl-shader>

       custom_shader_path
	   Specifies a path to a custom shader file to load at runtime.

       custom_shader_bin
	   Specifies a complete custom shader as a raw string.

       Debugging / performance

       All of the options in this section default off. They may be of
       assistance when attempting to squeeze the maximum performance at the
       cost of quality.

       skip_aa
	   Disable anti-aliasing when downscaling.

       polar_cutoff
	   Truncate polar (EWA) scaler kernels below this absolute magnitude,
	   between 0.0 and 1.0.

       disable_linear
	   Disable linear light scaling.

       disable_builtin
	   Disable built-in GPU sampling (forces LUT).

       disable_fbos
	   Forcibly disable FBOs, resulting in loss of almost all
	   functionality, but offering the maximum possible speed.

       Commands

       This filter supports almost all of the above options as commands.

       Examples

       o   Tone-map input to standard gamut BT.709 output:

		   libplacebo=colorspace=bt709:color_primaries=bt709:color_trc=bt709:range=tv

       o   Rescale input to fit into standard 1080p, with high quality
	   scaling:

		   libplacebo=w=1920:h=1080:force_original_aspect_ratio=decrease:normalize_sar=true:upscaler=ewa_lanczos:downscaler=ewa_lanczos

       o   Interpolate low FPS / VFR input to smoothed constant 60 fps output:

		   libplacebo=fps=60:frame_mixer=mitchell_clamp

       o   Convert input to standard sRGB JPEG:

		   libplacebo=format=yuv420p:colorspace=bt470bg:color_primaries=bt709:color_trc=iec61966-2-1:range=pc

       o   Use higher quality debanding settings:

		   libplacebo=deband=true:deband_iterations=3:deband_radius=8:deband_threshold=6

       o   Run this filter on the CPU, on systems with Mesa installed (and
	   with the most expensive options disabled):

		   ffmpeg ... -init_hw_device vulkan:llvmpipe ... -vf libplacebo=upscaler=none:downscaler=none:peak_detect=false

       o   Suppress CPU-based AV1/H.274 film grain application in the decoder,
	   in favor of doing it with this filter. Note that this is only a
	   gain if the frames are either already on the GPU, or if you're
	   using libplacebo for other purposes, since otherwise the VRAM
	   roundtrip will more than offset any expected speedup.

		   ffmpeg -export_side_data +film_grain ... -vf libplacebo=apply_filmgrain=true

       o   Interop with VAAPI hwdec to avoid round-tripping through RAM:

		   ffmpeg -init_hw_device vulkan -hwaccel vaapi -hwaccel_output_format vaapi ... -vf libplacebo

   libvmaf
       Calculate the VMAF (Video Multi-Method Assessment Fusion) score for a
       reference/distorted pair of input videos.

       The first input is the distorted video, and the second input is the
       reference video.

       The obtained 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".

       The filter has following options:

       model
	   A `|` delimited list of vmaf models. Each model can be configured
	   with a number of parameters.  Default value: "version=vmaf_v0.6.1"

       feature
	   A `|` delimited list of features. Each feature can be configured
	   with a number of parameters.

       log_path
	   Set the file path to be used to store log files.

       log_fmt
	   Set the format of the log file (xml, json, csv, or sub).

       pool
	   Set the pool method to be used for computing vmaf.  Options are
	   "min", "harmonic_mean" or "mean" (default).

       n_threads
	   Set number of threads to be used when initializing libvmaf.
	   Default value: 0, no threads.

       n_subsample
	   Set frame subsampling interval to be used.

       This filter also supports the framesync options.

       Examples

       o   In the examples below, a distorted video distorted.mpg is compared
	   with a reference file reference.mpg.

       o   Basic usage:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf=log_path=output.xml -f null -

       o   Example with multiple models:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='model=version=vmaf_v0.6.1\\:name=vmaf|version=vmaf_v0.6.1neg\\:name=vmaf_neg' -f null -

       o   Example with multiple additional features:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null -

       o   Example with options and different containers:

		   ffmpeg -i distorted.mpg -i reference.mkv -lavfi "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]libvmaf=log_fmt=json:log_path=output.json" -f null -

   libvmaf_cuda
       This is the CUDA variant of the libvmaf filter. It only accepts CUDA
       frames.

       It requires Netflix's vmaf library (libvmaf) as a pre-requisite.  After
       installing the library it can be enabled using: "./configure
       --enable-nonfree --enable-ffnvcodec --enable-libvmaf".

       Examples

       o   Basic usage showing CUVID hardware decoding and CUDA scaling with
	   scale_cuda:

		   ffmpeg \
		       -hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid -i dis.obu \
		       -hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid -i ref.obu \
		       -filter_complex "
			   [0:v]scale_cuda=format=yuv420p[ref]; \
			   [1:v]scale_cuda=format=yuv420p[dis]; \
			   [dis][ref]libvmaf_cuda=log_fmt=json:log_path=output.json
		       " \
		       -f null -

   limitdiff
       Apply limited difference filter using second and optionally third video
       stream.

       The filter accepts the following options:

       threshold
	   Set the threshold to use when allowing certain differences between
	   video streams.  Any absolute difference value lower or exact than
	   this threshold will pick pixel components from first video stream.

       elasticity
	   Set the elasticity of soft thresholding when processing video
	   streams.  This value multiplied with first one sets second
	   threshold.  Any absolute difference value greater or exact than
	   second threshold will pick pixel components from second video
	   stream. For values between those two threshold linear interpolation
	   between first and second video stream will be used.

       reference
	   Enable the reference (third) video stream processing. By default is
	   disabled.  If set, this video stream will be used for calculating
	   absolute difference with first video stream.

       planes
	   Specify which planes will be processed. Defaults to all available.

       Commands

       This filter supports the all above options as commands except option
       reference.

   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.

       Commands

       This filter supports the all above options as commands.

   loop
       Loop video frames.

       The filter accepts the following options:

       loop
	   Set the number of loops. Setting this value to -1 will result in
	   infinite loops.  Default is 0.

       size
	   Set maximal size in number of frames. Default is 0.

       start
	   Set first frame of loop. Default is 0.

       time
	   Set the time of loop start in seconds.  Only used if option named
	   start is set to "-1".

       Examples

       o   Loop single first frame infinitely:

		   loop=loop=-1:size=1:start=0

       o   Loop single first frame 10 times:

		   loop=loop=10:size=1:start=0

       o   Loop 10 first frames 5 times:

		   loop=loop=5:size=10:start=0

   lut1d
       Apply a 1D LUT to an input video.

       The filter accepts the following options:

       file
	   Set the 1D LUT file name.

	   Currently supported formats:

	   cube
	       Iridas

	   csp cineSpace

       interp
	   Select interpolation mode.

	   Available values are:

	   nearest
	       Use values from the nearest defined point.

	   linear
	       Interpolate values using the linear interpolation.

	   cosine
	       Interpolate values using the cosine interpolation.

	   cubic
	       Interpolate values using the cubic interpolation.

	   spline
	       Interpolate values using the spline interpolation.

       Commands

       This filter supports the all above options as commands.

   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

	   csp cineSpace

       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.

	   pyramid
	       Interpolate values using a pyramid.

	   prism
	       Interpolate values using a prism.

       Commands

       This filter supports the "interp" option as commands.

   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
	   0.01.

       softness
	   Set the range of softness. Default value is 0.  Use this to control
	   gradual transition from zero to full transparency.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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/luma 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 "clipval".

       Commands

       This filter supports same commands as options.

       Examples

       o   Negate input video:

		   lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
		   lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

	   The above is the same as:

		   lutrgb="r=negval:g=negval:b=negval"
		   lutyuv="y=negval:u=negval:v=negval"

       o   Negate luma:

		   lutyuv=y=negval

       o   Remove chroma components, turning the video into a graytone image:

		   lutyuv="u=128:v=128"

       o   Apply a luma burning effect:

		   lutyuv="y=2*val"

       o   Remove green and blue components:

		   lutrgb="g=0:b=0"

       o   Set a constant alpha channel value on input:

		   format=rgba,lutrgb=a="maxval-minval/2"

       o   Correct luma gamma by a factor of 0.5:

		   lutyuv=y=gammaval(0.5)

       o   Discard least significant bits of luma:

		   lutyuv=y='bitand(val, 128+64+32)'

       o   Technicolor like effect:

		   lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'

   lut2, tlut2
       The "lut2" filter takes two input streams and outputs one stream.

       The "tlut2" (time lut2) filter takes two consecutive frames from one
       single stream.

       This filter accepts the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
	   component

       d   set output bit depth, only available for "lut2" filter. By default
	   is 0, which means bit depth is automatically picked from first
	   input format.

       The "lut2" filter also supports the framesync options.

       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".

       Commands

       This filter supports the all above options as commands except option
       "d".

       Examples

       o   Highlight differences between two RGB video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

       o   Highlight differences between two YUV video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

       o   Show max difference between two video streams:

		   lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'

   maskedclamp
       Clamp the first input stream with the second input and third input
       stream.

       Returns the value of first stream to be between second input stream -
       "undershoot" and third input stream + "overshoot".

       This filter accepts the following options:

       undershoot
	   Default value is 0.

       overshoot
	   Default value is 0.

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

       Commands

       This filter supports the all above options as commands.

   maskedmax
       Merge the second and third input stream into output stream using
       absolute differences between second input stream and first input stream
       and absolute difference between third input stream and first input
       stream. The picked value will be from second input stream if second
       absolute difference is greater than first one or from third input
       stream otherwise.

       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.

       Commands

       This filter supports the all above options as commands.

   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.

       Commands

       This filter supports the all above options as commands.

   maskedmin
       Merge the second and third input stream into output stream using
       absolute differences between second input stream and first input stream
       and absolute difference between third input stream and first input
       stream. The picked value will be from second input stream if second
       absolute difference is less than first one or from third input stream
       otherwise.

       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.

       Commands

       This filter supports the all above options as commands.

   maskedthreshold
       Pick pixels comparing absolute difference of two video streams with
       fixed threshold.

       If absolute difference between pixel component of first and second
       video stream is equal or lower than user supplied threshold than pixel
       component from first video stream is picked, otherwise pixel component
       from second video stream is picked.

       This filter accepts the following options:

       threshold
	   Set threshold used when picking pixels from absolute difference
	   from two input video streams.

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

       mode
	   Set mode of filter operation. Can be "abs" or "diff".  Default is
	   "abs".

       Commands

       This filter supports the all above options as commands.

   maskfun
       Create mask from input video.

       For example it is useful to create motion masks after "tblend" filter.

       This filter accepts the following options:

       low Set low threshold. Any pixel component lower or exact than this
	   value will be set to 0.

       high
	   Set high threshold. Any pixel component higher than this value will
	   be set to max value allowed for current pixel format.

       planes
	   Set planes to filter, by default all available planes are filtered.

       fill
	   Fill all frame pixels with this value.

       sum Set max average pixel value for frame. If sum of all pixel
	   components is higher that this average, output frame will be
	   completely filled with value set by fill option.  Typically useful
	   for scene changes when used in combination with "tblend" filter.

       Commands

       This filter supports the all above options as commands.

   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.

   median
       Pick median pixel from certain rectangle defined by radius.

       This filter accepts the following options:

       radius
	   Set horizontal radius size. Default value is 1.  Allowed range is
	   integer from 1 to 127.

       planes
	   Set which planes to process. Default is 15, which is all available
	   planes.

       radiusV
	   Set vertical radius size. Default value is 0.  Allowed range is
	   integer from 0 to 127.  If it is 0, value will be picked from
	   horizontal "radius" option.

       percentile
	   Set median percentile. Default value is 0.5.  Default value of 0.5
	   will pick always median values, while 0 will pick minimum values,
	   and 1 maximum values.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

   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".

       map0s
       map1s
       map2s
       map3s
	   Set input to output stream mapping for output Nth plane. Default is
	   0.

       map0p
       map1p
       map2p
       map3p
	   Set input to output plane mapping for output Nth plane. Default is
	   0.

       Examples

       o   Merge three gray video streams of same width and height into single
	   video stream:

		   [a0][a1][a2]mergeplanes=0x001020:yuv444p

       o   Merge 1st yuv444p stream and 2nd gray video stream into yuva444p
	   video stream:

		   [a0][a1]mergeplanes=0x00010210:yuva444p

       o   Swap Y and A plane in yuva444p stream:

		   format=yuva444p,mergeplanes=0x03010200:yuva444p

       o   Swap U and V plane in yuv420p stream:

		   format=yuv420p,mergeplanes=0x000201:yuv420p

       o   Cast a rgb24 clip to yuv444p:

		   format=rgb24,mergeplanes=0x000102:yuv444p

   mestimate
       Estimate and export motion vectors using block matching algorithms.
       Motion vectors are stored in frame side data to be used by other
       filters.

       This filter accepts the following options:

       method
	   Specify the motion estimation method. Accepts one of the following
	   values:

	   esa Exhaustive search algorithm.

	   tss Three step search algorithm.

	   tdls
	       Two dimensional logarithmic search algorithm.

	   ntss
	       New three step search algorithm.

	   fss Four step search algorithm.

	   ds  Diamond search algorithm.

	   hexbs
	       Hexagon-based search algorithm.

	   epzs
	       Enhanced predictive zonal search algorithm.

	   umh Uneven multi-hexagon search algorithm.

	   Default value is esa.

       mb_size
	   Macroblock size. Default 16.

       search_param
	   Search parameter. Default 7.

   midequalizer
       Apply Midway Image Equalization effect using two video streams.

       Midway Image Equalization adjusts a pair of images to have the same
       histogram, while maintaining their dynamics as much as possible. It's
       useful for e.g. matching exposures from a pair of stereo cameras.

       This filter has two inputs and one output, which must be of same pixel
       format, but may be of different sizes. The output of filter is first
       input adjusted with midway histogram of both inputs.

       This filter accepts the following option:

       planes
	   Set which planes to process. Default is 15, which is all available
	   planes.

   minterpolate
       Convert the video to specified frame rate using motion interpolation.

       This filter accepts the following options:

       fps Specify the output frame rate. This can be rational e.g.
	   "60000/1001". Frames are dropped if fps is lower than source fps.
	   Default 60.

       mi_mode
	   Motion interpolation mode. Following values are accepted:

	   dup Duplicate previous or next frame for interpolating new ones.

	   blend
	       Blend source frames. Interpolated frame is mean of previous and
	       next frames.

	   mci Motion compensated interpolation. Following options are
	       effective when this mode is selected:

	       mc_mode
		   Motion compensation mode. Following values are accepted:

		   obmc
		       Overlapped block motion compensation.

		   aobmc
		       Adaptive overlapped block motion compensation. Window
		       weighting coefficients are controlled adaptively
		       according to the reliabilities of the neighboring
		       motion vectors to reduce oversmoothing.

		   Default mode is obmc.

	       me_mode
		   Motion estimation mode. Following values are accepted:

		   bidir
		       Bidirectional motion estimation. Motion vectors are
		       estimated for each source frame in both forward and
		       backward directions.

		   bilat
		       Bilateral motion estimation. Motion vectors are
		       estimated directly for interpolated frame.

		   Default mode is bilat.

	       me  The algorithm to be used for motion estimation. Following
		   values are accepted:

		   esa Exhaustive search algorithm.

		   tss Three step search algorithm.

		   tdls
		       Two dimensional logarithmic search algorithm.

		   ntss
		       New three step search algorithm.

		   fss Four step search algorithm.

		   ds  Diamond search algorithm.

		   hexbs
		       Hexagon-based search algorithm.

		   epzs
		       Enhanced predictive zonal search algorithm.

		   umh Uneven multi-hexagon search algorithm.

		   Default algorithm is epzs.

	       mb_size
		   Macroblock size. Default 16.

	       search_param
		   Motion estimation search parameter. Default 32.

	       vsbmc
		   Enable variable-size block motion compensation. Motion
		   estimation is applied with smaller block sizes at object
		   boundaries in order to make them less blurry. 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 10..

   mix
       Mix several video input streams into one video stream.

       A description of the accepted options follows.

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

       weights
	   Specify weight of each input video stream as sequence.  Each weight
	   is separated by space. If number of weights is smaller than number
	   of frames last specified weight will be used for all remaining
	   unset weights.

       scale
	   Specify scale, if it is set it will be multiplied with sum of each
	   weight multiplied with pixel values to give final destination pixel
	   value. By default scale is auto scaled to sum of weights.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       duration
	   Specify how end of stream is determined.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

       Commands

       This filter supports the following commands:

       weights
       scale
       planes
	   Syntax is same as option with same name.

   monochrome
       Convert video to gray using custom color filter.

       A description of the accepted options follows.

       cb  Set the chroma blue spot. Allowed range is from -1 to 1.  Default
	   value is 0.

       cr  Set the chroma red spot. Allowed range is from -1 to 1.  Default
	   value is 0.

       size
	   Set the color filter size. Allowed range is from .1 to 10.  Default
	   value is 1.

       high
	   Set the highlights strength. Allowed range is from 0 to 1.  Default
	   value is 0.

       Commands

       This filter supports the all above options as commands.

   morpho
       This filter allows to apply main morphological grayscale transforms,
       erode and dilate with arbitrary structures set in second input stream.

       Unlike naive implementation and much slower performance in erosion and
       dilation filters, when speed is critical "morpho" filter should be used
       instead.

       A description of accepted options follows,

       mode
	   Set morphological transform to apply, can be:

	   erode
	   dilate
	   open
	   close
	   gradient
	   tophat
	   blackhat

	   Default is "erode".

       planes
	   Set planes to filter, by default all planes except alpha are
	   filtered.

       structure
	   Set which structure video frames will be processed from second
	   input stream, can be first or all. Default is all.

       The "morpho" filter also supports the framesync options.

       Commands

       This filter supports same commands as options.

   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.

       keep
	   Set the maximum number of consecutive similar frames to ignore
	   before to start dropping them.  If the value is 0, the frame is
	   dropped disregarding the number of previous sequentially similar
	   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.

   msad
       Obtain the MSAD (Mean Sum of Absolute Differences) between two input
       videos.

       This filter takes two input videos.

       Both input videos 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 per component, average, min and max MSAD is printed
       through the logging system.

       The filter stores the calculated MSAD of each frame in frame metadata.

       This filter also supports the framesync options.

       In the below example the input file main.mpg being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi msad -f null -

   multiply
       Multiply first video stream pixels values with second video stream
       pixels values.

       The filter accepts the following options:

       scale
	   Set the scale applied to second video stream. By default is 1.
	   Allowed range is from 0 to 9.

       offset
	   Set the offset applied to second video stream. By default is 0.5.
	   Allowed range is from "-1" to 1.

       planes
	   Specify planes from input video stream that will be processed.  By
	   default all planes are processed.

       Commands

       This filter supports same commands as options.

   negate
       Negate (invert) the input video.

       It accepts the following option:

       components
	   Set components to negate.

	   Available values for components are:

	   y
	   u
	   v
	   a
	   r
	   g
	   b
       negate_alpha
	   With value 1, it negates the alpha component, if present. Default
	   value is 0.

       Commands

       This filter supports same commands as options.

   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. Default is 1.0. Must be in range [1.0,
	   30.0].

       p   Set patch size. Default is 7. Must be odd number in range [0, 99].

       pc  Same as p but for chroma planes.

	   The default value is 0 and means automatic.

       r   Set research size. Default is 15. Must be odd number in range [0,
	   99].

       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, abs
	       weights trained to minimize absolute error

	   s, mse
	       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
	   new2
	   new3

	   Default is "new".

       Commands

       This filter supports same commands as options, excluding weights
       option.

   noformat
       Force libavfilter not to use any of the specified pixel formats for the
       input to the next filter.

       It accepts the following parameters:

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

       Examples

       o   Force libavfilter to use a format different from yuv420p for the
	   input to the vflip filter:

		   noformat=pix_fmts=yuv420p,vflip

       o   Convert the input video to any of the formats not contained in the
	   list:

		   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
	   Set noise seed for specific pixel component or all pixel components
	   in case of all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
	   Set noise strength for specific pixel component or all pixel
	   components in case all_strength. Default value is 0. Allowed range
	   is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
	   Set pixel component flags or set flags for all components if
	   all_flags.  Available values for component flags are:

	   a   averaged temporal noise (smoother)

	   p   mix random noise with a (semi)regular pattern

	   t   temporal noise (noise pattern changes between frames)

	   u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

	       noise=alls=20:allf=t+u

   normalize
       Normalize RGB video (aka histogram stretching, contrast stretching).
       See: https://en.wikipedia.org/wiki/Normalization_(image_processing)

       For each channel of each frame, the filter computes the input range and
       maps it linearly to the user-specified output range. The output range
       defaults to the full dynamic range from pure black to pure white.

       Temporal smoothing can be used on the input range to reduce flickering
       (rapid changes in brightness) caused when small dark or bright objects
       enter or leave the scene. This is similar to the auto-exposure
       (automatic gain control) on a video camera, and, like a video camera,
       it may cause a period of over- or under-exposure of the video.

       The R,G,B channels can be normalized independently, which may cause
       some color shifting, or linked together as a single channel, which
       prevents color shifting. Linked normalization preserves hue.
       Independent normalization does not, so it can be used to remove some
       color casts. Independent and linked normalization can be combined in
       any ratio.

       The normalize filter accepts the following options:

       blackpt
       whitept
	   Colors which define the output range. The minimum input value is
	   mapped to the blackpt. The maximum input value is mapped to the
	   whitept.  The defaults are black and white respectively. Specifying
	   white for blackpt and black for whitept will give color-inverted,
	   normalized video. Shades of grey can be used to reduce the dynamic
	   range (contrast). Specifying saturated colors here can create some
	   interesting effects.

       smoothing
	   The number of previous frames to use for temporal smoothing. The
	   input range of each channel is smoothed using a rolling average
	   over the current frame and the smoothing previous frames. The
	   default is 0 (no temporal smoothing).

       independence
	   Controls the ratio of independent (color shifting) channel
	   normalization to linked (color preserving) normalization. 0.0 is
	   fully linked, 1.0 is fully independent. Defaults to 1.0 (fully
	   independent).

       strength
	   Overall strength of the filter. 1.0 is full strength. 0.0 is a
	   rather expensive no-op. Defaults to 1.0 (full strength).

       Commands

       This filter supports same commands as options, excluding smoothing
       option.	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

       Stretch video contrast to use the full dynamic range, with no temporal
       smoothing; may flicker depending on the source content:

	       normalize=blackpt=black:whitept=white:smoothing=0

       As above, but with 50 frames of temporal smoothing; flicker should be
       reduced, depending on the source content:

	       normalize=blackpt=black:whitept=white:smoothing=50

       As above, but with hue-preserving linked channel normalization:

	       normalize=blackpt=black:whitept=white:smoothing=50:independence=0

       As above, but with half strength:

	       normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5

       Map the darkest input color to red, the brightest input color to cyan:

	       normalize=blackpt=red:whitept=cyan

   null
       Pass the video source unchanged to the output.

   ocr
       Optical Character Recognition

       This filter uses Tesseract for optical character recognition. To enable
       compilation of this filter, you need to configure FFmpeg with
       "--enable-libtesseract".

       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".  The filter exports confidence of recognized words as
       the frame metadata "lavfi.ocr.confidence".

   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 depends 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.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

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

       Examples

       o   Inspect full first row of video frame.

		   oscilloscope=x=0.5:y=0:s=1

       o   Inspect full last row of video frame.

		   oscilloscope=x=0.5:y=1:s=1

       o   Inspect full 5th line of video frame of height 1080.

		   oscilloscope=x=0.5:y=5/1080:s=1

       o   Inspect full last column of video frame.

		   oscilloscope=x=1:y=0.5:s=1:t=1

   overlay
       Overlay one video on top of another.

       It takes two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.

       It accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and y coordinates of the overlaid
	   video on the main video. Default value is "0" for both expressions.
	   In case the expression is invalid, it is set to a huge value
	   (meaning that the overlay will not be displayed within the output
	   visible area).

       eof_action
	   See framesync.

       eval
	   Set when the expressions for x, and y are evaluated.

	   It accepts the following values:

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

	   frame
	       evaluate expressions for each incoming frame

	   Default value is frame.

       shortest
	   See framesync.

       format
	   Set the format for the output video.

	   It accepts the following values:

	   yuv420
	       force YUV 4:2:0 8-bit planar output

	   yuv420p10
	       force YUV 4:2:0 10-bit planar output

	   yuv422
	       force YUV 4:2:2 8-bit planar output

	   yuv422p10
	       force YUV 4:2:2 10-bit planar output

	   yuv444
	       force YUV 4:4:4 8-bit planar output

	   yuv444p10
	       force YUV 4:4:4 10-bit planar output

	   rgb force RGB 8-bit packed output

	   gbrp
	       force RGB 8-bit planar output

	   auto
	       automatically pick format

	   Default value is yuv420.

       repeatlast
	   See framesync.

       alpha
	   Set format of alpha of the overlaid video, it can be straight or
	   premultiplied. Default is straight.

       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;
	   deprecated, do not use

       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, t variables are available only when evaluation is done
       per frame, and will evaluate to NAN when eval is set to init.

       Be aware that frames are taken from each input video in timestamp
       order, hence, if their initial timestamps differ, it is a good idea to
       pass the two inputs through a setpts=PTS-STARTPTS filter to have them
       begin in the same zero timestamp, as the example for the movie filter
       does.

       You can chain together more overlays but you should test the efficiency
       of such approach.

       Commands

       This filter supports the following commands:

       x
       y   Modify the x and y of the overlay input.  The command accepts the
	   same syntax of the corresponding option.

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

       Examples

       o   Draw the overlay at 10 pixels from the bottom right corner of the
	   main video:

		   overlay=main_w-overlay_w-10:main_h-overlay_h-10

	   Using named options the example above becomes:

		   overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

       o   Insert a transparent PNG logo in the bottom left corner of the
	   input, using the ffmpeg tool with the "-filter_complex" option:

		   ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

       o   Insert 2 different transparent PNG logos (second logo on bottom
	   right corner) using the ffmpeg tool:

		   ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

       o   Add a transparent color layer on top of the main video; "WxH" must
	   specify the size of the main input to the overlay filter:

		   color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

       o   Play an original video and a filtered version (here with the
	   deshake filter) side by side using the ffplay tool:

		   ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

	   The above command is the same as:

		   ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

       o   Make a sliding overlay appearing from the left to the right top
	   part of the screen starting since time 2:

		   overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

       o   Compose output by putting two input videos side to side:

		   ffmpeg -i left.avi -i right.avi -filter_complex "
		   nullsrc=size=200x100 [background];
		   [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
		   [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
		   [background][left]	    overlay=shortest=1	     [background+left];
		   [background+left][right] overlay=shortest=1:x=100 [left+right]
		   "

       o   Mask 10-20 seconds of a video by applying the delogo filter to a
	   section

		   ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
		   -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
		   masked.avi

       o   Chain several overlays in cascade:

		   nullsrc=s=200x200 [bg];
		   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
		   [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
		   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
		   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
		   [in3] null,	     [mid2] overlay=100:100 [out0]

   overlay_cuda
       Overlay one video on top of another.

       This is the CUDA variant of the overlay filter.	It only accepts CUDA
       frames. The underlying input pixel formats have to match.

       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:

       x
       y   Set expressions for the x and y coordinates of the overlaid video
	   on the main video.

	   They 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.

	   n   The ordinal index of the main input frame, starting from 0.

	   pos The byte offset position in the file of the main input frame,
	       NAN if unknown.	Deprecated, do not use.

	   t   The timestamp of the main input frame, expressed in seconds,
	       NAN if unknown.

	   Default value is "0" for both expressions.

       eval
	   Set when the expressions for x and y are evaluated.

	   It accepts the following values:

	   init
	       Evaluate expressions once during filter initialization or when
	       a command is processed.

	   frame
	       Evaluate expressions for each incoming frame

	   Default value is frame.

       eof_action
	   See framesync.

       shortest
	   See framesync.

       repeatlast
	   See framesync.

       This filter also supports the framesync options.

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
	   Set depth.

	   Larger depth values will denoise lower frequency components more,
	   but slow down filtering.

	   Must be an int in the range 8-16, default is 8.

       luma_strength, ls
	   Set luma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
	   Set chroma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add paddings to the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following parameters:

       width, w
       height, h
	   Specify an expression for the size of the output image with the
	   paddings added. If the value for width or height is 0, the
	   corresponding input size is used for the output.

	   The width expression can reference the value set by the height
	   expression, and vice versa.

	   The default value of width and height is 0.

       x
       y   Specify the offsets to place the input image at within the padded
	   area, with respect to the top/left border of the output image.

	   The x expression can reference the value set by the y expression,
	   and vice versa.

	   The default value of x and y is 0.

	   If x or y evaluate to a negative number, they'll be changed so the
	   input image is centered on the padded area.

       color
	   Specify the color of the padded area. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of color is "black".

       eval
	   Specify when to evaluate  width, height, x and y expression.

	   It accepts the following values:

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

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

       aspect
	   Pad to aspect instead to a resolution.

       The value for the width, height, x, and y options are expressions
       containing the following constants:

       in_w
       in_h
	   The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output width and height (the size of the padded area), as
	   specified by the width and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN
	   if not yet specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
	   The horizontal and vertical chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       o   Add paddings with the color "violet" to the input video. The output
	   video size is 640x480, and the top-left corner of the input video
	   is placed at column 0, row 40

		   pad=640:480:0:40:violet

	   The example above is equivalent to the following command:

		   pad=width=640:height=480:x=0:y=40:color=violet

       o   Pad the input to get an output with dimensions increased by 3/2,
	   and put the input video at the center of the padded area:

		   pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

       o   Pad the input to get a squared output with size equal to the
	   maximum value between the input width and height, and put the input
	   video at the center of the padded area:

		   pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

       o   Pad the input to get a final w/h ratio of 16:9:

		   pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

       o   In case of anamorphic video, in order to set the output display
	   aspect correctly, it is necessary to use sar in the expression,
	   according to the relation:

		   (ih * X / ih) * sar = output_dar
		   X = output_dar / sar

	   Thus the previous example needs to be modified to:

		   pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

       o   Double the output size and put the input video in the bottom-right
	   corner of the output padded area:

		   pad="2*iw:2*ih:ow-iw:oh-ih"

   palettegen
       Generate one palette for a whole video stream.

       It accepts the following options:

       max_colors
	   Set the maximum number of colors to quantize in the palette.  Note:
	   the palette will still contain 256 colors; the unused palette
	   entries will be black.

       reserve_transparent
	   Create a palette of 255 colors maximum and reserve the last one for
	   transparency. Reserving the transparency color is useful for GIF
	   optimization.  If not set, the maximum of colors in the palette
	   will be 256. You probably want to disable this option for a
	   standalone image.  Set by default.

       transparency_color
	   Set the color that will be used as background for transparency.

       stats_mode
	   Set statistics mode.

	   It accepts the following values:

	   full
	       Compute full frame histograms.

	   diff
	       Compute histograms only for the part that differs from previous
	       frame. This might be relevant to give more importance to the
	       moving part of your input if the background is static.

	   single
	       Compute new histogram for each frame.

	   Default value is full.

       The filter also exports the frame metadata "lavfi.color_quant_ratio"
       ("nb_color_in / nb_color_out") which you can use to evaluate the degree
       of color quantization of the palette. This information is also visible
       at info logging level.

       Examples

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

		   ffmpeg -i input.mkv -vf palettegen palette.png

   paletteuse
       Use a palette to downsample an input video stream.

       The filter takes two inputs: one video stream and a palette. The
       palette must be a 256 pixels image.

       It accepts the following options:

       dither
	   Select dithering mode. Available algorithms are:

	   bayer
	       Ordered 8x8 bayer dithering (deterministic)

	   heckbert
	       Dithering as defined by Paul Heckbert in 1982 (simple error
	       diffusion).  Note: this dithering is sometimes considered
	       "wrong" and is included as a reference.

	   floyd_steinberg
	       Floyd and Steingberg dithering (error diffusion)

	   sierra2
	       Frankie Sierra dithering v2 (error diffusion)

	   sierra2_4a
	       Frankie Sierra dithering v2 "Lite" (error diffusion)

	   sierra3
	       Frankie Sierra dithering v3 (error diffusion)

	   burkes
	       Burkes dithering (error diffusion)

	   atkinson
	       Atkinson dithering by Bill Atkinson at Apple Computer (error
	       diffusion)

	   none
	       Disable dithering.

	   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.

       alpha_threshold
	   Sets the alpha threshold for transparency. Alpha values above this
	   threshold will be treated as completely opaque, and values below
	   this threshold will be treated as completely transparent.

	   The option must be an integer value in the range [0,255]. Default
	   is 128.

       Examples

       o   Use a palette (generated for example with palettegen) to encode a
	   GIF using ffmpeg:

		   ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

   perspective
       Correct perspective of video not recorded perpendicular to the screen.

       A description of the accepted parameters follows.

       x0
       y0
       x1
       y1
       x2
       y2
       x3
       y3  Set coordinates expression for top left, top right, bottom left and
	   bottom right corners.  Default values are "0:0:W:0:0:H:W:H" with
	   which perspective will remain unchanged.  If the "sense" option is
	   set to "source", then the specified points will be sent to the
	   corners of the destination. If the "sense" option is set to
	   "destination", then the corners of the source will be sent to the
	   specified coordinates.

	   The expressions can use the following variables:

	   W
	   H   the width and height of video frame.

	   in  Input frame count.

	   on  Output frame count.

       interpolation
	   Set interpolation for perspective correction.

	   It accepts the following values:

	   linear
	   cubic

	   Default value is linear.

       sense
	   Set interpretation of coordinate options.

	   It accepts the following values:

	   0, source
	       Send point in the source specified by the given coordinates to
	       the corners of the destination.

	   1, destination
	       Send the corners of the source to the point in the destination
	       specified by the given coordinates.

	       Default value is source.

       eval
	   Set when the expressions for coordinates x0,y0,...x3,y3 are
	   evaluated.

	   It accepts the following values:

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

	   frame
	       evaluate expressions for each incoming frame

	   Default value is init.

   phase
       Delay interlaced video by one field time so that the field order
       changes.

       The intended use is to fix PAL movies that have been captured with the
       opposite field order to the film-to-video transfer.

       A description of the accepted parameters follows.

       mode
	   Set phase mode.

	   It accepts the following values:

	   t   Capture field order top-first, transfer bottom-first.  Filter
	       will delay the bottom field.

	   b   Capture field order bottom-first, transfer top-first.  Filter
	       will delay the top field.

	   p   Capture and transfer with the same field order. This mode only
	       exists for the documentation of the other options to refer to,
	       but if you actually select it, the filter will faithfully do
	       nothing.

	   a   Capture field order determined automatically by field flags,
	       transfer opposite.  Filter selects among t and b modes on a
	       frame by frame basis using field flags. If no field information
	       is available, then this works just like u.

	   u   Capture unknown or varying, transfer opposite.  Filter selects
	       among t and b on a frame by frame basis by analyzing the images
	       and selecting the alternative that produces best match between
	       the fields.

	   T   Capture top-first, transfer unknown or varying.	Filter selects
	       among t and p using image analysis.

	   B   Capture bottom-first, transfer unknown or varying.  Filter
	       selects among b and p using image analysis.

	   A   Capture determined by field flags, transfer unknown or varying.
	       Filter selects among t, b and p using field flags and image
	       analysis. If no field information is available, then this works
	       just like U. This is the default mode.

	   U   Both capture and transfer unknown or varying.  Filter selects
	       among t, b and p using image analysis only.

       Commands

       This filter supports the all above options as commands.

   photosensitivity
       Reduce various flashes in video, so to help users with epilepsy.

       It accepts the following options:

       frames, f
	   Set how many frames to use when filtering. Default is 30.

       threshold, t
	   Set detection threshold factor. Default is 1.  Lower is stricter.

       skip
	   Set how many pixels to skip when sampling frames. Default is 1.
	   Allowed range is from 1 to 1024.

       bypass
	   Leave frames unchanged. Default is disabled.

   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.

   pixelize
       Apply pixelization to video stream.

       The filter accepts the following options:

       width, w
       height, h
	   Set block dimensions that will be used for pixelization.  Default
	   value is 16.

       mode, m
	   Set the mode of pixelization used.

	   Possible values are:

	   avg
	   min
	   max

	   Default value is "avg".

       planes, p
	   Set what planes to filter. Default is to filter all planes.

       Commands

       This filter supports all options as commands.

   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.

       Commands

       This filter supports same commands as options.

   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 luma filtering only (no chrominance).

       n/noluma
	   Do chrominance filtering only (no luma).

       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 luma to "0-255".

       lb/linblenddeint
	   Linear blend deinterlacing filter that deinterlaces the given block
	   by filtering all lines with a "(1 2 1)" filter.

       li/linipoldeint
	   Linear interpolating deinterlacing filter that deinterlaces the
	   given block by linearly interpolating every second line.

       ci/cubicipoldeint
	   Cubic interpolating deinterlacing filter deinterlaces the given
	   block by cubically interpolating every second line.

       md/mediandeint
	   Median deinterlacing filter that deinterlaces the given block by
	   applying a median filter to every second line.

       fd/ffmpegdeint
	   FFmpeg deinterlacing filter that deinterlaces the given block by
	   filtering every second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
	   Vertically applied FIR lowpass deinterlacing filter that
	   deinterlaces the given block by filtering all lines with a "(-1 2 6
	   2 -1)" filter.

       fq/forceQuant[|quantizer]
	   Overrides the quantizer table from the input with the constant
	   quantizer you specify.

	   quantizer
	       Quantizer to use

       de/default
	   Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
	   Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       o   Apply horizontal and vertical deblocking, deringing and automatic
	   brightness/contrast:

		   pp=hb/vb/dr/al

       o   Apply default filters without brightness/contrast correction:

		   pp=de/-al

       o   Apply default filters and temporal denoiser:

		   pp=default/tmpnoise|1|2|3

       o   Apply deblocking on luma 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.

       Commands

       This filter supports the all above options as commands.

   pseudocolor
       Alter frame colors in video with pseudocolors.

       This filter accepts 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

       index, i
	   set component to use as base for altering colors

       preset, p
	   Pick one of built-in LUTs. By default is set to none.

	   Available LUTs:

	   magma
	   inferno
	   plasma
	   viridis
	   turbo
	   cividis
	   range1
	   range2
	   shadows
	   highlights
	   solar
	   nominal
	   preferred
	   total
	   spectral
	   cool
	   heat
	   fiery
	   blues
	   green
	   helix
       opacity
	   Set opacity of output colors. Allowed range is from 0 to 1.
	   Default value is set to 1.

       Each of the expression options 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".

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Change too high luma values to gradient:

		   pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"

   psnr
       Obtain the average, maximum and minimum PSNR (Peak Signal to Noise
       Ratio) between two input videos.

       This filter takes in input two input videos, the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the PSNR.

       Both video inputs must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average PSNR is printed through the logging system.

       The filter stores the accumulated MSE (mean squared error) of each
       frame, and at the end of the processing it is averaged across all
       frames equally, and the following formula is applied to obtain the
       PSNR:

	       PSNR = 10*log10(MAX^2/MSE)

       Where MAX is the average of the maximum values of each component of the
       image.

       The description of the accepted parameters follows.

       stats_file, f
	   If specified the filter will use the named file to save the PSNR of
	   each individual frame. When filename equals "-" the data is sent to
	   standard output.

       stats_version
	   Specifies which version of the stats file format to use. Details of
	   each format are written below.  Default value is 1.

       stats_add_max
	   Determines whether the max value is output to the stats log.
	   Default value is 0.	Requires stats_version >= 2. If this is set
	   and stats_version < 2, the filter will return an error.

       This filter also supports the framesync options.

       The file printed if stats_file is selected, contains a sequence of
       key/value pairs of the form key:value for each compared couple of
       frames.

       If a stats_version greater than 1 is specified, a header line precedes
       the list of per-frame-pair stats, with key value pairs following the
       frame format with the following parameters:

       psnr_log_version
	   The version of the log file format. Will match stats_version.

       fields
	   A comma separated list of the per-frame-pair parameters included in
	   the log.

       A description of each shown per-frame-pair parameter follows:

       n   sequential number of the input frame, starting from 1

       mse_avg
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames, averaged over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_b, 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.

       Examples

       o   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.

       o   Another example with different containers:

		   ffmpeg -i main.mpg -i ref.mkv -lavfi  "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]psnr" -f null -

   pullup
       Pulldown reversal (inverse telecine) filter, capable of handling mixed
       hard-telecine, 24000/1001 fps progressive, and 30000/1001 fps
       progressive content.

       The pullup filter is designed to take advantage of future context in
       making its decisions. This filter is stateless in the sense that it
       does not lock onto a pattern to follow, but it instead looks forward to
       the following fields in order to identify matches and rebuild
       progressive frames.

       To produce content with an even framerate, insert the fps filter after
       pullup, use "fps=24000/1001" if the input frame rate is 29.97fps,
       "fps=24" for 30fps and the (rare) telecined 25fps input.

       The filter accepts the following options:

       jl
       jr
       jt
       jb  These options set the amount of "junk" to ignore at the left,
	   right, top, and bottom of the image, respectively. Left and right
	   are in units of 8 pixels, while top and bottom are in units of 2
	   lines.  The default is 8 pixels on each side.

       sb  Set the strict breaks. Setting this option to 1 will reduce the
	   chances of filter generating an occasional mismatched frame, but it
	   may also cause an excessive number of frames to be dropped during
	   high motion sequences.  Conversely, setting it to -1 will make
	   filter match fields more easily.  This may help processing of video
	   where there is slight blurring between the fields, but may also
	   cause there to be interlaced frames in the output.  Default value
	   is 0.

       mp  Set the metric plane to use. It accepts the following values:

	   l   Use luma plane.

	   u   Use chroma blue plane.

	   v   Use chroma red plane.

	   This option may be set to use chroma plane instead of the default
	   luma plane for doing filter's computations. This may improve
	   accuracy on very clean source material, but more likely will
	   decrease accuracy, especially if there is chroma noise (rainbow
	   effect) or any grayscale video.  The main purpose of setting mp to
	   a chroma plane is to reduce CPU load and make pullup usable in
	   realtime on slow machines.

       For best results (without duplicated frames in the output file) it is
       necessary to change the output frame rate. For example, to inverse
       telecine NTSC input:

	       ffmpeg -i input -vf pullup -r 24000/1001 ...

   qp
       Change video quantization parameters (QP).

       The filter accepts the following option:

       qp  Set expression for quantization parameter.

       The expression is evaluated through the eval API and can contain, among
       others, the following constants:

       known
	   1 if index is not 129, 0 otherwise.

       qp  Sequential index starting from -129 to 128.

       Examples

       o   Some equation like:

		   qp=2+2*sin(PI*qp)

   qrencode
       Generate a QR code using the libqrencode library (see
       <https://fukuchi.org/works/qrencode/>), and overlay it on top of the
       current frame.

       To enable the compilation of this filter, you need to configure FFmpeg
       with "--enable-libqrencode".

       The QR code is generated from the provided text or text pattern. The
       corresponding QR code is scaled and overlayed into the video output
       according to the specified options.

       In case no text is specified, no QR code is overlaied.

       This filter accepts the following options:

       qrcode_width, q
       padded_qrcode_width, Q
	   Specify an expression for the width of the rendered QR code, with
	   and without padding. The qrcode_width expression can reference the
	   value set by the padded_qrcode_width expression, and vice versa.
	   By default padded_qrcode_width is set to qrcode_width, meaning that
	   there is no padding.

	   These expressions are evaluated for each new frame.

	   See the qrencode Expressions section for details.

       x
       y   Specify an expression for positioning the padded QR code top-left
	   corner.  The x expression can reference the value set by the y
	   expression, and vice.

	   By default x and y are set set to 0, meaning that the QR code is
	   placed in the top left corner of the input.

	   These expressions are evaluated for each new frame.

	   See the qrencode Expressions section for details.

       case_sensitive, cs
	   Instruct libqrencode to use case sensitive encoding. This is
	   enabled by default. This can be disabled to reduce the QR encoding
	   size.

       level, l
	   Specify the QR encoding error correction level. With an higher
	   correction level, the encoding size will increase but the code will
	   be more robust to corruption.  Lower level is L.

	   It accepts the following values:

	   L
	   M
	   Q
	   H
       expansion
	   Select how the input text is expanded. Can be either "none", or
	   "normal" (default). See the qrencode Text expansion section below
	   for details.

       text
       textfile
	   Define the text to be rendered. In case neither is specified, no QR
	   is encoded (just an empty colored frame).

	   In case expansion is enabled, the text is treated as a text
	   template, using the qrencode expansion mechanism. See the qrencode
	   Text expansion section below for details.

       background_color, bc
       foreground_color, fc
	   Set the QR code and background color. The default value of
	   foreground_color is "black", the default value of background_color
	   is "white".

	   For the syntax of the color options, check the "Color" section in
	   the ffmpeg-utils manual.

       qrencode Expressions

       The expressions set by the options contain the following constants and
       functions.

       dar input display aspect ratio, it is the same as (w / h) * sar

       duration
	   the current frame's duration, in seconds

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       main_h, H
	   the input height

       main_w, W
	   the input width

       n   the number of input frame, starting from 0

       pict_type
	   a number representing the picture type

       qr_w, w
	   the width of the encoded QR code

       rendered_qr_w, q
       rendered_padded_qr_w, Q
	   the width of the rendered QR code, without and without padding.

	   These parameters allow the q and Q expressions to refer to each
	   other, so you can for example specify "q=3/4*Q".

       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

       x
       y   the x and y offset coordinates where the text is drawn.

	   These parameters allow the x and y expressions to refer to each
	   other, so you can for example specify "y=x/dar".

       qrencode Text expansion

       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 with the textfile option
       avoids these problems.

       The following functions are available:

       n, frame_num
	   return the frame number

       pts Return the presentation timestamp of the current frame.

	   It can take up to two 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. If the format is
	   set to "hms24hh", the time is formatted in 24h format (00-23).

	   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" C function format string. By default,
	   YYYY-MM-DD HH:MM:SS format will be used.

       expr, e
	   Evaluate the expression's value and output as a double.

	   It must take one argument specifying the expression to be
	   evaluated, accepting the constants and functions defined in
	   qrencode_expressions.

       expr_formatted, ef
	   Evaluate the expression's value and output as a formatted string.

	   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" C function format string.  The
	   format string is extended to support the variable %[1-6]N which
	   prints fractions of the second with optionally specified number of
	   digits.

       localtime
	   The time at which the filter is running, expressed in the local
	   time zone.  It can accept an argument: a "strftime" C function
	   format string.  The format string is extended to support the
	   variable %[1-6]N which prints fractions of the second with
	   optionally specified number of digits.

       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.

	   Available metadata can be identified by inspecting entries starting
	   with TAG included within each frame section printed by running
	   "ffprobe -show_frames".

	   String metadata generated in filters leading to the qrencode filter
	   are also available.

       rand(min, max)
	   return a random number included between min and max

       Examples

       o   Generate a QR code encoding the specified text with the default
	   size, overalaid in the top left corner of the input video, with the
	   default size:

		   qrencode=text=www.ffmpeg.org

       o   Same as below, but select blue on pink colors:

		   qrencode=text=www.ffmpeg.org:bc=pink@0.5:fc=blue

       o   Place the QR code in the bottom right corner of the input video:

		   qrencode=text=www.ffmpeg.org:x=W-Q:y=H-Q

       o   Generate a QR code with width of 200 pixels and padding, making the
	   padded width 4/3 of the QR code width:

		   qrencode=text=www.ffmpeg.org:q=200:Q=4/3*q

       o   Generate a QR code with padded width of 200 pixels and padding,
	   making the QR code width 3/4 of the padded width:

		   qrencode=text=www.ffmpeg.org:Q=200:q=3/4*Q

       o   Make the QR code a fraction of the input video width:

		   qrencode=text=www.ffmpeg.org:q=W/5

       o   Generate a QR code encoding the frame number:

		   qrencode=text=%{n}

       o   Generate a QR code encoding the GMT timestamp:

		   qrencode=text=%{gmtime}

       o   Generate a QR code encoding the timestamp expressed as a float:

		   qrencode=text=%{pts}

   quirc
       Identify and decode a QR code using the libquirc library (see
       <https://github.com/dlbeer/quirc/>), and print the identified QR codes
       positions and payload as metadata.

       To enable the compilation of this filter, you need to configure FFmpeg
       with "--enable-libquirc".

       For each found QR code in the input video, some metadata entries are
       added with the prefix lavfi.quirc.N, where N is the index, starting
       from 0, associated to the QR code.

       A description of each metadata value follows:

       lavfi.quirc.count
	   the number of found QR codes, it is not set in case none was found

       lavfi.quirc.N.corner.M.x
       lavfi.quirc.N.coreer.M.y
	   the x/y positions of the four corners of the square containing the
	   QR code, where M is the index of the corner starting from 0

       lavfi.quirc.N.payload
	   the payload of the QR code

   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.

       spw Set the ratio of width reserved for sync code detection.  Default
	   is 0.27. Allowed range is "[0.1 - 0.7]".

       chp Enable checking the parity bit. In the event of a parity error, the
	   filter will output 0x00 for that character. Default is false.

       lp  Lowpass lines prior to further processing. Default is enabled.

       Commands

       This filter supports the all above options as commands.

       Examples

       o   Output a csv with presentation time and the first two lines of
	   identified EIA-608 captioning data.

		   ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv

   readvitc
       Read vertical interval timecode (VITC) information from the top lines
       of a video frame.

       The filter adds frame metadata key "lavfi.readvitc.tc_str" with the
       timecode value, if a valid timecode has been detected. Further metadata
       key "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
       data has been found or not.

       This filter accepts the following options:

       scan_max
	   Set the maximum number of lines to scan for VITC data. If the value
	   is set to "-1" the full video frame is scanned. Default is 45.

       thr_b
	   Set the luma threshold for black. Accepts float numbers in the
	   range [0.0,1.0], default value is 0.2. The value must be equal or
	   less than "thr_w".

       thr_w
	   Set the luma threshold for white. Accepts float numbers in the
	   range [0.0,1.0], default value is 0.6. The value must be equal or
	   greater than "thr_b".

       Examples

       o   Detect and draw VITC data onto the video frame; if no valid VITC is
	   detected, draw "--:--:--:--" as a placeholder:

		   ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

   remap
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x, y)
       position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
       out of range, zero value for pixel will be used for destination pixel.

       Xmap and Ymap input video streams must be of same dimensions. Output
       video stream will have Xmap/Ymap video stream dimensions.  Xmap and
       Ymap input video streams are 16bit depth, single channel.

       format
	   Specify pixel format of output from this filter. Can be "color" or
	   "gray".  Default is "color".

       fill
	   Specify the color of the unmapped pixels. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.
	   Default color is "black".

   removegrain
       The removegrain filter is a spatial denoiser for progressive video.

       m0  Set mode for the first plane.

       m1  Set mode for the second plane.

       m2  Set mode for the third plane.

       m3  Set mode for the fourth plane.

       Range of mode is from 0 to 24. Description of each mode follows:

       0   Leave input plane unchanged. Default.

       1   Clips the pixel with the minimum and maximum of the 8 neighbour
	   pixels.

       2   Clips the pixel with the second minimum and maximum of the 8
	   neighbour pixels.

       3   Clips the pixel with the third minimum and maximum of the 8
	   neighbour pixels.

       4   Clips the pixel with the fourth minimum and maximum of the 8
	   neighbour pixels.  This is equivalent to a median filter.

       5   Line-sensitive clipping giving the minimal change.

       6   Line-sensitive clipping, intermediate.

       7   Line-sensitive clipping, intermediate.

       8   Line-sensitive clipping, intermediate.

       9   Line-sensitive clipping on a line where the neighbours pixels are
	   the closest.

       10  Replaces the target pixel with the closest neighbour.

       11  [1 2 1] horizontal and vertical kernel blur.

       12  Same as mode 11.

       13  Bob mode, interpolates top field from the line where the neighbours
	   pixels are the closest.

       14  Bob mode, interpolates bottom field from the line where the
	   neighbours pixels are the closest.

       15  Bob mode, interpolates top field. Same as 13 but with a more
	   complicated interpolation formula.

       16  Bob mode, interpolates bottom field. Same as 14 but with a more
	   complicated interpolation formula.

       17  Clips the pixel with the minimum and maximum of respectively the
	   maximum and minimum of each pair of opposite neighbour pixels.

       18  Line-sensitive clipping using opposite neighbours whose greatest
	   distance from the current pixel is minimal.

       19  Replaces the pixel with the average of its 8 neighbours.

       20  Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

       21  Clips pixels using the averages of opposite neighbour.

       22  Same as mode 21 but simpler and faster.

       23  Small edge and halo removal, but reputed useless.

       24  Similar as 23.

   removelogo
       Suppress a TV station logo, using an image file to determine which
       pixels comprise the logo. It works by filling in the pixels that
       comprise the logo with neighboring pixels.

       The filter accepts the following options:

       filename, f
	   Set the filter bitmap file, which can be any image format supported
	   by libavformat. The width and height of the image file must match
	   those of the video stream being processed.

       Pixels in the provided bitmap image with a value of zero are not
       considered part of the logo, non-zero pixels are considered part of the
       logo. If you use white (255) for the logo and black (0) for the rest,
       you will be safe. For making the filter bitmap, it is recommended to
       take a screen capture of a black frame with the logo visible, and then
       using a threshold filter followed by the erode filter once or twice.

       If needed, little splotches can be fixed manually. Remember that if
       logo pixels are not covered, the filter quality will be much reduced.
       Marking too many pixels as part of the logo does not hurt as much, but
       it will increase the amount of blurring needed to cover over the image
       and will destroy more information than necessary, and extra pixels will
       slow things down on a large logo.

   repeatfields
       This filter uses the repeat_field flag from the Video ES headers and
       hard repeats fields based on its value.

   reverse
       Reverse a video clip.

       Warning: This filter requires memory to buffer the entire clip, so
       trimming is suggested.

       Examples

       o   Take the first 5 seconds of a clip, and reverse it.

		   trim=end=5,reverse

   rgbashift
       Shift R/G/B/A pixels horizontally and/or vertically.

       The filter accepts the following options:

       rh  Set amount to shift red horizontally.

       rv  Set amount to shift red vertically.

       gh  Set amount to shift green horizontally.

       gv  Set amount to shift green vertically.

       bh  Set amount to shift blue horizontally.

       bv  Set amount to shift blue vertically.

       ah  Set amount to shift alpha horizontally.

       av  Set amount to shift alpha vertically.

       edge
	   Set edge mode, can be smear, default, or warp.

       Commands

       This filter supports the all above options as commands.

   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.

       Commands

       This filter supports the all above options as commands.

   rotate
       Rotate video by an arbitrary angle expressed in radians.

       The filter accepts the following options:

       A description of the optional parameters follows.

       angle, a
	   Set an expression for the angle by which to rotate the input video
	   clockwise, expressed as a number of radians. A negative value will
	   result in a counter-clockwise rotation. By default it is set to
	   "0".

	   This expression is evaluated for each frame.

       out_w, ow
	   Set the output width expression, default value is "iw".  This
	   expression is evaluated just once during configuration.

       out_h, oh
	   Set the output height expression, default value is "ih".  This
	   expression is evaluated just once during configuration.

       bilinear
	   Enable bilinear interpolation if set to 1, a value of 0 disables
	   it. Default value is 1.

       fillcolor, c
	   Set the color used to fill the output area not covered by the
	   rotated image. For the general syntax of this option, check the
	   "Color" section in the ffmpeg-utils manual.	If the special value
	   "none" is selected then no background is printed (useful for
	   example if the background is never shown).

	   Default value is "black".

       The expressions for the angle and the output size can contain the
       following constants and functions:

       n   sequential number of the input frame, starting from 0. It is always
	   NAN before the first frame is filtered.

       t   time in seconds of the input frame, it is set to 0 when the filter
	   is configured. It is always NAN before the first frame is filtered.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_w, iw
       in_h, ih
	   the input video width and height

       out_w, ow
       out_h, oh
	   the output width and height, that is the size of the padded area as
	   specified by the width and height expressions

       rotw(a)
       roth(a)
	   the minimal width/height required for completely containing the
	   input video rotated by a radians.

	   These are only available when computing the out_w and out_h
	   expressions.

       Examples

       o   Rotate the input by PI/6 radians clockwise:

		   rotate=PI/6

       o   Rotate the input by PI/6 radians counter-clockwise:

		   rotate=-PI/6

       o   Rotate the input by 45 degrees clockwise:

		   rotate=45*PI/180

       o   Apply a constant rotation with period T, starting from an angle of
	   PI/3:

		   rotate=PI/3+2*PI*t/T

       o   Make the input video rotation oscillating with a period of T
	   seconds and an amplitude of A radians:

		   rotate=A*sin(2*PI/T*t)

       o   Rotate the video, output size is chosen so that the whole rotating
	   input video is always completely contained in the output:

		   rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

       o   Rotate the video, reduce the output size so that no background is
	   ever shown:

		   rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

       Commands

       The filter supports the following commands:

       a, angle
	   Set the angle expression.  The command accepts the same syntax of
	   the corresponding option.

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

   sab
       Apply Shape Adaptive Blur.

       The filter accepts the following options:

       luma_radius, lr
	   Set luma blur filter strength, must be a value in range 0.1-4.0,
	   default value is 1.0. A greater value will result in a more blurred
	   image, and in slower processing.

       luma_pre_filter_radius, lpfr
	   Set luma pre-filter radius, must be a value in the 0.1-2.0 range,
	   default value is 1.0.

       luma_strength, ls
	   Set luma maximum difference between pixels to still be considered,
	   must be a value in the 0.1-100.0 range, default value is 1.0.

       chroma_radius, cr
	   Set chroma blur filter strength, must be a value in range -0.9-4.0.
	   A greater value will result in a more blurred image, and in slower
	   processing.

       chroma_pre_filter_radius, cpfr
	   Set chroma pre-filter radius, must be a value in the -0.9-2.0
	   range.

       chroma_strength, cs
	   Set chroma maximum difference between pixels to still be
	   considered, must be a value in the -0.9-100.0 range.

       Each chroma option value, if not explicitly specified, is set to the
       corresponding luma option value.

   scale
       Scale (resize) the input video, using the libswscale library.

       The scale filter forces the output display aspect ratio to be the same
       of the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the scale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options, any of the options supported
       by the libswscale scaler, as well as any of the framesync options.

       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
	   bt470
	   smpte170m
	       Set color space conforming to:

	       o   ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

	       o   ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

	       o   Society of Motion Picture and Television Engineers (SMPTE)
		   ST 170:2004

	   smpte240m
	       Set color space conforming to SMPTE ST 240:1999.

	   bt2020
	       Set color space conforming to ITU-R BT.2020 non-constant
	       luminance system.

       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/unknown
	       Choose automatically.

	   jpeg/full/pc
	       Set full range (0-255 in case of 8-bit luma).

	   mpeg/limited/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.

       force_divisible_by
	   Ensures that both the output dimensions, width and height, are
	   divisible by the given integer when used together with
	   force_original_aspect_ratio. This works similar to using "-n" in
	   the w and h options.

	   This option respects the value set for force_original_aspect_ratio,
	   increasing or decreasing the resolution accordingly. The video's
	   aspect ratio may be slightly modified.

	   This option can be handy if you need to have a video fit within or
	   exceed a defined resolution using force_original_aspect_ratio but
	   also have encoder restrictions on width or height divisibility.

       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.

       n   The (sequential) number of the input frame, starting from 0.  Only
	   available with "eval=frame".

       t   The presentation timestamp of the input frame, expressed as a
	   number of seconds. Only available with "eval=frame".

       pos The position (byte offset) of the frame in the input stream, or NaN
	   if this information is unavailable and/or meaningless (for example
	   in case of synthetic video).  Only available with "eval=frame".
	   Deprecated, do not use.

       ref_w, rw
       ref_h, rh
       ref_a
       ref_dar, rdar
       ref_n
       ref_t
       ref_pos
	   Eqvuialent to the above, but for a second reference input. If any
	   of these variables are present, this filter accepts two inputs.

       Examples

       o   Scale the input video to a size of 200x100

		   scale=w=200:h=100

	   This is equivalent to:

		   scale=200:100

	   or:

		   scale=200x100

       o   Specify a size abbreviation for the output size:

		   scale=qcif

	   which can also be written as:

		   scale=size=qcif

       o   Scale the input to 2x:

		   scale=w=2*iw:h=2*ih

       o   The above is the same as:

		   scale=2*in_w:2*in_h

       o   Scale the input to 2x with forced interlaced scaling:

		   scale=2*iw:2*ih:interl=1

       o   Scale the input to half size:

		   scale=w=iw/2:h=ih/2

       o   Increase the width, and set the height to the same size:

		   scale=3/2*iw:ow

       o   Seek Greek harmony:

		   scale=iw:1/PHI*iw
		   scale=ih*PHI:ih

       o   Increase the height, and set the width to 3/2 of the height:

		   scale=w=3/2*oh:h=3/5*ih

       o   Increase the size, making the size a multiple of the chroma
	   subsample values:

		   scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

       o   Increase the width to a maximum of 500 pixels, keeping the same
	   aspect ratio as the input:

		   scale=w='min(500\, iw*3/2):h=-1'

       o   Make pixels square by combining scale and setsar:

		   scale='trunc(ih*dar):ih',setsar=1/1

       o   Make pixels square by combining scale and setsar, making sure the
	   resulting resolution is even (required by some codecs):

		   scale='trunc(ih*dar/2)*2:trunc(ih/2)*2',setsar=1/1

       o   Scale a subtitle stream (sub) to match the main video (main) in
	   size before overlaying. ("scale2ref")

		   '[main]split[a][b]; [ref][a]scale=rw:rh[c]; [b][c]overlay'

       o   Scale a logo to 1/10th the height of a video, while preserving its
	   display aspect ratio.

		   [logo-in][video-in]scale=w=oh*dar:h=rh/10[logo-out]

       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_cuda
       Scale (resize) and convert (pixel format) the input video, using
       accelerated CUDA kernels.  Setting the output width and height works in
       the same way as for the scale filter.

       The filter accepts the following options:

       w
       h   Set the output video dimension expression. Default value is the
	   input dimension.

	   Allows for the same expressions as the scale filter.

       interp_algo
	   Sets the algorithm used for scaling:

	   nearest
	       Nearest neighbour

	       Used by default if input parameters match the desired output.

	   bilinear
	       Bilinear

	   bicubic
	       Bicubic

	       This is the default.

	   lanczos
	       Lanczos

       format
	   Controls the output pixel format. By default, or if none is
	   specified, the input pixel format is used.

	   The filter does not support converting between YUV and RGB pixel
	   formats.

       passthrough
	   If set to 0, every frame is processed, even if no conversion is
	   necessary.  This mode can be useful to use the filter as a buffer
	   for a downstream frame-consumer that exhausts the limited decoder
	   frame pool.

	   If set to 1, frames are passed through as-is if they match the
	   desired output parameters. This is the default behaviour.

       param
	   Algorithm-Specific parameter.

	   Affects the curves of the bicubic algorithm.

       force_original_aspect_ratio
       force_divisible_by
	   Work the same as the identical scale filter options.

       Examples

       o   Scale input to 720p, keeping aspect ratio and ensuring the output
	   is yuv420p.

		   scale_cuda=-2:720:format=yuv420p

       o   Upscale to 4K using nearest neighbour algorithm.

		   scale_cuda=4096:2160:interp_algo=nearest

       o   Don't do any conversion or scaling, but copy all input frames into
	   newly allocated ones.  This can be useful to deal with a filter and
	   encode chain that otherwise exhausts the decoders frame pool.

		   scale_cuda=passthrough=0

   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
       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.

       force_divisible_by
	   Ensures that both the output dimensions, width and height, are
	   divisible by the given integer when used together with
	   force_original_aspect_ratio. This works similar to using "-n" in
	   the w and h options.

	   This option respects the value set for force_original_aspect_ratio,
	   increasing or decreasing the resolution accordingly. The video's
	   aspect ratio may be slightly modified.

	   This option can be handy if you need to have a video fit within or
	   exceed a defined resolution using force_original_aspect_ratio but
	   also have encoder restrictions on width or height divisibility.

       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.

       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".

       n   The (sequential) number of the input frame, starting from 0.  Only
	   available with "eval=frame".

       t   The presentation timestamp of the input frame, expressed as a
	   number of seconds. Only available with "eval=frame".

       pos The position (byte offset) of the frame in the input stream, or NaN
	   if this information is unavailable and/or meaningless (for example
	   in case of synthetic video).  Only available with "eval=frame".
	   Deprecated, do not use.

   scale2ref_npp
       Use the NVIDIA Performance Primitives (libnpp) to scale (resize) the
       input video, based on a reference video.

       See the scale_npp filter for available options, scale2ref_npp supports
       the same but uses the reference video instead of the main input as
       basis. scale2ref_npp 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_n
	   The (sequential) number of the main input frame, starting from 0.
	   Only available with "eval=frame".

       main_t
	   The presentation timestamp of the main input frame, expressed as a
	   number of seconds. Only available with "eval=frame".

       main_pos
	   The position (byte offset) of the frame in the main input stream,
	   or NaN if this information is unavailable and/or meaningless (for
	   example in case of synthetic video).  Only available with
	   "eval=frame".

       Examples

       o   Scale a subtitle stream (b) to match the main video (a) in size
	   before overlaying

		   'scale2ref_npp[b][a];[a][b]overlay_cuda'

       o   Scale a logo to 1/10th the height of a video, while preserving its
	   display aspect ratio.

		   [logo-in][video-in]scale2ref_npp=w=oh*mdar:h=ih/10[logo-out][video-out]

   scale_vt
       Scale and convert the color parameters using VTPixelTransferSession.

       The filter accepts the following options:

       w
       h   Set the output video dimension expression. Default value is the
	   input dimension.

       color_matrix
	   Set the output colorspace matrix.

       color_primaries
	   Set the output color primaries.

       color_transfer
	   Set the output transfer characteristics.

   scharr
       Apply scharr 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.

       Commands

       This filter supports the all above options as commands.

   scroll
       Scroll input video horizontally and/or vertically by constant speed.

       The filter accepts the following options:

       horizontal, h
	   Set the horizontal scrolling speed. Default is 0. Allowed range is
	   from -1 to 1.  Negative values changes scrolling direction.

       vertical, v
	   Set the vertical scrolling speed. Default is 0. Allowed range is
	   from -1 to 1.  Negative values changes scrolling direction.

       hpos
	   Set the initial horizontal scrolling position. Default is 0.
	   Allowed range is from 0 to 1.

       vpos
	   Set the initial vertical scrolling position. Default is 0. Allowed
	   range is from 0 to 1.

       Commands

       This filter supports the following commands:

       horizontal, h
	   Set the horizontal scrolling speed.

       vertical, v
	   Set the vertical scrolling speed.

   scdet
       Detect video scene change.

       This filter sets frame metadata with mafd between frame, the scene
       score, and forward the frame to the next filter, so they can use these
       metadata to detect scene change or others.

       In addition, this filter logs a message and sets frame metadata when it
       detects a scene change by threshold.

       "lavfi.scd.mafd" metadata keys are set with mafd for every frame.

       "lavfi.scd.score" metadata keys are set with scene change score for
       every frame to detect scene change.

       "lavfi.scd.time" metadata keys are set with current filtered frame time
       which detect scene change with threshold.

       The filter accepts the following options:

       threshold, t
	   Set the scene change detection threshold as a percentage of maximum
	   change. Good values are in the "[8.0, 14.0]" range. The range for
	   threshold is "[0., 100.]".

	   Default value is 10..

       sc_pass, s
	   Set the flag to pass scene change frames to the next filter.
	   Default value is 0 You can enable it if you want to get snapshot of
	   scene change frames only.

   selectivecolor
       Adjust cyan, magenta, yellow and black (CMYK) to certain ranges of
       colors (such as "reds", "yellows", "greens", "cyans", ...). The
       adjustment range is defined by the "purity" of the color (that is, how
       saturated it already is).

       This filter is similar to the Adobe Photoshop Selective Color tool.

       The filter accepts the following options:

       correction_method
	   Select color correction method.

	   Available values are:

	   absolute
	       Specified adjustments are applied "as-is" (added/subtracted to
	       original pixel component value).

	   relative
	       Specified adjustments are relative to the original component
	       value.

	   Default is "absolute".

       reds
	   Adjustments for red pixels (pixels where the red component is the
	   maximum)

       yellows
	   Adjustments for yellow pixels (pixels where the blue component is
	   the minimum)

       greens
	   Adjustments for green pixels (pixels where the green component is
	   the maximum)

       cyans
	   Adjustments for cyan pixels (pixels where the red component is the
	   minimum)

       blues
	   Adjustments for blue pixels (pixels where the blue component is the
	   maximum)

       magentas
	   Adjustments for magenta pixels (pixels where the green component is
	   the minimum)

       whites
	   Adjustments for white pixels (pixels where all components are
	   greater than 128)

       neutrals
	   Adjustments for all pixels except pure black and pure white

       blacks
	   Adjustments for black pixels (pixels where all components are
	   lesser than 128)

       psfile
	   Specify a Photoshop selective color file (".asv") to import the
	   settings from.

       All the adjustment settings (reds, yellows, ...) accept up to 4 space
       separated floating point adjustment values in the [-1,1] range,
       respectively to adjust the amount of cyan, magenta, yellow and black
       for the pixels of its range.

       Examples

       o   Increase cyan by 50% and reduce yellow by 33% in every green areas,
	   and increase magenta by 27% in blue areas:

		   selectivecolor=greens=.5 0 -.33 0:blues=0 .27

       o   Use a Photoshop selective color preset:

		   selectivecolor=psfile=MySelectiveColorPresets/Misty.asv

   separatefields
       The "separatefields" takes a frame-based video input and splits each
       frame into its components fields, producing a new half height clip with
       twice the frame rate and twice the frame count.

       This filter use field-dominance information in frame to decide which of
       each pair of fields to place first in the output.  If it gets it wrong
       use setfield filter before "separatefields" filter.

   setdar, setsar
       The "setdar" filter sets the Display Aspect Ratio for the filter output
       video.

       This is done by changing the specified Sample (aka Pixel) Aspect Ratio,
       according to the following equation:

	       <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

       Keep in mind that the "setdar" filter does not modify the pixel
       dimensions of the video frame. Also, the display aspect ratio set by
       this filter may be changed by later filters in the filterchain, e.g. in
       case of scaling or if another "setdar" or a "setsar" filter is applied.

       The "setsar" filter sets the Sample (aka Pixel) Aspect Ratio for the
       filter output video.

       Note that as a consequence of the application of this filter, the
       output display aspect ratio will change according to the equation
       above.

       Keep in mind that the sample aspect ratio set by the "setsar" filter
       may be changed by later filters in the filterchain, e.g. if another
       "setsar" or a "setdar" filter is applied.

       It accepts the following parameters:

       r, ratio, dar ("setdar" only), sar ("setsar" only)
	   Set the aspect ratio used by the filter.

	   The parameter can be a floating point number string, or an
	   expression. If the parameter is not specified, the value "0" is
	   assumed, meaning that the same input value is used.

       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:

       w, h
	   The input width and height.

       a   Same as w / h.

       sar The input sample aspect ratio.

       dar The input display aspect ratio. It is the same as (w / h) * sar.

       hsub, vsub
	   Horizontal and vertical chroma subsample values. For example, for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       o   To change the display aspect ratio to 16:9, specify one of the
	   following:

		   setdar=dar=1.77777
		   setdar=dar=16/9

       o   To change the sample aspect ratio to 10:11, specify:

		   setsar=sar=10/11

       o   To set a display aspect ratio of 16:9, and specify a maximum
	   integer value of 1000 in the aspect ratio reduction, use the
	   command:

		   setdar=ratio=16/9:max=1000

   setfield
       Force field for the output video frame.

       The "setfield" filter marks the interlace type field for the output
       frames. It does not change the input frame, but only sets the
       corresponding property, which affects how the frame is treated by
       following filters (e.g. "fieldorder" or "yadif").

       The filter accepts the following options:

       mode
	   Available values are:

	   auto
	       Keep the same field property.

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

   setparams
       Force frame parameter for the output video frame.

       The "setparams" filter marks interlace and color range 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
       filters/encoders.

       field_mode
	   Available values are:

	   auto
	       Keep the same field property (default).

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

       range
	   Available values are:

	   auto
	       Keep the same color range property (default).

	   unspecified, unknown
	       Mark the frame as unspecified color range.

	   limited, tv, mpeg
	       Mark the frame as limited range.

	   full, pc, jpeg
	       Mark the frame as full range.

       color_primaries
	   Set the color primaries.  Available values are:

	   auto
	       Keep the same color primaries property (default).

	   bt709
	   unknown
	   bt470m
	   bt470bg
	   smpte170m
	   smpte240m
	   film
	   bt2020
	   smpte428
	   smpte431
	   smpte432
	   jedec-p22
       color_trc
	   Set the color transfer.  Available values are:

	   auto
	       Keep the same color trc property (default).

	   bt709
	   unknown
	   bt470m
	   bt470bg
	   smpte170m
	   smpte240m
	   linear
	   log100
	   log316
	   iec61966-2-4
	   bt1361e
	   iec61966-2-1
	   bt2020-10
	   bt2020-12
	   smpte2084
	   smpte428
	   arib-std-b67
       colorspace
	   Set the colorspace.	Available values are:

	   auto
	       Keep the same colorspace property (default).

	   gbr
	   bt709
	   unknown
	   fcc
	   bt470bg
	   smpte170m
	   smpte240m
	   ycgco
	   bt2020nc
	   bt2020c
	   smpte2085
	   chroma-derived-nc
	   chroma-derived-c
	   ictcp

   sharpen_npp
       Use the NVIDIA Performance Primitives (libnpp) to perform image
       sharpening with border control.

       The following additional options are accepted:

       border_type
	   Type of sampling to be used ad frame borders. One of the following:

	   replicate
	       Replicate pixel values.

   shear
       Apply shear transform to input video.

       This filter supports the following options:

       shx Shear factor in X-direction. Default value is 0.  Allowed range is
	   from -2 to 2.

       shy Shear factor in Y-direction. Default value is 0.  Allowed range is
	   from -2 to 2.

       fillcolor, c
	   Set the color used to fill the output area not covered by the
	   transformed video. 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".

       interp
	   Set interpolation type. Can be "bilinear" or "nearest". Default is
	   "bilinear".

       Commands

       This filter supports the all above options as commands.

   showinfo
       Show a line containing various information for each input video frame.
       The input video is not modified.

       This filter supports the following options:

       checksum
	   Calculate checksums of each plane. By default enabled.

       udu_sei_as_ascii
	   Try to print user data unregistered SEI as ascii character when
	   possible, in hex format otherwise.

       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.

       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]".

       mean
	   The mean value of pixels in each plane of the input frame,
	   expressed in the form "[mean0 mean1 mean2 mean3]".

       stdev
	   The standard deviation of pixel values in each plane of the input
	   frame, expressed in the form "[stdev0 stdev1 stdev2 stdev3]".

   showpalette
       Displays the 256 colors palette of each frame. This filter is only
       relevant for pal8 pixel format frames.

       It accepts the following option:

       s   Set the size of the box used to represent one palette color entry.
	   Default is 30 (for a "30x30" pixel box).

   shuffleframes
       Reorder and/or duplicate and/or drop video frames.

       It accepts the following parameters:

       mapping
	   Set the destination indexes of input frames.  This is space or '|'
	   separated list of indexes that maps input frames to output frames.
	   Number of indexes also sets maximal value that each index may have.
	   '-1' index have special meaning and that is to drop frame.

       The first frame has the index 0. The default is to keep the input
       unchanged.

       Examples

       o   Swap second and third frame of every three frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT

       o   Swap 10th and 1st frame of every ten frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT

   shufflepixels
       Reorder pixels in video frames.

       This filter accepts the following options:

       direction, d
	   Set shuffle direction. Can be forward or inverse direction.
	   Default direction is forward.

       mode, m
	   Set shuffle mode. Can be horizontal, vertical or block mode.

       width, w
       height, h
	   Set shuffle block_size. In case of horizontal shuffle mode only
	   width part of size is used, and in case of vertical shuffle mode
	   only height part of size is used.

       seed, s
	   Set random seed used with shuffling pixels. Mainly useful to set to
	   be able to reverse filtering process to get original input.	For
	   example, to reverse forward shuffle you need to use same parameters
	   and exact same seed and to set direction to inverse.

   shuffleplanes
       Reorder and/or duplicate video planes.

       It accepts the following parameters:

       map0
	   The index of the input plane to be used as the first output plane.

       map1
	   The index of the input plane to be used as the second output plane.

       map2
	   The index of the input plane to be used as the third output plane.

       map3
	   The index of the input plane to be used as the fourth output plane.

       The first plane has the index 0. The default is to keep the input
       unchanged.

       Examples

       o   Swap the second and third planes of the input:

		   ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

   signalstats
       Evaluate various visual metrics that assist in determining issues
       associated with the digitization of analog video media.

       By default the filter will log these metadata values:

       YMIN
	   Display the minimal Y value contained within the input frame.
	   Expressed in range of [0-255].

       YLOW
	   Display the Y value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       YAVG
	   Display the average Y value within the input frame. Expressed in
	   range of [0-255].

       YHIGH
	   Display the Y value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       YMAX
	   Display the maximum Y value contained within the input frame.
	   Expressed in range of [0-255].

       UMIN
	   Display the minimal U value contained within the input frame.
	   Expressed in range of [0-255].

       ULOW
	   Display the U value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       UAVG
	   Display the average U value within the input frame. Expressed in
	   range of [0-255].

       UHIGH
	   Display the U value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       UMAX
	   Display the maximum U value contained within the input frame.
	   Expressed in range of [0-255].

       VMIN
	   Display the minimal V value contained within the input frame.
	   Expressed in range of [0-255].

       VLOW
	   Display the V value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       VAVG
	   Display the average V value within the input frame. Expressed in
	   range of [0-255].

       VHIGH
	   Display the V value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       VMAX
	   Display the maximum V value contained within the input frame.
	   Expressed in range of [0-255].

       SATMIN
	   Display the minimal saturation value contained within the input
	   frame.  Expressed in range of [0-~181.02].

       SATLOW
	   Display the saturation value at the 10% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATAVG
	   Display the average saturation value within the input frame.
	   Expressed in range of [0-~181.02].

       SATHIGH
	   Display the saturation value at the 90% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATMAX
	   Display the maximum saturation value contained within the input
	   frame.  Expressed in range of [0-~181.02].

       HUEMED
	   Display the median value for hue within the input frame. Expressed
	   in range of [0-360].

       HUEAVG
	   Display the average value for hue within the input frame. Expressed
	   in range of [0-360].

       YDIF
	   Display the average of sample value difference between all values
	   of the Y plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       UDIF
	   Display the average of sample value difference between all values
	   of the U plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       VDIF
	   Display the average of sample value difference between all values
	   of the V plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       YBITDEPTH
	   Display bit depth of Y plane in current frame.  Expressed in range
	   of [0-16].

       UBITDEPTH
	   Display bit depth of U plane in current frame.  Expressed in range
	   of [0-16].

       VBITDEPTH
	   Display bit depth of V plane in current frame.  Expressed in range
	   of [0-16].

       The filter accepts the following options:

       stat
       out stat specify an additional form of image analysis.  out output
	   video with the specified type of pixel highlighted.

	   Both options accept the following values:

	   tout
	       Identify temporal outliers pixels. A temporal outlier is a
	       pixel unlike the neighboring pixels of the same field. Examples
	       of temporal outliers include the results of video dropouts,
	       head clogs, or tape tracking issues.

	   vrep
	       Identify vertical line repetition. Vertical line repetition
	       includes similar rows of pixels within a frame. In born-digital
	       video vertical line repetition is common, but this pattern is
	       uncommon in video digitized from an analog source. When it
	       occurs in video that results from the digitization of an analog
	       source it can indicate concealment from a dropout compensator.

	   brng
	       Identify pixels that fall outside of legal broadcast range.

       color, c
	   Set the highlight color for the out option. The default color is
	   yellow.

       Examples

       o   Output data of various video metrics:

		   ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

       o   Output specific data about the minimum and maximum values of the Y
	   plane per frame:

		   ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

       o   Playback video while highlighting pixels that are outside of
	   broadcast range in red.

		   ffplay example.mov -vf signalstats="out=brng:color=red"

       o   Playback video with signalstats metadata drawn over the frame.

		   ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

	   The contents of signalstat_drawtext.txt used in the command are:

		   time %{pts:hms}
		   Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
		   U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
		   V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
		   saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

   signature
       Calculates the MPEG-7 Video Signature. The filter can handle more than
       one input. In this case the matching between the inputs can be
       calculated additionally.  The filter always passes through the first
       input. The signature of each stream can be written into a file.

       It accepts the following options:

       detectmode
	   Enable or disable the matching process.

	   Available values are:

	   off Disable the calculation of a matching (default).

	   full
	       Calculate the matching for the whole video and output whether
	       the whole video matches or only parts.

	   fast
	       Calculate only until a matching is found or the video ends.
	       Should be faster in some cases.

       nb_inputs
	   Set the number of inputs. The option value must be a non negative
	   integer.  Default value is 1.

       filename
	   Set the path to which the output is written. If there is more than
	   one input, the path must be a prototype, i.e. must contain %d or
	   %0nd (where n is a positive integer), that will be replaced with
	   the input number. If no filename is specified, no output will be
	   written. This is the default.

       format
	   Choose the output format.

	   Available values are:

	   binary
	       Use the specified binary representation (default).

	   xml Use the specified xml representation.

       th_d
	   Set threshold to detect one word as similar. The option value must
	   be an integer greater than zero. The default value is 9000.

       th_dc
	   Set threshold to detect all words as similar. The option value must
	   be an integer greater than zero. The default value is 60000.

       th_xh
	   Set threshold to detect frames as similar. The option value must be
	   an integer greater than zero. The default value is 116.

       th_di
	   Set the minimum length of a sequence in frames to recognize it as
	   matching sequence. The option value must be a non negative integer
	   value.  The default value is 0.

       th_it
	   Set the minimum relation, that matching frames to all frames must
	   have.  The option value must be a double value between 0 and 1. The
	   default value is 0.5.

       Examples

       o   To calculate the signature of an input video and store it in
	   signature.bin:

		   ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -

       o   To detect whether two videos match and store the signatures in XML
	   format in signature0.xml and signature1.xml:

		   ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -

   siti
       Calculate Spatial Information (SI) and Temporal Information (TI) scores
       for a video, as defined in ITU-T Rec. P.910 (11/21): Subjective video
       quality assessment methods for multimedia applications. Available PDF
       at <https://www.itu.int/rec/T-REC-P.910-202111-S/en>.  Note that this
       is a legacy implementation that corresponds to a superseded
       recommendation.	Refer to ITU-T Rec. P.910 (07/22) for the latest
       version: <https://www.itu.int/rec/T-REC-P.910-202207-I/en>

       It accepts the following option:

       print_summary
	   If set to 1, Summary statistics will be printed to the console.
	   Default 0.

       Examples

       o   To calculate SI/TI metrics and print summary:

		   ffmpeg -i input.mp4 -vf siti=print_summary=1 -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.

       alpha_radius, ar
	   Set the alpha 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.

       alpha_strength, as
	   Set the alpha 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.

       alpha_threshold, at
	   Set the alpha 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 or alpha option is not explicitly set, the corresponding
       luma value is set.

   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.

       Commands

       This filter supports the all above options as commands.

   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).

       Commands

       This filter supports the following commands:

       quality, level
	   Set quality level. The value "max" can be used to set the maximum
	   level, currently 6.

   sr
       Scale the input by applying one of the super-resolution methods based
       on convolutional neural networks. Supported models:

       o   Super-Resolution Convolutional Neural Network model (SRCNN).  See
	   <https://arxiv.org/abs/1501.00092>.

       o   Efficient Sub-Pixel Convolutional Neural Network model (ESPCN).
	   See <https://arxiv.org/abs/1609.05158>.

       Training scripts as well as scripts for model file (.pb) saving can be
       found at <https://github.com/XueweiMeng/sr/tree/sr_dnn_native>.
       Original repository is at
       <https://github.com/HighVoltageRocknRoll/sr.git>.

       The filter accepts the following options:

       dnn_backend
	   Specify which DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to install
	       the TensorFlow for C library (see
	       <https://www.tensorflow.org/install/lang_c>) and configure
	       FFmpeg with "--enable-libtensorflow"

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats. TensorFlow, OpenVINO backend can load files for only its
	   format.

       scale_factor
	   Set scale factor for SRCNN model. Allowed values are 2, 3 and 4.
	   Default value is 2. Scale factor is necessary for SRCNN model,
	   because it accepts input upscaled using bicubic upscaling with
	   proper scale factor.

       To get full functionality (such as async execution), please use the
       dnn_processing filter.

   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.

       Examples

       o   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.

       o   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 -

       o   Another example with different containers:

		   ffmpeg -i main.mpg -i ref.mkv -lavfi  "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]ssim" -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
	   tbl above-below (left eye above, right eye below)

	   abr
	   tbr above-below (right eye above, left eye below)

	   ab2l
	   tb2l
	       above-below with half height resolution (left eye above, right
	       eye below)

	   ab2r
	   tb2r
	       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
	   tbl above-below (left eye above, right eye below)

	   abr
	   tbr above-below (right eye above, left eye below)

	   ab2l
	   tb2l
	       above-below with half height resolution (left eye above, right
	       eye below)

	   ab2r
	   tb2r
	       above-below with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	   irl interleaved rows (left eye has top row, right eye starts on
	       next row)

	   irr interleaved rows (right eye has top row, left eye starts on
	       next row)

	   arbg
	       anaglyph red/blue gray (red filter on left eye, blue filter on
	       right eye)

	   argg
	       anaglyph red/green gray (red filter on left eye, green filter
	       on right eye)

	   arcg
	       anaglyph red/cyan gray (red filter on left eye, cyan filter on
	       right eye)

	   arch
	       anaglyph red/cyan half colored (red filter on left eye, cyan
	       filter on right eye)

	   arcc
	       anaglyph red/cyan color (red filter on left eye, cyan filter on
	       right eye)

	   arcd
	       anaglyph red/cyan color optimized with the least squares
	       projection of dubois (red filter on left eye, cyan filter on
	       right eye)

	   agmg
	       anaglyph green/magenta gray (green filter on left eye, magenta
	       filter on right eye)

	   agmh
	       anaglyph green/magenta half colored (green filter on left eye,
	       magenta filter on right eye)

	   agmc
	       anaglyph green/magenta colored (green filter on left eye,
	       magenta filter on right eye)

	   agmd
	       anaglyph green/magenta color optimized with the least squares
	       projection of dubois (green filter on left eye, magenta filter
	       on right eye)

	   aybg
	       anaglyph yellow/blue gray (yellow filter on left eye, blue
	       filter on right eye)

	   aybh
	       anaglyph yellow/blue half colored (yellow filter on left eye,
	       blue filter on right eye)

	   aybc
	       anaglyph yellow/blue colored (yellow filter on left eye, blue
	       filter on right eye)

	   aybd
	       anaglyph yellow/blue color optimized with the least squares
	       projection of dubois (yellow filter on left eye, blue filter on
	       right eye)

	   ml  mono output (left eye only)

	   mr  mono output (right eye only)

	   chl checkerboard, left eye first

	   chr checkerboard, right eye first

	   icl interleaved columns, left eye first

	   icr interleaved columns, right eye first

	   hdmi
	       HDMI frame pack

	   Default value is arcd.

       Examples

       o   Convert input video from side by side parallel to anaglyph
	   yellow/blue dubois:

		   stereo3d=sbsl:aybd

       o   Convert input video from above below (left eye above, right eye
	   below) to side by side crosseye.

		   stereo3d=abl:sbsr

   streamselect, astreamselect
       Select video or audio streams.

       The filter accepts the following options:

       inputs
	   Set number of inputs. Default is 2.

       map Set input indexes to remap to outputs.

       Commands

       The "streamselect" and "astreamselect" filter supports the following
       commands:

       map Set input indexes to remap to outputs.

       Examples

       o   Select first 5 seconds 1st stream and rest of time 2nd stream:

		   sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0

       o   Same as above, but for audio:

		   asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0

   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 ",".

       wrap_unicode
	   Break lines according to the Unicode Line Breaking Algorithm.
	   Availability requires at least libass release 0.17.0 (or
	   LIBASS_VERSION 0x01600010), and libass must have been built with
	   libunibreak.

	   The option is enabled by default except for native ASS.

       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 80% transparent
       blue "DejaVu Serif", use:

	       subtitles=sub.srt:force_style='Fontname=DejaVu Serif,PrimaryColour=&HCCFF0000'

   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;
	   deprecated, do not use

       Commands

       This filter supports the all above options as commands.

   swapuv
       Swap U & V plane.

   tblend
       Blend successive video frames.

       See blend

   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

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

       Unlike histogram video filter which only shows histogram of single
       input frame at certain time, this filter shows also past histograms of
       number of frames defined by "width" option.

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

       The filter accepts the following options:

       width, w
	   Set width of single color component output. Default value is 0.
	   Value of 0 means width will be picked from input video.  This also
	   set number of passed histograms to keep.  Allowed range is [0,
	   8192].

       display_mode, d
	   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, m
	   Set mode. Can be either "linear", or "logarithmic".	Default is
	   "linear".

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

       bgopacity, b
	   Set background opacity. Default is 0.9.

       envelope, e
	   Show envelope. Default is disabled.

       ecolor, ec
	   Set envelope color. Default is "gold".

       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 "replace".

   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:

       Commands

       This filter supports the all options as commands.

       Examples

       o   Binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi

       o   Inverted binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi

       o   Truncate binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi

       o   Threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi

       o   Inverted threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi

   thumbnail
       Select the most representative frame in a given sequence of consecutive
       frames.

       The filter accepts the following options:

       n   Set the frames batch size to analyze; in a set of n frames, the
	   filter will pick one of them, and then handle the next batch of n
	   frames until the end. Default is 100.

       log Set the log level to display picked frame stats.  Default is
	   "info".

       Since the filter keeps track of the whole frames sequence, a bigger n
       value will result in a higher memory usage, so a high value is not
       recommended.

       Examples

       o   Extract one picture each 50 frames:

		   thumbnail=50

       o   Complete example of a thumbnail creation with ffmpeg:

		   ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

   tile
       Tile several successive frames together.

       The untile filter can do the reverse.

       The filter accepts the following options:

       layout
	   Set the grid size in the form "COLUMNSxROWS". Range is up to
	   UINT_MAX cells.  Default is "6x5".

       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. Range is 0 to 1024. Default
	   is 0.

       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.
	   Range is 0 to 1024. Default is 0.

       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".

       overlap
	   Set the number of frames to overlap when tiling several successive
	   frames together.  The value must be between 0 and nb_frames - 1.
	   Default is 0.

       init_padding
	   Set the number of frames to initially be empty before displaying
	   first output frame.	This controls how soon will one get first
	   output frame.  The value must be between 0 and nb_frames - 1.
	   Default is 0.

       Examples

       o   Produce 8x8 PNG tiles of all keyframes (-skip_frame nokey) in a
	   movie:

		   ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

	   The -vsync 0 is necessary to prevent ffmpeg from duplicating each
	   output frame to accommodate the originally detected frame rate.

       o   Display 5 pictures in an area of "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

   tiltandshift
       Apply tilt-and-shift effet.

       What happens when you invert time and space?

       Normally a video is composed of several frames that represent a
       different instant of time and shows a scene that evolves in the space
       captured by the frame. This filter is the antipode of that concept,
       taking inspiration by tilt and shift photography.

       A filtered frame contains the whole timeline of events composing the
       sequence, and this is obtained by placing a slice of pixels from each
       frame into a single one. However, since there are no infinite-width
       frames, this is done up the width of the input frame, and a video is
       recomposed by shifting away one column for each subsequent frame. In
       order to map space to time, the filter tilts each input frame as well,
       so that motion is preserved. This is accomplished by progressively
       selecting a different column from each input frame.

       The end result is a sort of inverted parallax, so that far away objects
       move much faster that the ones in the front. The ideal conditions for
       this video effect are when there is either very little motion and the
       backgroud is static, or when there is a lot of motion and a very wide
       depth of field (e.g. wide panorama, while moving on a train).

       The filter accepts the following parameters:

       tilt
	   Tilt video while shifting (default). When unset, video will be
	   sliding a static image, composed of the first column of each frame.

       start
	   What to do at the start of filtering (see below).

       end What to do at the end of filtering (see below).

       hold
	   How many columns should pass through before start of filtering.

       pad How many columns should be inserted before end of filtering.

       Normally the filter shifts and tils from the very first frame, and
       stops when the last one is received. However, before filtering starts,
       normal video may be preseved, so that the effect is slowly shifted in
       its place. Similarly, the last video frame may be reconstructed at the
       end. Alternatively it is possible to just start and end with black.

       none
	   Filtering is starts immediately and ends when the last frame is
	   received.

       frame
	   The first frames or the very last frame are kept intact during
	   processing.

       black
	   Black is padded at the beginning or at the end of filtering.

   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, vlpf
	       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, cvlpf
	       Enable complex vertical low-pass filtering.  This will slightly
	       less reduce interlace 'twitter' and Moire patterning but better
	       retain detail and subjective sharpness impression.

	   bypass_il
	       Bypass already interlaced frames, only adjust the frame rate.

	   Vertical low-pass filtering and bypassing already interlaced frames
	   can only be enabled for mode interleave_top and interleave_bottom.

   tmedian
       Pick median pixels from several successive input video frames.

       The filter accepts the following options:

       radius
	   Set radius of median filter.  Default is 1. Allowed range is from 1
	   to 127.

       planes
	   Set which planes to filter. Default value is 15, by which all
	   planes are processed.

       percentile
	   Set median percentile. Default value is 0.5.  Default value of 0.5
	   will pick always median values, while 0 will pick minimum values,
	   and 1 maximum values.

       Commands

       This filter supports all above options as commands, excluding option
       "radius".

   tmidequalizer
       Apply Temporal Midway Video Equalization effect.

       Midway Video Equalization adjusts a sequence of video frames to have
       the same histograms, while maintaining their dynamics as much as
       possible. It's useful for e.g. matching exposures from a video frames
       sequence.

       This filter accepts the following option:

       radius
	   Set filtering radius. Default is 5. Allowed range is from 1 to 127.

       sigma
	   Set filtering sigma. Default is 0.5. This controls strength of
	   filtering.  Setting this option to 0 effectively does nothing.

       planes
	   Set which planes to process. Default is 15, which is all available
	   planes.

   tmix
       Mix successive video frames.

       A description of the accepted options follows.

       frames
	   The number of successive frames to mix. If unspecified, it defaults
	   to 3.

       weights
	   Specify weight of each input video frame.  Each weight is separated
	   by space. If number of weights is smaller than number of frames
	   last specified weight will be used for all remaining unset weights.

       scale
	   Specify scale, if it is set it will be multiplied with sum of each
	   weight multiplied with pixel values to give final destination pixel
	   value. By default scale is auto scaled to sum of weights.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Examples

       o   Average 7 successive frames:

		   tmix=frames=7:weights="1 1 1 1 1 1 1"

       o   Apply simple temporal convolution:

		   tmix=frames=3:weights="-1 3 -1"

       o   Similar as above but only showing temporal differences:

		   tmix=frames=3:weights="-1 2 -1":scale=1

       Commands

       This filter supports the following commands:

       weights
       scale
       planes
	   Syntax is same as option with same name.

   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.

   tpad
       Temporarily pad video frames.

       The filter accepts the following options:

       start
	   Specify number of delay frames before input video stream. Default
	   is 0.

       stop
	   Specify number of padding frames after input video stream.  Set to
	   -1 to pad indefinitely. Default is 0.

       start_mode
	   Set kind of frames added to beginning of stream.  Can be either add
	   or clone.  With add frames of solid-color are added.  With clone
	   frames are clones of first frame.  Default is add.

       stop_mode
	   Set kind of frames added to end of stream.  Can be either add or
	   clone.  With add frames of solid-color are added.  With clone
	   frames are clones of last frame.  Default is add.

       start_duration, stop_duration
	   Specify the duration of the start/stop delay. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.
	   These options override start and stop. Default is 0.

       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".

   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

   transpose_npp
       Transpose rows with columns in the input video and optionally flip it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate by 90 degrees counterclockwise and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

       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. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

   trim
       Trim the input so that the output contains one continuous subpart of
       the input.

       It accepts the following parameters:

       start
	   Specify the time of the start of the kept section, i.e. the frame
	   with the timestamp start will be the first frame in the output.

       end Specify the time of the first frame that will be dropped, i.e. the
	   frame immediately preceding the one with the timestamp end will be
	   the last frame in the output.

       start_pts
	   This is the same as start, except this option sets the start
	   timestamp in timebase units instead of seconds.

       end_pts
	   This is the same as end, except this option sets the end timestamp
	   in timebase units instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_frame
	   The number of the first frame that should be passed to the output.

       end_frame
	   The number of the first frame that should be dropped.

       start, end, and duration are expressed as time duration specifications;
       see the Time duration section in the ffmpeg-utils(1) manual for the
       accepted syntax.

       Note that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the _frame variants simply
       count the frames that pass through the filter. Also note that this
       filter does not modify the timestamps. If you wish for the output
       timestamps to start at zero, insert a setpts filter after the trim
       filter.

       If multiple start or end options are set, this filter tries to be
       greedy and keep all the frames that match at least one of the specified
       constraints. To keep only the part that matches all the constraints at
       once, chain multiple trim filters.

       The defaults are such that all the input is kept. So it is possible to
       set e.g.  just the end values to keep everything before the specified
       time.

       Examples:

       o   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -vf trim=60:120

       o   Keep only the first second:

		   ffmpeg -i INPUT -vf trim=duration=1

   unpremultiply
       Apply alpha unpremultiply effect to input video stream using first
       plane of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
	   Set which planes will be processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

	   If the format has 1 or 2 components, then luma is bit 0.  If the
	   format has 3 or 4 components: for RGB formats bit 0 is green, bit 1
	   is blue and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
	   chroma-U and bit 2 is chroma-V.  If present, the alpha channel is
	   always the last bit.

       inplace
	   Do not require 2nd input for processing, instead use alpha plane
	   from input stream.

   unsharp
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
	   Set the luma matrix horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       luma_msize_y, ly
	   Set the luma matrix vertical size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       luma_amount, la
	   Set the luma effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 1.0.

       chroma_msize_x, cx
	   Set the chroma matrix horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       chroma_msize_y, cy
	   Set the chroma matrix vertical size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 0.0.

       alpha_msize_x, ax
	   Set the alpha matrix horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       alpha_msize_y, ay
	   Set the alpha matrix vertical size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       alpha_amount, aa
	   Set the alpha 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.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       o   Apply strong luma sharpen effect:

		   unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

       o   Apply a strong blur of both luma and chroma parameters:

		   unsharp=7:7:-2:7:7:-2

   untile
       Decompose a video made of tiled images into the individual images.

       The frame rate of the output video is the frame rate of the input video
       multiplied by the number of tiles.

       This filter does the reverse of tile.

       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.

       Examples

       o   Produce a 1-second video from a still image file made of 25 frames
	   stacked vertically, like an analogic film reel:

		   ffmpeg -r 1 -i image.jpg -vf untile=1x25 movie.mkv

   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.

       This filter is only available in ffmpeg version 4.4 or earlier.

       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).

       codec
	   Use specified codec instead of snow.

   v360
       Convert 360 videos between various formats.

       The filter accepts the following options:

       input
       output
	   Set format of the input/output video.

	   Available formats:

	   e
	   equirect
	       Equirectangular projection.

	   c3x2
	   c6x1
	   c1x6
	       Cubemap with 3x2/6x1/1x6 layout.

	       Format specific options:

	       in_pad
	       out_pad
		   Set padding proportion for the input/output cubemap. Values
		   in decimals.

		   Example values:

		   0   No padding.

		   0.01
		       1% of face is padding. For example, with 1920x1280
		       resolution face size would be 640x640 and padding would
		       be 3 pixels from each side. (640 * 0.01 = 6 pixels)

		   Default value is @samp{0}.  Maximum value is @samp{0.1}.

	       fin_pad
	       fout_pad
		   Set fixed padding for the input/output cubemap. Values in
		   pixels.

		   Default value is @samp{0}. If greater than zero it
		   overrides other padding options.

	       in_forder
	       out_forder
		   Set order of faces for the input/output cubemap. Choose one
		   direction for each position.

		   Designation of directions:

		   r   right

		   l   left

		   u   up

		   d   down

		   f   forward

		   b   back

		   Default value is @samp{rludfb}.

	       in_frot
	       out_frot
		   Set rotation of faces for the input/output cubemap. Choose
		   one angle for each position.

		   Designation of angles:

		   0   0 degrees clockwise

		   1   90 degrees clockwise

		   2   180 degrees clockwise

		   3   270 degrees clockwise

		   Default value is @samp{000000}.

	   eac Equi-Angular Cubemap.

	   flat
	   gnomonic
	   rectilinear
	       Regular video.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   dfisheye
	       Dual fisheye.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   barrel
	   fb
	   barrelsplit
	       Facebook's 360 formats.

	   sg  Stereographic format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   mercator
	       Mercator format.

	   ball
	       Ball format, gives significant distortion toward the back.

	   hammer
	       Hammer-Aitoff map projection format.

	   sinusoidal
	       Sinusoidal map projection format.

	   fisheye
	       Fisheye projection.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   pannini
	       Pannini projection.

	       Format specific options:

	       h_fov
		   Set output pannini parameter.

	       ih_fov
		   Set input pannini parameter.

	   cylindrical
	       Cylindrical projection.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   perspective
	       Perspective projection. (output only)

	       Format specific options:

	       v_fov
		   Set perspective parameter.

	   tetrahedron
	       Tetrahedron projection.

	   tsp Truncated square pyramid projection.

	   he
	   hequirect
	       Half equirectangular projection.

	   equisolid
	       Equisolid format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   og  Orthographic format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set output horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set input horizontal/vertical/diagonal field of view.
		   Values in degrees.

		   If diagonal field of view is set it overrides horizontal
		   and vertical field of view.

	   octahedron
	       Octahedron projection.

	   cylindricalea
	       Cylindrical Equal Area projection.

       interp
	   Set interpolation method.Note: more complex interpolation methods
	   require much more memory to run.

	   Available methods:

	   near
	   nearest
	       Nearest neighbour.

	   line
	   linear
	       Bilinear interpolation.

	   lagrange9
	       Lagrange9 interpolation.

	   cube
	   cubic
	       Bicubic interpolation.

	   lanc
	   lanczos
	       Lanczos interpolation.

	   sp16
	   spline16
	       Spline16 interpolation.

	   gauss
	   gaussian
	       Gaussian interpolation.

	   mitchell
	       Mitchell interpolation.

	   Default value is @samp{line}.

       w
       h   Set the output video resolution.

	   Default resolution depends on formats.

       in_stereo
       out_stereo
	   Set the input/output stereo format.

	   2d  2D mono

	   sbs Side by side

	   tb  Top bottom

	   Default value is @samp{2d} for input and output format.

       yaw
       pitch
       roll
	   Set rotation for the output video. Values in degrees.

       rorder
	   Set rotation order for the output video. Choose one item for each
	   position.

	   y, Y
	       yaw

	   p, P
	       pitch

	   r, R
	       roll

	   Default value is @samp{ypr}.

       h_flip
       v_flip
       d_flip
	   Flip the output video horizontally(swaps
	   left-right)/vertically(swaps up-down)/in-depth(swaps back-forward).
	   Boolean values.

       ih_flip
       iv_flip
	   Set if input video is flipped horizontally/vertically. Boolean
	   values.

       in_trans
	   Set if input video is transposed. Boolean value, by default
	   disabled.

       out_trans
	   Set if output video needs to be transposed. Boolean value, by
	   default disabled.

       h_offset
       v_offset
	   Set output horizontal/vertical off-axis offset. Default is set to
	   0.  Allowed range is from -1 to 1.

       alpha_mask
	   Build mask in alpha plane for all unmapped pixels by marking them
	   fully transparent. Boolean value, by default disabled.

       reset_rot
	   Reset rotation of output video. Boolean value, by default disabled.

       Examples

       o   Convert equirectangular video to cubemap with 3x2 layout and 1%
	   padding using bicubic interpolation:

		   ffmpeg -i input.mkv -vf v360=e:c3x2:cubic:out_pad=0.01 output.mkv

       o   Extract back view of Equi-Angular Cubemap:

		   ffmpeg -i input.mkv -vf v360=eac:flat:yaw=180 output.mkv

       o   Convert transposed and horizontally flipped Equi-Angular Cubemap in
	   side-by-side stereo format to equirectangular top-bottom stereo
	   format:

		   v360=eac:equirect:in_stereo=sbs:in_trans=1:ih_flip=1:out_stereo=tb

       Commands

       This filter supports subset of above options as commands.

   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.

       type
	   The threshold type the filter will use.

	   It accepts the following values:

	   universal
	       Threshold used is same for all decompositions.

	   bayes
	       Threshold used depends also on each decomposition coefficients.

	   Default is universal.

   varblur
       Apply variable blur filter by using 2nd video stream to set blur
       radius.	The 2nd stream must have the same dimensions.

       This filter accepts the following options:

       min_r
	   Set min allowed radius. Allowed range is from 0 to 254. Default is
	   0.

       max_r
	   Set max allowed radius. Allowed range is from 1 to 255. Default is
	   8.

       planes
	   Set which planes to process. By default, all are used.

       The "varblur" filter also supports the framesync options.

       Commands

       This filter supports all the above options as commands.

   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
	   tint
	       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
	   invert
       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.

       tint0, t0
       tint1, t1
	   Set color tint for gray/tint vectorscope mode. By default both
	   options are zero.  This means no tint, and output will remain gray.

   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.

       fileformat
	   Format for the transforms data file to be written.  Acceptable
	   values are

	   ascii
	       Human-readable plain text

	   binary
	       Binary format, roughly 40% smaller than "ascii". (default)

       Examples

       o   Use default values:

		   vidstabdetect

       o   Analyze strongly shaky movie and put the results in file
	   mytransforms.trf:

		   vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

       o   Visualize the result of internal transformations in the resulting
	   video:

		   vidstabdetect=show=1

       o   Analyze a video with medium shakiness using ffmpeg:

		   ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

   vidstabtransform
       Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
       1.

       Read a file with transform information for each frame and
       apply/compensate them. Together with the vidstabdetect filter this can
       be used to deshake videos. See also
       <http://public.hronopik.de/vid.stab>. It is important to also use the
       unsharp filter, see below.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libvidstab".

       Options

       input
	   Set path to the file used to read the transforms. Default value is
	   transforms.trf.

       smoothing
	   Set the number of frames (value*2 + 1) used for lowpass filtering
	   the camera movements. Default value is 10.

	   For example a number of 10 means that 21 frames are used (10 in the
	   past and 10 in the future) to smoothen the motion in the video. A
	   larger value leads to a smoother video, but limits the acceleration
	   of the camera (pan/tilt movements). 0 is a special case where a
	   static camera is simulated.

       optalgo
	   Set the camera path optimization algorithm.

	   Accepted values are:

	   gauss
	       gaussian kernel low-pass filter on camera motion (default)

	   avg averaging on transformations

       maxshift
	   Set maximal number of pixels to translate frames. Default value is
	   -1, meaning no limit.

       maxangle
	   Set maximal angle in radians (degree*PI/180) to rotate frames.
	   Default value is -1, meaning no limit.

       crop
	   Specify how to deal with borders that may be visible due to
	   movement compensation.

	   Available values are:

	   keep
	       keep image information from previous frame (default)

	   black
	       fill the border black

       invert
	   Invert transforms if set to 1. Default value is 0.

       relative
	   Consider transforms as relative to previous frame if set to 1,
	   absolute if set to 0. Default value is 0.

       zoom
	   Set percentage to zoom. A positive value will result in a zoom-in
	   effect, a negative value in a zoom-out effect. Default value is 0
	   (no zoom).

       optzoom
	   Set optimal zooming to avoid borders.

	   Accepted values are:

	   0   disabled

	   1   optimal static zoom value is determined (only very strong
	       movements will lead to visible borders) (default)

	   2   optimal adaptive zoom value is determined (no borders will be
	       visible), see zoomspeed

	   Note that the value given at zoom is added to the one calculated
	   here.

       zoomspeed
	   Set percent to zoom maximally each frame (enabled when optzoom is
	   set to 2). Range is from 0 to 5, default value is 0.25.

       interpol
	   Specify type of interpolation.

	   Available values are:

	   no  no interpolation

	   linear
	       linear only horizontal

	   bilinear
	       linear in both directions (default)

	   bicubic
	       cubic in both directions (slow)

       tripod
	   Enable virtual tripod mode if set to 1, which is equivalent to
	   "relative=0:smoothing=0". Default value is 0.

	   Use also "tripod" option of vidstabdetect.

       debug
	   Increase log verbosity if set to 1. Also the detected global
	   motions are written to the temporary file global_motions.trf.
	   Default value is 0.

       Examples

       o   Use ffmpeg for a typical stabilization with default values:

		   ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

	   Note the use of the unsharp filter which is always recommended.

       o   Zoom in a bit more and load transform data from a given file:

		   vidstabtransform=zoom=5:input="mytransforms.trf"

       o   Smoothen the video even more:

		   vidstabtransform=smoothing=30

   vflip
       Flip the input video vertically.

       For example, to vertically flip a video with ffmpeg:

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

   vfrdet
       Detect variable frame rate video.

       This filter tries to detect if the input is variable or constant frame
       rate.

       At end it will output number of frames detected as having variable
       delta pts, and ones with constant delta pts.  If there was frames with
       variable delta, than it will also show min, max and average delta
       encountered.

   vibrance
       Boost or alter saturation.

       The filter accepts the following options:

       intensity
	   Set strength of boost if positive value or strength of alter if
	   negative value.  Default is 0. Allowed range is from -2 to 2.

       rbal
	   Set the red balance. Default is 1. Allowed range is from -10 to 10.

       gbal
	   Set the green balance. Default is 1. Allowed range is from -10 to
	   10.

       bbal
	   Set the blue balance. Default is 1. Allowed range is from -10 to
	   10.

       rlum
	   Set the red luma coefficient.

       glum
	   Set the green luma coefficient.

       blum
	   Set the blue luma coefficient.

       alternate
	   If "intensity" is negative and this is set to 1, colors will
	   change, otherwise colors will be less saturated, more towards gray.

       Commands

       This filter supports the all above options as commands.

   vif
       Obtain the average VIF (Visual Information Fidelity) between two input
       videos.

       This filter takes two input videos.

       Both input videos 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 VIF score is printed through the logging system.

       The filter stores the calculated VIF score of each frame.

       This filter also supports the framesync options.

       In the below example the input file main.mpg being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi vif -f null -

   vignette
       Make or reverse a natural vignetting effect.

       The filter accepts the following options:

       angle, a
	   Set lens angle expression as a number of radians.

	   The value is clipped in the "[0,PI/2]" range.

	   Default value: "PI/5"

       x0
       y0  Set center coordinates expressions. Respectively "w/2" and "h/2" by
	   default.

       mode
	   Set forward/backward mode.

	   Available modes are:

	   forward
	       The larger the distance from the central point, the darker the
	       image becomes.

	   backward
	       The larger the distance from the central point, the brighter
	       the image becomes.  This can be used to reverse a vignette
	       effect, though there is no automatic detection to extract the
	       lens angle and other settings (yet). It can also be used to
	       create a burning effect.

	   Default value is forward.

       eval
	   Set evaluation mode for the expressions (angle, x0, y0).

	   It accepts the following values:

	   init
	       Evaluate expressions only once during the filter
	       initialization.

	   frame
	       Evaluate expressions for each incoming frame. This is way
	       slower than the init mode since it requires all the scalers to
	       be re-computed, but it allows advanced dynamic expressions.

	   Default value is init.

       dither
	   Set dithering to reduce the circular banding effects. Default is 1
	   (enabled).

       aspect
	   Set vignette aspect. This setting allows one to adjust the shape of
	   the vignette.  Setting this value to the SAR of the input will make
	   a rectangular vignetting following the dimensions of the video.

	   Default is "1/1".

       Expressions

       The alpha, x0 and y0 expressions can contain the following parameters.

       w
       h   input width and height

       n   the number of input frame, starting from 0

       pts the PTS (Presentation TimeStamp) time of the filtered video frame,
	   expressed in TB units, NAN if undefined

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

       t   the PTS (Presentation TimeStamp) of the filtered video frame,
	   expressed in seconds, NAN if undefined

       tb  time base of the input video

       Examples

       o   Apply simple strong vignetting effect:

		   vignette=PI/4

       o   Make a flickering vignetting:

		   vignette='PI/4+random(1)*PI/50':eval=frame

   vmafmotion
       Obtain the average VMAF motion score of a video.  It is one of the
       component metrics of VMAF.

       The obtained average motion score is printed through the logging
       system.

       The filter accepts the following options:

       stats_file
	   If specified, the filter will use the named file to save the motion
	   score of each frame with respect to the previous frame.  When
	   filename equals "-" the data is sent to standard output.

       Example:

	       ffmpeg -i ref.mpg -vf 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 accepts the following options:

       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.

       This filter uses 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 "w3fdif" filter.

       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.

       mode
	   The interlacing mode to adopt. It accepts one of the following
	   values:

	   frame
	       Output one frame for each frame.

	   field
	       Output one frame for each field.

	   The default value is "field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   tff Assume the top field is first.

	   bff Assume the bottom field is first.

	   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. Accepts one of the following
	   values:

	   all Deinterlace all frames,

	   interlaced
	       Only deinterlace frames marked as interlaced.

	   Default value is all.

       Commands

       This filter supports same commands as options.

   waveform
       Video waveform monitor.

       The waveform monitor plots color component intensity. By default luma
       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 luma 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.

	   xflat
	       Similar as above, but use different colors.

	   yflat
	       Similar as above, but again with different colors.

	   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.

	   orange
	       Display orange graticule showing legal broadcast ranges.

	   invert
	       Display invert 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.

       tint0, t0
       tint1, t1
	   Set tint for output.  Only used with lowpass filter and when
	   display is not overlay and input pixel formats are not RGB.

       fitmode, fm
	   Set sample aspect ratio of video output frames.  Can be used to
	   configure waveform so it is not streched too much in one of
	   directions.

	   none
	       Set sample aspect ration to 1/1.

	   size
	       Set sample aspect ratio to match input size of video

	   Default is none.

       input
	   Set input formats for filter to pick from.  Can be all, for
	   selecting from all available formats, or first, for selecting first
	   available format.  Default is first.

   weave, doubleweave
       The "weave" takes a field-based video input and join each two
       sequential fields into single frame, producing a new double height clip
       with half the frame rate and half the frame count.

       The "doubleweave" works same as "weave" but without halving frame rate
       and frame count.

       It accepts the following option:

       first_field
	   Set first field. Available values are:

	   top, t
	       Set the frame as top-field-first.

	   bottom, b
	       Set the frame as bottom-field-first.

       Examples

       o   Interlace video using select and separatefields filter:

		   separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave

   xbr
       Apply the xBR high-quality magnification filter which is designed for
       pixel art. It follows a set of edge-detection rules, see
       <https://forums.libretro.com/t/xbr-algorithm-tutorial/123>.

       It accepts the following option:

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

   xcorrelate
       Apply normalized cross-correlation between first and second input video
       stream.

       Second input video stream dimensions must be lower than first input
       video stream.

       The filter accepts the following options:

       planes
	   Set which planes to process.

       secondary
	   Set which secondary video frames will be processed from second
	   input video stream, can be first or all. Default is all.

       The "xcorrelate" filter also supports the framesync options.

   xfade
       Apply cross fade from one input video stream to another input video
       stream.	The cross fade is applied for specified duration.

       Both inputs must be constant frame-rate and have the same resolution,
       pixel format, frame rate and timebase.

       The filter accepts the following options:

       transition
	   Set one of available transition effects:

	   custom
	   fade
	   wipeleft
	   wiperight
	   wipeup
	   wipedown
	   slideleft
	   slideright
	   slideup
	   slidedown
	   circlecrop
	   rectcrop
	   distance
	   fadeblack
	   fadewhite
	   radial
	   smoothleft
	   smoothright
	   smoothup
	   smoothdown
	   circleopen
	   circleclose
	   vertopen
	   vertclose
	   horzopen
	   horzclose
	   dissolve
	   pixelize
	   diagtl
	   diagtr
	   diagbl
	   diagbr
	   hlslice
	   hrslice
	   vuslice
	   vdslice
	   hblur
	   fadegrays
	   wipetl
	   wipetr
	   wipebl
	   wipebr
	   squeezeh
	   squeezev
	   zoomin
	   fadefast
	   fadeslow
	   hlwind
	   hrwind
	   vuwind
	   vdwind
	   coverleft
	   coverright
	   coverup
	   coverdown
	   revealleft
	   revealright
	   revealup
	   revealdown

	   Default transition effect is fade.

       duration
	   Set cross fade duration in seconds.	Range is 0 to 60 seconds.
	   Default duration is 1 second.

       offset
	   Set cross fade start relative to first input stream in seconds.
	   Default offset is 0.

       expr
	   Set expression for custom transition effect.

	   The expressions can use the following variables and functions:

	   X
	   Y   The coordinates of the current sample.

	   W
	   H   The width and height of the image.

	   P   Progress of transition effect.

	   PLANE
	       Currently processed plane.

	   A   Return value of first input at current location and plane.

	   B   Return value of second input at current location and plane.

	   a0(x, y)
	   a1(x, y)
	   a2(x, y)
	   a3(x, y)
	       Return the value of the pixel at location (x,y) of the
	       first/second/third/fourth component of first input.

	   b0(x, y)
	   b1(x, y)
	   b2(x, y)
	   b3(x, y)
	       Return the value of the pixel at location (x,y) of the
	       first/second/third/fourth component of second input.

       Examples

       o   Cross fade from one input video to another input video, with fade
	   transition and duration of transition of 2 seconds starting at
	   offset of 5 seconds:

		   ffmpeg -i first.mp4 -i second.mp4 -filter_complex xfade=transition=fade:duration=2:offset=5 output.mp4

   xmedian
       Pick median pixels from several input videos.

       The filter accepts the following options:

       inputs
	   Set number of inputs.  Default is 3. Allowed range is from 3 to
	   255.  If number of inputs is even number, than result will be mean
	   value between two median values.

       planes
	   Set which planes to filter. Default value is 15, by which all
	   planes are processed.

       percentile
	   Set median percentile. Default value is 0.5.  Default value of 0.5
	   will pick always median values, while 0 will pick minimum values,
	   and 1 maximum values.

       Commands

       This filter supports all above options as commands, excluding option
       "inputs".

   xstack
       Stack video inputs into custom layout.

       All streams must be of same pixel format.

       The filter accepts the following options:

       inputs
	   Set number of input streams. Default is 2.

       layout
	   Specify layout of inputs.  This option requires the desired layout
	   configuration to be explicitly set by the user.  This sets position
	   of each video input in output. Each input is separated by '|'.  The
	   first number represents the column, and the second number
	   represents the row.	Numbers start at 0 and are separated by '_'.
	   Optionally one can use wX and hX, where X is video input from which
	   to take width or height.  Multiple values can be used when
	   separated by '+'. In such case values are summed together.

	   Note that if inputs are of different sizes gaps may appear, as not
	   all of the output video frame will be filled. Similarly, videos can
	   overlap each other if their position doesn't leave enough space for
	   the full frame of adjoining videos.

	   For 2 inputs, a default layout of "0_0|w0_0" (equivalent to
	   "grid=2x1") is set. In all other cases, a layout or a grid must be
	   set by the user. Either "grid" or "layout" can be specified at a
	   time.  Specifying both will result in an error.

       grid
	   Specify a fixed size grid of inputs.  This option is used to create
	   a fixed size grid of the input streams. Set the grid size in the
	   form "COLUMNSxROWS". There must be "ROWS * COLUMNS" input streams
	   and they will be arranged as a grid with "ROWS" rows and "COLUMNS"
	   columns. When using this option, each input stream within a row
	   must have the same height and all the rows must have the same
	   width.

	   If "grid" is set, then "inputs" option is ignored and is implicitly
	   set to "ROWS * COLUMNS".

	   For 2 inputs, a default grid of "2x1" (equivalent to
	   "layout=0_0|w0_0") is set. In all other cases, a layout or a grid
	   must be set by the user. Either "grid" or "layout" can be specified
	   at a time.  Specifying both will result in an error.

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

       fill
	   If set to valid color, all unused pixels will be filled with that
	   color.  By default fill is set to none, so it is disabled.

       Examples

       o   Display 4 inputs into 2x2 grid.

	   Layout:

		   input1(0, 0)  | input3(w0, 0)
		   input2(0, h0) | input4(w0, h0)

		   xstack=inputs=4:layout=0_0|0_h0|w0_0|w0_h0

	   Note that if inputs are of different sizes, gaps or overlaps may
	   occur.

       o   Display 4 inputs into 1x4 grid.

	   Layout:

		   input1(0, 0)
		   input2(0, h0)
		   input3(0, h0+h1)
		   input4(0, h0+h1+h2)

		   xstack=inputs=4:layout=0_0|0_h0|0_h0+h1|0_h0+h1+h2

	   Note that if inputs are of different widths, unused space will
	   appear.

       o   Display 9 inputs into 3x3 grid.

	   Layout:

		   input1(0, 0)       | input4(w0, 0)	   | input7(w0+w3, 0)
		   input2(0, h0)      | input5(w0, h0)	   | input8(w0+w3, h0)
		   input3(0, h0+h1)   | input6(w0, h0+h1)  | input9(w0+w3, h0+h1)

		   xstack=inputs=9:layout=0_0|0_h0|0_h0+h1|w0_0|w0_h0|w0_h0+h1|w0+w3_0|w0+w3_h0|w0+w3_h0+h1

	   Note that if inputs are of different sizes, gaps or overlaps may
	   occur.

       o   Display 16 inputs into 4x4 grid.

	   Layout:

		   input1(0, 0)       | input5(w0, 0)	    | input9 (w0+w4, 0)       | input13(w0+w4+w8, 0)
		   input2(0, h0)      | input6(w0, h0)	    | input10(w0+w4, h0)      | input14(w0+w4+w8, h0)
		   input3(0, h0+h1)   | input7(w0, h0+h1)   | input11(w0+w4, h0+h1)   | input15(w0+w4+w8, h0+h1)
		   input4(0, h0+h1+h2)| input8(w0, h0+h1+h2)| input12(w0+w4, h0+h1+h2)| input16(w0+w4+w8, h0+h1+h2)

		   xstack=inputs=16:layout=0_0|0_h0|0_h0+h1|0_h0+h1+h2|w0_0|w0_h0|w0_h0+h1|w0_h0+h1+h2|w0+w4_0|
		   w0+w4_h0|w0+w4_h0+h1|w0+w4_h0+h1+h2|w0+w4+w8_0|w0+w4+w8_h0|w0+w4+w8_h0+h1|w0+w4+w8_h0+h1+h2

	   Note that if inputs are of different sizes, gaps or overlaps may
	   occur.

   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. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   yadif_cuda
       Deinterlace the input video using the yadif algorithm, but implemented
       in CUDA so that it can work as part of a GPU accelerated pipeline with
       nvdec and/or nvenc.

       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. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   yaepblur
       Apply blur filter while preserving edges ("yaepblur" means "yet another
       edge preserving blur filter").  The algorithm is described in "J. S.
       Lee, Digital image enhancement and noise filtering by use of local
       statistics, IEEE Trans. Pattern Anal. Mach. Intell. PAMI-2, 1980."

       It accepts the following parameters:

       radius, r
	   Set the window radius. Default value is 3.

       planes, p
	   Set which planes to filter. Default is only the first plane.

       sigma, s
	   Set blur strength. Default value is 128.

       Commands

       This filter supports same commands as options.

   zoompan
       Apply Zoom & Pan effect.

       This filter accepts the following options:

       zoom, z
	   Set the zoom expression. Range is 1-10. 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.
	   Default is 90.

       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.

       in_time, it
	   The input timestamp expressed in seconds. It's NAN if the input
	   timestamp is unknown.

       out_time, time, ot
	   The output timestamp expressed in seconds.

       x
       y   Last calculated 'x' and 'y' position from 'x' and 'y' expression
	   for current input frame.

       px
       py  'x' and 'y' of last output frame of previous input frame or 0 when
	   there was not yet such frame (first input frame).

       zoom
	   Last calculated zoom from 'z' expression for current input frame.

       pzoom
	   Last calculated zoom of last output frame of previous input frame.

       duration
	   Number of output frames for current input frame. Calculated from
	   'd' expression for each input frame.

       pduration
	   number of output frames created for previous input frame

       a   Rational number: input width / input height

       sar sample aspect ratio

       dar display aspect ratio

       Examples

       o   Zoom in up to 1.5x and pan at same time to some spot near center of
	   picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

       o   Zoom in up to 1.5x and pan always at center of picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       o   Same as above but without pausing:

		   zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       o   Zoom in 2x into center of picture only for the first second of the
	   input video:

		   zoompan=z='if(between(in_time,0,1),2,1)':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>. To enable compilation of this
       filter, you need to configure FFmpeg with "--enable-libzimg".

       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.

       param_a
	   Parameter A for scaling filters. Parameter "b" for bicubic, and the
	   number of filter taps for lanczos.

       param_b
	   Parameter B for scaling filters. Parameter "c" for bicubic.

       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.

       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.

OPENCL VIDEO FILTERS
       Below is a description of the currently available OpenCL video filters.

       To enable compilation of these filters you need to configure FFmpeg
       with "--enable-opencl".

       Running OpenCL filters requires you to initialize a hardware device and
       to pass that device to all filters in any filter graph.

       -init_hw_device opencl[=name][:device[,key=value...]]
	   Initialise a new hardware device of type opencl called name, using
	   the given device parameters.

       -filter_hw_device name
	   Pass the hardware device called name to all filters in any filter
	   graph.

       For more detailed information see
       <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

       o   Example of choosing the first device on the second platform and
	   running avgblur_opencl filter with default parameters on it.

		   -init_hw_device opencl=gpu:1.0 -filter_hw_device gpu -i INPUT -vf "hwupload, avgblur_opencl, hwdownload" OUTPUT

       Since OpenCL filters are not able to access frame data in normal
       memory, all frame data needs to be uploaded(hwupload) to hardware
       surfaces connected to the appropriate device before being used and then
       downloaded(hwdownload) back to normal memory. Note that hwupload will
       upload to a surface with the same layout as the software frame, so it
       may be necessary to add a format filter immediately before to get the
       input into the right format and hwdownload does not support all formats
       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.

   avgblur_opencl
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.	Range is "[1, 1024]" and default value
	   is 1.

       planes
	   Set which planes to filter. Default value is 0xf, by which all
	   planes are processed.

       sizeY
	   Set vertical radius size. Range is "[1, 1024]" and default value is
	   0. If zero, "sizeX" value will be used.

       Example

       o   Apply average blur filter with horizontal and vertical size of 3,
	   setting each pixel of the output to the average value of the 7x7
	   region centered on it in the input. For pixels on the edges of the
	   image, the region does not extend beyond the image boundaries, and
	   so out-of-range coordinates are not used in the calculations.

		   -i INPUT -vf "hwupload, avgblur_opencl=3, hwdownload" OUTPUT

   boxblur_opencl
       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 boxblur filter, setting each pixel of the output to the average
       value of box-radiuses luma_radius, chroma_radius, alpha_radius for each
       plane respectively. The filter will apply luma_power, chroma_power,
       alpha_power times onto the corresponding plane. For pixels on the edges
       of the image, the radius does not extend beyond the image boundaries,
       and so out-of-range coordinates are not used in the calculations.

       o   Apply a boxblur filter with the luma, chroma, and alpha radius set
	   to 2 and luma, chroma, and alpha power set to 3. The filter will
	   run 3 times with box-radius set to 2 for every plane of the image.

		   -i INPUT -vf "hwupload, boxblur_opencl=luma_radius=2:luma_power=3, hwdownload" OUTPUT
		   -i INPUT -vf "hwupload, boxblur_opencl=2:3, hwdownload" OUTPUT

       o   Apply a boxblur filter with luma radius set to 2, luma_power to 1,
	   chroma_radius to 4, chroma_power to 5, alpha_radius to 3 and
	   alpha_power to 7.

	   For the luma plane, a 2x2 box radius will be run once.

	   For the chroma plane, a 4x4 box radius will be run 5 times.

	   For the alpha plane, a 3x3 box radius will be run 7 times.

		   -i INPUT -vf "hwupload, boxblur_opencl=2:1:4:5:3:7, hwdownload" OUTPUT

   colorkey_opencl
       RGB colorspace color keying.

       The filter accepts the following options:

       color
	   The color which will be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01 matches only the exact key color, while 1.0 matches
	   everything.

       blend
	   Blend percentage.

	   0.0 makes pixels either fully transparent, or not transparent at
	   all.

	   Higher values result in semi-transparent pixels, with a higher
	   transparency the more similar the pixels color is to the key color.

       Examples

       o   Make every semi-green pixel in the input transparent with some
	   slight blending:

		   -i INPUT -vf "hwupload, colorkey_opencl=green:0.3:0.1, hwdownload" OUTPUT

   convolution_opencl
       Apply convolution of 3x3, 5x5, 7x7 matrix.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set matrix for each plane.  Matrix is sequence of 9, 25 or 49
	   signed numbers.  Default value for each plane is "0 0 0 0 1 0 0 0
	   0".

       0rdiv
       1rdiv
       2rdiv
       3rdiv
	   Set multiplier for calculated value for each plane.	If unset or 0,
	   it will be sum of all matrix elements.  The option value must be a
	   float number greater or equal to 0.0. Default value is 1.0.

       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.  The option value must be a float number greater or equal
	   to 0.0. Default value is 0.0.

       Examples

       o   Apply sharpen:

		   -i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

       o   Apply blur:

		   -i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

       o   Apply edge enhance:

		   -i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

       o   Apply edge detect:

		   -i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

       o   Apply laplacian edge detector which includes diagonals:

		   -i INPUT -vf "hwupload, convolution_opencl=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, hwdownload" OUTPUT

       o   Apply emboss:

		   -i INPUT -vf "hwupload, convolution_opencl=-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, hwdownload" OUTPUT

   erosion_opencl
       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. Range is "[0, 65535]" and
	   default value is 65535.  If 0, plane will remain unchanged.

       coordinates
	   Flag which specifies the pixel to refer to.	Range is "[0, 255]"
	   and default value is 255, i.e. all eight pixels are used.

	   Flags to local 3x3 coordinates region centered on "x":

	       1 2 3

	       4 x 5

	       6 7 8

       Example

       o   Apply erosion filter with threshold0 set to 30, threshold1 set 40,
	   threshold2 set to 50 and coordinates set to 231, setting each pixel
	   of the output to the local minimum between pixels: 1, 2, 3, 6, 7, 8
	   of the 3x3 region centered on it in the input. If the difference
	   between input pixel and local minimum is more then threshold of the
	   corresponding plane, output pixel will be set to input pixel -
	   threshold of corresponding plane.

		   -i INPUT -vf "hwupload, erosion_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

   deshake_opencl
       Feature-point based video stabilization filter.

       The filter accepts the following options:

       tripod
	   Simulates a tripod by preventing any camera movement whatsoever
	   from the original frame. Defaults to 0.

       debug
	   Whether or not additional debug info should be displayed, both in
	   the processed output and in the console.

	   Note that in order to see console debug output you will also need
	   to pass "-v verbose" to ffmpeg.

	   Viewing point matches in the output video is only supported for RGB
	   input.

	   Defaults to 0.

       adaptive_crop
	   Whether or not to do a tiny bit of cropping at the borders to cut
	   down on the amount of mirrored pixels.

	   Defaults to 1.

       refine_features
	   Whether or not feature points should be refined at a sub-pixel
	   level.

	   This can be turned off for a slight performance gain at the cost of
	   precision.

	   Defaults to 1.

       smooth_strength
	   The strength of the smoothing applied to the camera path from 0.0
	   to 1.0.

	   1.0 is the maximum smoothing strength while values less than that
	   result in less smoothing.

	   0.0 causes the filter to adaptively choose a smoothing strength on
	   a per-frame basis.

	   Defaults to 0.0.

       smooth_window_multiplier
	   Controls the size of the smoothing window (the number of frames
	   buffered to determine motion information from).

	   The size of the smoothing window is determined by multiplying the
	   framerate of the video by this number.

	   Acceptable values range from 0.1 to 10.0.

	   Larger values increase the amount of motion data available for
	   determining how to smooth the camera path, potentially improving
	   smoothness, but also increase latency and memory usage.

	   Defaults to 2.0.

       Examples

       o   Stabilize a video with a fixed, medium smoothing strength:

		   -i INPUT -vf "hwupload, deshake_opencl=smooth_strength=0.5, hwdownload" OUTPUT

       o   Stabilize a video with debugging (both in console and in rendered
	   video):

		   -i INPUT -filter_complex "[0:v]format=rgba, hwupload, deshake_opencl=debug=1, hwdownload, format=rgba, format=yuv420p" -v verbose OUTPUT

   dilation_opencl
       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. Range is "[0, 65535]" and
	   default value is 65535.  If 0, plane will remain unchanged.

       coordinates
	   Flag which specifies the pixel to refer to.	Range is "[0, 255]"
	   and default value is 255, i.e. all eight pixels are used.

	   Flags to local 3x3 coordinates region centered on "x":

	       1 2 3

	       4 x 5

	       6 7 8

       Example

       o   Apply dilation filter with threshold0 set to 30, threshold1 set 40,
	   threshold2 set to 50 and coordinates set to 231, setting each pixel
	   of the output to the local maximum between pixels: 1, 2, 3, 6, 7, 8
	   of the 3x3 region centered on it in the input. If the difference
	   between input pixel and local maximum is more then threshold of the
	   corresponding plane, output pixel will be set to input pixel +
	   threshold of corresponding plane.

		   -i INPUT -vf "hwupload, dilation_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

   nlmeans_opencl
       Non-local Means denoise filter through OpenCL, this filter accepts same
       options as nlmeans.

   overlay_opencl
       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.  This filter requires same
       memory layout for all the inputs. So, format conversion may be needed.

       The filter accepts the following options:

       x   Set the x coordinate of the overlaid video on the main video.
	   Default value is 0.

       y   Set the y coordinate of the overlaid video on the main video.
	   Default value is 0.

       Examples

       o   Overlay an image LOGO at the top-left corner of the INPUT video.
	   Both inputs are yuv420p format.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

       o   The inputs have same memory layout for color channels , the overlay
	   has additional alpha plane, like INPUT is yuv420p, and the LOGO is
	   yuva420p.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

   pad_opencl
       Add paddings to the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following options:

       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.

       aspect
	   Pad to an 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

   prewitt_opencl
       Apply the Prewitt operator
       (<https://en.wikipedia.org/wiki/Prewitt_operator>) to input video
       stream.

       The filter accepts the following option:

       planes
	   Set which planes to filter. Default value is 0xf, by which all
	   planes are processed.

       scale
	   Set value which will be multiplied with filtered result.  Range is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set value which will be added to filtered result.  Range is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       o   Apply the Prewitt operator with scale set to 2 and delta set to 10.

		   -i INPUT -vf "hwupload, prewitt_opencl=scale=2:delta=10, hwdownload" OUTPUT

   program_opencl
       Filter video using an OpenCL program.

       source
	   OpenCL program source file.

       kernel
	   Kernel name in program.

       inputs
	   Number of inputs to the filter.  Defaults to 1.

       size, s
	   Size of output frames.  Defaults to the same as the first input.

       The "program_opencl" filter also supports the framesync options.

       The program source file must contain a kernel function with the given
       name, which will be run once for each plane of the output.  Each run on
       a plane gets enqueued as a separate 2D global NDRange with one work-
       item for each pixel to be generated.  The global ID offset for each
       work-item is therefore the coordinates of a pixel in the destination
       image.

       The kernel function needs to take the following arguments:

       o   Destination image, __write_only image2d_t.

	   This image will become the output; the kernel should write all of
	   it.

       o   Frame index, unsigned int.

	   This is a counter starting from zero and increasing by one for each
	   frame.

       o   Source images, __read_only image2d_t.

	   These are the most recent images on each input.  The kernel may
	   read from them to generate the output, but they can't be written
	   to.

       Example programs:

       o   Copy the input to the output (output must be the same size as the
	   input).

		   __kernel void copy(__write_only image2d_t destination,
				      unsigned int index,
				      __read_only  image2d_t source)
		   {
		       const sampler_t sampler = CLK_NORMALIZED_COORDS_FALSE;

		       int2 location = (int2)(get_global_id(0), get_global_id(1));

		       float4 value = read_imagef(source, sampler, location);

		       write_imagef(destination, location, value);
		   }

       o   Apply a simple transformation, rotating the input by an amount
	   increasing with the index counter.  Pixel values are linearly
	   interpolated by the sampler, and the output need not have the same
	   dimensions as the input.

		   __kernel void rotate_image(__write_only image2d_t dst,
					      unsigned int index,
					      __read_only  image2d_t src)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);

		       float angle = (float)index / 100.0f;

		       float2 dst_dim = convert_float2(get_image_dim(dst));
		       float2 src_dim = convert_float2(get_image_dim(src));

		       float2 dst_cen = dst_dim / 2.0f;
		       float2 src_cen = src_dim / 2.0f;

		       int2   dst_loc = (int2)(get_global_id(0), get_global_id(1));

		       float2 dst_pos = convert_float2(dst_loc) - dst_cen;
		       float2 src_pos = {
			   cos(angle) * dst_pos.x - sin(angle) * dst_pos.y,
			   sin(angle) * dst_pos.x + cos(angle) * dst_pos.y
		       };
		       src_pos = src_pos * src_dim / dst_dim;

		       float2 src_loc = src_pos + src_cen;

		       if (src_loc.x < 0.0f	 || src_loc.y < 0.0f ||
			   src_loc.x > src_dim.x || src_loc.y > src_dim.y)
			   write_imagef(dst, dst_loc, 0.5f);
		       else
			   write_imagef(dst, dst_loc, read_imagef(src, sampler, src_loc));
		   }

       o   Blend two inputs together, with the amount of each input used
	   varying with the index counter.

		   __kernel void blend_images(__write_only image2d_t dst,
					      unsigned int index,
					      __read_only  image2d_t src1,
					      __read_only  image2d_t src2)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);

		       float blend = (cos((float)index / 50.0f) + 1.0f) / 2.0f;

		       int2  dst_loc = (int2)(get_global_id(0), get_global_id(1));
		       int2 src1_loc = dst_loc * get_image_dim(src1) / get_image_dim(dst);
		       int2 src2_loc = dst_loc * get_image_dim(src2) / get_image_dim(dst);

		       float4 val1 = read_imagef(src1, sampler, src1_loc);
		       float4 val2 = read_imagef(src2, sampler, src2_loc);

		       write_imagef(dst, dst_loc, val1 * blend + val2 * (1.0f - blend));
		   }

   remap_opencl
       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 32bit float pixel format, single channel.

       interp
	   Specify interpolation used for remapping of pixels.	Allowed values
	   are "near" and "linear".  Default value is "linear".

       fill
	   Specify the color of the unmapped pixels. For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.
	   Default color is "black".

   roberts_opencl
       Apply the Roberts cross operator
       (<https://en.wikipedia.org/wiki/Roberts_cross>) to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes to filter. Default value is 0xf, by which all
	   planes are processed.

       scale
	   Set value which will be multiplied with filtered result.  Range is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set value which will be added to filtered result.  Range is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       o   Apply the Roberts cross operator with scale set to 2 and delta set
	   to 10

		   -i INPUT -vf "hwupload, roberts_opencl=scale=2:delta=10, hwdownload" OUTPUT

   sobel_opencl
       Apply the Sobel operator
       (<https://en.wikipedia.org/wiki/Sobel_operator>) to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes to filter. Default value is 0xf, by which all
	   planes are processed.

       scale
	   Set value which will be multiplied with filtered result.  Range is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set value which will be added to filtered result.  Range is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       o   Apply sobel operator with scale set to 2 and delta set to 10

		   -i INPUT -vf "hwupload, sobel_opencl=scale=2:delta=10, hwdownload" OUTPUT

   tonemap_opencl
       Perform HDR(PQ/HLG) to SDR conversion with tone-mapping.

       It accepts the following parameters:

       tonemap
	   Specify the tone-mapping operator to be used. Same as tonemap
	   option in tonemap.

       param
	   Tune the tone mapping algorithm. same as param option in tonemap.

       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 value is 0.5, and the algorithm here is a little
	   different from the cpu version tonemap currently. A setting of 0.0
	   disables this option.

       threshold
	   The tonemapping algorithm parameters is fine-tuned per each scene.
	   And a threshold is used to detect whether the scene has changed or
	   not. If the distance between the current frame average brightness
	   and the current running average exceeds a threshold value, we would
	   re-calculate scene average and peak brightness.  The default value
	   is 0.2.

       format
	   Specify the output pixel format.

	   Currently supported formats are:

	   p010
	   nv12
       range, r
	   Set the output color range.

	   Possible values are:

	   tv/mpeg
	   pc/jpeg

	   Default is same as input.

       primaries, p
	   Set the output color primaries.

	   Possible values are:

	   bt709
	   bt2020

	   Default is same as input.

       transfer, t
	   Set the output transfer characteristics.

	   Possible values are:

	   bt709
	   bt2020

	   Default is bt709.

       matrix, m
	   Set the output colorspace matrix.

	   Possible value are:

	   bt709
	   bt2020

	   Default is same as input.

       Example

       o   Convert HDR(PQ/HLG) video to bt2020-transfer-characteristic p010
	   format using linear operator.

		   -i INPUT -vf "format=p010,hwupload,tonemap_opencl=t=bt2020:tonemap=linear:format=p010,hwdownload,format=p010" OUTPUT

   unsharp_opencl
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
	   Set the luma matrix horizontal size.  Range is "[1, 23]" and
	   default value is 5.

       luma_msize_y, ly
	   Set the luma matrix vertical size.  Range is "[1, 23]" and default
	   value is 5.

       luma_amount, la
	   Set the luma effect strength.  Range is "[-10, 10]" and default
	   value is 1.0.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

       chroma_msize_x, cx
	   Set the chroma matrix horizontal size.  Range is "[1, 23]" and
	   default value is 5.

       chroma_msize_y, cy
	   Set the chroma matrix vertical size.  Range is "[1, 23]" and
	   default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength.  Range is "[-10, 10]" and default
	   value is 0.0.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       o   Apply strong luma sharpen effect:

		   -i INPUT -vf "hwupload, unsharp_opencl=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5, hwdownload" OUTPUT

       o   Apply a strong blur of both luma and chroma parameters:

		   -i INPUT -vf "hwupload, unsharp_opencl=7:7:-2:7:7:-2, hwdownload" OUTPUT

   xfade_opencl
       Cross fade two videos with custom transition effect by using OpenCL.

       It accepts the following options:

       transition
	   Set one of possible transition effects.

	   custom
	       Select custom transition effect, the actual transition
	       description will be picked from source and kernel options.

	   fade
	   wipeleft
	   wiperight
	   wipeup
	   wipedown
	   slideleft
	   slideright
	   slideup
	   slidedown
	       Default transition is fade.

       source
	   OpenCL program source file for custom transition.

       kernel
	   Set name of kernel to use for custom transition from program source
	   file.

       duration
	   Set duration of video transition.

       offset
	   Set time of start of transition relative to first video.

       The program source file must contain a kernel function with the given
       name, which will be run once for each plane of the output.  Each run on
       a plane gets enqueued as a separate 2D global NDRange with one work-
       item for each pixel to be generated.  The global ID offset for each
       work-item is therefore the coordinates of a pixel in the destination
       image.

       The kernel function needs to take the following arguments:

       o   Destination image, __write_only image2d_t.

	   This image will become the output; the kernel should write all of
	   it.

       o   First Source image, __read_only image2d_t.  Second Source image,
	   __read_only image2d_t.

	   These are the most recent images on each input.  The kernel may
	   read from them to generate the output, but they can't be written
	   to.

       o   Transition progress, float. This value is always between 0 and 1
	   inclusive.

       Example programs:

       o   Apply dots curtain transition effect:

		   __kernel void blend_images(__write_only image2d_t dst,
					      __read_only  image2d_t src1,
					      __read_only  image2d_t src2,
					      float progress)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);
		       int2  p = (int2)(get_global_id(0), get_global_id(1));
		       float2 rp = (float2)(get_global_id(0), get_global_id(1));
		       float2 dim = (float2)(get_image_dim(src1).x, get_image_dim(src1).y);
		       rp = rp / dim;

		       float2 dots = (float2)(20.0, 20.0);
		       float2 center = (float2)(0,0);
		       float2 unused;

		       float4 val1 = read_imagef(src1, sampler, p);
		       float4 val2 = read_imagef(src2, sampler, p);
		       bool next = distance(fract(rp * dots, &unused), (float2)(0.5, 0.5)) < (progress / distance(rp, center));

		       write_imagef(dst, p, next ? val1 : val2);
		   }

VAAPI VIDEO FILTERS
       VAAPI Video filters are usually used with VAAPI decoder and VAAPI
       encoder. Below is a description of VAAPI video filters.

       To enable compilation of these filters you need to configure FFmpeg
       with "--enable-vaapi".

       To use vaapi filters, you need to setup the vaapi device correctly. For
       more information, please read
       <https://trac.ffmpeg.org/wiki/Hardware/VAAPI>

   overlay_vaapi
       Overlay one video on the 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.

       The filter accepts the following options:

       x
       y   Set expressions for the x and y coordinates of the overlaid video
	   on the main video.

	   Default value is "0" for both expressions.

       w
       h   Set expressions for the width and height the overlaid video on the
	   main video.

	   Default values are 'overlay_iw' for 'w' and
	   'overlay_ih*w/overlay_iw' for 'h'.

	   The expressions can contain the following parameters:

	   main_w, W
	   main_h, H
	       The main input width and height.

	   overlay_iw
	   overlay_ih
	       The overlay input width and height.

	   overlay_w, w
	   overlay_h, h
	       The overlay output width and height.

	   overlay_x, x
	   overlay_y, y
	       Position of the overlay layer inside of main

       alpha
	   Set transparency of overlaid video. Allowed range is 0.0 to 1.0.
	   Higher value means lower transparency.  Default value is 1.0.

       eof_action
	   See framesync.

       shortest
	   See framesync.

       repeatlast
	   See framesync.

       This filter also supports the framesync options.

       Examples

       o   Overlay an image LOGO at the top-left corner of the INPUT video.
	   Both inputs for this filter are yuv420p format.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_vaapi" OUTPUT

       o   Overlay an image LOGO at the offset (200, 100) from the top-left
	   corner of the INPUT video.  The inputs have same memory layout for
	   color channels, the overlay has additional alpha plane, like INPUT
	   is yuv420p, and the LOGO is yuva420p.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_vaapi=x=200:y=100:w=400:h=300:alpha=1.0, hwdownload, format=nv12" OUTPUT

   tonemap_vaapi
       Perform HDR-to-SDR or HDR-to-HDR tone-mapping.  It currently only
       accepts HDR10 as input.

       It accepts the following parameters:

       format
	   Specify the output pixel format.

	   Default is nv12 for HDR-to-SDR tone-mapping and p010 for HDR-to-HDR
	   tone-mapping.

       primaries, p
	   Set the output color primaries.

	   Default is bt709 for HDR-to-SDR tone-mapping and same as input for
	   HDR-to-HDR tone-mapping.

       transfer, t
	   Set the output transfer characteristics.

	   Default is bt709 for HDR-to-SDR tone-mapping and same as input for
	   HDR-to-HDR tone-mapping.

       matrix, m
	   Set the output colorspace matrix.

	   Default is bt709 for HDR-to-SDR tone-mapping and same as input for
	   HDR-to-HDR tone-mapping.

       display
	   Set the output mastering display colour volume. It is given by a
	   '|'-separated list of two values, two values are space separated.
	   It set display primaries x & y in G, B, R order, then white point x
	   & y, the nominal minimum & maximum display luminances.

	   HDR-to-HDR tone-mapping will be performed when this option is set.

       light
	   Set the output content light level information. It accepts 2 space-
	   separated values, the first input is the maximum light level and
	   the second input is the maximum average light level.

	   It is ignored for HDR-to-SDR tone-mapping, and optional for HDR-to-
	   HDR tone-mapping.

       Example

       o   Convert HDR(HDR10) video to bt2020-transfer-characteristic p010
	   format

		   tonemap_vaapi=format=p010:t=bt2020-10

       o   Convert HDR video to HDR video

		   tonemap_vaapi=display=7500\ 3000|34000\ 16000|13250\ 34500|15635\ 16450|500\ 10000000

   hstack_vaapi
       Stack input videos horizontally.

       This is the VA-API variant of the hstack filter, each input stream may
       have different height, this filter will scale down/up each input stream
       while keeping the original aspect.

       It accepts the following options:

       inputs
	   See hstack.

       shortest
	   See hstack.

       height
	   Set height of output. If set to 0, this filter will set height of
	   output to height of the first input stream. Default value is 0.

   vstack_vaapi
       Stack input videos vertically.

       This is the VA-API variant of the vstack filter, each input stream may
       have different width, this filter will scale down/up each input stream
       while keeping the original aspect.

       It accepts the following options:

       inputs
	   See vstack.

       shortest
	   See vstack.

       width
	   Set width of output. If set to 0, this filter will set width of
	   output to width of the first input stream. Default value is 0.

   xstack_vaapi
       Stack video inputs into custom layout.

       This is the VA-API variant of the xstack filter,  each input stream may
       have different size, this filter will scale down/up each input stream
       to the given output size, or the size of the first input stream.

       It accepts the following options:

       inputs
	   See xstack.

       shortest
	   See xstack.

       layout
	   See xstack.	Moreover, this permits the user to supply output size
	   for each input stream.

		   xstack_vaapi=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

       grid
	   See xstack.

       grid_tile_size
	   Set output size for each input stream when grid is set. If this
	   option is not set, this filter will set output size by default to
	   the size of the first input stream. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       fill
	   See xstack.

   pad_vaapi
       Add paddings to the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following options:

       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.

       aspect
	   Pad to an 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

   drawbox_vaapi
       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.

       thickness, t
	   The expression which sets the thickness of the box edge.  A value
	   of "fill" will create a filled box. Default value is 3.

	   See below for the list of accepted constants.

       replace
	   With value 1, the pixels of the painted box will overwrite the
	   video's color and alpha pixels.  Default is 0, which composites the
	   box onto the input video.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       in_h, ih
       in_w, iw
	   The input width and height.

       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.

       Examples

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

		   drawbox

       o   Draw a box with color red and an opacity of 50%:

		   drawbox=10:20:200:60:red@0.5

	   The previous example can be specified as:

		   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       o   Fill the box with pink color:

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

       o   Draw a 2-pixel red 2.40:1 mask:

		   drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

VULKAN VIDEO FILTERS
       Below is a description of the currently available Vulkan video filters.

       To enable compilation of these filters you need to configure FFmpeg
       with "--enable-vulkan" and either "--enable-libglslang" or
       "--enable-libshaderc".

       Running Vulkan filters requires you to initialize a hardware device and
       to pass that device to all filters in any filter graph.

       -init_hw_device vulkan[=name][:device[,key=value...]]
	   Initialise a new hardware device of type vulkan called name, using
	   the given device parameters and options in key=value. The following
	   options are supported:

	   debug
	       Switches validation layers on if set to 1.

	   linear_images
	       Allocates linear images. Does not apply to decoding.

	   disable_multiplane
	       Disables multiplane images. Does not apply to decoding.

       -filter_hw_device name
	   Pass the hardware device called name to all filters in any filter
	   graph.

       For more detailed information see
       <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

       o   Example of choosing the first device and running nlmeans_vulkan
	   filter with default parameters on it.

		   -init_hw_device vulkan=vk:0 -filter_hw_device vk -i INPUT -vf "hwupload,nlmeans_vulkan,hwdownload" OUTPUT

       As Vulkan filters are not able to access frame data in normal memory,
       all frame data needs to be uploaded (hwupload) to hardware surfaces
       connected to the appropriate device before being used and then
       downloaded (hwdownload) back to normal memory. Note that hwupload will
       upload to a frame with the same layout as the software frame, so it may
       be necessary to add a format filter immediately before to get the input
       into the right format and hwdownload does not support all formats on
       the output - it is usually necessary to insert an additional format
       filter immediately following in the graph to get the output in a
       supported format.

   avgblur_vulkan
       Apply an average blur filter, implemented on the GPU using Vulkan.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.	Range is "[1, 32]" and default value
	   is 3.

       sizeY
	   Set vertical radius size. Range is "[1, 32]" and default value is
	   3.

       planes
	   Set which planes to filter. Default value is 0xf, by which all
	   planes are processed.

   blend_vulkan
       Blend two Vulkan 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.

       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:

	   normal
	   multiply

   bwdif_vulkan
       Deinterlacer using bwdif, the "Bob Weaver Deinterlacing Filter"
       algorithm, implemented on the GPU using Vulkan.

       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. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   chromaber_vulkan
       Apply an effect that emulates chromatic aberration. Works best with RGB
       inputs, but provides a similar effect with YCbCr inputs too.

       dist_x
	   Horizontal displacement multiplier. Each chroma pixel's position
	   will be multiplied by this amount, starting from the center of the
	   image. Default is 0.

       dist_y
	   Similarly, this sets the vertical displacement multiplier. Default
	   is 0.

   color_vulkan
       Video source that creates a Vulkan frame of a solid color.  Useful for
       benchmarking, or overlaying.

       It accepts the following parameters:

       color
	   The color to use. Either a name, or a hexadecimal value.  The
	   default value is "black".

       size
	   The size of the output frame. Default value is "1920x1080".

       rate
	   The framerate to output at. Default value is 60 frames per second.

       duration
	   The video duration. Default value is "-0.000001".

       sar The video signal aspect ratio. Default value is "1/1".

       format
	   The pixel format of the output Vulkan frames. Default value is
	   "yuv444p".

       out_range
	   Set the 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/unknown
	       Choose automatically.

	   jpeg/full/pc
	       Set full range (0-255 in case of 8-bit luma).

	   mpeg/limited/tv
	       Set "MPEG" range (16-235 in case of 8-bit luma).

   vflip_vulkan
       Flips an image vertically.

   hflip_vulkan
       Flips an image horizontally.

   flip_vulkan
       Flips an image along both the vertical and horizontal axis.

   gblur_vulkan
       Apply Gaussian blur filter on Vulkan frames.

       The filter accepts the following options:

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

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

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

       size
	   Set the kernel size along the horizontal axis. Default is 19.

       sizeV
	   Set the kernel size along the vertical axis. Default is 0, which
	   sets to use the same value as size.

   nlmeans_vulkan
       Denoise frames using Non-Local Means algorithm, implemented on the GPU
       using Vulkan.  Supports more pixel formats than nlmeans or
       nlmeans_opencl, including alpha channel support.

       The filter accepts the following options.

       s   Set denoising strength for all components. Default is 1.0. Must be
	   in range [1.0, 100.0].

       p   Set patch size for all planes. Default is 7. Must be odd number in
	   range [0, 99].

       r   Set research size. Default is 15. Must be odd number in range [0,
	   99].

       t   Set parallelism. Default is 36. Must be a number in the range [1,
	   168].  Larger values may speed up processing, at the cost of more
	   VRAM.  Lower values will slow it down, reducing VRAM usage.	Only
	   supported on GPUs with atomic float operations (RDNA3+, Ampere+).

       s0
       s1
       s2
       s3  Set denoising strength for a specific component. Default is 1,
	   equal to s.	Must be odd number in range [1, 100].

       p0
       p1
       p2
       p3  Set patch size for a specific component. Default is 7, equal to p.
	   Must be odd number in range [0, 99].

   overlay_vulkan
       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.  This filter requires all
       inputs to use the same pixel format. So, format conversion may be
       needed.

       The filter accepts the following options:

       x   Set the x coordinate of the overlaid video on the main video.
	   Default value is 0.

       y   Set the y coordinate of the overlaid video on the main video.
	   Default value is 0.

   transpose_vt
       Transpose rows with columns in the input video and optionally flip it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate by 90 degrees counterclockwise and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

	   hflip
	       Flip the input video horizontally.

	   vflip
	       Flip the input video vertically.

       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. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

   transpose_vulkan
       Transpose rows with columns in the input video and optionally flip it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate by 90 degrees counterclockwise and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

       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. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

QSV VIDEO FILTERS
       Below is a description of the currently available QSV video filters.

       To enable compilation of these filters you need to configure FFmpeg
       with "--enable-libmfx" or "--enable-libvpl".

       To use QSV filters, you need to setup the QSV device correctly. For
       more information, please read
       <https://trac.ffmpeg.org/wiki/Hardware/QuickSync>

   hstack_qsv
       Stack input videos horizontally.

       This is the QSV variant of the hstack filter, each input stream may
       have different height, this filter will scale down/up each input stream
       while keeping the original aspect.

       It accepts the following options:

       inputs
	   See hstack.

       shortest
	   See hstack.

       height
	   Set height of output. If set to 0, this filter will set height of
	   output to height of the first input stream. Default value is 0.

   vstack_qsv
       Stack input videos vertically.

       This is the QSV variant of the vstack filter, each input stream may
       have different width, this filter will scale down/up each input stream
       while keeping the original aspect.

       It accepts the following options:

       inputs
	   See vstack.

       shortest
	   See vstack.

       width
	   Set width of output. If set to 0, this filter will set width of
	   output to width of the first input stream. Default value is 0.

   xstack_qsv
       Stack video inputs into custom layout.

       This is the QSV variant of the xstack filter.

       It accepts the following options:

       inputs
	   See xstack.

       shortest
	   See xstack.

       layout
	   See xstack.	Moreover, this permits the user to supply output size
	   for each input stream.

		   xstack_qsv=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

       grid
	   See xstack.

       grid_tile_size
	   Set output size for each input stream when grid is set. If this
	   option is not set, this filter will set output size by default to
	   the size of the first input stream. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       fill
	   See xstack.

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/buffersrc.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.

       colorspace
	   A string representing the color space of the buffered video frames.
	   It may be a number corresponding to a color space, or a color space
	   name.

       range
	   A string representing the color range of the buffered video frames.
	   It may be a number corresponding to a color range, or a color range
	   name.

       pixel_aspect, sar
	   The sample (pixel) aspect ratio of the input video.

       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

   cellauto
       Create a pattern generated by an elementary cellular automaton.

       The initial state of the cellular automaton can be defined through the
       filename and pattern options. If such options are not specified an
       initial state is created randomly.

       At each new frame a new row in the video is filled with the result of
       the cellular automaton next generation. The behavior when the whole
       frame is filled is defined by the scroll option.

       This source accepts the following options:

       filename, f
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified file.  In the file, each non-whitespace
	   character is considered an alive cell, a newline will terminate the
	   row, and further characters in the file will be ignored.

       pattern, p
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified string.

	   Each non-whitespace character in the string is considered an alive
	   cell, a newline will terminate the row, and further characters in
	   the string will be ignored.

       rate, r
	   Set the video rate, that is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial cellular automaton row.
	   It is a floating point number value ranging from 0 to 1, defaults
	   to 1/PHI.

	   This option is ignored when a file or a pattern is specified.

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

       rule
	   Set the cellular automaton rule, it is a number ranging from 0 to
	   255.  Default value is 110.

       size, s
	   Set the size of the output video. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If filename or pattern is specified, the size is set by default to
	   the width of the specified initial state row, and the height is set
	   to width * PHI.

	   If size is set, it must contain the width of the specified pattern
	   string, and the specified pattern will be centered in the larger
	   row.

	   If a filename or a pattern string is not specified, the size value
	   defaults to "320x518" (used for a randomly generated initial
	   state).

       scroll
	   If set to 1, scroll the output upward when all the rows in the
	   output have been already filled. If set to 0, the new generated row
	   will be written over the top row just after the bottom row is
	   filled.  Defaults to 1.

       start_full, full
	   If set to 1, completely fill the output with generated rows before
	   outputting the first frame.	This is the default behavior, for
	   disabling set the value to 0.

       stitch
	   If set to 1, stitch the left and right row edges together.  This is
	   the default behavior, for disabling set the value to 0.

       Examples

       o   Read the initial state from pattern, and specify an output of size
	   200x400.

		   cellauto=f=pattern:s=200x400

       o   Generate a random initial row with a width of 200 cells, with a
	   fill ratio of 2/3:

		   cellauto=ratio=2/3:s=200x200

       o   Create a pattern generated by rule 18 starting by a single alive
	   cell centered on an initial row with width 100:

		   cellauto=p=@s=100x400:full=0:rule=18

       o   Specify a more elaborated initial pattern:

		   cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

   coreimagesrc
       Video source generated on GPU using Apple's CoreImage API on OSX.

       This video source is a specialized version of the coreimage video
       filter.	Use a core image generator at the beginning of the applied
       filterchain to generate the content.

       The coreimagesrc video source accepts the following options:

       list_generators
	   List all available generators along with all their respective
	   options as well as possible minimum and maximum values along with
	   the default values.

		   list_generators=true

       size, s
	   Specify the size of the sourced video. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils manual.
	   The default value is "320x240".

       rate, r
	   Specify the frame rate of the sourced video, as the number of
	   frames generated per second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default value
	   is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
	   Set the duration of the sourced video. See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       Additionally, all options of the coreimage video filter are accepted.
       A complete filterchain can be used for further processing of the
       generated input without CPU-HOST transfer. See coreimage documentation
       and examples for details.

       Examples

       o   Use CIQRCodeGenerator to create a QR code for the FFmpeg homepage,
	   given as complete and escaped command-line for Apple's standard
	   bash shell:

		   ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

	   This example is equivalent to the QRCode example of coreimage
	   without the need for a nullsrc video source.

   ddagrab
       Captures the Windows Desktop via Desktop Duplication API.

       The filter exclusively returns D3D11 Hardware Frames, for on-gpu
       encoding or processing. So an explicit hwdownload is needed for any
       kind of software processing.

       It accepts the following options:

       output_idx
	   DXGI Output Index to capture.

	   Usually corresponds to the index Windows has given the screen minus
	   one, so it's starting at 0.

	   Defaults to output 0.

       draw_mouse
	   Whether to draw the mouse cursor.

	   Defaults to true.

	   Only affects hardware cursors. If a game or application renders its
	   own cursor, it'll always be captured.

       framerate
	   Maximum framerate at which the desktop will be captured - the
	   interval between successive frames will not be smaller than the
	   inverse of the framerate. When dup_frames is true (the default) and
	   the desktop is not being updated often enough, the filter will
	   duplicate a previous frame. Note that there is no background
	   buffering going on, so when the filter is not polled often enough
	   then the actual inter-frame interval may be significantly larger.

	   Defaults to 30 FPS.

       video_size
	   Specify the size of the captured video.

	   Defaults to the full size of the screen.

	   Cropped from the bottom/right if smaller than screen size.

       offset_x
	   Horizontal offset of the captured video.

       offset_y
	   Vertical offset of the captured video.

       output_fmt
	   Desired filter output format.  Defaults to 8 Bit BGRA.

	   It accepts the following values:

	   auto
	       Passes all supported output formats to DDA and returns what DDA
	       decides to use.

	   8bit
	   bgra
	       8 Bit formats always work, and DDA will convert to them if
	       necessary.

	   10bit
	   x2bgr10
	       Filter initialization will fail if 10 bit format is requested
	       but unavailable.

       dup_frames
	   When this option is set to true (the default), the filter will
	   duplicate frames when the desktop has not been updated in order to
	   maintain approximately constant target framerate. When this option
	   is set to false, the filter will wait for the desktop to be updated
	   (inter-frame intervals may vary significantly in this case).

       Examples

       Capture primary screen and encode using nvenc:

	       ffmpeg -f lavfi -i ddagrab -c:v h264_nvenc -cq 18 output.mp4

       You can also skip the lavfi device and directly use the filter.	Also
       demonstrates downloading the frame and encoding with libx264.  Explicit
       output format specification is required in this case:

	       ffmpeg -filter_complex ddagrab=output_idx=1:framerate=60,hwdownload,format=bgra -c:v libx264 -crf 18 output.mp4

       If you want to capture only a subsection of the desktop, this can be
       achieved by specifying a smaller size and its offsets into the screen:

	       ddagrab=video_size=800x600:offset_x=100:offset_y=100

   gradients
       Generate several gradients.

       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".

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       c0, c1, c2, c3, c4, c5, c6, c7
	   Set 8 colors. Default values for colors is to pick random one.

       x0, y0, y0, y1
	   Set gradient line source and destination points. If negative or out
	   of range, random ones are picked.

       nb_colors, n
	   Set number of colors to use at once. Allowed range is from 2 to 8.
	   Default value is 2.

       seed
	   Set seed for picking gradient line points.

       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.

       speed
	   Set speed of gradients rotation.

       type, t
	   Set type of gradients.  Available values are:

	   linear
	   radial
	   circular
	   spiral
	   square

	   Default type is linear.

       Commands

       This source supports the some above options as commands.

   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 count 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
	   max_frames, m
	       Set the maximum number of frames generated for each test,
	       default value is 30.

	   Default value is "all", which will cycle through the list of all
	   tests.

       Some examples:

	       mptestsrc=t=dc_luma

       will generate a "dc_luma" test pattern.

   frei0r_src
       Provide a frei0r source.

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

       This source accepts the following parameters:

       size
	   The size of the video to generate. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       framerate
	   The framerate of the generated video. It may be a string of the
	   form num/den or a frame rate abbreviation.

       filter_name
	   The name to the frei0r source to load. For more information
	   regarding frei0r and how to set the parameters, read the frei0r
	   section in the video filters documentation.

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

       For example, to generate a frei0r partik0l source with size 200x200 and
       frame rate 10 which is overlaid on the overlay filter main input:

	       frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

   life
       Generate a life pattern.

       This source is based on a generalization of John Conway's life game.

       The sourced input represents a life grid, each pixel represents a cell
       which can be in one of two possible states, alive or dead. Every cell
       interacts with its eight neighbours, which are the cells that are
       horizontally, vertically, or diagonally adjacent.

       At each interaction the grid evolves according to the adopted rule,
       which specifies the number of neighbor alive cells which will make a
       cell stay alive or born. The rule option allows one to specify the rule
       to adopt.

       This source accepts the following options:

       filename, f
	   Set the file from which to read the initial grid state. In the
	   file, each non-whitespace character is considered an alive cell,
	   and newline is used to delimit the end of each row.

	   If this option is not specified, the initial grid is generated
	   randomly.

       rate, r
	   Set the video rate, that is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial random grid. It is a
	   floating point number value ranging from 0 to 1, defaults to 1/PHI.
	   It is ignored when a file is specified.

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

       rule
	   Set the life rule.

	   A rule can be specified with a code of the kind "SNS/BNB", where NS
	   and NB are sequences of numbers in the range 0-8, NS specifies the
	   number of alive neighbor cells which make a live cell stay alive,
	   and NB the number of alive neighbor cells which make a dead cell to
	   become alive (i.e. to "born").  "s" and "b" can be used in place of
	   "S" and "B", respectively.

	   Alternatively a rule can be specified by an 18-bits integer. The 9
	   high order bits are used to encode the next cell state if it is
	   alive for each number of neighbor alive cells, the low order bits
	   specify the rule for "borning" new cells. Higher order bits encode
	   for an higher number of neighbor cells.  For example the number
	   6153 = "(12<<9)+9" specifies a stay alive rule of 12 and a born
	   rule of 9, which corresponds to "S23/B03".

	   Default value is "S23/B3", which is the original Conway's game of
	   life rule, and will keep a cell alive if it has 2 or 3 neighbor
	   alive cells, and will born a new cell if there are three alive
	   cells around a dead cell.

       size, s
	   Set the size of the output video. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If filename is specified, the size is set by default to the same
	   size of the input file. If size is set, it must contain the size
	   specified in the input file, and the initial grid defined in that
	   file is centered in the larger resulting area.

	   If a filename is not specified, the size value defaults to
	   "320x240" (used for a randomly generated initial grid).

       stitch
	   If set to 1, stitch the left and right grid edges together, and the
	   top and bottom edges also. Defaults to 1.

       mold
	   Set cell mold speed. If set, a dead cell will go from death_color
	   to mold_color with a step of mold. mold can have a value from 0 to
	   255.

       life_color
	   Set the color of living (or new born) cells.

       death_color
	   Set the color of dead cells. If mold is set, this is the first
	   color used to represent a dead cell.

       mold_color
	   Set mold color, for definitely dead and moldy cells.

	   For the syntax of these 3 color options, check the "Color" section
	   in the ffmpeg-utils manual.

       Examples

       o   Read a grid from pattern, and center it on a grid of size 300x300
	   pixels:

		   life=f=pattern:s=300x300

       o   Generate a random grid of size 200x200, with a fill ratio of 2/3:

		   life=ratio=2/3:s=200x200

       o   Specify a custom rule for evolving a randomly generated grid:

		   life=rule=S14/B34

       o   Full example with slow death effect (mold) using ffplay:

		   ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

   qrencodesrc
       Generate a QR code using the libqrencode library (see
       <https://fukuchi.org/works/qrencode/>).

       To enable the compilation of this source, you need to configure FFmpeg
       with "--enable-libqrencode".

       The QR code is generated from the provided text or text pattern. The
       corresponding QR code is scaled and put in the video output according
       to the specified output size options.

       In case no text is specified, the QR code is not generated, but an
       empty colored output is returned instead.

       This source accepts the following options:

       qrcode_width, q
       padded_qrcode_width, Q
	   Specify an expression for the width of the rendered QR code, with
	   and without padding. The qrcode_width expression can reference the
	   value set by the padded_qrcode_width expression, and vice versa.
	   By default padded_qrcode_width is set to qrcode_width, meaning that
	   there is no padding.

	   These expressions are evaluated only once, when initializing the
	   source.  See the qrencode Expressions section for details.

	   Note that some of the constants are missing for the source (for
	   example the x or t or Xn), since they only makes sense when
	   evaluating the expression for each frame rather than at
	   initialization time.

       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".

       case_sensitive, cs
	   Instruct libqrencode to use case sensitive encoding. This is
	   enabled by default. This can be disabled to reduce the QR encoding
	   size.

       level, l
	   Specify the QR encoding error correction level. With an higher
	   correction level, the encoding size will increase but the code will
	   be more robust to corruption.  Lower level is L.

	   It accepts the following values:

	   L
	   M
	   Q
	   H
       expansion
	   Select how the input text is expanded. Can be either "none", or
	   "normal" (default). See the qrencode Text expansion section for
	   details.

       text
       textfile
	   Define the text to be rendered. In case neither is specified, no QR
	   is encoded (just an empty colored frame).

	   In case expansion is enabled, the text is treated as a text
	   template, using the qrencode expansion mechanism. See the qrencode
	   Text expansion section for details.

       background_color, bc
       foreground_color, fc
	   Set the QR code and background color. The default value of
	   foreground_color is "black", the default value of background_color
	   is "white".

	   For the syntax of the color options, check the "Color" section in
	   the ffmpeg-utils manual.

       Examples

       o   Generate a QR code encoding the specified text with the default
	   size:

		   qrencodesrc=text=www.ffmpeg.org

       o   Same as below, but select blue on pink colors:

		   qrencodesrc=text=www.ffmpeg.org:bc=pink:fc=blue

       o   Generate a QR code with width of 200 pixels and padding, making the
	   padded width 4/3 of the QR code width:

		   qrencodesrc=text=www.ffmpeg.org:q=200:Q=4/3*q

       o   Generate a QR code with padded width of 200 pixels and padding,
	   making the QR code width 3/4 of the padded width:

		   qrencodesrc=text=www.ffmpeg.org:Q=200:q=3/4*Q

       o   Generate a QR code encoding the frame number:

		   qrencodesrc=text=%{n}

       o   Generate a QR code encoding the GMT timestamp:

		   qrencodesrc=text=%{gmtime}

       o   Generate a QR code encoding the timestamp expressed as a float:

		   qrencodesrc=text=%{pts}

   allrgb, allyuv, color, colorchart, colorspectrum, haldclutsrc, nullsrc,
       pal75bars, pal100bars, 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 "colorchart" source provides a colors checker chart.

       The "colorspectrum" source provides a color spectrum 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 "pal75bars" source generates a color bars pattern, based on EBU PAL
       recommendations with 75% color levels.

       The "pal100bars" source generates a color bars pattern, based on EBU
       PAL recommendations with 100% color levels.

       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:

       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.

       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.

       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 "allrgb", "allyuv", and
	   "haldclutsrc" filters.

       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.

	   Since the frame rate is used as time base, all frames including the
	   last one will have their full duration. If the specified duration
	   is not a multiple of the frame duration, it will be rounded up.

       sar Set the sample aspect ratio of the sourced video.

       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).

       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.

       type
	   Set the type of the color spectrum, only available in the
	   "colorspectrum" source. Can be one of the following:

	   black
	   white
	   all
       patch_size
	   Set patch size of single color patch, only available in the
	   "colorchart" source. Default is "64x64".

       preset
	   Set colorchecker colors preset, only available in the "colorchart"
	   source.

	   Available values are:

	   reference
	   skintones

	   Default value is "reference".

       Examples

       o   Generate a video with a duration of 5.3 seconds, with size 176x144
	   and a frame rate of 10 frames per second:

		   testsrc=duration=5.3:size=qcif:rate=10

       o   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

       o   If the input content is to be ignored, "nullsrc" can be used. The
	   following command generates noise in the luma 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.

   openclsrc
       Generate video using an OpenCL program.

       source
	   OpenCL program source file.

       kernel
	   Kernel name in program.

       size, s
	   Size of frames to generate.	This must be set.

       format
	   Pixel format to use for the generated frames.  This must be set.

       rate, r
	   Number of frames generated every second.  Default value is '25'.

       For details of how the program loading works, see the program_opencl
       filter.

       Example programs:

       o   Generate a colour ramp by setting pixel values from the position of
	   the pixel in the output image.  (Note that this will work with all
	   pixel formats, but the generated output will not be the same.)

		   __kernel void ramp(__write_only image2d_t dst,
				      unsigned int index)
		   {
		       int2 loc = (int2)(get_global_id(0), get_global_id(1));

		       float4 val;
		       val.xy = val.zw = convert_float2(loc) / convert_float2(get_image_dim(dst));

		       write_imagef(dst, loc, val);
		   }

       o   Generate a Sierpinski carpet pattern, panning by a single pixel
	   each frame.

		   __kernel void sierpinski_carpet(__write_only image2d_t dst,
						   unsigned int index)
		   {
		       int2 loc = (int2)(get_global_id(0), get_global_id(1));

		       float4 value = 0.0f;
		       int x = loc.x + index;
		       int y = loc.y + index;
		       while (x > 0 || y > 0) {
			   if (x % 3 == 1 && y % 3 == 1) {
			       value = 1.0f;
			       break;
			   }
			   x /= 3;
			   y /= 3;
		       }

		       write_imagef(dst, loc, value);
		   }

   sierpinski
       Generate a Sierpinski carpet/triangle fractal, and randomly pan around.

       This source accepts the following options:

       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".

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       seed
	   Set seed which is used for random panning.

       jump
	   Set max jump for single pan destination. Allowed range is from 1 to
	   10000.

       type
	   Set fractal type, can be default "carpet" or "triangle".

   zoneplate
       Generate a zoneplate test video pattern.

       This source accepts the following options:

       size, s
	   Set frame size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual. Default value is
	   "320x240".

       rate, r
	   Set frame rate, expressed as number of frames per second. 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.

       sar Set the sample aspect ratio of the sourced video.

       precision
	   Set precision in bits for look-up table for sine calculations.
	   Default value is 10.  Allowed range is from 4 to 16.

       xo  Set horizontal axis offset for output signal. Default value is 0.

       yo  Set vertical axis offset for output signal. Default value is 0.

       to  Set time axis offset for output signal. Default value is 0.

       k0  Set 0-order, constant added to signal phase. Default value is 0.

       kx  Set 1-order, phase factor multiplier for horizontal axis. Default
	   value is 0.

       ky  Set 1-order, phase factor multiplier for vertical axis. Default
	   value is 0.

       kt  Set 1-order, phase factor multiplier for time axis. Default value
	   is 0.

       kxt, kyt, kxy
	   Set phase factor multipliers for combination of spatial and
	   temporal axis.  Default value is 0.

       kx2 Set 2-order, phase factor multiplier for horizontal axis. Default
	   value is 0.

       ky2 Set 2-order, phase factor multiplier for vertical axis. Default
	   value is 0.

       kt2 Set 2-order, phase factor multiplier for time axis. Default value
	   is 0.

       ku  Set the constant added to final phase to produce chroma-blue
	   component of signal.  Default value is 0.

       kv  Set the constant added to final phase to produce chroma-red
	   component of signal.  Default value is 0.

       Commands

       This source supports the some above options as commands.

       Examples

       o   Generate horizontal color sine sweep:

		   zoneplate=ku=512:kv=0:kt2=0:kx2=256:s=wvga:xo=-426:kt=11

       o   Generate vertical color sine sweep:

		   zoneplate=ku=512:kv=0:kt2=0:ky2=156:s=wvga:yo=-240:kt=11

       o   Generate circular zone-plate:

		   zoneplate=ku=512:kv=100:kt2=0:ky2=256:kx2=556:s=wvga:yo=0:kt=11

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.

   a3dscope
       Convert input audio to 3d scope video output.

       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 "hd720".

       fov Set the camera field of view. Default is 90 degrees.  Allowed range
	   is from 40 to 150.

       roll
	   Set the camera roll.

       pitch
	   Set the camera pitch.

       yaw Set the camera yaw.

       xzoom
	   Set the camera zoom on X-axis.

       yzoom
	   Set the camera zoom on Y-axis.

       zzoom
	   Set the camera zoom on Z-axis.

       xpos
	   Set the camera position on X-axis.

       ypos
	   Set the camera position on Y-axis.

       zpos
	   Set the camera position on Z-axis.

       length
	   Set the length of displayed audio waves in number of frames.

       Commands

       Filter supports the some above options as commands.

   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.

       mode, m
	   Set output mode. Can be "bars" or "trace". Default is "bars".

   adrawgraph
       Draw a graph using input audio metadata.

       See drawgraph

   agraphmonitor
       See graphmonitor.

   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.  Default 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".

       hmode
	   Set histogram mode.

	   It accepts the following values:

	   abs Use absolute values of samples.

	   sign
	       Use untouched values of samples.

	   Default is "abs".

   aphasemeter
       Measures phase of input audio, which is exported as metadata
       "lavfi.aphasemeter.phase", representing mean phase of current audio
       frame. A video output can also be produced and is enabled by default.
       The audio is passed through as first output.

       Audio will be rematrixed to stereo if it has a different channel
       layout. Phase value is in range "[-1, 1]" where "-1" means left and
       right channels are completely out of phase and 1 means channels are in
       phase.

       The filter accepts the following options, all related to its video
       output:

       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.

       phasing detection

       The filter also detects out of phase and mono sequences in stereo
       streams.  It logs the sequence start, end and duration when it lasts
       longer or as long as the minimum set.

       The filter accepts the following options for this detection:

       phasing
	   Enable mono and out of phase detection. Default is disabled.

       tolerance, t
	   Set phase tolerance for mono detection, in amplitude ratio. Default
	   is 0.  Allowed range is "[0, 1]".

       angle, a
	   Set angle threshold for out of phase detection, in degree. Default
	   is 170.  Allowed range is "[90, 180]".

       duration, d
	   Set mono or out of phase duration until notification, expressed in
	   seconds. Default is 2.

       Examples

       o   Complete example with ffmpeg to detect 1 second of mono with 0.001
	   phase tolerance:

		   ffmpeg -i stereo.wav -af aphasemeter=video=0:phasing=1:duration=1:tolerance=0.001 -f null -

   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 monaural 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.

	   aaline
	       Draw anti-aliased 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.

       swap
	   Swap left channel axis with right channel axis.

       mirror
	   Mirror axis.

	   none
	       No mirror.

	   x   Mirror only x axis.

	   y   Mirror only y axis.

	   xy  Mirror both axis.

       Examples

       o   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]'

       Commands

       This filter supports the all above options as commands except options
       "size" and "rate".

   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

       o   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

       o   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

       o   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.

       Commands

       This filter supports the following commands:

       next
	   Close the current segment and step to the next one

   ebur128
       EBU R128 scanner filter. This filter takes an audio stream and analyzes
       its loudness level. 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 can only analyze streams which have sample format is double-
       precision floating point. The input stream will be converted to this
       specification, if needed. Users may need to insert aformat and/or
       aresample filters after this filter to obtain the original parameters.

       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),
       but can optionally be configured to instead display short-term loudness
       (see gauge).

       The green area marks a  +/- 1LU target range around the target loudness
       (-23LUFS by default, unless modified through target).

       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:

	   quiet
	       logging disabled

	   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.

       target
	   Set a specific target level (in LUFS) used as relative zero in the
	   visualization.  This parameter is optional and has a default value
	   of -23LUFS as specified by EBU R128. However, material published
	   online may prefer a level of -16LUFS (e.g. for use with podcasts or
	   video platforms).

       gauge
	   Set the value displayed by the gauge. Valid values are "momentary"
	   and s "shortterm". By default the momentary value will be used, but
	   in certain scenarios it may be more useful to observe the short
	   term value instead (e.g.  live mixing).

       scale
	   Sets the display scale for the loudness. Valid parameters are
	   "absolute" (in LUFS) or "relative" (LU) relative to the target.
	   This only affects the video output, not the summary or continuous
	   log output.

       integrated
	   Read-only exported value for measured integrated loudness, in LUFS.

       range
	   Read-only exported value for measured loudness range, in LU.

       lra_low
	   Read-only exported value for measured LRA low, in LUFS.

       lra_high
	   Read-only exported value for measured LRA high, in LUFS.

       sample_peak
	   Read-only exported value for measured sample peak, in dBFS.

       true_peak
	   Read-only exported value for measured true peak, in dBFS.

       Examples

       o   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]"

       o   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.

       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.

       Examples

       o   Interleave frames belonging to different streams using ffmpeg:

		   ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

       o   Add flickering blur effect:

		   select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

   latency, alatency
       Measure filtering latency.

       Report previous filter filtering latency, delay in number of audio
       samples for audio filters or number of video frames for video filters.

       On end of input stream, filter will report min and max measured latency
       for previous running filter in filtergraph.

   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.

	   ends_with
	       Values are interpreted as strings, returns true if metadata
	       value ends with the "value" option string.

       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, FRAMEVAL
	       Float representation of "value" from metadata key.

	   VALUE2, USERVAL
	       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.

       direct
	   Reduces buffering in print mode when output is written to a URL set
	   using file.

       Examples

       o   Print all metadata values for frames with key
	   "lavfi.signalstats.YDIF" with values between 0 and 1.

		   signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

       o   Print silencedetect output to file metadata.txt.

		   silencedetect,ametadata=mode=print:file=metadata.txt

       o   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.

       speed
	   Speed factor for processing. The value must be a float larger than
	   zero.  Values larger than 1.0 will result in faster than realtime
	   processing, smaller will slow processing down. The limit is
	   automatically adapted accordingly. Default is 1.0.

	   A processing speed faster than what is possible without these
	   filters cannot be achieved.

       Commands

       Both filters supports the all above options as commands.

   segment, asegment
       Split single input stream into multiple streams.

       This filter does opposite of concat filters.

       "segment" works on video frames, "asegment" on audio samples.

       This filter accepts the following options:

       timestamps
	   Timestamps of output segments separated by '|'. The first segment
	   will run from the beginning of the input stream. The last segment
	   will run until the end of the input stream

       frames, samples
	   Exact frame/sample count to split the segments.

       In all cases, prefixing an each segment with '+' will make it relative
       to the previous segment.

       Examples

       o   Split input audio stream into three output audio streams, starting
	   at start of input audio stream and storing that in 1st output audio
	   stream, then following at 60th second and storing than in 2nd
	   output audio stream, and last after 150th second of input audio
	   stream store in 3rd output audio stream:

		   asegment=timestamps="60|150"

   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 frame, expressed
	   in TB units. It's NAN if undefined.

       t   The PTS of the filtered frame, expressed in seconds. It's NAN if
	   undefined.

       prev_pts
	   The PTS of the previously filtered frame. It's NAN if undefined.

       prev_selected_pts
	   The PTS of the last previously filtered frame. It's NAN if
	   undefined.

       prev_selected_t
	   The PTS of the last previously selected frame, expressed in
	   seconds. It's NAN if undefined.

       start_pts
	   The first PTS in the stream which is not NAN. It remains NAN if not
	   found.

       start_t
	   The first PTS, in seconds, in the stream which is not NAN. It
	   remains NAN if not found.

       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);
	   deprecated, do not use

       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.

       iw (video only)
	   Represents the width of the input video frame.

       ih (video only)
	   Represents the height of the input video frame.

       The default value of the select expression is "1".

       Examples

       o   Select all frames in input:

		   select

	   The example above is the same as:

		   select=1

       o   Skip all frames:

		   select=0

       o   Select only I-frames:

		   select='eq(pict_type\,I)'

       o   Select one frame every 100:

		   select='not(mod(n\,100))'

       o   Select only frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)

       o   Select only I-frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)*eq(pict_type\,I)

       o   Select frames with a minimum distance of 10 seconds:

		   select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

       o   Use aselect to select only audio frames with samples number > 100:

		   aselect='gt(samples_n\,100)'

       o   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.

       o   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

       o   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.

       expr
	   The command ARG is interpreted as expression and result of
	   expression is passed as ARG.

	   The expression is evaluated through the eval API and can contain
	   the following constants:

	   POS Original position in the file of the frame, or undefined if
	       undefined for the current frame. Deprecated, do not use.

	   PTS The presentation timestamp in input.

	   N   The count of the input frame for video or audio, starting from
	       0.

	   T   The time in seconds of the current frame.

	   TS  The start time in seconds of the current command interval.

	   TE  The end time in seconds of the current command interval.

	   TI  The interpolated time of the current command interval, TI = (T
	       - TS) / (TE - TS).

	   W   The video frame width.

	   H   The video frame height.

       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

       o   Specify audio tempo change at second 4:

		   asendcmd=c='4.0 atempo tempo 1.5',atempo

       o   Target a specific filter instance:

		   asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my

       o   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, FR
	   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; deprecated, do not use

       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.

       T_CHANGE
	   Time of the first frame after command was applied or time of the
	   first frame if no commands.

       Examples

       o   Start counting PTS from zero

		   setpts=PTS-STARTPTS

       o   Apply fast motion effect:

		   setpts=0.5*PTS

       o   Apply slow motion effect:

		   setpts=2.0*PTS

       o   Set fixed rate of 25 frames per second:

		   setpts=N/(25*TB)

       o   Apply a random jitter effect of +/-100 TB units:

		   setpts=PTS+randomi(0, -100\,100)

       o   Set fixed rate 25 fps with some jitter:

		   setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

       o   Apply an offset of 10 seconds to the input PTS:

		   setpts=PTS+10/TB

       o   Generate timestamps from a "live source" and rebase onto the
	   current timebase:

		   setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

       o   Generate timestamps by counting samples:

		   asetpts=N/SR/TB

       Commands

       Both filters support all above options as commands.

   setrange
       Force color range for the output video frame.

       The "setrange" filter marks the color range property 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.

       The filter accepts the following options:

       range
	   Available values are:

	   auto
	       Keep the same color range property.

	   unspecified, unknown
	       Set the color range as unspecified.

	   limited, tv, mpeg
	       Set the color range as limited.

	   full, pc, jpeg
	       Set the color range as full.

   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

       o   Set the timebase to 1/25:

		   settb=expr=1/25

       o   Set the timebase to 1/10:

		   settb=expr=0.1

       o   Set the timebase to 1001/1000:

		   settb=1+0.001

       o   Set the timebase to 2*intb:

		   settb=2*intb

       o   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

       o   Playing audio while showing the spectrum:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

       o   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]'

       o   Playing at 1280x720:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'

       o   Disable sonogram display:

		   sono_h=0

       o   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]'

       o   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]'

       o   Custom volume:

		   bar_v=10:sono_v=bar_v*a_weighting(f)

       o   Custom gamma, now spectrum is linear to the amplitude.

		   bar_g=2:sono_g=2

       o   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)))'

       o   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

       o   Custom font using fontconfig:

		   font='Courier New,Monospace,mono|bold'

       o   Custom frequency range with custom axis using image file:

		   axisfile=myaxis.png:basefreq=40:endfreq=10000

   showcwt
       Convert input audio to video output representing frequency spectrum
       using Continuous Wavelet Transform and Morlet wavelet.

       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".

       rate, r
	   Set the output frame rate. Default value is 25.

       scale
	   Set the frequency scale used. Allowed values are:

	   linear
	   log
	   bark
	   mel
	   erbs
	   sqrt
	   cbrt
	   qdrt
	   fm

	   Default value is "linear".

       iscale
	   Set the intensity scale used. Allowed values are:

	   linear
	   log
	   sqrt
	   cbrt
	   qdrt

	   Default value is "log".

       min Set the minimum frequency that will be used in output.  Default is
	   20 Hz.

       max Set the maximum frequency that will be used in output.  Default is
	   20000 Hz. The real frequency upper limit depends on input audio's
	   sample rate and such will be enforced on this value when it is set
	   to value greater than Nyquist frequency.

       imin
	   Set the minimum intensity that will be used in output.

       imax
	   Set the maximum intensity that will be used in output.

       logb
	   Set the logarithmic basis for brightness strength when mapping
	   calculated magnitude values to pixel values.  Allowed range is from
	   0 to 1.  Default value is 0.0001.

       deviation
	   Set the frequency deviation.  Lower values than 1 are more
	   frequency oriented, while higher values than 1 are more time
	   oriented.  Allowed range is from 0 to 10.  Default value is 1.

       pps Set the number of pixel output per each second in one row.  Allowed
	   range is from 1 to 1024.  Default value is 64.

       mode
	   Set the output visual mode. Allowed values are:

	   magnitude
	       Show magnitude.

	   phase
	       Show only phase.

	   magphase
	       Show combination of magnitude and phase.  Magnitude is mapped
	       to brightness and phase to color.

	   channel
	       Show unique color per channel magnitude.

	   stereo
	       Show unique color per stereo difference.

	   Default value is "magnitude".

       slide
	   Set the output slide method. Allowed values are:

	   replace
	   scroll
	   frame
       direction
	   Set the direction method for output slide method. Allowed values
	   are:

	   lr  Direction from left to right.

	   rl  Direction from right to left.

	   ud  Direction from up to down.

	   du  Direction from down to up.

       bar Set the ratio of bargraph display to display size. Default is 0.

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

   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".

       rate, r
	   Set video rate. Default is 25.

       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. Allowed range is from 16 to 65536.

	   Default is 2048

       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
	   bohman
	   kaiser

	   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.

       data
	   Set data display mode.

	   It accepts the following values:

	   magnitude
	   phase
	   delay

	   Default is "magnitude".

       channels
	   Set channels to use when processing audio. By default all are
	   processed.

   showspatial
       Convert stereo input audio to a video output, representing the spatial
       relationship between two channels.

       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 "512x512".

       win_size
	   Set window size. Allowed range is from 1024 to 65536. Default size
	   is 4096.

       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
	   bohman
	   kaiser

	   Default value is "hann".

       rate, r
	   Set output framerate.

   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

	   lreplace
	       the samples start again on the right when they reach the left

	   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

	   magma
	       each channel is displayed using the magma color scheme

	   green
	       each channel is displayed using the green color scheme

	   viridis
	       each channel is displayed using the viridis color scheme

	   plasma
	       each channel is displayed using the plasma color scheme

	   cividis
	       each channel is displayed using the cividis color scheme

	   terrain
	       each channel is displayed using the terrain 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.

       fscale
	   Specify frequency scale.

	   It accepts the following values:

	   lin linear

	   log logarithmic

	   Default value is lin.

       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
	   bohman
	   kaiser

	   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",
	   or unwrapped phase: "uphase".

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

       start
	   Set start frequency from which to display spectrogram. Default is
	   0.

       stop
	   Set stop frequency to which to display spectrogram. Default is 0.

       fps Set upper frame rate limit. Default is "auto", unlimited.

       legend
	   Draw time and frequency axes and legends. Default is disabled.

       drange
	   Set dynamic range used to calculate intensity color values. Default
	   is 120 dBFS.  Allowed range is from 10 to 200.

       limit
	   Set upper limit of input audio samples volume in dBFS. Default is 0
	   dBFS.  Allowed range is from -100 to 100.

       opacity
	   Set opacity strength when using pixel format output with alpha
	   component.

       The usage is very similar to the showwaves filter; see the examples in
       that section.

       Examples

       o   Large window with logarithmic color scaling:

		   showspectrum=s=1280x480:scale=log

       o   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

	   magma
	       each channel is displayed using the magma color scheme

	   green
	       each channel is displayed using the green color scheme

	   viridis
	       each channel is displayed using the viridis color scheme

	   plasma
	       each channel is displayed using the plasma color scheme

	   cividis
	       each channel is displayed using the cividis color scheme

	   terrain
	       each channel is displayed using the terrain 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.

       fscale
	   Specify frequency scale.

	   It accepts the following values:

	   lin linear

	   log logarithmic

	   Default value is lin.

       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
	   bohman
	   kaiser

	   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.

       start
	   Set start frequency from which to display spectrogram. Default is
	   0.

       stop
	   Set stop frequency to which to display spectrogram. Default is 0.

       drange
	   Set dynamic range used to calculate intensity color values. Default
	   is 120 dBFS.  Allowed range is from 10 to 200.

       limit
	   Set upper limit of input audio samples volume in dBFS. Default is 0
	   dBFS.  Allowed range is from -100 to 100.

       opacity
	   Set opacity strength when using pixel format output with alpha
	   component.

       Examples

       o   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, 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: "h" or vertical: "v", default
	   is "h".

       s   Set step size, allowed range is [0, 5]. Default is 0, which means
	   step is disabled.

       p   Set background opacity, allowed range is [0, 1]. Default is 0.

       m   Set metering mode, can be peak: "p" or rms: "r", default is "p".

       ds  Set display scale, can be linear: "lin" or log: "log", default is
	   "lin".

       dm  In second.  If set to > 0., display a line for the max level in the
	   previous seconds.  default is disabled: 0.

       dmc The color of the max line. Use when "dm" option is set to > 0.
	   default is: "orange"

   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.

       draw
	   Set the draw mode. This is mostly useful to set for high n.

	   Available values are:

	   scale
	       Scale pixel values for each drawn sample.

	   full
	       Draw every sample directly.

	   Default value is "scale".

       Examples

       o   Output the input file audio and the corresponding video
	   representation at the same time:

		   amovie=a.mp3,asplit[out0],showwaves[out1]

       o   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.

       draw
	   Set the draw mode.

	   Available values are:

	   scale
	       Scale pixel values for each drawn sample.

	   full
	       Draw every sample directly.

	   Default value is "scale".

       filter
	   Set the filter mode.

	   Available values are:

	   average
	       Use average samples values for each drawn sample.

	   peak
	       Use peak samples values for each drawn sample.

	   Default value is "average".

       Examples

       o   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
       Synthesize 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 it's 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

       o   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

       o   Create two separate outputs from the same input:

		   [in] split [out0][out1]

       o   To create 3 or more outputs, you need to specify the number of
	   outputs, like in:

		   [in] asplit=3 [out0][out1][out2]

       o   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];

       o   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. Both are
       capable to send messages to any filter type.

       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
       (or the abbreviation "b") option.  Default value of this option is
       tcp://localhost:5555. You may want to alter this value to your needs,
       but do not forget to escape any ':' signs (see filtergraph escaping).

       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. The default filter
       instance name uses the pattern Parsed_<filter_name>_<index>, but you
       can override this by using the filter_name@id syntax (see Filtergraph
       syntax).

       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.	In this
       example the last overlay filter has an instance name. All other filters
       will have default instance names.

	       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@my=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

       To change the position of the right side:

	       echo overlay@my x 150 | 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.

   avsynctest
       Generate an Audio/Video Sync Test.

       Generated stream periodically shows flash video frame and emits beep in
       audio.  Useful to inspect A/V sync issues.

       It accepts the following options:

       size, s
	   Set output video size. Default value is "hd720".

       framerate, fr
	   Set output video frame rate. Default value is 30.

       samplerate, sr
	   Set output audio sample rate. Default value is 44100.

       amplitude, a
	   Set output audio beep amplitude. Default value is 0.7.

       period, p
	   Set output audio beep period in seconds. Default value is 3.

       delay, dl
	   Set output video flash delay in number of frames. Default value is
	   0.

       cycle, c
	   Enable cycling of video delays, by default is disabled.

       duration, d
	   Set stream output duration. By default duration is unlimited.

       fg, bg, ag
	   Set foreground/background/additional color.

       Commands

       This source supports the some above options as commands.

   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.

       dec_threads
	   Specifies the number of threads for decoding

       format_opts
	   Specify format options for the opened file. Format options can be
	   specified as a list of key=value pairs separated by ':'. The
	   following example shows how to add protocol_whitelist and
	   protocol_blacklist options:

		   ffplay -f lavfi
		   "movie=filename='1.sdp':format_opts='protocol_whitelist=file,rtp,udp\:protocol_blacklist=http'"

       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

       o   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]

       o   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]

       o   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

	   o   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.

	   o   timestamp: Timestamp in AVStream.time_base units or, if no
	       stream is specified, in AV_TIME_BASE units.

	   o   flags: Flags which select direction and seeking mode.

       get_duration
	   Get movie duration in AV_TIME_BASE units.

EXTERNAL LIBRARIES
       FFmpeg can be hooked up with a number of external libraries to add
       support for more formats. None of them are used by default, their use
       has to be explicitly requested by passing the appropriate flags to
       ./configure.

   Alliance for Open Media (AOM)
       FFmpeg can make use of the AOM library for AV1 decoding and encoding.

       Go to <http://aomedia.org/> and follow the instructions for installing
       the library. Then pass "--enable-libaom" to configure to enable it.

   AMD AMF/VCE
       FFmpeg can use the AMD Advanced Media Framework library for accelerated
       H.264 and HEVC(only windows) encoding on hardware with Video Coding
       Engine (VCE).

       To enable support you must obtain the AMF framework header
       files(version 1.4.9+) from
       <https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git>.

       Create an "AMF/" directory in the system include path.  Copy the
       contents of "AMF/amf/public/include/" into that directory.  Then
       configure FFmpeg with "--enable-amf".

       Initialization of amf encoder occurs in this order: 1) trying to
       initialize through dx11(only windows) 2) trying to initialize through
       dx9(only windows) 3) trying to initialize through vulkan

       To use h.264(AMD VCE) encoder on linux amdgru-pro version 19.20+ and
       amf-amdgpu-pro package(amdgru-pro contains, but does not install
       automatically) are required.

       This driver can be installed using amdgpu-pro-install script in
       official amd driver archive.

   AviSynth
       FFmpeg can read AviSynth scripts as input. To enable support, pass
       "--enable-avisynth" to configure after installing the headers provided
       by <https://github.com/AviSynth/AviSynthPlus>.  AviSynth+ can be
       configured to install only the headers by either passing
       "-DHEADERS_ONLY:bool=on" to the normal CMake-based build system, or by
       using the supplied "GNUmakefile".

       For Windows, supported AviSynth variants are <http://avisynth.nl> for
       32-bit builds and <http://avisynth.nl/index.php/AviSynth+> for 32-bit
       and 64-bit builds.

       For Linux, macOS, and BSD, the only supported AviSynth variant is
       <https://github.com/AviSynth/AviSynthPlus>, starting with version 3.5.

	   In 2016, AviSynth+ added support for building with GCC. However,
	   due to the eccentricities of Windows' calling conventions, 32-bit
	   GCC builds of AviSynth+ are not compatible with typical 32-bit
	   builds of FFmpeg.

	   By default, FFmpeg assumes compatibility with 32-bit MSVC builds of
	   AviSynth+ since that is the most widely-used and entrenched build
	   configuration.  Users can override this and enable support for
	   32-bit GCC builds of AviSynth+ by passing "-DAVSC_WIN32_GCC32" to
	   "--extra-cflags" when configuring FFmpeg.

	   64-bit builds of FFmpeg are not affected, and can use either MSVC
	   or GCC builds of AviSynth+ without any special flags.

	   AviSynth(+) is loaded dynamically.  Distributors can build FFmpeg
	   with "--enable-avisynth", and the binaries will work regardless of
	   the end user having AviSynth installed.  If/when an end user would
	   like to use AviSynth scripts, then they can install AviSynth(+) and
	   FFmpeg will be able to find and use it to open scripts.

   Chromaprint
       FFmpeg can make use of the Chromaprint library for generating audio
       fingerprints.  Pass "--enable-chromaprint" to configure to enable it.
       See <https://acoustid.org/chromaprint>.

   codec2
       FFmpeg can make use of the codec2 library for codec2 decoding and
       encoding.  There is currently no native decoder, so libcodec2 must be
       used for decoding.

       Go to <http://freedv.org/>, download "Codec 2 source archive".  Build
       and install using CMake. Debian users can install the libcodec2-dev
       package instead.  Once libcodec2 is installed you can pass
       "--enable-libcodec2" to configure to enable it.

       The easiest way to use codec2 is with .c2 files, since they contain the
       mode information required for decoding.	To encode such a file, use a
       .c2 file extension and give the libcodec2 encoder the -mode option:
       "ffmpeg -i input.wav -mode 700C output.c2".  Playback is as simple as
       "ffplay output.c2".  For a list of supported modes, run "ffmpeg -h
       encoder=libcodec2".  Raw codec2 files are also supported.  To make
       sense of them the mode in use needs to be specified as a format option:
       "ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav".

   dav1d
       FFmpeg can make use of the dav1d library for AV1 video decoding.

       Go to <https://code.videolan.org/videolan/dav1d> and follow the
       instructions for installing the library. Then pass "--enable-libdav1d"
       to configure to enable it.

   davs2
       FFmpeg can make use of the davs2 library for AVS2-P2/IEEE1857.4 video
       decoding.

       Go to <https://github.com/pkuvcl/davs2> and follow the instructions for
       installing the library. Then pass "--enable-libdavs2" to configure to
       enable it.

	   libdavs2 is under the GNU Public License Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for
	   details), you must upgrade FFmpeg's license to GPL in order to use
	   it.

   uavs3d
       FFmpeg can make use of the uavs3d library for AVS3-P2/IEEE1857.10 video
       decoding.

       Go to <https://github.com/uavs3/uavs3d> and follow the instructions for
       installing the library. Then pass "--enable-libuavs3d" to configure to
       enable it.

   Game Music Emu
       FFmpeg can make use of the Game Music Emu library to read audio from
       supported video game music file formats. Pass "--enable-libgme" to
       configure to enable it. See
       <https://bitbucket.org/mpyne/game-music-emu/overview>.

   Intel QuickSync Video
       FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and
       encoding of multiple codecs. To use QSV, FFmpeg must be linked against
       the "libmfx" dispatcher, which loads the actual decoding libraries.

       The dispatcher is open source and can be downloaded from
       <https://github.com/lu-zero/mfx_dispatch.git>. FFmpeg needs to be
       configured with the "--enable-libmfx" option and "pkg-config" needs to
       be able to locate the dispatcher's ".pc" files.

   Kvazaar
       FFmpeg can make use of the Kvazaar library for HEVC encoding.

       Go to <https://github.com/ultravideo/kvazaar> and follow the
       instructions for installing the library. Then pass
       "--enable-libkvazaar" to configure to enable it.

   LAME
       FFmpeg can make use of the LAME library for MP3 encoding.

       Go to <http://lame.sourceforge.net/> and follow the instructions for
       installing the library.	Then pass "--enable-libmp3lame" to configure
       to enable it.

   libilbc
       iLBC is a narrowband speech codec that has been made freely available
       by Google as part of the WebRTC project. libilbc is a packaging
       friendly copy of the iLBC codec. FFmpeg can make use of the libilbc
       library for iLBC decoding and encoding.

       Go to <https://github.com/TimothyGu/libilbc> and follow the
       instructions for installing the library. Then pass "--enable-libilbc"
       to configure to enable it.

   libjxl
       JPEG XL is an image format intended to fully replace legacy JPEG for an
       extended period of life. See <https://jpegxl.info/> for more
       information, and see <https://github.com/libjxl/libjxl> for the library
       source. You can pass "--enable-libjxl" to configure in order enable the
       libjxl wrapper.

   libvpx
       FFmpeg can make use of the libvpx library for VP8/VP9 decoding and
       encoding.

       Go to <http://www.webmproject.org/> and follow the instructions for
       installing the library. Then pass "--enable-libvpx" to configure to
       enable it.

   ModPlug
       FFmpeg can make use of this library, originating in Modplug-XMMS, to
       read from MOD-like music files.	See
       <https://github.com/Konstanty/libmodplug>. Pass "--enable-libmodplug"
       to configure to enable it.

   OpenCORE, VisualOn, and Fraunhofer libraries
       Spun off Google Android sources, OpenCore, VisualOn and Fraunhofer
       libraries provide encoders for a number of audio codecs.

	   OpenCORE and VisualOn libraries are under the Apache License 2.0
	   (see <http://www.apache.org/licenses/LICENSE-2.0> for details),
	   which is incompatible to the LGPL version 2.1 and GPL version 2.
	   You have to upgrade FFmpeg's license to LGPL version 3 (or if you
	   have enabled GPL components, GPL version 3) by passing
	   "--enable-version3" to configure in order to use it.

	   The license of the Fraunhofer AAC library is incompatible with the
	   GPL.  Therefore, for GPL builds, you have to pass
	   "--enable-nonfree" to configure in order to use it. To the best of
	   our knowledge, it is compatible with the LGPL.

       OpenCORE AMR

       FFmpeg can make use of the OpenCORE libraries for AMR-NB
       decoding/encoding and AMR-WB decoding.

       Go to <http://sourceforge.net/projects/opencore-amr/> and follow the
       instructions for installing the libraries.  Then pass
       "--enable-libopencore-amrnb" and/or "--enable-libopencore-amrwb" to
       configure to enable them.

       VisualOn AMR-WB encoder library

       FFmpeg can make use of the VisualOn AMR-WBenc library for AMR-WB
       encoding.

       Go to <http://sourceforge.net/projects/opencore-amr/> and follow the
       instructions for installing the library.  Then pass
       "--enable-libvo-amrwbenc" to configure to enable it.

       Fraunhofer AAC library

       FFmpeg can make use of the Fraunhofer AAC library for AAC decoding &
       encoding.

       Go to <http://sourceforge.net/projects/opencore-amr/> and follow the
       instructions for installing the library.  Then pass
       "--enable-libfdk-aac" to configure to enable it.

       LC3 library

       FFmpeg can make use of the Google LC3 library for LC3 decoding &
       encoding.

       Go to <https://github.com/google/liblc3/> and follow the instructions
       for installing the library.  Then pass "--enable-liblc3" to configure
       to enable it.

   OpenH264
       FFmpeg can make use of the OpenH264 library for H.264 decoding and
       encoding.

       Go to <http://www.openh264.org/> and follow the instructions for
       installing the library. Then pass "--enable-libopenh264" to configure
       to enable it.

       For decoding, this library is much more limited than the built-in
       decoder in libavcodec; currently, this library lacks support for
       decoding B-frames and some other main/high profile features. (It
       currently only supports constrained baseline profile and CABAC.) Using
       it is mostly useful for testing and for taking advantage of Cisco's
       patent portfolio license
       (<http://www.openh264.org/BINARY_LICENSE.txt>).

   OpenJPEG
       FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos.
       Go to <http://www.openjpeg.org/> to get the libraries and follow the
       installation instructions.  To enable using OpenJPEG in FFmpeg, pass
       "--enable-libopenjpeg" to ./configure.

   rav1e
       FFmpeg can make use of rav1e (Rust AV1 Encoder) via its C bindings to
       encode videos.  Go to <https://github.com/xiph/rav1e/> and follow the
       instructions to build the C library. To enable using rav1e in FFmpeg,
       pass "--enable-librav1e" to ./configure.

   SVT-AV1
       FFmpeg can make use of the Scalable Video Technology for AV1 library
       for AV1 encoding.

       Go to <https://gitlab.com/AOMediaCodec/SVT-AV1/> and follow the
       instructions for installing the library. Then pass "--enable-libsvtav1"
       to configure to enable it.

   TwoLAME
       FFmpeg can make use of the TwoLAME library for MP2 encoding.

       Go to <http://www.twolame.org/> and follow the instructions for
       installing the library.	Then pass "--enable-libtwolame" to configure
       to enable it.

   VapourSynth
       FFmpeg can read VapourSynth scripts as input. To enable support, pass
       "--enable-vapoursynth" to configure. Vapoursynth is detected via
       "pkg-config". Versions 42 or greater supported.	See
       <http://www.vapoursynth.com/>.

       Due to security concerns, Vapoursynth scripts will not be autodetected
       so the input format has to be forced. For ff* CLI tools, add "-f
       vapoursynth" before the input "-i yourscript.vpy".

   x264
       FFmpeg can make use of the x264 library for H.264 encoding.

       Go to <http://www.videolan.org/developers/x264.html> and follow the
       instructions for installing the library. Then pass "--enable-libx264"
       to configure to enable it.

	   x264 is under the GNU Public License Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for
	   details), you must upgrade FFmpeg's license to GPL in order to use
	   it.

   x265
       FFmpeg can make use of the x265 library for HEVC encoding.

       Go to <http://x265.org/developers.html> and follow the instructions for
       installing the library. Then pass "--enable-libx265" to configure to
       enable it.

	   x265 is under the GNU Public License Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for
	   details), you must upgrade FFmpeg's license to GPL in order to use
	   it.

   xavs
       FFmpeg can make use of the xavs library for AVS encoding.

       Go to <http://xavs.sf.net/> and follow the instructions for installing
       the library. Then pass "--enable-libxavs" to configure to enable it.

   xavs2
       FFmpeg can make use of the xavs2 library for AVS2-P2/IEEE1857.4 video
       encoding.

       Go to <https://github.com/pkuvcl/xavs2> and follow the instructions for
       installing the library. Then pass "--enable-libxavs2" to configure to
       enable it.

	   libxavs2 is under the GNU Public License Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html> for
	   details), you must upgrade FFmpeg's license to GPL in order to use
	   it.

   eXtra-fast Essential Video Encoder (XEVE)
       FFmpeg can make use of the XEVE library for EVC video encoding.

       Go to <https://github.com/mpeg5/xeve> and follow the instructions for
       installing the XEVE library. Then pass "--enable-libxeve" to configure
       to enable it.

   eXtra-fast Essential Video Decoder (XEVD)
       FFmpeg can make use of the XEVD library for EVC video decoding.

       Go to <https://github.com/mpeg5/xevd> and follow the instructions for
       installing the XEVD library. Then pass "--enable-libxevd" to configure
       to enable it.

   ZVBI
       ZVBI is a VBI decoding library which can be used by FFmpeg to decode
       DVB teletext pages and DVB teletext subtitles.

       Go to <http://sourceforge.net/projects/zapping/> and follow the
       instructions for installing the library. Then pass "--enable-libzvbi"
       to configure to enable it.

SUPPORTED FILE FORMATS
       You can use the "-formats" and "-codecs" options to have an exhaustive
       list.

   File Formats
       FFmpeg supports the following file formats through the "libavformat"
       library:

       Name  :	Encoding @tab Decoding @tab Comments
       3dostr			  :    @tab X
       4xm			  :    @tab X
	       @tab 4X Technologies format, used in some games.

       8088flex TMV		  :    @tab X
       AAX			  :    @tab X
	       @tab Audible Enhanced Audio format, used in audiobooks.

       AA			  :    @tab X
	       @tab Audible Format 2, 3, and 4, used in audiobooks.

       ACT Voice		  :    @tab X
	       @tab contains G.729 audio

       Adobe Filmstrip		  :  X @tab X
       Audio IFF (AIFF) 	  :  X @tab X
       American Laser Games MM	  :    @tab X
	       @tab Multimedia format used in games like Mad Dog McCree.

       3GPP AMR 		  :  X @tab X
       Amazing Studio Packed Animation File   :    @tab X
	       @tab Multimedia format used in game Heart Of Darkness.

       Apple HTTP Live Streaming  :    @tab X
       Artworx Data Format	  :    @tab X
       Interplay ACM		  :    @tab X
	       @tab Audio only format used in some Interplay games.

       ADP			  :    @tab X
	       @tab Audio format used on the Nintendo Gamecube.

       AFC			  :    @tab X
	       @tab Audio format used on the Nintendo Gamecube.

       ADS/SS2			  :    @tab X
	       @tab Audio format used on the PS2.

       APNG			  :  X @tab X
       ASF			  :  X @tab X
	       @tab Advanced / Active Streaming Format.

       AST			  :  X @tab X
	       @tab Audio format used on the Nintendo Wii.

       AVI			  :  X @tab X
       AviSynth 		  :    @tab X
       AVR			  :    @tab X
	       @tab Audio format used on Mac.

       AVS			  :    @tab X
	       @tab Multimedia format used by the Creature Shock game.

       Beam Software SIFF	  :    @tab X
	       @tab Audio and video format used in some games by Beam Software.

       Bethesda Softworks VID	  :    @tab X
	       @tab Used in some games from Bethesda Softworks.

       Binary text		  :    @tab X
       Bink			  :    @tab X
	       @tab Multimedia format used by many games.

       Bink Audio		  :    @tab X
	       @tab Audio only multimedia format used by some games.

       Bitmap Brothers JV	  :    @tab X
	       @tab Used in Z and Z95 games.

       BRP			  :    @tab X
	       @tab Argonaut Games format.

       Brute Force & Ignorance	  :    @tab X
	       @tab Used in the game Flash Traffic: City of Angels.

       BFSTM			  :    @tab X
	       @tab Audio format used on the Nintendo WiiU (based on BRSTM).

       BRSTM			  :    @tab X
	       @tab Audio format used on the Nintendo Wii.

       BW64			  :    @tab X
	       @tab Broadcast Wave 64bit.

       BWF			  :  X @tab X
       codec2 (raw)		  :  X @tab X
	       @tab Must be given -mode format option to decode correctly.

       codec2 (.c2 files)	  :  X @tab X
	       @tab Contains header with version and mode info, simplifying playback.

       CRI ADX			  :  X @tab X
	       @tab Audio-only format used in console video games.

       CRI AIX			  :    @tab X
       CRI HCA			  :    @tab X
	       @tab Audio-only format used in console video games.

       Discworld II BMV 	  :    @tab X
       Interplay C93		  :    @tab X
	       @tab Used in the game Cyberia from Interplay.

       Delphine Software International CIN  :	 @tab X
	       @tab Multimedia format used by Delphine Software games.

       Digital Speech Standard (DSS)  :    @tab X
       CD+G			  :    @tab X
	       @tab Video format used by CD+G karaoke disks

       Phantom Cine		  :    @tab X
       Commodore CDXL		  :    @tab X
	       @tab Amiga CD video format

       Core Audio Format	  :  X @tab X
	       @tab Apple Core Audio Format

       CRC testing format	  :  X @tab
       Creative Voice		  :  X @tab X
	       @tab Created for the Sound Blaster Pro.

       CRYO APC 		  :    @tab X
	       @tab Audio format used in some games by CRYO Interactive Entertainment.

       D-Cinema audio		  :  X @tab X
       Deluxe Paint Animation	  :    @tab X
       DCSTR			  :    @tab X
       DFA			  :    @tab X
	       @tab This format is used in Chronomaster game

       DirectDraw Surface	  :    @tab X
       DSD Stream File (DSF)	  :    @tab X
       DV video 		  :  X @tab X
       DXA			  :    @tab X
	       @tab This format is used in the non-Windows version of the Feeble Files
		    game and different game cutscenes repacked for use with ScummVM.

       Electronic Arts cdata   :     @tab X
       Electronic Arts Multimedia   :	  @tab X
	       @tab Used in various EA games; files have extensions like WVE and UV2.

       Ensoniq Paris Audio File   :    @tab X
       FFM (FFserver live feed)   :  X @tab X
       Flash (SWF)		  :  X @tab X
       Flash 9 (AVM2)		  :  X @tab X
	       @tab Only embedded audio is decoded.

       FLI/FLC/FLX animation	  :    @tab X
	       @tab .fli/.flc files

       Flash Video (FLV)	  :  X @tab X
	       @tab Macromedia Flash video files

       framecrc testing format	  :  X @tab
       FunCom ISS		  :    @tab X
	       @tab Audio format used in various games from FunCom like The Longest Journey.

       G.723.1			  :  X @tab X
       G.726			  :    @tab X @tab Both left- and right-
       justified.
       G.729 BIT		  :  X @tab X
       G.729 raw		  :    @tab X
       GENH			  :    @tab X
	       @tab Audio format for various games.

       GIF Animation		  :  X @tab X
       GXF			  :  X @tab X
	       @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
		    playout servers.

       HNM  :	 @tab X
	       @tab Only version 4 supported, used in some games from Cryo Interactive

       iCEDraw File		  :    @tab X
       ICO			  :  X @tab X
	       @tab Microsoft Windows ICO

       id Quake II CIN video	  :    @tab X
       id RoQ			  :  X @tab X
	       @tab Used in Quake III, Jedi Knight 2 and other computer games.

       IEC61937 encapsulation  :  X @tab X
       IFF			  :    @tab X
	       @tab Interchange File Format

       IFV			  :    @tab X
	       @tab A format used by some old CCTV DVRs.

       iLBC			  :  X @tab X
       Interplay MVE		  :    @tab X
	       @tab Format used in various Interplay computer games.

       Iterated Systems ClearVideo  :	   @tab  X
	       @tab I-frames only

       IV8			  :    @tab X
	       @tab A format generated by IndigoVision 8000 video server.

       IVF (On2)		  :  X @tab X
	       @tab A format used by libvpx

       Internet Video Recording   :    @tab X
       IRCAM			  :  X @tab X
       LAF			  :    @tab X
	       @tab Limitless Audio Format

       LATM			  :  X @tab X
       LMLM4			  :    @tab X
	       @tab Used by Linux Media Labs MPEG-4 PCI boards

       LOAS			  :    @tab X
	       @tab contains LATM multiplexed AAC audio

       LRC			  :  X @tab X
       LVF			  :    @tab X
       LXF			  :    @tab X
	       @tab VR native stream format, used by Leitch/Harris' video servers.

       Magic Lantern Video (MLV)  :    @tab X
       Matroska 		  :  X @tab X
       Matroska audio		  :  X @tab
       FFmpeg metadata		  :  X @tab X
	       @tab Metadata in text format.

       MAXIS XA 		  :    @tab X
	       @tab Used in Sim City 3000; file extension .xa.

       MCA			  :    @tab X
	       @tab Used in some games from Capcom; file extension .mca.

       MD Studio		  :    @tab X
       Metal Gear Solid: The Twin Snakes  :  @tab X
       Megalux Frame		  :    @tab X
	       @tab Used by Megalux Ultimate Paint

       MobiClip MODS		  :    @tab X
       MobiClip MOFLEX		  :    @tab X
       Mobotix .mxg		  :    @tab X
       Monkey's Audio		  :    @tab X
       Motion Pixels MVI	  :    @tab X
       MOV/QuickTime/MP4	  :  X @tab X
	       @tab 3GP, 3GP2, PSP, iPod variants supported

       MP2			  :  X @tab X
       MP3			  :  X @tab X
       MPEG-1 System		  :  X @tab X
	       @tab muxed audio and video, VCD format supported

       MPEG-PS (program stream)   :  X @tab X
	       @tab also known as C<VOB> file, SVCD and DVD format supported

       MPEG-TS (transport stream)  :  X @tab X
	       @tab also known as DVB Transport Stream

       MPEG-4			  :  X @tab X
	       @tab MPEG-4 is a variant of QuickTime.

       MSF			  :    @tab X
	       @tab Audio format used on the PS3.

       Mirillis FIC video	  :    @tab X
	       @tab No cursor rendering.

       MIDI Sample Dump Standard  :    @tab X
       MIME multipart JPEG	  :  X @tab
       MSN TCP webcam		  :    @tab X
	       @tab Used by MSN Messenger webcam streams.

       MTV			  :    @tab X
       Musepack 		  :    @tab X
       Musepack SV8		  :    @tab X
       Material eXchange Format (MXF)  :  X @tab X
	       @tab SMPTE 377M, used by D-Cinema, broadcast industry.

       Material eXchange Format (MXF), D-10 Mapping  :	X @tab X
	       @tab SMPTE 386M, D-10/IMX Mapping.

       NC camera feed		  :    @tab X
	       @tab NC (AVIP NC4600) camera streams

       NIST SPeech HEader REsources  :	  @tab X
       Computerized Speech Lab NSP  :	 @tab X
       NTT TwinVQ (VQF) 	  :    @tab X
	       @tab Nippon Telegraph and Telephone Corporation TwinVQ.

       Nullsoft Streaming Video   :    @tab X
       NuppelVideo		  :    @tab X
       NUT			  :  X @tab X
	       @tab NUT Open Container Format

       Ogg			  :  X @tab X
       Playstation Portable PMP   :    @tab X
       Portable Voice Format	  :    @tab X
       RK Audio (RKA)		  :    @tab X
       TechnoTrend PVA		  :    @tab X
	       @tab Used by TechnoTrend DVB PCI boards.

       QCP			  :    @tab X
       raw ADTS (AAC)		  :  X @tab X
       raw AC-3 		  :  X @tab X
       raw AMR-NB		  :    @tab X
       raw AMR-WB		  :    @tab X
       raw APAC 		  :    @tab X
       raw aptX 		  :  X @tab X
       raw aptX HD		  :  X @tab X
       raw Bonk 		  :    @tab X
       raw Chinese AVS video	  :  X @tab X
       raw DFPWM		  :  X @tab X
       raw Dirac		  :  X @tab X
       raw DNxHD		  :  X @tab X
       raw DTS			  :  X @tab X
       raw DTS-HD		  :    @tab X
       raw E-AC-3		  :  X @tab X
       raw EVC			  :  X @tab X
       raw FLAC 		  :  X @tab X
       raw GSM			  :    @tab X
       raw H.261		  :  X @tab X
       raw H.263		  :  X @tab X
       raw H.264		  :  X @tab X
       raw HEVC 		  :  X @tab X
       raw Ingenient MJPEG	  :    @tab X
       raw MJPEG		  :  X @tab X
       raw MLP			  :    @tab X
       raw MPEG 		  :    @tab X
       raw MPEG-1		  :    @tab X
       raw MPEG-2		  :    @tab X
       raw MPEG-4		  :  X @tab X
       raw NULL 		  :  X @tab
       raw video		  :  X @tab X
       raw id RoQ		  :  X @tab
       raw OBU			  :  X @tab X
       raw OSQ			  :    @tab X
       raw SBC			  :  X @tab X
       raw Shorten		  :    @tab X
       raw TAK			  :    @tab X
       raw TrueHD		  :  X @tab X
       raw VC-1 		  :  X @tab X
       raw PCM A-law		  :  X @tab X
       raw PCM mu-law		  :  X @tab X
       raw PCM Archimedes VIDC	  :  X @tab X
       raw PCM signed 8 bit	  :  X @tab X
       raw PCM signed 16 bit big-endian   :  X @tab X
       raw PCM signed 16 bit little-endian   :	X @tab X
       raw PCM signed 24 bit big-endian   :  X @tab X
       raw PCM signed 24 bit little-endian   :	X @tab X
       raw PCM signed 32 bit big-endian   :  X @tab X
       raw PCM signed 32 bit little-endian   :	X @tab X
       raw PCM signed 64 bit big-endian   :  X @tab X
       raw PCM signed 64 bit little-endian   :	X @tab X
       raw PCM unsigned 8 bit	  :  X @tab X
       raw PCM unsigned 16 bit big-endian   :  X @tab X
       raw PCM unsigned 16 bit little-endian   :  X @tab X
       raw PCM unsigned 24 bit big-endian   :  X @tab X
       raw PCM unsigned 24 bit little-endian   :  X @tab X
       raw PCM unsigned 32 bit big-endian   :  X @tab X
       raw PCM unsigned 32 bit little-endian   :  X @tab X
       raw PCM 16.8 floating point little-endian  :    @tab X
       raw PCM 24.0 floating point little-endian  :    @tab X
       raw PCM floating-point 32 bit big-endian   :  X @tab X
       raw PCM floating-point 32 bit little-endian   :	X @tab X
       raw PCM floating-point 64 bit big-endian   :  X @tab X
       raw PCM floating-point 64 bit little-endian   :	X @tab X
       RDT			  :    @tab X
       REDCODE R3D		  :    @tab X
	       @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.

       RealMedia		  :  X @tab X
       Redirector		  :    @tab X
       RedSpark 		  :    @tab X
       Renderware TeXture Dictionary  :    @tab X
       Resolume DXV		  :  X @tab X
	       @tab Encoding is only supported for the DXT1 (Normal Quality, No Alpha) texture format.

       RF64			  :    @tab X
       RL2			  :    @tab X
	       @tab Audio and video format used in some games by Entertainment Software Partners.

       RPL/ARMovie		  :    @tab X
       Lego Mindstorms RSO	  :  X @tab X
       RSD			  :    @tab X
       RTMP			  :  X @tab X
	       @tab Output is performed by publishing stream to RTMP server

       RTP			  :  X @tab X
       RTSP			  :  X @tab X
       Sample Dump eXchange	  :    @tab X
       SAP			  :  X @tab X
       SBG			  :    @tab X
       SDNS			  :    @tab X
       SDP			  :    @tab X
       SER			  :    @tab X
       Digital Pictures SGA	  :    @tab X
       Sega FILM/CPK		  :  X @tab X
	       @tab Used in many Sega Saturn console games.

       Silicon Graphics Movie	  :    @tab X
       Sierra SOL		  :    @tab X
	       @tab .sol files used in Sierra Online games.

       Sierra VMD		  :    @tab X
	       @tab Used in Sierra CD-ROM games.

       Smacker			  :    @tab X
	       @tab Multimedia format used by many games.

       SMJPEG			  :  X @tab X
	       @tab Used in certain Loki game ports.

       SMPTE 337M encapsulation   :    @tab X
       Smush			  :    @tab X
	       @tab Multimedia format used in some LucasArts games.

       Sony OpenMG (OMA)	  :  X @tab X
	       @tab Audio format used in Sony Sonic Stage and Sony Vegas.

       Sony PlayStation STR	  :    @tab X
       Sony Wave64 (W64)	  :  X @tab X
       SoX native format	  :  X @tab X
       SUN AU format		  :  X @tab X
       SUP raw PGS subtitles	  :  X @tab X
       SVAG			  :    @tab X
	       @tab Audio format used in Konami PS2 games.

       TDSC			  :    @tab X
       Text files		  :    @tab X
       THP			  :    @tab X
	       @tab Used on the Nintendo GameCube.

       Tiertex Limited SEQ	  :    @tab X
	       @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.

       True Audio		  :  X @tab X
       VAG			  :    @tab X
	       @tab Audio format used in many Sony PS2 games.

       VC-1 test bitstream	  :  X @tab X
       Vidvox Hap		  :  X @tab X
       Vivo			  :    @tab X
       VPK			  :    @tab X
	       @tab Audio format used in Sony PS games.

       Marble WADY		  :    @tab X
       WAV			  :  X @tab X
       Waveform Archiver	  :    @tab X
       WavPack			  :  X @tab X
       WebM			  :  X @tab X
       Windows Televison (WTV)	  :  X @tab X
       Wing Commander III movie   :    @tab X
	       @tab Multimedia format used in Origin's Wing Commander III computer game.

       Westwood Studios audio	  :  X @tab X
	       @tab Multimedia format used in Westwood Studios games.

       Westwood Studios VQA	  :    @tab X
	       @tab Multimedia format used in Westwood Studios games.

       Wideband Single-bit Data (WSD)  :    @tab X
       WVE			  :    @tab X
       Konami XMD		  :    @tab X
       XMV			  :    @tab X
	       @tab Microsoft video container used in Xbox games.

       XVAG			  :    @tab X
	       @tab Audio format used on the PS3.

       xWMA			  :    @tab X
	       @tab Microsoft audio container used by XAudio 2.

       eXtended BINary text (XBIN)  :  @tab X
       YUV4MPEG pipe		  :  X @tab X
       Psygnosis YOP		  :    @tab X

       "X" means that the feature in that column (encoding / decoding) is
       supported.

   Image Formats
       FFmpeg can read and write images for each frame of a video sequence.
       The following image formats are supported:

       Name  :	Encoding @tab Decoding @tab Comments
       .Y.U.V	     :	X @tab X
	       @tab one raw file per component

       Alias PIX     :	X @tab X
	       @tab Alias/Wavefront PIX image format

       animated GIF  :	X @tab X
       APNG	     :	X @tab X
	       @tab Animated Portable Network Graphics

       BMP	     :	X @tab X
	       @tab Microsoft BMP image

       BRender PIX   :	  @tab X
	       @tab Argonaut BRender 3D engine image format.

       CRI	     :	  @tab X
	       @tab Cintel RAW

       DPX	     :	X @tab X
	       @tab Digital Picture Exchange

       EXR	     :	  @tab X
	       @tab OpenEXR

       FITS	     :	X @tab X
	       @tab Flexible Image Transport System

       HDR	     :	X @tab X
	       @tab Radiance HDR RGBE Image format

       IMG	     :	  @tab X
	       @tab GEM Raster image

       JPEG	     :	X @tab X
	       @tab Progressive JPEG is not supported.

       JPEG 2000     :	X @tab X
       JPEG-LS	     :	X @tab X
       LJPEG	     :	X @tab
	       @tab Lossless JPEG

       Media 100     :	  @tab X
       MSP	     :	  @tab X
	       @tab Microsoft Paint image

       PAM	     :	X @tab X
	       @tab PAM is a PNM extension with alpha support.

       PBM	     :	X @tab X
	       @tab Portable BitMap image

       PCD	     :	  @tab X
	       @tab PhotoCD

       PCX	     :	X @tab X
	       @tab PC Paintbrush

       PFM	     :	X @tab X
	       @tab Portable FloatMap image

       PGM	     :	X @tab X
	       @tab Portable GrayMap image

       PGMYUV	     :	X @tab X
	       @tab PGM with U and V components in YUV 4:2:0

       PGX	     :	  @tab X
	       @tab PGX file decoder

       PHM	     :	X @tab X
	       @tab Portable HalfFloatMap image

       PIC	     :	@tab X
	       @tab Pictor/PC Paint

       PNG	     :	X @tab X
	       @tab Portable Network Graphics image

       PPM	     :	X @tab X
	       @tab Portable PixelMap image

       PSD	     :	  @tab X
	       @tab Photoshop

       PTX	     :	  @tab X
	       @tab V.Flash PTX format

       QOI	     :	X @tab X
	       @tab Quite OK Image format

       SGI	     :	X @tab X
	       @tab SGI RGB image format

       Sun Rasterfile	:  X @tab X
	       @tab Sun RAS image format

       TIFF	     :	X @tab X
	       @tab YUV, JPEG and some extension is not supported yet.

       Truevision Targa   :  X @tab X
	       @tab Targa (.TGA) image format

       VBN   :	X @tab X
	       @tab Vizrt Binary Image format

       WBMP	     :	X @tab X
	       @tab Wireless Application Protocol Bitmap image format

       WebP	     :	E @tab X
	       @tab WebP image format, encoding supported through external library libwebp

       XBM   :	X @tab X
	       @tab X BitMap image format

       XFace  :  X @tab X
	       @tab X-Face image format

       XPM   :	  @tab X
	       @tab X PixMap image format

       XWD   :	X @tab X
	       @tab X Window Dump image format

       "X" means that the feature in that column (encoding / decoding) is
       supported.

       "E" means that support is provided through an external library.

   Video Codecs
       Name  :	Encoding @tab Decoding @tab Comments
       4X Movie 	       :      @tab  X
	       @tab Used in certain computer games.

       8088flex TMV	       :      @tab  X
       A64 multicolor	       :   X  @tab
	       @tab Creates video suitable to be played on a commodore 64 (multicolor mode).

       Amazing Studio PAF Video  :	@tab  X
       American Laser Games MM	 :     @tab X
	       @tab Used in games like Mad Dog McCree.

       Amuse Graphics Movie    :      @tab  X
       AMV Video	       :   X  @tab  X
	       @tab Used in Chinese MP3 players.

       ANSI/ASCII art	       :      @tab  X
       Apple Intermediate Codec  :	@tab  X
       Apple MJPEG-B	       :      @tab  X
       Apple Pixlet	       :      @tab  X
       Apple ProRes	       :   X  @tab  X
	       @tab fourcc: apch,apcn,apcs,apco,ap4h,ap4x

       Apple QuickDraw	       :      @tab  X
	       @tab fourcc: qdrw

       Argonaut Video	       :      @tab  X
	       @tab Used in some Argonaut games.

       Asus v1		       :   X  @tab  X
	       @tab fourcc: ASV1

       Asus v2		       :   X  @tab  X
	       @tab fourcc: ASV2

       ATI VCR1 	       :      @tab  X
	       @tab fourcc: VCR1

       ATI VCR2 	       :      @tab  X
	       @tab fourcc: VCR2

       Auravision Aura	       :      @tab  X
       Auravision Aura 2       :      @tab  X
       Autodesk Animator Flic video   :      @tab  X
       Autodesk RLE	       :      @tab  X
	       @tab fourcc: AASC

       AV1		       :   E  @tab  E
	       @tab Supported through external libraries libaom, libdav1d, librav1e and libsvtav1

       Avid 1:1 10-bit RGB Packer   :	X  @tab  X
	       @tab fourcc: AVrp

       AVS (Audio Video Standard) video   :	 @tab  X
	       @tab Video encoding used by the Creature Shock game.

       AVS2-P2/IEEE1857.4      :   E  @tab  E
	       @tab Supported through external libraries libxavs2 and libdavs2

       AVS3-P2/IEEE1857.10     :      @tab  E
	       @tab Supported through external library libuavs3d

       AYUV		       :   X  @tab  X
	       @tab Microsoft uncompressed packed 4:4:4:4

       Beam Software VB        :      @tab  X
       Bethesda VID video      :      @tab  X
	       @tab Used in some games from Bethesda Softworks.

       Bink Video	       :      @tab  X
       BitJazz SheerVideo      :      @tab  X
       Bitmap Brothers JV video   :    @tab X
       y41p Brooktree uncompressed 4:1:1 12-bit      :	 X  @tab  X
       Brooktree ProSumer Video   :	 @tab  X
	       @tab fourcc: BT20

       Brute Force & Ignorance	  :    @tab X
	       @tab Used in the game Flash Traffic: City of Angels.

       C93 video	       :      @tab  X
	       @tab Codec used in Cyberia game.

       CamStudio	       :      @tab  X
	       @tab fourcc: CSCD

       CD+G		       :      @tab  X
	       @tab Video codec for CD+G karaoke disks

       CDXL		       :      @tab  X
	       @tab Amiga CD video codec

       Chinese AVS video       :   E  @tab  X
	       @tab AVS1-P2, JiZhun profile, encoding through external library libxavs

       Delphine Software International CIN video   :	  @tab	X
	       @tab Codec used in Delphine Software International games.

       Discworld II BMV Video  :      @tab  X
       CineForm HD	       :   X  @tab  X
       Canopus HQ	       :      @tab  X
       Canopus HQA	       :      @tab  X
       Canopus HQX	       :      @tab  X
       Canopus Lossless Codec  :      @tab  X
       CDToons		       :      @tab  X
	       @tab Codec used in various Broderbund games.

       Cinepak		       :      @tab  X
       Cirrus Logic AccuPak    :   X  @tab  X
	       @tab fourcc: CLJR

       CPiA Video Format       :      @tab  X
       Creative YUV (CYUV)     :      @tab  X
       DFA		       :      @tab  X
	       @tab Codec used in Chronomaster game.

       Dirac		       :   E  @tab  X
	       @tab supported though the native vc2 (Dirac Pro) encoder

       Deluxe Paint Animation  :      @tab  X
       DNxHD		       :    X @tab  X
	       @tab aka SMPTE VC3

       Duck TrueMotion 1.0    :      @tab  X
	       @tab fourcc: DUCK

       Duck TrueMotion 2.0     :      @tab  X
	       @tab fourcc: TM20

       Duck TrueMotion 2.0 RT  :      @tab  X
	       @tab fourcc: TR20

       DV (Digital Video)      :   X  @tab  X
       Dxtory capture format   :      @tab  X
       Feeble Files/ScummVM DXA   :	 @tab  X
	       @tab Codec originally used in Feeble Files game.

       Electronic Arts CMV video   :	  @tab	X
	       @tab Used in NHL 95 game.

       Electronic Arts Madcow video   :      @tab  X
       Electronic Arts TGV video   :	  @tab	X
       Electronic Arts TGQ video   :	  @tab	X
       Electronic Arts TQI video   :	  @tab	X
       Escape 124	       :      @tab  X
       Escape 130	       :      @tab  X
       EVC / MPEG-5 Part 1     :   E  @tab  E
	       @tab encoding and decoding supported through external libraries libxeve and libxevd

       FFmpeg video codec #1   :   X  @tab  X
	       @tab lossless codec (fourcc: FFV1)

       Flash Screen Video v1   :   X  @tab  X
	       @tab fourcc: FSV1

       Flash Screen Video v2   :   X  @tab  X
       Flash Video (FLV)       :   X  @tab  X
	       @tab Sorenson H.263 used in Flash

       FM Screen Capture Codec	 :	@tab  X
       Forward Uncompressed    :      @tab  X
       Fraps		       :      @tab  X
       Go2Meeting	       :      @tab  X
	       @tab fourcc: G2M2, G2M3

       Go2Webinar	       :      @tab  X
	       @tab fourcc: G2M4

       Gremlin Digital Video   :      @tab  X
       H.261		       :   X  @tab  X
       H.263 / H.263-1996      :   X  @tab  X
       H.263+ / H.263-1998 / H.263 version 2   :   X  @tab  X
       H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10   :   E  @tab	X
	       @tab encoding supported through external library libx264 and OpenH264

       HEVC		       :   X  @tab  X
	       @tab encoding supported through external library libx265 and libkvazaar

       HNM version 4	       :      @tab  X
       HuffYUV		       :   X  @tab  X
       HuffYUV FFmpeg variant  :   X  @tab  X
       IBM Ultimotion	       :      @tab  X
	       @tab fourcc: ULTI

       id Cinematic video      :      @tab  X
	       @tab Used in Quake II.

       id RoQ video	       :   X  @tab  X
	       @tab Used in Quake III, Jedi Knight 2, other computer games.

       IFF ILBM 	       :      @tab  X
	       @tab IFF interleaved bitmap

       IFF ByteRun1	       :      @tab  X
	       @tab IFF run length encoded bitmap

       Infinity IMM4	       :      @tab  X
       Intel H.263	       :      @tab  X
       Intel Indeo 2	       :      @tab  X
       Intel Indeo 3	       :      @tab  X
       Intel Indeo 4	       :      @tab  X
       Intel Indeo 5	       :      @tab  X
       Interplay C93	       :      @tab  X
	       @tab Used in the game Cyberia from Interplay.

       Interplay MVE video     :      @tab  X
	       @tab Used in Interplay .MVE files.

       J2K  :	X  @tab  X
       Karl Morton's video codec   :	  @tab	X
	       @tab Codec used in Worms games.

       Kega Game Video (KGV1)  :       @tab  X
	       @tab Kega emulator screen capture codec.

       Lagarith 	       :      @tab  X
       LCL (LossLess Codec Library) MSZH   :	  @tab	X
       LCL (LossLess Codec Library) ZLIB   :   E  @tab	E
       LEAD MCMP	       :      @tab  X
       LOCO		       :      @tab  X
       LucasArts SANM/Smush    :      @tab  X
	       @tab Used in LucasArts games / SMUSH animations.

       lossless MJPEG	       :   X  @tab  X
       MagicYUV Video	       :   X  @tab  X
       Mandsoft Screen Capture Codec   :      @tab  X
       Microsoft ATC Screen    :      @tab  X
	       @tab Also known as Microsoft Screen 3.

       Microsoft Expression Encoder Screen   :	    @tab  X
	       @tab Also known as Microsoft Titanium Screen 2.

       Microsoft RLE	       :   X  @tab  X
       Microsoft Screen 1      :      @tab  X
	       @tab Also known as Windows Media Video V7 Screen.

       Microsoft Screen 2      :      @tab  X
	       @tab Also known as Windows Media Video V9 Screen.

       Microsoft Video 1       :      @tab  X
       Mimic		       :      @tab  X
	       @tab Used in MSN Messenger Webcam streams.

       Miro VideoXL	       :      @tab  X
	       @tab fourcc: VIXL

       MJPEG (Motion JPEG)     :   X  @tab  X
       Mobotix MxPEG video     :      @tab  X
       Motion Pixels video     :      @tab  X
       MPEG-1 video	       :   X  @tab  X
       MPEG-2 video	       :   X  @tab  X
       MPEG-4 part 2	       :   X  @tab  X
	       @tab libxvidcore can be used alternatively for encoding.

       MPEG-4 part 2 Microsoft variant version 1   :	  @tab	X
       MPEG-4 part 2 Microsoft variant version 2   :   X  @tab	X
       MPEG-4 part 2 Microsoft variant version 3   :   X  @tab	X
       Newtek SpeedHQ		     :	 X  @tab  X
       Nintendo Gamecube THP video   :	    @tab  X
       NotchLC		       :      @tab  X
       NuppelVideo/RTjpeg      :      @tab  X
	       @tab Video encoding used in NuppelVideo files.

       On2 VP3		       :      @tab  X
	       @tab still experimental

       On2 VP4		       :      @tab  X
	       @tab fourcc: VP40

       On2 VP5		       :      @tab  X
	       @tab fourcc: VP50

       On2 VP6		       :      @tab  X
	       @tab fourcc: VP60,VP61,VP62

       On2 VP7		       :      @tab  X
	       @tab fourcc: VP70,VP71

       VP8		       :   E  @tab  X
	       @tab fourcc: VP80, encoding supported through external library libvpx

       VP9		       :   E  @tab  X
	       @tab encoding supported through external library libvpx

       Pinnacle TARGA CineWave YUV16  :      @tab  X
	       @tab fourcc: Y216

       Q-team QPEG	       :      @tab  X
	       @tab fourccs: QPEG, Q1.0, Q1.1

       QuickTime 8BPS video    :      @tab  X
       QuickTime Animation (RLE) video	 :   X	@tab  X
	       @tab fourcc: 'rle '

       QuickTime Graphics (SMC)   :   X  @tab  X
	       @tab fourcc: 'smc '

       QuickTime video (RPZA)  :   X  @tab  X
	       @tab fourcc: rpza

       R10K AJA Kona 10-bit RGB Codec	   :   X  @tab	X
       R210 Quicktime Uncompressed RGB 10-bit	   :   X  @tab	X
       Raw Video	       :   X  @tab  X
       RealVideo 1.0	       :   X  @tab  X
       RealVideo 2.0	       :   X  @tab  X
       RealVideo 3.0	       :      @tab  X
	       @tab still far from ideal

       RealVideo 4.0	       :      @tab  X
       Renderware TXD (TeXture Dictionary)   :	    @tab  X
	       @tab Texture dictionaries used by the Renderware Engine.

       RivaTuner Video	       :      @tab  X
	       @tab fourcc: 'RTV1'

       RL2 video	       :      @tab  X
	       @tab used in some games by Entertainment Software Partners

       ScreenPressor	       :      @tab  X
       Screenpresso	       :      @tab  X
       Screen Recorder Gold Codec   :	   @tab  X
       Sierra VMD video        :      @tab  X
	       @tab Used in Sierra VMD files.

       Silicon Graphics Motion Video Compressor 1 (MVC1)   :	  @tab	X
       Silicon Graphics Motion Video Compressor 2 (MVC2)   :	  @tab	X
       Silicon Graphics RLE 8-bit video   :	 @tab  X
       Smacker video	       :      @tab  X
	       @tab Video encoding used in Smacker.

       SMPTE VC-1	       :      @tab  X
       Snow		       :   X  @tab  X
	       @tab experimental wavelet codec (fourcc: SNOW)

       Sony PlayStation MDEC (Motion DECoder)	:      @tab  X
       Sorenson Vector Quantizer 1   :	 X  @tab  X
	       @tab fourcc: SVQ1

       Sorenson Vector Quantizer 3   :	    @tab  X
	       @tab fourcc: SVQ3

       Sunplus JPEG (SP5X)     :      @tab  X
	       @tab fourcc: SP5X

       TechSmith Screen Capture Codec	:      @tab  X
	       @tab fourcc: TSCC

       TechSmith Screen Capture Codec 2   :	 @tab  X
	       @tab fourcc: TSC2

       Theora		       :   E  @tab  X
	       @tab encoding supported through external library libtheora

       Tiertex Limited SEQ video   :	  @tab	X
	       @tab Codec used in DOS CD-ROM FlashBack game.

       Ut Video 	       :   X  @tab  X
       v210 QuickTime uncompressed 4:2:2 10-bit      :	 X  @tab  X
       v308 QuickTime uncompressed 4:4:4	     :	 X  @tab  X
       v408 QuickTime uncompressed 4:4:4:4	     :	 X  @tab  X
       v410 QuickTime uncompressed 4:4:4 10-bit      :	 X  @tab  X
       VBLE Lossless Codec     :      @tab  X
       vMix Video	       :      @tab  X
	       @tab fourcc: 'VMX1'

       VMware Screen Codec / VMware Video   :	   @tab  X
	       @tab Codec used in videos captured by VMware.

       Westwood Studios VQA (Vector Quantized Animation) video	 :	@tab
       X
       Windows Media Image     :      @tab  X
       Windows Media Video 7   :   X  @tab  X
       Windows Media Video 8   :   X  @tab  X
       Windows Media Video 9   :      @tab  X
	       @tab not completely working

       Wing Commander III / Xan   :	 @tab  X
	       @tab Used in Wing Commander III .MVE files.

       Wing Commander IV / Xan	 :	@tab  X
	       @tab Used in Wing Commander IV.

       Winnov WNV1	       :      @tab  X
       WMV7		       :   X  @tab  X
       YAMAHA SMAF	       :   X  @tab  X
       Psygnosis YOP Video     :      @tab  X
       yuv4		       :   X  @tab  X
	       @tab libquicktime uncompressed packed 4:2:0

       ZeroCodec Lossless Video  :	@tab  X
       ZLIB		       :   X  @tab  X
	       @tab part of LCL, encoder experimental

       Zip Motion Blocks Video	 :    X @tab  X
	       @tab Encoder works only in PAL8.

       "X" means that the feature in that column (encoding / decoding) is
       supported.

       "E" means that support is provided through an external library.

   Audio Codecs
       Name  :	Encoding @tab Decoding @tab Comments
       8SVX exponential        :      @tab  X
       8SVX fibonacci	       :      @tab  X
       AAC		       :  EX  @tab  X
	       @tab encoding supported through internal encoder and external library libfdk-aac

       AAC+		       :   E  @tab  IX
	       @tab encoding supported through external library libfdk-aac

       AC-3		       :  IX  @tab  IX
       ACELP.KELVIN	       :      @tab  X
       ADPCM 4X Movie	       :      @tab  X
       ADPCM Yamaha AICA       :      @tab  X
       ADPCM AmuseGraphics Movie  :	@tab  X
       ADPCM Argonaut Games    :  X   @tab  X
       ADPCM CDROM XA	       :      @tab  X
       ADPCM Creative Technology  :	 @tab  X
	       @tab 16 -E<gt> 4, 8 -E<gt> 4, 8 -E<gt> 3, 8 -E<gt> 2

       ADPCM Electronic Arts   :      @tab  X
	       @tab Used in various EA titles.

       ADPCM Electronic Arts Maxis CDROM XS   :      @tab  X
	       @tab Used in Sim City 3000.

       ADPCM Electronic Arts R1   :	 @tab  X
       ADPCM Electronic Arts R2   :	 @tab  X
       ADPCM Electronic Arts R3   :	 @tab  X
       ADPCM Electronic Arts XAS  :	 @tab  X
       ADPCM G.722	       :   X  @tab  X
       ADPCM G.726	       :   X  @tab  X
       ADPCM IMA Acorn Replay  :      @tab  X
       ADPCM IMA AMV	       :   X  @tab  X
	       @tab Used in AMV files

       ADPCM IMA Cunning Developments	:      @tab  X
       ADPCM IMA Electronic Arts EACS	:      @tab  X
       ADPCM IMA Electronic Arts SEAD	:      @tab  X
       ADPCM IMA Funcom        :      @tab  X
       ADPCM IMA High Voltage Software ALP	 :   X	@tab  X
       ADPCM IMA Mobiclip MOFLEX   :	  @tab	X
       ADPCM IMA QuickTime     :   X  @tab  X
       ADPCM IMA Simon & Schuster Interactive	 :   X	@tab  X
       ADPCM IMA Ubisoft APM   :   X  @tab  X
       ADPCM IMA Loki SDL MJPEG   :	 @tab  X
       ADPCM IMA WAV	       :   X  @tab  X
       ADPCM IMA Westwood      :      @tab  X
       ADPCM ISS IMA	       :      @tab  X
	       @tab Used in FunCom games.

       ADPCM IMA Dialogic      :      @tab  X
       ADPCM IMA Duck DK3      :      @tab  X
	       @tab Used in some Sega Saturn console games.

       ADPCM IMA Duck DK4      :      @tab  X
	       @tab Used in some Sega Saturn console games.

       ADPCM IMA Radical       :      @tab  X
       ADPCM Microsoft	       :   X  @tab  X
       ADPCM MS IMA	       :   X  @tab  X
       ADPCM Nintendo Gamecube AFC   :	    @tab  X
       ADPCM Nintendo Gamecube DTK   :	    @tab  X
       ADPCM Nintendo THP   :	   @tab  X
       ADPCM Playstation       :      @tab  X
       ADPCM QT IMA	       :   X  @tab  X
       ADPCM SEGA CRI ADX      :   X  @tab  X
	       @tab Used in Sega Dreamcast games.

       ADPCM Shockwave Flash   :   X  @tab  X
       ADPCM Sound Blaster Pro 2-bit   :      @tab  X
       ADPCM Sound Blaster Pro 2.6-bit	 :	@tab  X
       ADPCM Sound Blaster Pro 4-bit   :      @tab  X
       ADPCM VIMA	       :      @tab  X
	       @tab Used in LucasArts SMUSH animations.

       ADPCM Konami XMD        :      @tab  X
       ADPCM Westwood Studios IMA	:   X @tab  X
	       @tab Used in Westwood Studios games like Command and Conquer.

       ADPCM Yamaha	       :   X  @tab  X
       ADPCM Zork	       :      @tab  X
       AMR-NB		       :   E  @tab  X
	       @tab encoding supported through external library libopencore-amrnb

       AMR-WB		       :   E  @tab  X
	       @tab encoding supported through external library libvo-amrwbenc

       Amazing Studio PAF Audio  :	@tab  X
       Apple lossless audio    :   X  @tab  X
	       @tab QuickTime fourcc 'alac'

       aptX		       :   X  @tab  X
	       @tab Used in Bluetooth A2DP

       aptX HD		       :   X  @tab  X
	       @tab Used in Bluetooth A2DP

       ATRAC1		       :      @tab  X
       ATRAC3		       :      @tab  X
       ATRAC3+		       :      @tab  X
       ATRAC9		       :      @tab  X
       Bink Audio	       :      @tab  X
	       @tab Used in Bink and Smacker files in many games.

       Bonk audio	       :      @tab  X
       CELT		       :      @tab  E
	       @tab decoding supported through external library libcelt

       codec2		       :   E  @tab  E
	       @tab en/decoding supported through external library libcodec2

       CRI HCA		       :      @tab X
       Delphine Software International CIN audio   :	  @tab	X
	       @tab Codec used in Delphine Software International games.

       DFPWM		       :   X  @tab  X
       Digital Speech Standard - Standard Play mode (DSS SP)  :      @tab  X
       Discworld II BMV Audio  :      @tab  X
       COOK		       :      @tab  X
	       @tab All versions except 5.1 are supported.

       DCA (DTS Coherent Acoustics)   :   X  @tab  X
	       @tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)

       Dolby E	 :	@tab  X
       DPCM Cuberoot-Delta-Exact  :   @tab  X
	       @tab Used in few games.

       DPCM Gremlin	       :      @tab  X
       DPCM id RoQ	       :   X  @tab  X
	       @tab Used in Quake III, Jedi Knight 2 and other computer games.

       DPCM Marble WADY        :      @tab  X
       DPCM Interplay	       :      @tab  X
	       @tab Used in various Interplay computer games.

       DPCM Squareroot-Delta-Exact   :	 @tab  X
	       @tab Used in various games.

       DPCM Sierra Online      :      @tab  X
	       @tab Used in Sierra Online game audio files.

       DPCM Sol 	       :      @tab  X
       DPCM Xan 	       :      @tab  X
	       @tab Used in Origin's Wing Commander IV AVI files.

       DPCM Xilam DERF	       :      @tab  X
       DSD (Direct Stream Digital), least significant bit first   :   @tab  X
       DSD (Direct Stream Digital), most significant bit first	  :   @tab  X
       DSD (Direct Stream Digital), least significant bit first, planar   :
       @tab  X
       DSD (Direct Stream Digital), most significant bit first, planar	  :
       @tab  X
       DSP Group TrueSpeech    :      @tab  X
       DST (Direct Stream Transfer)  :	 @tab  X
       DV audio 	       :      @tab  X
       Enhanced AC-3	       :   X  @tab  X
       EVRC (Enhanced Variable Rate Codec)  :	   @tab  X
       FLAC (Free Lossless Audio Codec)   :   X  @tab  IX
       FTR Voice	       :      @tab  X
       G.723.1		       :  X   @tab  X
       G.729		       :      @tab  X
       GSM		       :   E  @tab  X
	       @tab encoding supported through external library libgsm

       GSM Microsoft variant   :   E  @tab  X
	       @tab encoding supported through external library libgsm

       IAC (Indeo Audio Coder)	 :	@tab  X
       iLBC (Internet Low Bitrate Codec)  :   E  @tab  EX
	       @tab encoding and decoding supported through external library libilbc

       IMC (Intel Music Coder)	 :	@tab  X
       Interplay ACM		 :	@tab  X
       LC3		       :  E  @tab  E
	       @tab supported through external library liblc3

       MACE (Macintosh Audio Compression/Expansion) 6:1   :	 @tab  X
       Marian's A-pac audio	 :	@tab  X
       MI-SC4 (Micronas SC-4 Audio)   :      @tab  X
       MLP (Meridian Lossless Packing)	 :   X	@tab  X
	       @tab Used in DVD-Audio discs.

       Monkey's Audio	       :      @tab  X
       MP1 (MPEG audio layer 1)   :	 @tab IX
       MP2 (MPEG audio layer 2)   :  IX  @tab IX
	       @tab encoding supported also through external library TwoLAME

       MP3 (MPEG audio layer 3)   :   E  @tab IX
	       @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported

       MPEG-4 Audio Lossless Coding (ALS)   :	   @tab  X
       MobiClip FastAudio      :      @tab  X
       Musepack SV7	       :      @tab  X
       Musepack SV8	       :      @tab  X
       Nellymoser Asao	       :   X  @tab  X
       On2 AVC (Audio for Video Codec)	:      @tab  X
       Opus		       :   E  @tab  X
	       @tab encoding supported through external library libopus

       OSQ (Original Sound Quality)   :      @tab  X
       PCM A-law	       :   X  @tab  X
       PCM mu-law	       :   X  @tab  X
       PCM Archimedes VIDC     :   X  @tab  X
       PCM signed 8-bit planar	 :   X	@tab  X
       PCM signed 16-bit big-endian planar   :	 X  @tab  X
       PCM signed 16-bit little-endian planar	:   X  @tab  X
       PCM signed 24-bit little-endian planar	:   X  @tab  X
       PCM signed 32-bit little-endian planar	:   X  @tab  X
       PCM 32-bit floating point big-endian   :   X  @tab  X
       PCM 32-bit floating point little-endian	 :   X	@tab  X
       PCM 64-bit floating point big-endian   :   X  @tab  X
       PCM 64-bit floating point little-endian	 :   X	@tab  X
       PCM D-Cinema audio signed 24-bit    :   X  @tab	X
       PCM signed 8-bit        :   X  @tab  X
       PCM signed 16-bit big-endian   :   X  @tab  X
       PCM signed 16-bit little-endian	 :   X	@tab  X
       PCM signed 24-bit big-endian   :   X  @tab  X
       PCM signed 24-bit little-endian	 :   X	@tab  X
       PCM signed 32-bit big-endian   :   X  @tab  X
       PCM signed 32-bit little-endian	 :   X	@tab  X
       PCM signed 16/20/24-bit big-endian in MPEG-TS   :      @tab  X
       PCM unsigned 8-bit      :   X  @tab  X
       PCM unsigned 16-bit big-endian	:   X  @tab  X
       PCM unsigned 16-bit little-endian   :   X  @tab	X
       PCM unsigned 24-bit big-endian	:   X  @tab  X
       PCM unsigned 24-bit little-endian   :   X  @tab	X
       PCM unsigned 32-bit big-endian	:   X  @tab  X
       PCM unsigned 32-bit little-endian   :   X  @tab	X
       PCM SGA		       :      @tab  X
       QCELP / PureVoice       :      @tab  X
       QDesign Music Codec 1   :      @tab  X
       QDesign Music Codec 2   :      @tab  X
	       @tab There are still some distortions.

       RealAudio 1.0 (14.4K)   :   X  @tab  X
	       @tab Real 14400 bit/s codec

       RealAudio 2.0 (28.8K)   :      @tab  X
	       @tab Real 28800 bit/s codec

       RealAudio 3.0 (dnet)    :  IX  @tab  X
	       @tab Real low bitrate AC-3 codec

       RealAudio Lossless      :      @tab  X
       RealAudio SIPR / ACELP.NET  :	  @tab	X
       RK Audio (RKA)	       :      @tab  X
       SBC (low-complexity subband codec)  :   X  @tab	X
	       @tab Used in Bluetooth A2DP

       Shorten		       :      @tab  X
       Sierra VMD audio        :      @tab  X
	       @tab Used in Sierra VMD files.

       Smacker audio	       :      @tab  X
       SMPTE 302M AES3 audio   :   X  @tab  X
       Sonic		       :   X  @tab  X
	       @tab experimental codec

       Sonic lossless	       :   X  @tab  X
	       @tab experimental codec

       Speex		       :   E  @tab  EX
	       @tab supported through external library libspeex

       TAK (Tom's lossless Audio Kompressor)   :      @tab  X
       True Audio (TTA)        :   X  @tab  X
       TrueHD		       :   X  @tab  X
	       @tab Used in HD-DVD and Blu-Ray discs.

       TwinVQ (VQF flavor)     :      @tab  X
       VIMA		       :      @tab  X
	       @tab Used in LucasArts SMUSH animations.

       ViewQuest VQC	       :      @tab  X
       Vorbis		       :   E  @tab  X
	       @tab A native but very primitive encoder exists.

       Voxware MetaSound       :      @tab  X
       Waveform Archiver       :      @tab  X
       WavPack		       :   X  @tab  X
       Westwood Audio (SND1)   :      @tab  X
       Windows Media Audio 1   :   X  @tab  X
       Windows Media Audio 2   :   X  @tab  X
       Windows Media Audio Lossless  :	 @tab  X
       Windows Media Audio Pro	:     @tab  X
       Windows Media Audio Voice  :   @tab  X
       Xbox Media Audio 1      :      @tab  X
       Xbox Media Audio 2      :      @tab  X

       "X" means that the feature in that column (encoding / decoding) is
       supported.

       "E" means that support is provided through an external library.

       "I" means that an integer-only version is available, too (ensures high
       performance on systems without hardware floating point support).

   Subtitle Formats
       Name  :	Muxing @tab Demuxing @tab Encoding @tab Decoding
       3GPP Timed Text	 :    @tab   @tab X @tab X
       AQTitle		 :    @tab X @tab   @tab X
       DVB		 :  X @tab X @tab X @tab X
       DVB teletext	 :    @tab X @tab   @tab E
       DVD		 :  X @tab X @tab X @tab X
       JACOsub		 :  X @tab X @tab   @tab X
       MicroDVD 	 :  X @tab X @tab   @tab X
       MPL2		 :    @tab X @tab   @tab X
       MPsub (MPlayer)	 :    @tab X @tab   @tab X
       PGS		 :    @tab   @tab   @tab X
       PJS (Phoenix)	 :    @tab X @tab   @tab X
       RealText 	 :    @tab X @tab   @tab X
       SAMI		 :    @tab X @tab   @tab X
       Spruce format (STL)  :	 @tab X @tab   @tab X
       SSA/ASS		 :  X @tab X @tab X @tab X
       SubRip (SRT)	 :  X @tab X @tab X @tab X
       SubViewer v1	 :    @tab X @tab   @tab X
       SubViewer	 :    @tab X @tab   @tab X
       TED Talks captions  :  @tab X @tab   @tab X
       TTML		 :  X @tab   @tab X @tab
       VobSub (IDX+SUB)  :    @tab X @tab   @tab X
       VPlayer		 :    @tab X @tab   @tab X
       WebVTT		 :  X @tab X @tab X @tab X
       XSUB		 :    @tab   @tab X @tab X

       "X" means that the feature is supported.

       "E" means that support is provided through an external library.

   Network Protocols
       Name	     :	Support
       AMQP	     :	E
       file	     :	X
       FTP	     :	X
       Gopher	     :	X
       Gophers	     :	X
       HLS	     :	X
       HTTP	     :	X
       HTTPS	     :	X
       Icecast	     :	X
       MMSH	     :	X
       MMST	     :	X
       pipe	     :	X
       Pro-MPEG FEC  :	X
       RTMP	     :	X
       RTMPE	     :	X
       RTMPS	     :	X
       RTMPT	     :	X
       RTMPTE	     :	X
       RTMPTS	     :	X
       RTP	     :	X
       SAMBA	     :	E
       SCTP	     :	X
       SFTP	     :	E
       TCP	     :	X
       TLS	     :	X
       UDP	     :	X
       ZMQ	     :	E

       "X" means that the protocol is supported.

       "E" means that support is provided through an external library.

   Input/Output Devices
       Name		  :  Input  @tab Output
       ALSA		  :  X	    @tab X
       BKTR		  :  X	    @tab
       caca		  :	    @tab X
       DV1394		  :  X	    @tab
       Lavfi virtual device  :	X   @tab
       Linux framebuffer  :  X	    @tab X
       JACK		  :  X	    @tab
       LIBCDIO		  :  X
       LIBDC1394	  :  X	    @tab
       OpenAL		  :  X
       OpenGL		  :	    @tab X
       OSS		  :  X	    @tab X
       PulseAudio	  :  X	    @tab X
       SDL		  :	    @tab X
       Video4Linux2	  :  X	    @tab X
       VfW capture	  :  X	    @tab
       X11 grabbing	  :  X	    @tab
       Win32 grabbing	  :  X	    @tab

       "X" means that input/output is supported.

   Timecode
       Codec/format	  :  Read   @tab Write
       AVI		  :  X	    @tab X
       DV		  :  X	    @tab X
       GXF		  :  X	    @tab X
       MOV		  :  X	    @tab X
       MPEG1/2		  :  X	    @tab X
       MXF		  :  X	    @tab X

SEE ALSO
       ffmpeg(1), ffplay(1), ffprobe(1), ffmpeg-utils(1), ffmpeg-scaler(1),
       ffmpeg-resampler(1), ffmpeg-codecs(1), ffmpeg-bitstream-filters(1),
       ffmpeg-formats(1), ffmpeg-devices(1), ffmpeg-protocols(1),
       ffmpeg-filters(1)

AUTHORS
       The FFmpeg developers.

       For details about the authorship, see the Git history of the project
       (https://git.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
       the FFmpeg source directory, or browsing the online repository at
       <https://git.ffmpeg.org/ffmpeg>.

       Maintainers for the specific components are listed in the file
       MAINTAINERS in the source code tree.

								 FFMPEG-ALL(1)
