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

NAME
       ffmpeg -	ffmpeg video converter

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

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

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

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

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

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

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

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

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

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

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

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

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

       The format option may be	needed for raw input files.

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

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

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

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

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

       Simple filtergraphs

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

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

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

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

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

       Complex filtergraphs

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

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

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

       The -lavfi option is equivalent to -filter_complex.

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

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

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

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

STREAM SELECTION
       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.

   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 program ID	is also	specified. In this
	   case	it is based on the ordering of the streams in the 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.

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

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

       -version
	   Show	version.

       -formats
	   Show	available formats (including devices).

       -demuxers
	   Show	available demuxers.

       -muxers
	   Show	available muxers.

       -devices
	   Show	available devices.

       -codecs
	   Show	all codecs known to libavcodec.

	   Note	that the term 'codec' is used throughout this documentation as
	   a shortcut for what is more correctly called	a media	bitstream
	   format.

       -decoders
	   Show	available decoders.

       -encoders
	   Show	all available encoders.

       -bsfs
	   Show	available bitstream filters.

       -protocols
	   Show	available protocols.

       -filters
	   Show	available libavfilter filters.

       -pix_fmts
	   Show	available pixel	formats.

       -sample_fmts
	   Show	available sample formats.

       -layouts
	   Show	channel	names and standard channel layouts.

       -colors
	   Show	recognized color names.

       -sources	device[,opt1=val1[,opt2=val2]...]
	   Show	autodetected sources of	the input device.  Some	devices	may
	   provide system-dependent source names that cannot be	autodetected.
	   The returned	list cannot be assumed to be always complete.

		   ffmpeg -sources pulse,server=192.168.0.4

       -sinks device[,opt1=val1[,opt2=val2]...]
	   Show	autodetected sinks of the output device.  Some devices may
	   provide system-dependent sink names that cannot be autodetected.
	   The returned	list cannot be assumed to be always complete.

		   ffmpeg -sinks pulse,server=192.168.0.4

       -loglevel [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

   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.

       -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.

       -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.

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

	   The following dispositions are recognized:

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

	   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.

       -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

       -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.

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

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

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

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

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

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

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

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

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

	   See also the	option "-fdebug	ts".

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

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

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

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

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

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

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

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

		   ffmpeg -dump_attachment:t ""	-i INPUT

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

       -noautorotate
	   Disable automatically rotating video	based on file metadata.

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

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

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

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

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

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

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

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

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

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

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

       -vn (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.

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

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

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

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

       -psnr
	   Calculate PSNR of compressed	frames.

       -vstats
	   Dump	video coding statistics	to vstats_HHMMSS.log.

       -vstats_file file
	   Dump	video coding statistics	to file.

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

	   version = 1 :

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

	   version > 1:

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

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

       -dc precision
	   Intra_dc_precision.

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

       -qphist (global)
	   Show	QP histogram

       -vbsf bitstream_filter
	   Deprecated see -bsf

       -force_key_frames[:stream_specifier] time[,time...] (output,per-stream)
       -force_key_frames[:stream_specifier] expr:expr (output,per-stream)
       -force_key_frames[: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.

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

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

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

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

	   cuda
	       device is the number of the CUDA	device.

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

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

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

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

	       auto
	       sw
	       hw
	       auto_any
	       hw_any
	       hw2
	       hw3
	       hw4

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

	   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.

	   vaapi
	       Use VAAPI (Video	Acceleration API) hardware acceleration.

	   qsv Use the Intel QuickSync Video acceleration for video
	       transcoding.

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

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

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

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

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

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

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

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

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

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

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

       -an (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".

       -absf bitstream_filter
	   Deprecated, see -bsf

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

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

       -sn (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.

       -sbsf bitstream_filter
	   Deprecated, see -bsf

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

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

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

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

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

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

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

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

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

		   ffmpeg -i INPUT -map	0 output

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

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

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

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

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

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

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

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

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

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

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

	   To pick the English audio stream:

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

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

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

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

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

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

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

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

		   ffmpeg -i INPUT -map_channel	0.0.1 -map_channel 0.0.0 OUTPUT

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	   If metadata specifier is omitted, it	defaults to global.

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

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

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

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

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

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

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

       -benchmark (global)
	   Show	benchmarking information at the	end of an encode.  Shows 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.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	   1   Use the demuxer timebase.

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

	   0   Use the decoder timebase.

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

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

	   Default value is -1.

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

	   0   Assign a	default	value according	to the media type.

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

	   -1  Use the input stream timebase when possible.

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

	   >0  Use the provided	number as the timebase.

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

	   Default value is 0.

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

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

       -dts_delta_threshold
	   Timestamp discontinuity delta threshold.

       -dts_error_threshold seconds
	   Timestamp error delta threshold. This threshold use to discard
	   crazy/damaged timestamps and	the default is 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 (output,per-stream)
	   Set bitstream filters for matching streams. bitstream_filters is a
	   comma-separated list	of bitstream filters. Use the "-bsfs" option
	   to get the list of bitstream	filters.

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

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

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

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

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

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

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

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

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

	   For example,	to overlay an image over video

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       -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.

       -xerror (global)
	   Stop	and exit on error

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

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

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

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

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

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

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

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

       ffpreset	files

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

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

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

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

       avpreset	files

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

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

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

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

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

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

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

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

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

   X11 grabbing
       Grab the	X11 display with ffmpeg	via

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

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

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

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

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

       Examples:

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

       In both expressions, the	optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

       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

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

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

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

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

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

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

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

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

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

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

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

       downmix
	   DL+DR

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

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

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

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

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

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

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

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

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

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

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

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

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

       The following functions are available:

       abs(x)
	   Compute absolute value of x.

       acos(x)
	   Compute arccosine of	x.

       asin(x)
	   Compute arcsine of x.

       atan(x)
	   Compute arctangent of x.

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

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

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

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

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

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

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

       cos(x)
	   Compute cosine of x.

       cosh(x)
	   Compute hyperbolic cosine of	x.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       log(x)
	   Compute natural logarithm of	x.

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

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

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

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

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

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

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

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

	   Prints t with loglevel l

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

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

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

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

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

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

       tan(x)
	   Compute tangent of x.

       tanh(x)
	   Compute hyperbolic tangent of x.

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

	   When	the series does	not converge the result	is undefined.

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

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

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

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

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

       The following constants are available:

       PI  area	of the unit disc, approximately	3.14

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

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

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

       "*" works like AND

       "+" works like OR

       For example the construct:

	       if (A AND B) then C

       is equivalent to:

	       if(A*B, C)

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

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

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

       y   10^-24 / 2^-80

       z   10^-21 / 2^-70

       a   10^-18 / 2^-60

       f   10^-15 / 2^-50

       p   10^-12 / 2^-40

       n   10^-9 / 2^-30

       u   10^-6 / 2^-20

       m   10^-3 / 2^-10

       c   10^-2

       d   10^-1

       h   10^2

       k   10^3	/ 2^10

       K   10^3	/ 2^10

       M   10^6	/ 2^20

       G   10^9	/ 2^30

       T   10^12 / 2^40

       P   10^15 / 2^40

       E   10^18 / 2^50

       Z   10^21 / 2^60

       Y   10^24 / 2^70

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.

	   emu_edge
	       Do not draw edges.

	   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.

	   cbp Deprecated, use mpegvideo private options instead.

	   qprd
	       Deprecated, use mpegvideo private options instead.

	   ilme
	       Apply interlaced	motion estimation.

	   cgop
	       Use closed gop.

	   output_corrupt
	       Output even potentially corrupted frames.

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

	   Possible values:

	   zero
	       zero motion estimation (fastest)

	   full
	       full motion estimation (slowest)

	   epzs
	       EPZS motion estimation (default)

	   esa esa motion estimation (alias for	full)

	   tesa
	       tesa motion estimation

	   dia dia motion estimation (alias for	epzs)

	   log log motion estimation

	   phods
	       phods motion estimation

	   x1  X1 motion estimation

	   hex hex motion estimation

	   umh umh motion estimation

	   iter
	       iter motion estimation

       extradata_size integer
	   Set extradata size.

       time_base rational number
	   Set codec time base.

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

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

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

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

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

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

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

       frame_number integer
	   Set the frame number.

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

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

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

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

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

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

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

	   Default value is 0.

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

       rc_strategy integer (encoding,video)
	   Set ratecontrol method.

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

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

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

	   Possible values:

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

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

	   ump4
	       (autodetected if	fourcc==UMP4)

	   no_padding
	       padding bug (autodetected)

	   amv
	   ac_vlc
	       illegal vlc bug (autodetected per fourcc)

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

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

	   edge
	       edge padding bug	(autodetected per fourcc/version)

	   hpel_chroma
	   dc_clip
	   ms  Workaround various bugs in microsoft broken decoders.

	   trunc
	       trancated frames

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

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

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

	   Possible values:

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

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

	   normal
	   unofficial
	       allow unofficial	extensions

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

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

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

	   Possible values:

	   crccheck
	       verify embedded CRCs

	   bitstream
	       detect bitstream	specification deviations

	   buffer
	       detect improper bitstream length

	   explode
	       abort decoding on minor error detection

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

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

	   compliant
	       consider	all spec non compliancies as errors

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

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

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

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

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

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

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

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

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

       rc_buf_aggressivity float (encoding,video)
	   Currently useless.

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

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

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

       dct integer (encoding,video)
	   Set DCT algorithm.

	   Possible values:

	   auto
	       autoselect a good one (default)

	   fastint
	       fast integer

	   int accurate	integer

	   mmx
	   altivec
	   faan
	       floating	point AAN DCT

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

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

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

       p_mask float (encoding,video)
	   Set inter masking.

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

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

	   Possible values:

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

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

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

	   Possible values:

	   guess_mvs
	       iterative motion	vector (MV) search (slow)

	   deblock
	       use strong deblock filter for damaged MBs

	   favor_inter
	       favor predicting	from the previous frame	instead	of the current

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

	   Possible values:

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

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

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

	   Possible values:

	   pict
	       picture info

	   rc  rate control

	   bitstream
	   mb_type
	       macroblock (MB) type

	   qp  per-block quantization parameter	(QP)

	   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.

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

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard	transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal,	slow

	   zero
	       0

	   vsad
	       sum of absolute vertical	differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving	sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

	   vlc variable	length coder / huffman coder

	   ac  arithmetic coder

	   raw raw (no encoding)

	   rle run-length coder

	   deflate
	       deflate-based coder

       context integer (encoding,video)
	   Set context model.

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

	   Possible values:

	   simple
	       use mbcmp (default)

	   bits
	       use fewest bits

	   rd  use best	rate distortion

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

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

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

       nr integer (encoding,video)
	   Set noise reduction.

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

       flags2 flags (decoding/encoding,audio,video,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.

       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.

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

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

	   Possible values:

	   auto, 0
	       automatically select the	number of threads to set

	   Default value is auto.

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

       mb_threshold integer (encoding,video)
	   Set macroblock threshold.

       dc integer (encoding,video)
	   Set intra_dc_precision.

       nssew integer (encoding,video)
	   Set nsse weight.

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

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

       profile integer (encoding,audio,video)
	   Set encoder codec profile. Default value is unknown.	Encoder
	   specific profiles are documented in the relevant encoder
	   documentation.

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

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

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

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

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

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard	transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal,	slow

	   zero
	       0

	   vsad
	       sum of absolute vertical	differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving	sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

	   Possible values:

	   none
	       Discard no frame.

	   default
	       Discard useless frames like 0-sized frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   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.

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

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

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

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

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

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

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

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

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

       request_channel_layout integer (decoding,audio)
	   Possible values:

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

	   bt709
	       BT.709

	   bt470m
	       BT.470 M

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   film
	       Film

	   bt2020
	       BT.2020

	   smpte428
	   smpte428_1
	       SMPTE ST	428-1

	   smpte431
	       SMPTE 431-2

	   smpte432
	       SMPTE 432-1

	   jedec-p22
	       JEDEC P22

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

	   bt709
	       BT.709

	   gamma22
	       BT.470 M

	   gamma28
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   linear
	       Linear

	   log
	   log100
	       Log

	   log_sqrt
	   log316
	       Log square root

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

	   bt1361
	   bt1361e
	       BT.1361

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

	   bt2020_10
	   bt2020_10bit
	       BT.2020 - 10 bit

	   bt2020_12
	   bt2020_12bit
	       BT.2020 - 12 bit

	   smpte2084
	       SMPTE ST	2084

	   smpte428
	   smpte428_1
	       SMPTE ST	428-1

	   arib-std-b67
	       ARIB STD-B67

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

	   rgb RGB

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   ycocg
	       YCOCG

	   bt2020nc
	   bt2020_ncl
	       BT.2020 NCL

	   bt2020c
	   bt2020_cl
	       BT.2020 CL

	   smpte2085
	       SMPTE 2085

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

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

	   pc
	   jpeg
	       JPEG (2^n-1)

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

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

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

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

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

	   Possible values:

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

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

	   frame
	       Decode more than	one frame at once.

	   Default value is slice+frame.

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

	   Possible values:

	   ma  Main Audio Service

	   ef  Effects

	   vi  Visually	Impaired

	   hi  Hearing Impaired

	   di  Dialogue

	   co  Commentary

	   em  Emergency

	   vo  Voice Over

	   ka  Karaoke

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

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

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

	   progressive
	       Progressive video

	   tt  Interlaced video, top field coded and displayed first

	   bb  Interlaced video, bottom	field coded and	displayed first

	   tb  Interlaced video, top coded first, bottom displayed first

	   bt  Interlaced video, bottom	coded first, top displayed first

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

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

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

   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).

       tilethreads
	   Set amount of tile threads to use during decoding. 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.

       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.

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

   ac3
       AC-3 audio decoder.

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

       AC-3 Decoder Options

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

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

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

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

   flac
       FLAC audio decoder.

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

       FLAC Decoder options

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

   ffwavesynth
       Internal	wave synthesizer.

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

   libcelt
       libcelt decoder wrapper.

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

   libgsm
       libgsm decoder wrapper.

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

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

   libilbc
       libilbc decoder wrapper.

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

       Options

       The following option is supported by the	libilbc	wrapper.

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

   libopencore-amrnb
       libopencore-amrnb decoder wrapper.

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

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

   libopencore-amrwb
       libopencore-amrwb decoder wrapper.

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

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

   libopus
       libopus decoder wrapper.

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

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

SUBTITLES DECODERS
   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.

   dvbsub
       Options

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

	   0   Never compute CLUT

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

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

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

       Options

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

	   The format for this option is a string containing 16	24-bits
	   hexadecimal numbers (without	0x prefix) separated by	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 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.
	       This is the default choice for a	coder

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

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

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

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

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

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

       profile
	   Sets	the encoding profile, possible values:

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

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

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

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

	   If this option is unspecified it is set to aac_low.

   ac3 and ac3_fixed
       AC-3 audio encoders.

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

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

       AC-3 Metadata

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

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

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

       Metadata	Control	Options

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

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

	   1   Metadata	values can be changed before encoding each frame.

       Downmix Levels

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

	   0.707
	       Apply -3dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6dB gain

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

	   0.707
	       Apply -3dB gain

	   0.500
	       Apply -6dB gain (default)

	   0.000
	       Silence Surround	Channel(s)

       Audio Production	Information

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

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

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   large
	       Large Room

	   2
	   small
	       Small Room

       Other Metadata Options

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

	   0
	   off No Copyright Exists (default)

	   1
	   on  Copyright Exists

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

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   off Not Dolby Surround Encoded

	   2
	   on  Dolby Surround Encoded

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

	   0
	   off Not Original Source

	   1
	   on  Original	Source (default)

       Extended	Bitstream Information

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

       Extended	Bitstream Information -	Part 1

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   ltrt
	       Lt/Rt Downmix Preferred

	   2
	   loro
	       Lo/Ro Downmix Preferred

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

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

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

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround	Channel(s)

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

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

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

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround	Channel(s)

       Extended	Bitstream Information -	Part 2

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Surround EX Off

	   2
	   off Dolby Surround EX On

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Headphone Off

	   2
	   off Dolby Headphone On

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

	   0
	   standard
	       Standard	A/D Converter (default)

	   1
	   hdcd
	       HDCD A/D	Converter

       Other AC-3 Encoding Options

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

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

       Floating-Point-Only AC-3	Encoding Options

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

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

	   -1
	   auto
	       Selected	by Encoder (default)

	   0
	   off Disable Channel Coupling

	   1
	   on  Enable Channel Coupling

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

	   -1
	   auto
	       Selected	by Encoder (default)

   flac
       FLAC (Free Lossless Audio Codec)	Encoder

       Options

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

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

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

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

       lpc_type
	   Sets	the first stage	LPC algorithm

	   none
	       LPC is not used

	   fixed
	       fixed LPC coefficients

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

       min_partition_order
	   The minimum partition order

       max_partition_order
	   The maximum partition order

       prediction_order_method
	   estimation
	   2level
	   4level
	   8level
	   search
	       Bruteforce search

	   log
       ch_mode
	   Channel mode

	   auto
	       The mode	is chosen automatically	for each frame

	   indep
	       Channels	are independently coded

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

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

   opus
       Opus encoder.

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

       Options

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

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

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

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

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

       This encoder 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.

       Examples

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

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

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

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

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

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

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

       Options

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

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

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

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

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

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

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

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

   libopencore-amrnb
       OpenCORE	Adaptive Multi-Rate Narrowband encoder.

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

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

       Options

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

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

   libopus
       libopus Opus Interactive	Audio Codec encoder wrapper.

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

       Option Mapping

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

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

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

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

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

	   constrained (cvbr)
	       Use constrained variable	bit rate encoding.

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

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

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

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

	   voip
	       Favor improved speech intelligibility.

	   audio
	       Favor faithfulness to the input (the default).

	   lowdelay
	       Restrict	to only	the lowest delay modes.

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

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

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

       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.

   libwavpack
       A wrapper providing WavPack encoding through libwavpack.

       Only lossless mode using	32-bit integer samples is supported currently.

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

       Note that a libavcodec-native encoder for the WavPack codec exists so
       users can encode	audios with this codec without using this encoder. See
       wavpackenc.

       Options

       wavpack command line utility's corresponding options are	listed in
       parentheses, if any.

       frame_size (--blocksize)
	   Default is 32768.

       compression_level
	   Set speed vs. compression tradeoff. Acceptable arguments are	listed
	   below:

	   0 (-f)
	       Fast mode.

	   1   Normal (default)	settings.

	   2 (-h)
	       High quality.

	   3 (-hh)
	       Very high quality.

	   4-8 (-hh -xEXTRAPROC)
	       Same as 3, but with extra processing enabled.

	       4 is the	same as	-x2 and	8 is the same as -x6.

   mjpeg
       Motion JPEG encoder.

       Options

       huffman
	   Set the huffman encoding strategy. Possible values:

	   default
	       Use the default huffman tables. This is the default strategy.

	   optimal
	       Compute and use optimal huffman tables.

   wavpack
       WavPack lossless	audio encoder.

       This is a libavcodec-native WavPack encoder. There is also an encoder
       based on	libwavpack, but	there is virtually no reason to	use that
       encoder.

       See also	libwavpack.

       Options

       The equivalent options for wavpack command line utility are listed in
       parentheses.

       Shared options

       The following shared options are	effective for this encoder. Only
       special notes about this	particular encoder will	be documented here.
       For the general meaning of the options, see the Codec Options chapter.

       frame_size (--blocksize)
	   For this encoder, the range for this	option is between 128 and
	   131072. Default is automatically decided based on sample rate and
	   number of channel.

	   For the complete formula of calculating default, see
	   libavcodec/wavpackenc.c.

       compression_level (-f, -h, -hh, and -x)
	   This	option's syntax	is consistent with libwavpack's.

       Private options

       joint_stereo (-j)
	   Set whether to enable joint stereo. Valid values are:

	   on (1)
	       Force mid/side audio encoding.

	   off (0)
	       Force left/right	audio encoding.

	   auto
	       Let the encoder decide automatically.

       optimize_mono
	   Set whether to enable optimization for mono.	This option is only
	   effective for non-mono streams. Available values:

	   on  enabled

	   off disabled

VIDEO ENCODERS
       A description of	some of	the currently available	video encoders
       follows.

   Hap
       Vidvox Hap video	encoder.

       Options

       format integer
	   Specifies the Hap format to encode.

	   hap
	   hap_alpha
	   hap_q

	   Default value is hap.

       chunks integer
	   Specifies the number	of chunks to split frames into,	between	1 and
	   64. This permits multithreaded decoding of large frames,
	   potentially at the cost of data-rate. The encoder may modify	this
	   value to divide frames evenly.

	   Default value is 1.

       compressor integer
	   Specifies the second-stage compressor to use. If set	to none,
	   chunks will be limited to 1,	as chunked uncompressed	frames offer
	   no benefit.

	   none
	   snappy

	   Default value is snappy.

   jpeg2000
       The native jpeg 2000 encoder is lossy by	default, the "-q:v" option can
       be used to set the encoding quality. Lossless encoding can be selected
       with "-pred 1".

       Options

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

   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.

   libkvazaar
       Kvazaar H.265/HEVC encoder.

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

       Options

       b   Set target video bitrate in bit/s and enable	rate control.

       kvazaar-params
	   Set kvazaar parameters as a list of name=value pairs	separated by
	   commas (,). See kvazaar documentation for a list of options.

   libopenh264
       Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libopenh264 headers and
       library during configuration. You need to explicitly configure the
       build with "--enable-libopenh264". The library is detected using	pkg-
       config.

       For more	information about the library see <http://www.openh264.org>.

       Options

       The following FFmpeg global options affect the configurations of	the
       libopenh264 encoder.

       b   Set the bitrate (as a number	of bits	per second).

       g   Set the GOP size.

       maxrate
	   Set the max bitrate (as a number of bits per	second).

       flags +global_header
	   Set global header in	the bitstream.

       slices
	   Set the number of slices, used in parallelized encoding. Default
	   value is 0. This is only used when slice_mode is set	to fixed.

       slice_mode
	   Set slice mode. Can assume one of the following possible values:

	   fixed
	       a fixed number of slices

	   rowmb
	       one slice per row of macroblocks

	   auto
	       automatic number	of slices according to number of threads

	   dyn dynamic slicing

	   Default value is auto.

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

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

       max_nal_size
	   Set maximum NAL size	in bytes.

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

   libtheora
       libtheora Theora	encoder	wrapper.

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

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

       Options

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

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

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

       g   Set the GOP size.

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

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

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

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

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

       Examples

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

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

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

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

   libvpx
       VP8/VP9 format supported	through	libvpx.

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

       Options

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

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

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

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

       g (kf-max-dist)
       keyint_min (kf-min-dist)
       qmin (min-q)
       qmax (max-q)
       bufsize (buf-sz,	buf-optimal-sz)
	   Set ratecontrol buffer size (in bits). Note vpxenc's	options	are
	   specified in	milliseconds, the libvpx wrapper converts this value
	   as follows: "buf-sz = bufsize * 1000	/ bitrate", "buf-optimal-sz =
	   bufsize * 1000 / bitrate * 5	/ 6".

       rc_init_occupancy (buf-initial-sz)
	   Set number of bits which should be loaded into the rc buffer	before
	   decoding starts. Note vpxenc's option is specified in milliseconds,
	   the libvpx wrapper converts this value as follows:
	   "rc_init_occupancy *	1000 / bitrate".

       undershoot-pct
	   Set datarate	undershoot (min) percentage of the target bitrate.

       overshoot-pct
	   Set datarate	overshoot (max)	percentage of the target bitrate.

       skip_threshold (drop-frame)
       qcomp (bias-pct)
       maxrate (maxsection-pct)
	   Set GOP max bitrate in bits/s. Note vpxenc's	option is specified as
	   a percentage	of the target bitrate, the libvpx wrapper converts
	   this	value as follows: "(maxrate * 100 / bitrate)".

       minrate (minsection-pct)
	   Set GOP min bitrate in bits/s. Note vpxenc's	option is specified as
	   a percentage	of the target bitrate, the libvpx wrapper converts
	   this	value as follows: "(minrate * 100 / bitrate)".

       minrate,	maxrate, b end-usage=cbr
	   "(minrate ==	maxrate	== bitrate)".

       crf (end-usage=cq, cq-level)
       tune (tune)
	   psnr	(psnr)
	   ssim	(ssim)
       quality,	deadline (deadline)
	   best
	       Use best	quality	deadline. Poorly named and quite slow, this
	       option should be	avoided	as it may give worse quality output
	       than good.

	   good
	       Use good	quality	deadline. This is a good trade-off between
	       speed and quality when used with	the cpu-used option.

	   realtime
	       Use realtime quality deadline.

       speed, cpu-used (cpu-used)
	   Set quality/speed ratio modifier. Higher values speed up the	encode
	   at the cost of quality.

       nr (noise-sensitivity)
       static-thresh
	   Set a change	threshold on blocks below which	they will be skipped
	   by the encoder.

       slices (token-parts)
	   Note	that FFmpeg's slices option gives the total number of
	   partitions, while vpxenc's token-parts is given as
	   "log2(partitions)".

       max-intra-rate
	   Set maximum I-frame bitrate as a percentage of the target bitrate.
	   A value of 0	means unlimited.

       force_key_frames
	   "VPX_EFLAG_FORCE_KF"

       Alternate reference frame related
	   auto-alt-ref
	       Enable use of alternate reference frames	(2-pass	only).	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.

       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.

       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.

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

   libwebp
       libwebp WebP Image encoder wrapper

       libwebp is Google's official encoder for	WebP images. It	can encode in
       either lossy or lossless	mode. Lossy images are essentially a wrapper
       around a	VP8 frame. Lossless images are a separate codec	developed by
       Google.

       Pixel Format

       Currently, libwebp only supports	YUV420 for lossy and RGB for lossless
       due to limitations of the format	and libwebp. Alpha is supported	for
       either mode.  Because of	API limitations, if RGB	is passed in when
       encoding	lossy or YUV is	passed in for encoding lossless, the pixel
       format will automatically be converted using functions from libwebp.
       This is not ideal and is	done only for convenience.

       Options

       -lossless boolean
	   Enables/Disables use	of lossless mode. Default is 0.

       -compression_level integer
	   For lossy, this is a	quality/speed tradeoff.	Higher values give
	   better quality for a	given size at the cost of increased encoding
	   time. For lossless, this is a size/speed tradeoff. Higher values
	   give	smaller	size at	the cost of increased encoding time. More
	   specifically, it controls the number	of extra algorithms and
	   compression tools used, and varies the combination of these tools.
	   This	maps to	the method option in libwebp. The valid	range is 0 to
	   6.  Default is 4.

       -qscale float
	   For lossy encoding, this controls image quality, 0 to 100. For
	   lossless encoding, this controls the	effort and time	spent at
	   compressing more. The default value is 75. Note that	for usage via
	   libavcodec, this option is called global_quality and	must be
	   multiplied by FF_QP2LAMBDA.

       -preset type
	   Configuration preset. This does some	automatic settings based on
	   the general type of the image.

	   none
	       Do not use a preset.

	   default
	       Use the encoder default.

	   picture
	       Digital picture,	like portrait, inner shot

	   photo
	       Outdoor photograph, with	natural	lighting

	   drawing
	       Hand or line drawing, with high-contrast	details

	   icon
	       Small-sized colorful images

	   text
	       Text-like

   libx264, libx264rgb
       x264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libx264 headers and library
       during configuration. You need to explicitly configure the build	with
       "--enable-libx264".

       libx264 supports	an impressive number of	features, including 8x8	and
       4x4 adaptive spatial transform, adaptive	B-frame	placement, CAVLC/CABAC
       entropy coding, interlacing (MBAFF), lossless mode, psy optimizations
       for detail retention (adaptive quantization, psy-RD, psy-trellis).

       Many libx264 encoder options are	mapped to FFmpeg global	codec options,
       while unique encoder options are	provided through private options.
       Additionally the	x264opts and x264-params private options allows	one to
       pass a list of key=value	tuples as accepted by the libx264
       "x264_param_parse" function.

       The x264	project	website	is at
       <http://www.videolan.org/developers/x264.html>.

       The libx264rgb encoder is the same as libx264, except it	accepts	packed
       RGB pixel formats as input instead of YUV.

       Supported Pixel Formats

       x264 supports 8-	to 10-bit color	spaces.	The exact bit depth is
       controlled at x264's configure time. FFmpeg only	supports one bit depth
       in one particular build.	In other words,	it is not possible to build
       one FFmpeg with multiple	versions of x264 with different	bit depths.

       Options

       The following options are supported by the libx264 wrapper. The
       x264-equivalent options or values are listed in parentheses for easy
       migration.

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

       To get a	more accurate and extensive documentation of the libx264
       options,	invoke the command x264	--fullhelp or consult the libx264
       documentation.

       b (bitrate)
	   Set bitrate in bits/s. Note that FFmpeg's b option is expressed in
	   bits/s, while x264's	bitrate	is in kilobits/s.

       bf (bframes)
       g (keyint)
       qmin (qpmin)
	   Minimum quantizer scale.

       qmax (qpmax)
	   Maximum quantizer scale.

       qdiff (qpstep)
	   Maximum difference between quantizer	scales.

       qblur (qblur)
	   Quantizer curve blur

       qcomp (qcomp)
	   Quantizer curve compression factor

       refs (ref)
	   Number of reference frames each P-frame can use. The	range is from
	   0-16.

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

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

       nr  (nr)
       me_range	(merange)
	   Maximum range of the	motion search in pixels.

       me_method (me)
	   Set motion estimation method. Possible values in the	decreasing
	   order of speed:

	   dia (dia)
	   epzs	(dia)
	       Diamond search with radius 1 (fastest). epzs is an alias	for
	       dia.

	   hex (hex)
	       Hexagonal search	with radius 2.

	   umh (umh)
	       Uneven multi-hexagon search.

	   esa (esa)
	       Exhaustive search.

	   tesa	(tesa)
	       Hadamard	exhaustive search (slowest).

       forced-idr
	   Normally, when forcing a I-frame type, the encoder can select any
	   type	of I-frame. This option	forces it to choose an IDR-frame.

       subq (subme)
	   Sub-pixel motion estimation method.

       b_strategy (b-adapt)
	   Adaptive B-frame placement decision algorithm. Use only on first-
	   pass.

       keyint_min (min-keyint)
	   Minimum GOP size.

       coder
	   Set entropy encoder.	Possible values:

	   ac  Enable CABAC.

	   vlc Enable CAVLC and	disable	CABAC. It generates the	same effect as
	       x264's --no-cabac option.

       cmp Set full pixel motion estimation comparison algorithm. Possible
	   values:

	   chroma
	       Enable chroma in	motion estimation.

	   sad Ignore chroma in	motion estimation. It generates	the same
	       effect as x264's	--no-chroma-me option.

       threads (threads)
	   Number of encoding threads.

       thread_type
	   Set multithreading technique. Possible values:

	   slice
	       Slice-based multithreading. It generates	the same effect	as
	       x264's --sliced-threads option.

	   frame
	       Frame-based multithreading.

       flags
	   Set encoding	flags. It can be used to disable closed	GOP and	enable
	   open	GOP by setting it to "-cgop". The result is similar to the
	   behavior of x264's --open-gop option.

       rc_init_occupancy (vbv-init)
       preset (preset)
	   Set the encoding preset.

       tune (tune)
	   Set tuning of the encoding params.

       profile (profile)
	   Set profile restrictions.

       fastfirstpass
	   Enable fast settings	when encoding first pass, when set to 1. When
	   set to 0, it	has the	same effect of x264's --slow-firstpass option.

       crf (crf)
	   Set the quality for constant	quality	mode.

       crf_max (crf-max)
	   In CRF mode,	prevents VBV from lowering quality beyond this point.

       qp (qp)
	   Set constant	quantization rate control method parameter.

       aq-mode (aq-mode)
	   Set AQ method. Possible values:

	   none	(0)
	       Disabled.

	   variance (1)
	       Variance	AQ (complexity mask).

	   autovariance	(2)
	       Auto-variance AQ	(experimental).

       aq-strength (aq-strength)
	   Set AQ strength, reduce blocking and	blurring in flat and textured
	   areas.

       psy Use psychovisual optimizations when set to 1. When set to 0,	it has
	   the same effect as x264's --no-psy option.

       psy-rd  (psy-rd)
	   Set strength	of psychovisual	optimization, in psy-rd:psy-trellis
	   format.

       rc-lookahead (rc-lookahead)
	   Set number of frames	to look	ahead for frametype and	ratecontrol.

       weightb
	   Enable weighted prediction for B-frames when	set to 1. When set to
	   0, it has the same effect as	x264's --no-weightb option.

       weightp (weightp)
	   Set weighted	prediction method for P-frames.	Possible values:

	   none	(0)
	       Disabled

	   simple (1)
	       Enable only weighted refs

	   smart (2)
	       Enable both weighted refs and duplicates

       ssim (ssim)
	   Enable calculation and printing SSIM	stats after the	encoding.

       intra-refresh (intra-refresh)
	   Enable the use of Periodic Intra Refresh instead of IDR frames when
	   set to 1.

       avcintra-class (class)
	   Configure the encoder to generate AVC-Intra.	 Valid values are
	   50,100 and 200

       bluray-compat (bluray-compat)
	   Configure the encoder to be compatible with the bluray standard.
	   It is a shorthand for setting "bluray-compat=1 force-cfr=1".

       b-bias (b-bias)
	   Set the influence on	how often B-frames are used.

       b-pyramid (b-pyramid)
	   Set method for keeping of some B-frames as references. Possible
	   values:

	   none	(none)
	       Disabled.

	   strict (strict)
	       Strictly	hierarchical pyramid.

	   normal (normal)
	       Non-strict (not Blu-ray compatible).

       mixed-refs
	   Enable the use of one reference per partition, as opposed to	one
	   reference per macroblock when set to	1. When	set to 0, it has the
	   same	effect as x264's --no-mixed-refs option.

       8x8dct
	   Enable adaptive spatial transform (high profile 8x8 transform) when
	   set to 1. When set to 0, it has the same effect as x264's
	   --no-8x8dct option.

       fast-pskip
	   Enable early	SKIP detection on P-frames when	set to 1. When set to
	   0, it has the same effect as	x264's --no-fast-pskip option.

       aud (aud)
	   Enable use of access	unit delimiters	when set to 1.

       mbtree
	   Enable use macroblock tree ratecontrol when set to 1. When set to
	   0, it has the same effect as	x264's --no-mbtree option.

       deblock (deblock)
	   Set loop filter parameters, in alpha:beta form.

       cplxblur	(cplxblur)
	   Set fluctuations reduction in QP (before curve compression).

       partitions (partitions)
	   Set partitions to consider as a comma-separated list	of. Possible
	   values in the list:

	   p8x8
	       8x8 P-frame partition.

	   p4x4
	       4x4 P-frame partition.

	   b8x8
	       4x4 B-frame partition.

	   i8x8
	       8x8 I-frame partition.

	   i4x4
	       4x4 I-frame partition.  (Enabling p4x4 requires p8x8 to be
	       enabled.	Enabling i8x8 requires adaptive	spatial	transform
	       (8x8dct option) to be enabled.)

	   none	(none)
	       Do not consider any partitions.

	   all (all)
	       Consider	every partition.

       direct-pred (direct)
	   Set direct MV prediction mode. Possible values:

	   none	(none)
	       Disable MV prediction.

	   spatial (spatial)
	       Enable spatial predicting.

	   temporal (temporal)
	       Enable temporal predicting.

	   auto	(auto)
	       Automatically decided.

       slice-max-size (slice-max-size)
	   Set the limit of the	size of	each slice in bytes. If	not specified
	   but RTP payload size	(ps) is	specified, that	is used.

       stats (stats)
	   Set the file	name for multi-pass stats.

       nal-hrd (nal-hrd)
	   Set signal HRD information (requires	vbv-bufsize to be set).
	   Possible values:

	   none	(none)
	       Disable HRD information signaling.

	   vbr (vbr)
	       Variable	bit rate.

	   cbr (cbr)
	       Constant	bit rate (not allowed in MP4 container).

       x264opts	(N.A.)
	   Set any x264	option,	see x264 --fullhelp for	a list.

	   Argument is a list of key=value couples separated by	":". In	filter
	   and psy-rd options that use ":" as a	separator themselves, use ","
	   instead. They accept	it as well since long ago but this is kept
	   undocumented	for some reason.

	   For example to specify libx264 encoding options with	ffmpeg:

		   ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

       a53cc boolean
	   Import closed captions (which must be ATSC compatible format) into
	   output.  Only the mpeg2 and h264 decoders provide these. Default is
	   1 (on).

       x264-params (N.A.)
	   Override the	x264 configuration using a :-separated list of
	   key=value parameters.

	   This	option is functionally the same	as the x264opts, but is
	   duplicated for compatibility	with the Libav fork.

	   For example to specify libx264 encoding options with	ffmpeg:

		   ffmpeg -i INPUT -c:v	libx264	-x264-params level=30:bframes=0:weightp=0:\
		   cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
		   no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT

       Encoding	ffpresets for common usages are	provided so they can be	used
       with the	general	presets	system (e.g. passing the pre option).

   libx265
       x265 H.265/HEVC encoder wrapper.

       This encoder requires the presence of the libx265 headers and library
       during configuration. You need to explicitly configure the build	with
       --enable-libx265.

       Options

       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.

       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

   libxvid
       Xvid MPEG-4 Part	2 encoder wrapper.

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

       The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so	users
       can encode to this format without this library.

       Options

       The following options are supported by the libxvid wrapper. Some	of the
       following options are listed but	are not	documented, and	correspond to
       shared codec options. See the Codec Options chapter for their
       documentation. The other	shared options which are not listed have no
       effect for the libxvid encoder.

       b
       g
       qmin
       qmax
       mpeg_quant
       threads
       bf
       b_qfactor
       b_qoffset
       flags
	   Set specific	encoding flags.	Possible values:

	   mv4 Use four	motion vector by macroblock.

	   aic Enable high quality AC prediction.

	   gray
	       Only encode grayscale.

	   gmc Enable the use of global	motion compensation (GMC).

	   qpel
	       Enable quarter-pixel motion compensation.

	   cgop
	       Enable closed GOP.

	   global_header
	       Place global headers in extradata instead of every keyframe.

       trellis
       me_method
	   Set motion estimation method. Possible values in decreasing order
	   of speed and	increasing order of quality:

	   zero
	       Use no motion estimation	(default).

	   phods
	   x1
	   log Enable advanced diamond zonal search for	16x16 blocks and half-
	       pixel refinement	for 16x16 blocks. x1 and log are aliases for
	       phods.

	   epzs
	       Enable all of the things	described above, plus advanced diamond
	       zonal search for	8x8 blocks, half-pixel refinement for 8x8
	       blocks, and motion estimation on	chroma planes.

	   full
	       Enable all of the things	described above, plus extended 16x16
	       and 8x8 blocks search.

       mbd Set macroblock decision algorithm. Possible values in the
	   increasing order of quality:

	   simple
	       Use macroblock comparing	function algorithm (default).

	   bits
	       Enable rate distortion-based half pixel and quarter pixel
	       refinement for 16x16 blocks.

	   rd  Enable all of the things	described above, plus rate distortion-
	       based half pixel	and quarter pixel refinement for 8x8 blocks,
	       and rate	distortion-based search	using square pattern.

       lumi_aq
	   Enable lumi masking adaptive	quantization when set to 1. Default is
	   0 (disabled).

       variance_aq
	   Enable variance adaptive quantization when set to 1.	Default	is 0
	   (disabled).

	   When	combined with lumi_aq, the resulting quality will not be
	   better than any of the two specified	individually. In other words,
	   the resulting quality will be the worse one of the two effects.

       ssim
	   Set structural similarity (SSIM) displaying method. Possible
	   values:

	   off Disable displaying of SSIM information.

	   avg Output average SSIM at the end of encoding to stdout. The
	       format of showing the average SSIM is:

		       Average SSIM: %f

	       For users who are not familiar with C, %f means a float number,
	       or a decimal (e.g. 0.939232).

	   frame
	       Output both per-frame SSIM data during encoding and average
	       SSIM at the end of encoding to stdout. The format of per-frame
	       information is:

			      SSIM: avg: %1.3f min: %1.3f max: %1.3f

	       For users who are not familiar with C, %1.3f means a float
	       number rounded to 3 digits after	the dot	(e.g. 0.932).

       ssim_acc
	   Set SSIM accuracy. Valid options are	integers within	the range of
	   0-4,	while 0	gives the most accurate	result and 4 computes the
	   fastest.

   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).

   mpeg2
       MPEG-2 video encoder.

       Options

       profile integer
	   Select the mpeg2 profile to encode:

	   422
	   main
	   ss  Spatially Scalable

	   snr SNR Scalable

	   high
	   simple
       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 and VP9)

       The ratecontrol method is selected as follows:

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

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

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

	   -   ICQ -- intelligent constant quality otherwise.

       o   Otherwise, a	bitrate-based mode is used. For	all of those, you
	   should specify at least the desired average bitrate with the	b
	   option.

	   -   LA - VBR	with lookahead,	when the look_ahead option is
	       specified.

	   -   VCM - video conferencing	mode, when the vcm option is set.

	   -   CBR - constant bitrate, when maxrate is specified and equal to
	       the average bitrate.

	   -   VBR - variable bitrate, when maxrate is specified, but is
	       higher than the average bitrate.

	   -   AVBR - average VBR mode,	when maxrate is	not specified. This
	       mode is further configured by the avbr_accuracy and
	       avbr_convergence	options.

       Note that depending on your system, a different mode than the one you
       specified may be	selected by the	encoder. Set the verbosity level to
       verbose or higher to see	the actual settings used by the	QSV runtime.

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

       o   g/gop_size -> GopPicSize

       o   bf/max_b_frames+1 ->	GopRefDist

       o   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       o   slices -> NumSlice

       o   refs	-> NumRefFrame

       o   b_strategy/b_frame_strategy -> BRefType

       o   cgop/CLOSED_GOP codec flag -> GopOptFlag

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

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

   snow
       Options

       iterative_dia_size
	   dia size for	the iterative motion estimation

   VAAPI encoders
       Wrappers	for hardware encoders accessible via VAAPI.

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

       The following standard libavcodec options are used:

       o   g / gop_size

       o   bf /	max_b_frames

       o   profile

	   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.

       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.

       Each encoder also has its own specific options:

       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).

       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.

   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 (num_units_in_display_tick	/ time_scale) 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

   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.

       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

   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.

	   insert
	   remove
       sample_aspect_ratio
	   Set the sample aspect ratio of the stream in	the VUI	parameters.

       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 (num_units_in_tick	/ time_scale) 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).

       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.

       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.

       A new single global PPS is created, and all of the redundant PPSs
       within the stream are removed.

   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 (num_units_in_tick
	   / time_scale).  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).

       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.

   mp3decomp
       Decompress non-standard compressed MP3 audio headers.

   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
	   A numeral string, whose value is related to how often output	bytes
	   will	be modified. Therefore,	values below or	equal to 0 are
	   forbidden, and the lower the	more frequent bytes will be modified,
	   with	1 meaning every	byte is	modified.

       dropamount
	   A numeral string, whose value is related to how often packets will
	   be dropped.	Therefore, values below	or equal to 0 are forbidden,
	   and the lower the more frequent packets will	be dropped, with 1
	   meaning every packet	is dropped.

       The following example applies the modification to every byte but	does
       not drop	any packets.

	       ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

   null
       This bitstream filter passes the	packets	through	unchanged.

   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 -

   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.

   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 set. Inert when nofillin is	set.

	   ignidx
	       Ignore index.

	   keepside (deprecated,inert)
	   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.

	   latm	(deprecated,inert)
	   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 when calculated using PTS.	 At
	   present, applicable for MPEG-PS and MPEG-TS.

       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.

   apng
       Animated	Portable Network Graphics demuxer.

       This demuxer is used to demux APNG files.  All headers, but the PNG
       signature, up to	(but not including) the	first fcTL chunk are
       transmitted as extradata.  Frames are then split	as being all the
       chunks between two fcTL ones, or	between	the last fcTL and IEND chunks.

       -ignore_loop bool
	   Ignore the loop variable in the file	if set.

       -max_fps	int
	   Maximum framerate in	frames per second (0 for no limit).

       -default_fps int
	   Default framerate in	frames per second when none is specified in
	   the file (0 meaning as fast as possible).

   asf
       Advanced	Systems	Format demuxer.

       This demuxer is used to demux ASF files and MMS network streams.

       -no_resync_search bool
	   Do not try to resynchronize by looking for a	certain	optional start
	   code.

   concat
       Virtual concatenation script demuxer.

       This demuxer reads a list of files and other directives from a text
       file and	demuxes	them one after the other, as if	all their packets had
       been muxed together.

       The timestamps in the files are adjusted	so that	the first file starts
       at 0 and	each next file starts where the	previous one finishes. Note
       that it is done globally	and may	cause gaps if all streams do not have
       exactly the same	length.

       All files must have the same streams (same codecs, same time base,
       etc.).

       The duration of each file is used to adjust the timestamps of the next
       file: if	the duration is	incorrect (because it was computed using the
       bit-rate	or because the file is truncated, for example),	it can cause
       artifacts. The "duration" directive can be used to override the
       duration	stored in each file.

       Syntax

       The script is a text file in extended-ASCII, with one directive per
       line.  Empty lines, leading spaces and lines starting with '#' are
       ignored.	The following directive	is recognized:

       "file path"
	   Path	to a file to read; special characters and spaces must be
	   escaped with	backslash or single quotes.

	   All subsequent file-related directives apply	to that	file.

       "ffconcat version 1.0"
	   Identify the	script type and	version. It also sets the safe option
	   to 1	if it was -1.

	   To make FFmpeg recognize the	format automatically, this directive
	   must	appear exactly as is (no extra space or	byte-order-mark) on
	   the very first line of the script.

       "duration dur"
	   Duration of the file. This information can be specified from	the
	   file; specifying it here may	be more	efficient or help if the
	   information from the	file is	not available or accurate.

	   If the duration is set for all files, then it is possible to	seek
	   in the whole	concatenated video.

       "inpoint	timestamp"
	   In point of the file. When the demuxer opens	the file it instantly
	   seeks to the	specified timestamp. Seeking is	done so	that all
	   streams can be presented successfully at In point.

	   This	directive works	best with intra	frame codecs, because for non-
	   intra frame ones you	will usually get extra packets before the
	   actual In point and the decoded content will	most likely contain
	   frames before In point too.

	   For each file, packets before the file In point will	have
	   timestamps less than	the calculated start timestamp of the file
	   (negative in	case of	the first file), and the duration of the files
	   (if not specified by	the "duration" directive) will be reduced
	   based on their specified In point.

	   Because of potential	packets	before the specified In	point, packet
	   timestamps may overlap between two concatenated files.

       "outpoint timestamp"
	   Out point of	the file. When the demuxer reaches the specified
	   decoding timestamp in any of	the streams, it	handles	it as an end
	   of file condition and skips the current and all the remaining
	   packets from	all streams.

	   Out point is	exclusive, which means that the	demuxer	will not
	   output packets with a decoding timestamp greater or equal to	Out
	   point.

	   This	directive works	best with intra	frame codecs and formats where
	   all streams are tightly interleaved.	For non-intra frame codecs you
	   will	usually	get additional packets with presentation timestamp
	   after Out point therefore the decoded content will most likely
	   contain frames after	Out point too. If your streams are not tightly
	   interleaved you may not get all the packets from all	streams	before
	   Out point and you may only will be able to decode the earliest
	   stream until	Out point.

	   The duration	of the files (if not specified by the "duration"
	   directive) will be reduced based on their specified Out point.

       "file_packet_metadata key=value"
	   Metadata of the packets of the file.	The specified metadata will be
	   set for each	file packet. You can specify this directive multiple
	   times to add	multiple metadata entries.

       "stream"
	   Introduce a stream in the virtual file.  All	subsequent stream-
	   related directives apply to the last	introduced stream.  Some
	   streams properties must be set in order to allow identifying	the
	   matching streams in the subfiles.  If no streams are	defined	in the
	   script, the streams from the	first file are copied.

       "exact_stream_id	id"
	   Set the id of the stream.  If this directive	is given, the string
	   with	the corresponding id in	the subfiles will be used.  This is
	   especially useful for MPEG-PS (VOB) files, where the	order of the
	   streams is not reliable.

       Options

       This demuxer accepts the	following option:

       safe
	   If set to 1,	reject unsafe file paths. A file path is considered
	   safe	if it does not contain a protocol specification	and is
	   relative and	all components only contain characters from the
	   portable character set (letters, digits, period, underscore and
	   hyphen) and have no period at the beginning of a component.

	   If set to 0,	any file name is accepted.

	   The default is 1.

	   -1 is equivalent to 1 if the	format was automatically probed	and 0
	   otherwise.

       auto_convert
	   If set to 1,	try to perform automatic conversions on	packet data to
	   make	the streams concatenable.  The default is 1.

	   Currently, the only conversion is adding the	h264_mp4toannexb
	   bitstream filter to H.264 streams in	MP4 format. This is necessary
	   in particular if there are resolution changes.

       segment_time_metadata
	   If set to 1,	every packet will contain the lavf.concat.start_time
	   and the lavf.concat.duration	packet metadata	values which are the
	   start_time and the duration of the respective file segments in the
	   concatenated	output expressed in microseconds. The duration
	   metadata is only set	if it is known based on	the concat file.  The
	   default is 0.

       Examples

       o   Use absolute	filenames and include some comments:

		   # my	first filename
		   file	/mnt/share/file-1.wav
		   # my	second filename	including whitespace
		   file	'/mnt/share/file 2.wav'
		   # my	third filename including whitespace plus single	quote
		   file	'/mnt/share/file 3'\''.wav'

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

		   ffconcat version 1.0

		   file	file-1.wav
		   duration 20.0

		   file	subdir/file-2.wav

   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.

   flv,	live_flv
       Adobe Flash Video Format	demuxer.

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

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

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

       -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).

       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.

   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

       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).

       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.

   mpjpeg
       MJPEG encapsulated in multi-part	MIME demuxer.

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

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

   rawvideo
       Raw video demuxer.

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

       This demuxer accepts the	following options:

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

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

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

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

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

   sbg
       SBaGen script demuxer.

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

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

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

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

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

       This demuxer accepts the	following option:

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

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

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

   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.

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

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

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

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

       A description of	some of	the currently available	muxers follows.

   aiff
       Audio Interchange File Format muxer.

       Options

       It accepts the following	options:

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

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

   asf
       Advanced	Systems	Format muxer.

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

       Options

       It accepts the following	options:

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

   avi
       Audio Video Interleaved muxer.

       Options

       It accepts the following	options:

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

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

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

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

   chromaprint
       Chromaprint fingerprinter.

       This muxer feeds	audio data to the Chromaprint library, which generates
       a fingerprint for the provided audio data. See
       <https://acoustid.org/chromaprint>

       It takes	a single signed	native-endian 16-bit raw audio stream of at
       most 2 channels.

       Options

       silence_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.

       algorithm
	   Version of algorithm	to fingerprint with. Range is 0	to 4.  Version
	   3 enables silence detection.	Default	is 1.

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

	   raw Binary raw fingerprint

	   compressed
	       Binary compressed fingerprint

	   base64
	       Base64 compressed fingerprint (default)

   crc
       CRC (Cyclic Redundancy Check) testing format.

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

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

       See also	the framecrc muxer.

       Examples

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

	       ffmpeg -i INPUT -f crc out.crc

       You can print the CRC to	stdout with the	command:

	       ffmpeg -i INPUT -f crc -

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

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

   flv
       Adobe Flash Video Format	muxer.

       This muxer accepts the following	options:

       flvflags	flags
	   Possible values:

	   aac_seq_header_detect
	       Place AAC sequence header based on audio	stream data.

	   no_sequence_end
	       Disable sequence	end tag.

	   no_metadata
	       Disable metadata	tag.

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

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

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

       For more	information see:

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

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

       It creates a MPD	manifest file and segment files	for each stream.

       The segment filename might contain pre-defined identifiers used with
       SegmentTemplate as defined in section 5.3.9.4.4 of the standard.
       Available identifiers are "$RepresentationID$", "$Number$",
       "$Bandwidth$" and "$Time$".  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.,

	       ffmpeg -re -i <input> -map 0 -map 0 -c:a	libfdk_aac -c:v	libx264	\
	       -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline \
	       -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 \
	       -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 \
	       -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" \
	       -f dash /path/to/out.mpd

       min_seg_duration	microseconds
	   This	is a deprecated	option to set the segment length in
	   microseconds, use seg_duration instead.

       seg_duration duration
	   Set the segment length in seconds (fractional value can be set).
	   The value is	treated	as average segment duration when use_template
	   is enabled and

       frag_duration duration
	   Set the length in seconds of	fragments within segments (fractional
	   value can be	set).

       frag_type type
	   Set the type	of interval for	fragmentation.

       window_size size
	   Set the maximum number of segments kept in the manifest.

       extra_window_size size
	   Set the maximum number of segments kept outside of the manifest
	   before removing from	disk.

       remove_at_exit remove
	   Enable (1) or disable (0) removal of	all segments when finished.

       use_template template
	   Enable (1) or disable (0) use of SegmentTemplate instead of
	   SegmentList.

       use_timeline timeline
	   Enable (1) or disable (0) use of SegmentTimeline in
	   SegmentTemplate.

       single_file single_file
	   Enable (1) or disable (0) storing all segments in one file,
	   accessed using byte ranges.

       single_file_name	file_name
	   DASH-templated name to be used for baseURL. Implies single_file set
	   to "1". In the template, "$ext$" is replaced	with the file name
	   extension specific for the segment format.

       init_seg_name init_name
	   DASH-templated name to used for the initialization segment. Default
	   is "init-stream$RepresentationID$.$ext$". "$ext$" is	replaced with
	   the file name extension specific for	the segment format.

       media_seg_name segment_name
	   DASH-templated name to used 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.

       utc_timing_url utc_url
	   URL of the page that	will return the	UTC timestamp in ISO format.
	   Example: "https://time.akamai.com/?iso"

       method method
	   Use the given HTTP method to	create output files. Generally set to
	   PUT or POST.

       http_user_agent user_agent
	   Override User-Agent field in	HTTP header. Applicable	only for HTTP
	   output.

       http_persistent http_persistent
	   Use persistent HTTP connections. Applicable only for	HTTP output.

       hls_playlist hls_playlist
	   Generate HLS	playlist files as well.	The master playlist is
	   generated with the filename master.m3u8.  One media playlist	file
	   is generated	for each stream	with filenames media_0.m3u8,
	   media_1.m3u8, etc.

       streaming streaming
	   Enable (1) or disable (0) chunk streaming mode of output. In	chunk
	   streaming mode, each	frame will be a	moof fragment which forms a
	   chunk.

       adaptation_sets adaptation_sets
	   Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c
	   id=y,streams=d,e" with x and	y being	the IDs	of the adaptation sets
	   and a,b,c,d and e are the indices of	the mapped streams.

	   To map all video (or	audio) streams to an AdaptationSet, "v"	(or
	   "a")	can be used as stream identifier instead of IDs.

	   When	no assignment is defined, this defaults	to an AdaptationSet
	   for each stream.

	   Optional syntax is
	   "id=x,seg_duration=x,frag_duration=x,frag_type=type,descriptor=descriptor_string,streams=a,b,c
	   id=y,seg_duration=y,frag_type=type,streams=d,e" and so on,
	   descriptor is useful	to the scheme defined by ISO/IEC
	   23009-1:2014/Amd.2:2015.  For example, -adaptation_sets
	   "id=0,descriptor=<SupplementalProperty
	   schemeIdUri=\"urn:mpeg:dash:srd:2014\"
	   value=\"0,0,0,1,1,2,2\"/>,streams=v".  Please note that descriptor
	   string should be a self-closing xml tag.  seg_duration,
	   frag_duration and frag_type override	the global option values for
	   each	adaptation set.	 For example, -adaptation_sets
	   "id=0,seg_duration=2,frag_duration=1,frag_type=duration,streams=v
	   id=1,seg_duration=2,frag_type=none,streams=a" type_id marks an
	   adaptation set as containing	streams	meant to be used for Trick
	   Mode	for the	referenced adaptation set.  For	example,
	   -adaptation_sets "id=0,seg_duration=2,frag_type=none,streams=0
	   id=1,seg_duration=10,frag_type=none,trick_id=0,streams=1"

       timeout timeout
	   Set timeout for socket I/O operations. Applicable only for HTTP
	   output.

       index_correction	index_correction
	   Enable (1) or Disable (0) segment index correction logic.
	   Applicable only when	use_template is	enabled	and use_timeline is
	   disabled.

	   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.

       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.

       global_sidx global_sidx
	   Write global	SIDX atom. Applicable only for single file, mp4
	   output, non-streaming mode.

       dash_segment_type dash_segment_type
	   Possible values:

	   auto
	       If this flag is set, the	dash segment files format will be
	       selected	based on the stream codec. This	is the default mode.

	   mp4 If this flag is set, the	dash segment files will	be in in
	       ISOBMFF format.

	   webm
	       If this flag is set, the	dash segment files will	be in in WebM
	       format.

       ignore_io_errors	ignore_io_errors
	   Ignore IO errors during open	and write. Useful for long-duration
	   runs	with network output.

       lhls lhls
	   Enable Low-latency HLS(LHLS). Adds #EXT-X-PREFETCH tag with current
	   segment's URI.  Apple doesn't have an official spec for LHLS.
	   Meanwhile hls.js player folks are trying to standardize a open LHLS
	   spec. The draft spec	is available in
	   https://github.com/video-dev/hlsjs-rfcs/blob/lhls-spec/proposals/0001-lhls.md
	   This	option will also try to	comply with the	above open spec, till
	   Apple's spec	officially supports it.	 Applicable only when
	   streaming and hls_playlist options are enabled.  This is an
	   experimental	feature.

       ldash ldash
	   Enable Low-latency Dash by constraining the presence	and values of
	   some	elements.

       master_m3u8_publish_rate	master_m3u8_publish_rate
	   Publish master playlist repeatedly every after specified number of
	   segment intervals.

       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's set to auto by
	   default, in which case the muxer will attempt to enable it only in
	   modes that require it.

       mpd_profile mpd_profile
	   Set one or more manifest profiles.

       http_opts http_opts
	   A :-separated list of key=value options to pass to the underlying
	   HTTP	protocol. Applicable only for HTTP output.

       target_latency target_latency
	   Set an intended target latency in seconds (fractional value can be
	   set)	for serving. Applicable	only when streaming and	write_prft
	   options are enabled.	 This is an informative	fields clients can use
	   to measure the latency of the service.

       min_playback_rate min_playback_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.

       max_playback_rate max_playback_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.

   framecrc
       Per-packet CRC (Cyclic Redundancy Check)	testing	format.

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

       The output of the muxer consists	of a line for each audio and video
       packet of the form:

	       <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>

       CRC is a	hexadecimal number 0-padded to 8 digits	containing the CRC of
       the packet.

       Examples

       For example to compute the CRC of the audio and video frames in INPUT,
       converted to raw	audio and video	packets, and store it in the file
       out.crc:

	       ffmpeg -i INPUT -f framecrc out.crc

       To print	the information	to stdout, use the command:

	       ffmpeg -i INPUT -f framecrc -

       With ffmpeg, you	can select the output format to	which the audio	and
       video frames are	encoded	before computing the CRC for each packet by
       specifying the audio and	video codec. For example, to compute the CRC
       of each decoded input audio frame converted to PCM unsigned 8-bit and
       of each decoded input video frame converted to MPEG-2 video, use	the
       command:

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

       See also	the crc	muxer.

   framehash
       Per-packet hash testing format.

       This muxer computes and prints a	cryptographic hash for each audio and
       video packet. This can be used for packet-by-packet equality checks
       without having to individually do a binary comparison on	each.

       By default audio	frames are converted to	signed 16-bit raw audio	and
       video frames to raw video before	computing the hash, but	the output of
       explicit	conversions to other codecs can	also be	used. It uses the
       SHA-256 cryptographic hash function by default, but supports several
       other algorithms.

       The output of the muxer consists	of a line for each audio and video
       packet of the form:

	       <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>

       hash is a hexadecimal number representing the computed hash for the
       packet.

       hash algorithm
	   Use the cryptographic hash function specified by the	string
	   algorithm.  Supported values	include	"MD5", "murmur3", "RIPEMD128",
	   "RIPEMD160",	"RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
	   (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
	   and "adler32".

       Examples

       To compute the SHA-256 hash of the audio	and video frames in INPUT,
       converted to raw	audio and video	packets, and store it in the file
       out.sha256:

	       ffmpeg -i INPUT -f framehash out.sha256

       To print	the information	to stdout, using the MD5 hash function,	use
       the command:

	       ffmpeg -i INPUT -f framehash -hash md5 -

       See also	the hash muxer.

   framemd5
       Per-packet MD5 testing format.

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

       Examples

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

	       ffmpeg -i INPUT -f framemd5 out.md5

       To print	the information	to stdout, use the command:

	       ffmpeg -i INPUT -f framemd5 -

       See also	the framehash and md5 muxers.

   gif
       Animated	GIF muxer.

       It accepts the following	options:

       loop
	   Set the number of times to loop the output. Use "-1"	for no loop, 0
	   for looping indefinitely (default).

       final_delay
	   Force the delay (expressed in centiseconds) after the last frame.
	   Each	frame ends with	a delay	until the next frame. The default is
	   "-1", which is a special value to tell the muxer to re-use the
	   previous delay. In case of a	loop, you might	want to	customize this
	   value to mark a pause for instance.

       For example, to encode a	gif looping 10 times, with a 5 seconds delay
       between the loops:

	       ffmpeg -i INPUT -loop 10	-final_delay 500 out.gif

       Note 1: if you wish to extract the frames into separate GIF files, you
       need to force the image2	muxer:

	       ffmpeg -i INPUT -c:v gif	-f image2 "out%d.gif"

       Note 2: the GIF format has a very large time base: the delay between
       two frames can therefore	not be smaller than one	centi second.

   hash
       Hash testing format.

       This muxer computes and prints a	cryptographic hash of all the input
       audio and video frames. This can	be used	for equality checks without
       having to do a complete binary comparison.

       By default audio	frames are converted to	signed 16-bit raw audio	and
       video frames to raw video before	computing the hash, but	the output of
       explicit	conversions to other codecs can	also be	used. Timestamps are
       ignored.	It uses	the SHA-256 cryptographic hash function	by default,
       but supports several other algorithms.

       The output of the muxer consists	of a single line of the	form:
       algo=hash, where	algo is	a short	string representing the	hash function
       used, and hash is a hexadecimal number representing the computed	hash.

       hash algorithm
	   Use the cryptographic hash function specified by the	string
	   algorithm.  Supported values	include	"MD5", "murmur3", "RIPEMD128",
	   "RIPEMD160",	"RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
	   (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512", "CRC32"
	   and "adler32".

       Examples

       To compute the SHA-256 hash of the input	converted to raw audio and
       video, and store	it in the file out.sha256:

	       ffmpeg -i INPUT -f hash out.sha256

       To print	an MD5 hash to stdout use the command:

	       ffmpeg -i INPUT -f hash -hash md5 -

       See also	the framehash muxer.

   hls
       Apple HTTP Live Streaming muxer that segments MPEG-TS according to the
       HTTP Live Streaming (HLS) specification.

       It creates a playlist file, and one or more segment files. The output
       filename	specifies the playlist filename.

       By default, the muxer creates a file for	each segment produced. These
       files have the same name	as the playlist, followed by a sequential
       number and a .ts	extension.

       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

       This muxer supports the following options:

       hls_init_time seconds
	   Set the initial target segment length in seconds. Default value is
	   0.  Segment will be cut on the next key frame after this time has
	   passed on the first m3u8 list.  After the initial playlist is
	   filled ffmpeg will cut segments at duration equal to	"hls_time"

       hls_time	seconds
	   Set the target segment length in seconds. Default value is 2.
	   Segment will	be cut on the next key frame after this	time has
	   passed.

       hls_list_size size
	   Set the maximum number of playlist entries. If set to 0 the list
	   file	will contain all the segments. Default value is	5.

       hls_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_ts_options options_list
	   Set output format options using a :-separated list of key=value
	   parameters. Values containing ":" special characters	must be
	   escaped.

       hls_wrap	wrap
	   This	is a deprecated	option,	you can	use "hls_list_size" and
	   "hls_flags delete_segments" instead it

	   This	option is useful to avoid to fill the disk with	many segment
	   files, and limits the maximum number	of segment files written to
	   disk	to wrap.

       hls_start_number_source
	   Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE")
	   according to	the specified source.  Unless "hls_flags single_file"
	   is set, it also specifies source of starting	sequence numbers of
	   segment and subtitle	filenames. In any case,	if "hls_flags
	   append_list"	is set and read	playlist sequence number is greater
	   than	the specified start sequence number, then that value will be
	   used	as start value.

	   It accepts the following values:

	   generic (default)
	       Set the starting	sequence numbers according to start_number
	       option value.

	   epoch
	       The start number	will be	the seconds since epoch	(1970-01-01
	       00:00:00)

	   epoch_us
	       The start number	will be	the microseconds since epoch
	       (1970-01-01 00:00:00)

	   datetime
	       The start number	will be	based on the current date/time as
	       YYYYmmddHHMMSS. e.g. 20161231235759.

       start_number number
	   Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE")	from
	   the specified number	when hls_start_number_source value is generic.
	   (This is the	default	case.)	Unless "hls_flags single_file" is set,
	   it also specifies starting sequence numbers of segment and subtitle
	   filenames.  Default value is	0.

       hls_allow_cache allowcache
	   Explicitly set whether the client MAY (1) or	MUST NOT (0) cache
	   media segments.

       hls_base_url baseurl
	   Append baseurl to every entry in the	playlist.  Useful to generate
	   playlists with absolute paths.

	   Note	that the playlist sequence number must be unique for each
	   segment and it is not to be confused	with the segment filename
	   sequence number which can be	cyclic,	for example if the wrap	option
	   is specified.

       hls_segment_filename filename
	   Set the segment filename. Unless "hls_flags single_file" is set,
	   filename is used as a string	format with the	segment	number:

		   ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts'	out.m3u8

	   This	example	will produce the playlist, out.m3u8, and segment
	   files: file000.ts, file001.ts, file002.ts, etc.

	   filename may	contain	full path or relative path specification, but
	   only	the file name part without any path info will be contained in
	   the m3u8 segment list.  Should a relative path be specified,	the
	   path	of the created segment files will be relative to the current
	   working directory.  When 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", this string
	   specifies the position of variant stream index in the generated
	   segment file	names.

		   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

	   This	example	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.

		   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

	   This	example	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.

       use_localtime
	   Same	as strftime option, will be deprecated.

       strftime
	   Use strftime() on filename to expand	the segment filename with
	   localtime.  The segment number is also available in this mode, but
	   to use it, you need to specify second_level_segment_index hls_flag
	   and %%d will	be the specifier.

		   ffmpeg -i in.nut -strftime 1	-hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

	   This	example	will produce the playlist, out.m3u8, and segment
	   files: file-20160215-1455569023.ts, file-20160215-1455569024.ts,
	   etc.	 Note: On some systems/environments, the %s specifier is not
	   available. See
	     "strftime()" documentation.

		   ffmpeg -i in.nut -strftime 1	-hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

	   This	example	will produce the playlist, out.m3u8, and segment
	   files: file-20160215-0001.ts, file-20160215-0002.ts,	etc.

       use_localtime_mkdir
	   Same	as strftime_mkdir option, will be deprecated .

       strftime_mkdir
	   Used	together with -strftime_mkdir, it will create all
	   subdirectories which	is expanded in filename.

		   ffmpeg -i in.nut -strftime 1	-strftime_mkdir	1 -hls_segment_filename	'%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

	   This	example	will create a directory	201560215 (if it does not
	   exist), and then produce the	playlist, out.m3u8, and	segment	files:
	   20160215/file-20160215-1455569023.ts,
	   20160215/file-20160215-1455569024.ts, etc.

		   ffmpeg -i in.nut -strftime 1	-strftime_mkdir	1 -hls_segment_filename	'%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

	   This	example	will create a directory	hierarchy 2016/02/15 (if any
	   of them do not exist), and then produce the playlist, out.m3u8, and
	   segment files: 2016/02/15/file-20160215-1455569023.ts,
	   2016/02/15/file-20160215-1455569024.ts, etc.

       hls_key_info_file key_info_file
	   Use the information in key_info_file	for segment encryption.	The
	   first line of key_info_file specifies the key URI written to	the
	   playlist. The key URL is used to access the encryption key during
	   playback. The second	line specifies the path	to the key file	used
	   to obtain the key during the	encryption process. The	key file is
	   read	as a single packed array of 16 octets in binary	format.	The
	   optional third line specifies the initialization vector (IV)	as a
	   hexadecimal string to be used instead of the	segment	sequence
	   number (default) for	encryption. Changes to key_info_file will
	   result in segment encryption	with the new key/IV and	an entry in
	   the playlist	for the	new key	URI/IV if "hls_flags periodic_rekey"
	   is enabled.

	   Key info file format:

		   <key	URI>
		   <key	file path>
		   <IV>	(optional)

	   Example key URIs:

		   http://server/file.key
		   /path/to/file.key
		   file.key

	   Example key file paths:

		   file.key
		   /path/to/file.key

	   Example IV:

		   0123456789ABCDEF0123456789ABCDEF

	   Key info file example:

		   http://server/file.key
		   /path/to/file.key
		   0123456789ABCDEF0123456789ABCDEF

	   Example shell script:

		   #!/bin/sh
		   BASE_URL=${1:-'.'}
		   openssl rand	16 > file.key
		   echo	$BASE_URL/file.key > file.keyinfo
		   echo	file.key >> file.keyinfo
		   echo	$(openssl rand -hex 16)	>> file.keyinfo
		   ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
		     -hls_key_info_file	file.keyinfo out.m3u8

       -hls_enc	enc
	   Enable (1) or disable (0) the AES128	encryption.  When enabled
	   every segment generated is encrypted	and the	encryption key is
	   saved as playlist name.key.

       -hls_enc_key key
	   Hex-coded 16byte key	to encrypt the segments, by default it is
	   randomly generated.

       -hls_enc_key_url	keyurl
	   If set, keyurl is prepended instead of baseurl to the key filename
	   in the playlist.

       -hls_enc_iv iv
	   Hex-coded 16byte initialization vector for every segment instead of
	   the autogenerated ones.

       hls_segment_type	flags
	   Possible values:

	   mpegts
	       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	to the fragment	files header file, default filename is
	   init.mp4.

       hls_fmp4_init_resend
	   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.

	   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" 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" 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 keyframes. This
	       improves	behavior on some players when the time between
	       keyframes is inconsistent, but may make things worse on others,
	       and can cause some oddities during seeking. This	flag should be
	       used with the "hls_time"	option.

	   program_date_time
	       Generate	"EXT-X-PROGRAM-DATE-TIME" tags.

	   second_level_segment_index
	       Makes it	possible to use	segment	indexes	as %%d in
	       hls_segment_filename expression besides date/time values	when
	       strftime	is on.	To get fixed width numbers with	trailing
	       zeroes, %%0xd format is available where x is the	required
	       width.

	   second_level_segment_size
	       Makes it	possible to use	segment	sizes (counted in bytes) as
	       %%s in hls_segment_filename expression besides date/time	values
	       when 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
	       Makes it	possible to use	segment	duration (calculated  in
	       microseconds) as	%%t in hls_segment_filename 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.

		       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

	       This will produce segments like this:
	       segment_20170102194334_0003_00122200_0000003000000.ts,
	       segment_20170102194334_0004_00120072_0000003000000.ts etc.

	   temp_file
	       Write segment data to filename.tmp and rename to	filename only
	       once the	segment	is complete. A webserver serving up segments
	       can be configured to reject requests to *.tmp to	prevent	access
	       to in-progress segments before they have	been added to the m3u8
	       playlist. This flag also	affects	how m3u8 playlist files	are
	       created.	 If this flag is set, all playlist files will written
	       into temporary file and renamed after they are complete,
	       similarly as segments are handled.  But playlists with "file"
	       protocol	and with type ("hls_playlist_type") other than "vod"
	       are always written into temporary file regardless of this flag.
	       Master playlist files ("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 event
	   Emit	"#EXT-X-PLAYLIST-TYPE:EVENT" in	the m3u8 header. Forces
	   hls_list_size to 0; the playlist can	only be	appended to.

       hls_playlist_type vod
	   Emit	"#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces
	   hls_list_size to 0; the playlist must not change.

       method
	   Use the given HTTP method to	create the hls files.

		   ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

	   This	example	will upload all	the mpegts segment files to the	HTTP
	   server using	the HTTP PUT method, and update	the m3u8 files every
	   "refresh" times using the same method.  Note	that the HTTP server
	   must	support	the given method for uploading files.

       http_user_agent
	   Override User-Agent field in	HTTP header. Applicable	only for HTTP
	   output.

       var_stream_map
	   Map string which specifies 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.

		   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

	   This	example	creates	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. If you want	something meaningful text instead of indexes
	   in result names, you	may specify names for each or some of the
	   variants as in the following	example.

		   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

	   This	example	creates	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 \
		     -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

	   This	example	creates	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 -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

	   This	example	creates	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: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

	   This	example	creates	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, #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 \
		     -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

	   This	example	creates	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, #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,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

	   This	example	creates	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, #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 -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

	   This	example	adds "#EXT-X-MEDIA" tag	with "TYPE=SUBTITLES" in the
	   master playlist with	webvtt subtitle	group name 'subtitle'. Please
	   make	sure the input file has	one text subtitle stream at least.

       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. If
	   "var_stream_map" is not set,	then the first available ccgroup in
	   "cc_stream_map" is mapped to	the output variant stream. The
	   examples for	these two use cases are	given below.

		   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	example	adds "#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 adds	"CLOSED-CAPTIONS" attribute
	   with	group name 'cc'	for the	output variant stream.

		   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

	   This	example	adds two "#EXT-X-MEDIA"	tags with
	   "TYPE=CLOSED-CAPTIONS" in the master	playlist for the INSTREAM-IDs
	   'CC1' and 'CC2'. Also, it adds "CLOSED-CAPTIONS" attribute with
	   group name 'cc' for the two output variant streams.

       master_pl_name
	   Create HLS master playlist with the given name.

		   ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8

	   This	example	creates	HLS master playlist with name master.m3u8 and
	   it is published at http://example.com/live/

       master_pl_publish_rate
	   Publish master play list repeatedly every after specified number of
	   segment intervals.

		   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

	   This	example	creates	HLS master playlist with name master.m3u8 and
	   keep	publishing it repeatedly every after 30	segments i.e. every
	   after 60s.

       http_persistent
	   Use persistent HTTP connections. Applicable only for	HTTP output.

       timeout
	   Set timeout for socket I/O operations. Applicable only for HTTP
	   output.

       -ignore_io_errors
	   Ignore IO errors during open, write and delete. Useful for long-
	   duration runs with network output.

       headers
	   Set custom HTTP headers, can	override built in default headers.
	   Applicable only for HTTP output.

   ico
       ICO file	muxer.

       Microsoft's icon	file format (ICO) has some strict limitations that
       should be noted:

       o   Size	cannot exceed 256 pixels in any	dimension

       o   Only	BMP and	PNG images can be stored

       o   If a	BMP image is used, it must be one of the following pixel
	   formats:

		   BMP Bit Depth      FFmpeg Pixel Format
		   1bit		      pal8
		   4bit		      pal8
		   8bit		      pal8
		   16bit	      rgb555le
		   24bit	      bgr24
		   32bit	      bgra

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

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

   image2
       Image file muxer.

       The image file muxer writes video frames	to image files.

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

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

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

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

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

       Options

       frame_pts
	   If set to 1,	expand the filename with pts from pkt->pts.  Default
	   value is 0.

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

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

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

       protocol_opts options_list
	   Set protocol	options	as a :-separated list of key=value parameters.
	   Values containing the ":" special character must be escaped.

       Examples

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

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

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

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

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

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

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

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

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

       You can 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"

       A more complex example is to 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

   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

       This muxer supports the following options:

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

	   If this option is set to a non-zero value, the muxer	will reserve a
	   given amount	of space in the	file header and	then try to write the
	   cues	there when the muxing finishes.	If the 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.

       default_mode
	   This	option controls	how the	FlagDefault of the output tracks will
	   be set.  It influences which	tracks players should play by default.
	   The default mode is infer.

	   infer
	       In this mode, for each type of track (audio, video or
	       subtitle), if there is a	track with disposition default of this
	       type, then the first such track (i.e. the one with the lowest
	       index) will be marked as	default; if no such track exists, the
	       first track of this type	will be	marked as default instead (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.

       Examples

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

	       ffmpeg -i INPUT -f md5 out.md5

       You can print the MD5 to	stdout with the	command:

	       ffmpeg -i INPUT -f md5 -

       See also	the hash and framemd5 muxers.

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

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

       Options

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

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

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

       -frag_duration duration
	   Create fragments that are duration microseconds long.

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

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

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

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

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

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

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

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

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

       -movflags 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 global_sidx	flag is	enabled, this option will be
	   ignored.

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

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

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

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

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

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

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

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

       -write_prft
	   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.

	   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.

       Example

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

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

   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.

       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.

       tables_version integer
	   Set PAT, PMT	and SDT	version	(default 0, valid values are from 0 to
	   31, inclusively).  This option allows updating stream structure so
	   that	standard consumer may detect the change. To do so, reopen
	   output "AVFormatContext" (in	case of	API usage) or restart ffmpeg
	   instance, cyclically	changing tables_version	value:

		   ffmpeg -i source1.ts	-codec copy -f mpegts -tables_version 0	udp://1.1.1.1:1111
		   ffmpeg -i source2.ts	-codec copy -f mpegts -tables_version 1	udp://1.1.1.1:1111
		   ...
		   ffmpeg -i source3.ts	-codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
		   ffmpeg -i source1.ts	-codec copy -f mpegts -tables_version 0	udp://1.1.1.1:1111
		   ffmpeg -i source2.ts	-codec copy -f mpegts -tables_version 1	udp://1.1.1.1:1111
		   ...

       Example

	       ffmpeg -i file.mpg -c copy \
		    -mpegts_original_network_id	0x1122 \
		    -mpegts_transport_stream_id	0x3344 \
		    -mpegts_service_id 0x5566 \
		    -mpegts_pmt_start_pid 0x1500 \
		    -mpegts_start_pid 0x150 \
		    -metadata service_provider="Some provider" \
		    -metadata service_name="Some Channel" \
		    out.ts

   mxf,	mxf_d10, mxf_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.

   segment, stream_segment, ssegment
       Basic stream segmenter.

       This muxer outputs streams to a number of separate files	of nearly
       fixed duration. Output filename pattern can be set in a fashion similar
       to image2, or by	using a	"strftime" template if the strftime option is
       enabled.

       "stream_segment"	is a variant of	the muxer used to write	to streaming
       output formats, i.e. which do not require global	headers, and is
       recommended for outputting e.g. to MPEG transport stream	segments.
       "ssegment" is a shorter alias for "stream_segment".

       Every segment starts with a keyframe of the selected reference stream,
       which is	set through the	reference_stream option.

       Note that if you	want accurate splitting	for a video file, you need to
       make the	input key frames correspond to the exact splitting times
       expected	by the segmenter, or the segment muxer will start the new
       segment with the	key frame found	next after the specified start time.

       The segment muxer works best with a single constant frame rate video.

       Optionally it can generate a list of the	created	segments, by setting
       the option segment_list.	The list type is specified by the
       segment_list_type option. The entry filenames in	the segment list are
       set by default to the basename of the corresponding segment files.

       See also	the hls	muxer, which provides a	more specific implementation
       for HLS segmentation.

       Options

       The segment muxer supports the following	options:

       increment_tc 1|0
	   if set to 1,	increment timecode between each	segment	If this	is
	   selected, the input need to have a timecode in the first video
	   stream. Default value is 0.

       reference_stream	specifier
	   Set the reference stream, as	specified by the string	specifier.  If
	   specifier is	set to "auto", the reference is	chosen automatically.
	   Otherwise it	must be	a stream specifier (see	the ``Stream
	   specifiers''	chapter	in the ffmpeg manual) which specifies the
	   reference stream. The default value is "auto".

       segment_format format
	   Override the	inner container	format,	by default it is guessed by
	   the filename	extension.

       segment_format_options options_list
	   Set output format options using a :-separated list of key=value
	   parameters. Values containing the ":" special character must	be
	   escaped.

       segment_list name
	   Generate also a listfile named name.	If not specified no listfile
	   is generated.

       segment_list_flags flags
	   Set flags affecting the segment list	generation.

	   It currently	supports the following flags:

	   cache
	       Allow caching (only affects M3U8	list files).

	   live
	       Allow live-friendly file	generation.

       segment_list_size size
	   Update the list file	so that	it contains at most size segments. If
	   0 the list file will	contain	all the	segments. Default value	is 0.

       segment_list_entry_prefix prefix
	   Prepend prefix to each entry. Useful	to generate absolute paths.
	   By default no prefix	is applied.

       segment_list_type type
	   Select the listing format.

	   The following values	are recognized:

	   flat
	       Generate	a flat list for	the created segments, one segment per
	       line.

	   csv,	ext
	       Generate	a list for the created segments, one segment per line,
	       each line matching the format (comma-separated values):

		       <segment_filename>,<segment_start_time>,<segment_end_time>

	       segment_filename	is the name of the output file generated by
	       the muxer according to the provided pattern. CSV	escaping
	       (according to RFC4180) is applied if required.

	       segment_start_time and segment_end_time specify the segment
	       start and end time expressed in seconds.

	       A list file with	the suffix ".csv" or ".ext" will auto-select
	       this format.

	       ext is deprecated in favor or csv.

	   ffconcat
	       Generate	an ffconcat file for the created segments. The
	       resulting file can be read using	the FFmpeg concat demuxer.

	       A list file with	the suffix ".ffcat" or ".ffconcat" will	auto-
	       select this format.

	   m3u8
	       Generate	an extended M3U8 file, version 3, compliant with
	       <http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

	       A list file with	the suffix ".m3u8" will	auto-select this
	       format.

	   If not specified the	type is	guessed	from the list file name
	   suffix.

       segment_time time
	   Set segment duration	to time, the value must	be a duration
	   specification. Default value	is "2".	See also the segment_times
	   option.

	   Note	that splitting may not be accurate, unless you force the
	   reference stream key-frames at the given time. See the introductory
	   notice and the examples below.

       segment_atclocktime 1|0
	   If set to "1" split at regular clock	time intervals starting	from
	   00:00 o'clock. The time value specified in segment_time is used for
	   setting the length of the splitting interval.

	   For example with segment_time set to	"900" this makes it possible
	   to create files at 12:00 o'clock, 12:15, 12:30, etc.

	   Default value is "0".

       segment_clocktime_offset	duration
	   Delay the segment splitting times with the specified	duration when
	   using segment_atclocktime.

	   For example with segment_time set to	"900" and
	   segment_clocktime_offset set	to "300" this makes it possible	to
	   create files	at 12:05, 12:20, 12:35,	etc.

	   Default value is "0".

       segment_clocktime_wrap_duration duration
	   Force the segmenter to only start a new segment if a	packet reaches
	   the muxer within the	specified duration after the segmenting	clock
	   time. This way you can make the segmenter more resilient to
	   backward local time jumps, such as leap seconds or transition to
	   standard time from daylight savings time.

	   Default is the maximum possible duration which means	starting a new
	   segment regardless of the elapsed time since	the last clock time.

       segment_time_delta delta
	   Specify the accuracy	time when selecting the	start time for a
	   segment, expressed as a duration specification. Default value is
	   "0".

	   When	delta is specified a key-frame will start a new	segment	if its
	   PTS satisfies the relation:

		   PTS >= start_time - time_delta

	   This	option is useful when splitting	video content, which is	always
	   split at GOP	boundaries, in case a key frame	is found just before
	   the specified split time.

	   In particular may be	used in	combination with the ffmpeg option
	   force_key_frames. The key frame times specified by force_key_frames
	   may not be set accurately because of	rounding issues, with the
	   consequence that a key frame	time may result	set just before	the
	   specified time. For constant	frame rate videos a value of
	   1/(2*frame_rate) should address the worst case mismatch between the
	   specified time and the time set by force_key_frames.

       segment_times times
	   Specify a list of split points. times contains a list of comma
	   separated duration specifications, in increasing order. See also
	   the segment_time option.

       segment_frames frames
	   Specify a list of split video frame numbers.	frames contains	a list
	   of comma separated integer numbers, in increasing order.

	   This	option specifies to start a new	segment	whenever a reference
	   stream key frame is found and the sequential	number (starting from
	   0) of the frame is greater or equal to the next value in the	list.

       segment_wrap limit
	   Wrap	around segment index once it reaches limit.

       segment_start_number number
	   Set the sequence number of the first	segment. Defaults to 0.

       strftime	1|0
	   Use the "strftime" function to define the name of the new segments
	   to write. If	this is	selected, the output segment name must contain
	   a "strftime"	function template. Default value is 0.

       break_non_keyframes 1|0
	   If enabled, allow segments to start on frames other than keyframes.
	   This	improves behavior on some players when the time	between
	   keyframes is	inconsistent, but may make things worse	on others, and
	   can cause some oddities during seeking. Defaults to 0.

       reset_timestamps	1|0
	   Reset timestamps at the beginning of	each segment, so that each
	   segment will	start with near-zero timestamps. It is meant to	ease
	   the playback	of the generated segments. May not work	with some
	   combinations	of muxers/codecs. It is	set to 0 by default.

       initial_offset offset
	   Specify timestamp offset to apply to	the output packet timestamps.
	   The argument	must be	a time duration	specification, and defaults to
	   0.

       write_empty_segments 1|0
	   If enabled, write an	empty segment if there are no packets during
	   the period a	segment	would usually span. Otherwise, the segment
	   will	be filled with the next	packet written.	Defaults to 0.

       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.

   fifo
       The fifo	pseudo-muxer allows the	separation of encoding and muxing by
       using first-in-first-out	queue and running the actual muxer in a
       separate	thread.	This is	especially useful in combination with the tee
       muxer and can be	used to	send data to several destinations with
       different reliability/writing speed/latency.

       API users should	be aware that callback functions (interrupt_callback,
       io_open and io_close) used within its AVFormatContext must be thread-
       safe.

       The behavior of the fifo	muxer if the queue fills up or if the output
       fails is	selectable,

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

       o   encoding can	be blocked during temporary failure, or	continue
	   transparently dropping packets in case fifo queue fills up.

       fifo_format
	   Specify the format name. Useful if it cannot	be guessed from	the
	   output name suffix.

       queue_size
	   Specify size	of the queue (number of	packets). Default value	is 60.

       format_opts
	   Specify format options for the underlying muxer. Muxer options can
	   be specified	as a list of key=value pairs separated by ':'.

       drop_pkts_on_overflow bool
	   If set to 1 (true), in case the fifo	queue fills up,	packets	will
	   be dropped rather than blocking the encoder.	This makes it possible
	   to continue streaming without delaying the input, at	the cost of
	   omitting part of the	stream.	By default this	option is set to 0
	   (false), so in such cases the encoder will be blocked until the
	   muxer processes some	of the packets and none	of them	is lost.

       attempt_recovery	bool
	   If failure occurs, attempt to recover the output. This is
	   especially useful when used with network output, since it makes it
	   possible to restart streaming transparently.	 By default this
	   option is set to 0 (false).

       max_recovery_attempts
	   Sets	maximum	number of successive unsuccessful recovery attempts
	   after which the output fails	permanently. By	default	this option is
	   set to 0 (unlimited).

       recovery_wait_time duration
	   Waiting time	before the next	recovery attempt after previous
	   unsuccessful	recovery attempt. Default value	is 5 seconds.

       recovery_wait_streamtime	bool
	   If set to 0 (false),	the real time is used when waiting for the
	   recovery attempt (i.e. the recovery will be attempted after at
	   least recovery_wait_time seconds).  If set to 1 (true), the time of
	   the processed stream	is taken into account instead (i.e. the
	   recovery will be attempted after at least recovery_wait_time
	   seconds of the stream is omitted).  By default, this	option is set
	   to 0	(false).

       recover_any_error bool
	   If set to 1 (true), recovery	will be	attempted regardless of	type
	   of the error	causing	the failure. By	default	this option is set to
	   0 (false) and in case of certain (usually permanent)	errors the
	   recovery is not attempted even when attempt_recovery	is set to 1.

       restart_with_keyframe bool
	   Specify whether to wait for the keyframe after recovering from
	   queue overflow or failure. This option is set to 0 (false) by
	   default.

       Examples

       o   Stream something to rtmp server, continue processing	the stream at
	   real-time rate even in case of temporary failure (network outage)
	   and attempt to recover streaming every second indefinitely.

		   ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format	flv -map 0:v -map 0:a
		     -drop_pkts_on_overflow 1 -attempt_recovery	1 -recovery_wait_time 1	rtmp://example.com/live/stream_name

   tee
       The tee muxer can be used to write the same data	to several 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_dash_manifest
       WebM DASH Manifest muxer.

       This muxer implements the WebM DASH Manifest specification to generate
       the DASH	manifest XML. It also supports manifest	generation for DASH
       live streams.

       For more	information see:

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

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

       Options

       This muxer supports the following options:

       adaptation_sets
	   This	option has the following syntax: "id=x,streams=a,b,c
	   id=y,streams=d,e" where x and y are the unique identifiers of the
	   adaptation sets and a,b,c,d and e are the indices of	the
	   corresponding audio and video streams. Any number of	adaptation
	   sets	can be added using this	option.

       live
	   Set this to 1 to create a live stream DASH Manifest.	Default: 0.

       chunk_start_index
	   Start index of the first chunk. This	will go	in the startNumber
	   attribute of	the SegmentTemplate element in the manifest. Default:
	   0.

       chunk_duration_ms
	   Duration of each chunk in milliseconds. This	will go	in the
	   duration attribute of the SegmentTemplate element in	the manifest.
	   Default: 1000.

       utc_timing_url
	   URL of the page that	will return the	UTC timestamp in ISO format.
	   This	will go	in the value attribute of the UTCTiming	element	in the
	   manifest.  Default: None.

       time_shift_buffer_depth
	   Smallest time (in seconds) shifting buffer for which	any
	   Representation is guaranteed	to be available. This will go in the
	   timeShiftBufferDepth	attribute of the MPD element. Default: 60.

       minimum_update_period
	   Minimum update period (in seconds) of the manifest. This will go in
	   the minimumUpdatePeriod attribute of	the MPD	element. Default: 0.

       Example

	       ffmpeg -f webm_dash_manifest -i video1.webm \
		      -f webm_dash_manifest -i video2.webm \
		      -f webm_dash_manifest -i audio1.webm \
		      -f webm_dash_manifest -i audio2.webm \
		      -map 0 -map 1 -map 2 -map	3 \
		      -c copy \
		      -f webm_dash_manifest \
		      -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
		      manifest.xml

   webm_chunk
       WebM Live Chunk Muxer.

       This muxer writes out WebM headers and chunks as	separate files which
       can be consumed by clients that support WebM Live streams via DASH.

       Options

       This muxer supports the following options:

       chunk_start_index
	   Index of the	first chunk (defaults to 0).

       header
	   Filename of the header where	the initialization data	will be
	   written.

       audio_chunk_duration
	   Duration of each audio chunk	in milliseconds	(defaults to 5000).

       Example

	       ffmpeg -f v4l2 -i /dev/video0 \
		      -f alsa -i hw:0 \
		      -map 0:0 \
		      -c:v libvpx-vp9 \
		      -s 640x360 -keyint_min 30	-g 30 \
		      -f webm_chunk \
		      -header webm_live_video_360.hdr \
		      -chunk_start_index 1 \
		      webm_live_video_360_%d.chk \
		      -map 1:0 \
		      -c:a libvorbis \
		      -b:a 128k	\
		      -f webm_chunk \
		      -header webm_live_audio_128.hdr \
		      -chunk_start_index 1 \
		      -audio_chunk_duration 1000 \
		      webm_live_audio_128_%d.chk

METADATA
       FFmpeg is able to dump metadata from media files	into a simple
       UTF-8-encoded INI-like text file	and then load it back using the
       metadata	muxer/demuxer.

       The file	format is as follows:

       1.  A file consists of a	header and a number of metadata	tags divided
	   into	sections, each on its own line.

       2.  The header is a ;FFMETADATA string, followed	by a version number
	   (now	1).

       3.  Metadata tags are of	the form key=value

       4.  Immediately after header follows global metadata

       5.  After global	metadata there may be sections with
	   per-stream/per-chapter metadata.

       6.  A section starts with the section name in uppercase (i.e. STREAM or
	   CHAPTER) in brackets	([, ]) and ends	with next section or end of
	   file.

       7.  At the beginning of a chapter section there may be an optional
	   timebase to be used for start/end values. It	must be	in form
	   TIMEBASE=num/den, where num and den are integers. If	the timebase
	   is missing then start/end times are assumed to be in	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]

       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".

       Muliple subscribers may stream from the broker using the	command:

	       ffplay amqp://[[user]:[password]@]hostname[:port]

       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.

   async
       Asynchronous data filling wrapper for input stream.

       Fill data in a background thread, to decouple I/O operation from	demux
       thread.

	       async:<URL>
	       async:http://host/resource
	       async:cache:http://host/resource

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
	   BluRay angle

       chapter
	   Start chapter (1...N)

       playlist
	   Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

       Read longest playlist from BluRay mounted to /mnt/bluray:

	       bluray:/mnt/bluray

       Read angle 2 of playlist	4 from BluRay mounted to /mnt/bluray, start
       from chapter 2:

	       -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

   cache
       Caching wrapper for input stream.

       Cache the input stream to temporary file. It brings seeking capability
       to live streams.

	       cache:<URL>

   concat
       Physical	concatenation protocol.

       Read and	seek from many resources in sequence as	if they	were a unique
       resource.

       A URL accepted by this protocol has the syntax:

	       concat:<URL1>|<URL2>|...|<URLN>

       where URL1, URL2, ..., URLN are the urls	of the resource	to be
       concatenated, each one possibly specifying a distinct protocol.

       For example to read a sequence of files split1.mpeg, split2.mpeg,
       split3.mpeg with	ffplay use the command:

	       ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

       Note that you may need to escape	the character "|" which	is special for
       many shells.

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set the AES decryption key binary block from	given hexadecimal
	   representation.

       iv  Set the AES decryption initialization vector	binary block from
	   given hexadecimal representation.

       Accepted	URL formats:

	       crypto:<URL>
	       crypto+<URL>

   data
       Data in-line in the URI.	See
       <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given	inline with ffmpeg:

	       ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

	       file:<filename>

       where filename is the path of the file to read.

       An URL that does	not have a protocol prefix will	be assumed to be a
       file URL. Depending on the build, an URL	that looks like	a Windows path
       with the	drive letter at	the beginning will also	be assumed to be a
       file URL	(usually not the case in builds	for unix-like systems).

       For example to read from	a file input.mpeg with ffmpeg use the command:

	       ffmpeg -i file:input.mpeg output.mpeg

       This protocol accepts the following options:

       truncate
	   Truncate existing files on write, if	set to 1. A value of 0
	   prevents truncating.	Default	value is 1.

       blocksize
	   Set I/O operation maximum block size, in bytes. Default value is
	   "INT_MAX", which results in not limiting the	requested block	size.
	   Setting this	value reasonably low improves user termination request
	   reaction time, which	is valuable for	files on slow medium.

       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.

   hls
       Read Apple HTTP Live Streaming compliant	segmented stream as a uniform
       one. The	M3U8 playlists describing the segments can be remote HTTP
       resources or local files, accessed using	the standard file protocol.
       The nested protocol is declared by specifying "+proto" after the	hls
       URI scheme name,	where proto is either "file" or	"http".

	       hls+http://host/path/to/remote/resource.m3u8
	       hls+file://path/to/local/resource.m3u8

       Using this protocol is discouraged - the	hls demuxer should work	just
       as well (if not,	please report the issues) and is more complete.	 To
       use the hls demuxer instead, simply use the direct URLs to the m3u8
       files.

   http
       HTTP (Hyper Text	Transfer Protocol).

       This protocol accepts the following options:

       seekable
	   Control seekability of connection. If set to	1 the resource is
	   supposed to be seekable, if set to 0	it is assumed not to be
	   seekable, if	set to -1 it will try to autodetect if it is seekable.
	   Default value is -1.

       chunked_post
	   If set to 1 use chunked Transfer-Encoding for posts,	default	is 1.

       content_type
	   Set a specific content type for the POST messages or	for listen
	   mode.

       http_proxy
	   set HTTP proxy to tunnel through e.g. http://example.com:1234

       headers
	   Set custom HTTP headers, can	override built in default headers. The
	   value must be a string encoding the headers.

       multiple_requests
	   Use persistent connections if set to	1, default is 0.

       post_data
	   Set custom HTTP post	data.

       referer
	   Set the Referer header. Include 'Referer: URL' header in HTTP
	   request.

       user_agent
	   Override the	User-Agent header. If not specified the	protocol will
	   use a string	describing the libavformat build. ("Lavf/<version>")

       user-agent
	   This	is a deprecated	option,	you can	use user_agent instead it.

       timeout
	   Set timeout in microseconds of socket I/O operations	used by	the
	   underlying low level	operation. By default it is set	to -1, which
	   means that the timeout is not specified.

       reconnect_at_eof
	   If set then eof is treated like an error and	causes reconnection,
	   this	is useful for live / endless streams.

       reconnect_streamed
	   If set then even streamed/non seekable streams will be reconnected
	   on errors.

       reconnect_delay_max
	   Sets	the maximum delay in seconds after which to give up
	   reconnecting

       mime_type
	   Export the MIME type.

       http_version
	   Exports the HTTP response version number. Usually "1.0" or "1.1".

       icy If set to 1 request ICY (SHOUTcast) metadata	from the server. If
	   the server supports this, the metadata has to be retrieved by the
	   application by reading the icy_metadata_headers and
	   icy_metadata_packet options.	 The default is	1.

       icy_metadata_headers
	   If the server supports ICY metadata,	this contains the ICY-specific
	   HTTP	reply headers, separated by newline characters.

       icy_metadata_packet
	   If the server supports ICY metadata,	and icy	was set	to 1, this
	   contains the	last non-empty metadata	packet sent by the server. It
	   should be polled in regular intervals by applications interested in
	   mid-stream metadata updates.

       cookies
	   Set the cookies to be sent in future	requests. The format of	each
	   cookie is the same as the value of a	Set-Cookie HTTP	response
	   field. Multiple cookies can be delimited by a newline character.

       offset
	   Set initial byte offset.

       end_offset
	   Try to limit	the request to bytes preceding this offset.

       method
	   When	used as	a client option	it sets	the HTTP method	for the
	   request.

	   When	used as	a server option	it sets	the HTTP method	that is	going
	   to be expected from the client(s).  If the expected and the
	   received HTTP method	do not match the client	will be	given a	Bad
	   Request response.  When unset the HTTP method is not	checked	for
	   now.	This will be replaced by autodetection in the future.

       listen
	   If set to 1 enables experimental HTTP server. This can be used to
	   send	data when used as an output option, or read data from a	client
	   with	HTTP POST when used as an input	option.	 If set	to 2 enables
	   experimental	multi-client HTTP server. This is not yet implemented
	   in ffmpeg.c 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>

       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.

       HTTP Cookies

       Some HTTP requests will be denied unless	cookie values are passed in
       with the	request. The cookies option allows these cookies to be
       specified. At the very least, each cookie must specify a	value along
       with a path and domain.	HTTP requests that match both the domain and
       path will automatically include the cookie value	in the HTTP Cookie
       header field. Multiple cookies can be delimited by a newline.

       The required syntax to play a stream specifying a cookie	is:

	       ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

   Icecast
       Icecast protocol	(stream	to Icecast servers)

       This protocol accepts the following options:

       ice_genre
	   Set the stream genre.

       ice_name
	   Set the stream name.

       ice_description
	   Set the stream description.

       ice_url
	   Set the stream website URL.

       ice_public
	   Set if the stream should be public.	The default is 0 (not public).

       user_agent
	   Override the	User-Agent header. If not specified a string of	the
	   form	"Lavf/<version>" will be used.

       password
	   Set the Icecast mountpoint password.

       content_type
	   Set the stream content type.	This must be set if it is different
	   from	audio/mpeg.

       legacy_icecast
	   This	enables	support	for Icecast versions < 2.4.0, that do not
	   support the HTTP PUT	method but the SOURCE method.

	       icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

	       mmsh://<server>[:<port>][/<app>][/<playpath>]

   md5
       MD5 output protocol.

       Computes	the MD5	hash of	the data to be written,	and on close writes
       this to the designated output or	stdout if none is specified. It	can be
       used to test muxers without writing an actual file.

       Some examples follow.

	       # Write the MD5 hash of the encoded AVI file to the file	output.avi.md5.
	       ffmpeg -i input.flv -f avi -y md5:output.avi.md5

	       # Write the MD5 hash of the encoded AVI file to stdout.
	       ffmpeg -i input.flv -f avi -y md5:

       Note that some formats (typically MOV) require the output protocol to
       be seekable, so they will fail with the MD5 output protocol.

   pipe
       UNIX pipe access	protocol.

       Read and	write from UNIX	pipes.

       The accepted syntax is:

	       pipe:[<number>]

       number is the number corresponding to the file descriptor of the	pipe
       (e.g. 0 for stdin, 1 for	stdout,	2 for stderr).	If number is not
       specified, by default the stdout	file descriptor	will be	used for
       writing,	stdin for reading.

       For example to read from	stdin with ffmpeg:

	       cat test.wav | ffmpeg -i	pipe:0
	       # ...this is the	same as...
	       cat test.wav | ffmpeg -i	pipe:

       For writing to stdout with ffmpeg:

	       ffmpeg -i test.wav -f avi pipe:1	| cat >	test.avi
	       # ...this is the	same as...
	       ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       This protocol accepts the following options:

       blocksize
	   Set I/O operation maximum block size, in bytes. Default value is
	   "INT_MAX", which results in not limiting the	requested block	size.
	   Setting this	value reasonably low improves user termination request
	   reaction time, which	is valuable if data transmission is slow.

       Note that some formats (typically MOV), require the output protocol to
       be seekable, so they will fail with the pipe output protocol.

   prompeg
       Pro-MPEG	Code of	Practice #3 Release 2 FEC protocol.

       The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction
       mechanism for MPEG-2 Transport Streams sent over	RTP.

       This protocol must be used in conjunction with the "rtp_mpegts" muxer
       and the "rtp" protocol.

       The required syntax is:

	       -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

       The destination UDP ports are "port + 2"	for the	column FEC stream and
       "port + 4" for the row FEC stream.

       This protocol accepts the following options:

       l=n The number of columns (4-20,	LxD <= 100)

       d=n The number of rows (4-20, LxD <= 100)

       Example usage:

	       -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

   rtmp
       Real-Time Messaging Protocol.

       The Real-Time Messaging Protocol	(RTMP) is used for streaming
       multimedia content across a TCP/IP network.

       The required syntax is:

	       rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

       The accepted parameters are:

       username
	   An optional username	(mostly	for publishing).

       password
	   An optional password	(mostly	for publishing).

       server
	   The address of the RTMP server.

       port
	   The number of the TCP port to use (by default is 1935).

       app It is the name of the application to	access.	It usually corresponds
	   to the path where the application is	installed on the RTMP server
	   (e.g. /ondemand/, /flash/live/, etc.). You can override the value
	   parsed from the URI through the "rtmp_app" option, too.

       playpath
	   It is the path or name of the resource to play with reference to
	   the application specified in	app, may be prefixed by	"mp4:".	You
	   can override	the value parsed from the URI through the
	   "rtmp_playpath" option, too.

       listen
	   Act as a server, listening for an incoming connection.

       timeout
	   Maximum time	to wait	for the	incoming connection. Implies listen.

       Additionally, the following parameters can be set via command line
       options (or in code via "AVOption"s):

       rtmp_app
	   Name	of application to connect on the RTMP server. This option
	   overrides the parameter specified in	the URI.

       rtmp_buffer
	   Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
	   Extra arbitrary AMF connection parameters, parsed from a string,
	   e.g.	like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each
	   value is prefixed by	a single character denoting the	type, B	for
	   Boolean, N for number, S for	string,	O for object, or Z for null,
	   followed by a colon.	For Booleans the data must be either 0 or 1
	   for FALSE or	TRUE, respectively.  Likewise for Objects the data
	   must	be 0 or	1 to end or begin an object, respectively. Data	items
	   in subobjects may be	named, by prefixing the	type with 'N' and
	   specifying the name before the value	(i.e. "NB:myFlag:1"). This
	   option may be used multiple times to	construct arbitrary AMF
	   sequences.

       rtmp_flashver
	   Version of the Flash	plugin used to run the SWF player. The default
	   is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0
	   (compatible;	<libavformat version>).)

       rtmp_flush_interval
	   Number of packets flushed in	the same request (RTMPT	only). The
	   default is 10.

       rtmp_live
	   Specify that	the media is a live stream. No resuming	or seeking in
	   live	streams	is possible. The default value is "any", which means
	   the subscriber first	tries to play the live stream specified	in the
	   playpath. If	a live stream of that name is not found, it plays the
	   recorded stream. The	other possible values are "live" and
	   "recorded".

       rtmp_pageurl
	   URL of the web page in which	the media was embedded.	By default no
	   value will be sent.

       rtmp_playpath
	   Stream identifier to	play or	to publish. This option	overrides the
	   parameter specified in the URI.

       rtmp_subscribe
	   Name	of live	stream to subscribe to.	By default no value will be
	   sent.  It is	only sent if the option	is specified or	if rtmp_live
	   is set to live.

       rtmp_swfhash
	   SHA256 hash of the decompressed SWF file (32	bytes).

       rtmp_swfsize
	   Size	of the decompressed SWF	file, required for SWFVerification.

       rtmp_swfurl
	   URL of the SWF player for the media.	By default no value will be
	   sent.

       rtmp_swfverify
	   URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
	   URL of the target stream. Defaults to proto://host[:port]/app.

       For example to read with	ffplay a multimedia resource named "sample"
       from the	application "vod" from an RTMP server "myserver":

	       ffplay rtmp://myserver/vod/sample

       To publish to a password	protected server, passing the playpath and app
       names separately:

	       ffmpeg -re -i <input> -f	flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

   rtmpe
       Encrypted Real-Time Messaging Protocol.

       The Encrypted Real-Time Messaging Protocol (RTMPE) is used for
       streaming multimedia content within standard cryptographic primitives,
       consisting of Diffie-Hellman key	exchange and HMACSHA256, generating a
       pair of RC4 keys.

   rtmps
       Real-Time Messaging Protocol over a secure SSL connection.

       The Real-Time Messaging Protocol	(RTMPS)	is used	for streaming
       multimedia content across an encrypted connection.

   rtmpt
       Real-Time Messaging Protocol tunneled through HTTP.

       The Real-Time Messaging Protocol	tunneled through HTTP (RTMPT) is used
       for streaming multimedia	content	within HTTP requests to	traverse
       firewalls.

   rtmpte
       Encrypted Real-Time Messaging Protocol tunneled through HTTP.

       The Encrypted Real-Time Messaging Protocol tunneled through HTTP
       (RTMPTE)	is used	for streaming multimedia content within	HTTP requests
       to traverse firewalls.

   rtmpts
       Real-Time Messaging Protocol tunneled through HTTPS.

       The Real-Time Messaging Protocol	tunneled through HTTPS (RTMPTS)	is
       used for	streaming multimedia content within HTTPS requests to traverse
       firewalls.

   libsmbclient
       libsmbclient permits one	to manipulate CIFS/SMB network resources.

       Following syntax	is required.

	       smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

       This protocol accepts the following options.

       timeout
	   Set timeout in milliseconds of socket I/O operations	used by	the
	   underlying low level	operation. By default it is set	to -1, which
	   means that the timeout is not specified.

       truncate
	   Truncate existing files on write, if	set to 1. A value of 0
	   prevents truncating.	Default	value is 1.

       workgroup
	   Set the workgroup used for making connections. By default workgroup
	   is not specified.

       For more	information see: <http://www.samba.org/>.

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax	is required.

	       sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
	   Set timeout of socket I/O operations	used by	the underlying low
	   level operation. By default it is set to -1,	which means that the
	   timeout is not specified.

       truncate
	   Truncate existing files on write, if	set to 1. A value of 0
	   prevents truncating.	Default	value is 1.

       private_key
	   Specify the path of the file	containing private key to use during
	   authorization.  By default libssh searches for keys in the ~/.ssh/
	   directory.

       Example:	Play a file stored on remote server.

	       ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

   librtmp rtmp, rtmpe,	rtmps, rtmpt, rtmpte
       Real-Time Messaging Protocol and	its variants supported through
       librtmp.

       Requires	the presence of	the librtmp headers and	library	during
       configuration. You need to explicitly configure the build with
       "--enable-librtmp". If enabled this will	replace	the native RTMP
       protocol.

       This protocol provides most client functions and	a few server functions
       needed to support RTMP, RTMP tunneled in	HTTP (RTMPT), encrypted	RTMP
       (RTMPE),	RTMP over SSL/TLS (RTMPS) and tunneled variants	of these
       encrypted types (RTMPTE,	RTMPTS).

       The required syntax is:

	       <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

       where rtmp_proto	is one of the strings "rtmp", "rtmpt", "rtmpe",
       "rtmps",	"rtmpte", "rtmpts" corresponding to each RTMP variant, and
       server, port, app and playpath have the same meaning as specified for
       the RTMP	native protocol.  options contains a list of space-separated
       options of the form key=val.

       See the librtmp manual page (man	3 librtmp) for more information.

       For example, to stream a	file in	real-time to an	RTMP server using
       ffmpeg:

	       ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

       To play the same	stream using ffplay:

	       ffplay "rtmp://myserver/live/mystream live=1"

   rtp
       Real-time Transport Protocol.

       The required syntax for an RTP URL is:
       rtp://hostname[:port][?option=val...]

       port specifies the RTP port to use.

       The following URL options are supported:

       ttl=n
	   Set the TTL (Time-To-Live) value (for multicast only).

       rtcpport=n
	   Set the remote RTCP port to n.

       localrtpport=n
	   Set the local RTP port to n.

       localrtcpport=n'
	   Set the local RTCP port to n.

       pkt_size=n
	   Set max packet size (in bytes) to n.

       connect=0|1
	   Do a	"connect()" on the UDP socket (if set to 1) or not (if set to
	   0).

       sources=ip[,ip]
	   List	allowed	source IP addresses.

       block=ip[,ip]
	   List	disallowed (blocked) source IP addresses.

       write_to_source=0|1
	   Send	packets	to the source address of the latest received packet
	   (if set to 1) or to a default remote	address	(if set	to 0).

       localport=n
	   Set the local RTP port to n.

	   This	is a deprecated	option.	Instead, localrtpport should be	used.

       Important notes:

       1.  If rtcpport is not set the RTCP port	will be	set to the RTP port
	   value plus 1.

       2.  If localrtpport (the	local RTP port)	is not set any available port
	   will	be used	for the	local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it	will be	set to
	   the local RTP port value plus 1.

   rtsp
       Real-Time Streaming Protocol.

       RTSP is not technically a protocol handler in libavformat, it is	a
       demuxer and muxer. The demuxer supports both normal RTSP	(with data
       transferred over	RTP; this is used by e.g. Apple	and Microsoft) and
       Real-RTSP (with data transferred	over RDT).

       The muxer can be	used to	send a stream using RTSP ANNOUNCE to a server
       supporting it (currently	Darwin Streaming Server	and Mischa
       Spiegelmock's <https://github.com/revmischa/rtsp-server>).

       The required syntax for a RTSP url is:

	       rtsp://<hostname>[:<port>]/<path>

       Options can be set on the ffmpeg/ffplay command line, or	set in code
       via "AVOption"s or in "avformat_open_input".

       The following options are supported.

       initial_pause
	   Do not start	playing	the stream immediately if set to 1. Default
	   value is 0.

       rtsp_transport
	   Set RTSP transport protocols.

	   It accepts the following values:

	   udp Use UDP as lower	transport protocol.

	   tcp Use TCP (interleaving within the	RTSP control channel) as lower
	       transport protocol.

	   udp_multicast
	       Use UDP multicast as lower transport protocol.

	   http
	       Use HTTP	tunneling as lower transport protocol, which is	useful
	       for passing proxies.

	   Multiple lower transport protocols may be specified,	in that	case
	   they	are tried one at a time	(if the	setup of one fails, the	next
	   one is tried).  For the muxer, only the tcp and udp options are
	   supported.

       rtsp_flags
	   Set RTSP flags.

	   The following values	are accepted:

	   filter_src
	       Accept packets only from	negotiated peer	address	and port.

	   listen
	       Act as a	server,	listening for an incoming connection.

	   prefer_tcp
	       Try TCP for RTP transport first,	if TCP is available as RTSP
	       RTP transport.

	   Default value is none.

       allowed_media_types
	   Set media types to accept from the server.

	   The following flags are accepted:

	   video
	   audio
	   data

	   By default it accepts all media types.

       min_port
	   Set minimum local UDP port. Default value is	5000.

       max_port
	   Set maximum local UDP port. Default value is	65000.

       timeout
	   Set maximum timeout (in seconds) to wait for	incoming connections.

	   A value of -1 means infinite	(default). This	option implies the
	   rtsp_flags set to listen.

       reorder_queue_size
	   Set number of packets to buffer for handling	of reordered packets.

       stimeout
	   Set socket TCP I/O timeout in microseconds.

       user-agent
	   Override User-Agent header. If not specified, it defaults to	the
	   libavformat identifier string.

       When receiving data over	UDP, the demuxer tries to reorder received
       packets (since they may arrive out of order, or packets may get lost
       totally). This can be disabled by setting the maximum demuxing delay to
       zero (via the "max_delay" field of AVFormatContext).

       When watching multi-bitrate Real-RTSP streams with ffplay, the streams
       to display can be chosen	with "-vst" n and "-ast" n for video and audio
       respectively, and can be	switched on the	fly by pressing	"v" and	"a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       o   Watch a stream over UDP, with a max reordering delay	of 0.5
	   seconds:

		   ffplay -max_delay 500000 -rtsp_transport udp	rtsp://server/video.mp4

       o   Watch a stream tunneled over	HTTP:

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

       o   Send	a stream in realtime to	a RTSP server, for others to watch:

		   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       o   Receive a stream in realtime:

		   ffmpeg -rtsp_flags listen -i	rtsp://ownaddress/live.sdp <output>

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a
       protocol	handler	in libavformat,	it is a	muxer and demuxer.  It is used
       for signalling of RTP streams, by announcing the	SDP for	the streams
       regularly on a separate port.

       Muxer

       The syntax for a	SAP url	given to the muxer is:

	       sap://<destination>[:<port>][?<options>]

       The RTP packets are sent	to destination on port port, or	to port	5004
       if no port is specified.	 options is a "&"-separated list. The
       following options are supported:

       announce_addr=address
	   Specify the destination IP address for sending the announcements
	   to.	If omitted, the	announcements are sent to the commonly used
	   SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
	   or ff0e::2:7ffe if destination is an	IPv6 address.

       announce_port=port
	   Specify the port to send the	announcements on, defaults to 9875 if
	   not specified.

       ttl=ttl
	   Specify the time to live value for the announcements	and RTP
	   packets, defaults to	255.

       same_port=0|1
	   If set to 1,	send all RTP streams on	the same port pair. If zero
	   (the	default), all streams are sent on unique ports,	with each
	   stream on a port 2 numbers higher than the previous.	 VLC/Live555
	   requires this to be set to 1, to be able to receive the stream.
	   The RTP stack in libavformat	for receiving requires all streams to
	   be sent on unique ports.

       Example command lines follow.

       To broadcast a stream on	the local subnet, for watching in VLC:

	       ffmpeg -re -i <input> -f	sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

	       ffmpeg -re -i <input> -f	sap sap://224.0.0.255

       And for watching	in ffplay, over	IPv6:

	       ffmpeg -re -i <input> -f	sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a	SAP url	given to the demuxer is:

	       sap://[<address>][:<port>]

       address is the multicast	address	to listen for announcements on,	if
       omitted,	the default 224.2.127.254 (sap.mcast.net) is used. port	is the
       port that is listened on, 9875 if omitted.

       The demuxers listens for	announcements on the given address and port.
       Once an announcement is received, it tries to receive that particular
       stream.

       Example command lines follow.

       To play back the	first stream announced on the normal SAP multicast
       address:

	       ffplay sap://

       To play back the	first stream announced on one the default IPv6 SAP
       multicast address:

	       ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL	syntax is:

	       sctp://<host>:<port>[?<options>]

       The protocol accepts the	following options:

       listen
	   If set to any value,	listen for an incoming connection. Outgoing
	   connection is done by default.

       max_streams
	   Set the maximum number of streams. By default no limit is set.

   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".

       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

       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".

       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=1|0
	   Listen for an incoming connection. Default value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This	option is only relevant	in read	mode: if no data arrived in
	   more	than this time interval, raise error.

       listen_timeout=milliseconds
	   Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
	   Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
	   Set send buffer size, expressed bytes.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

       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.

       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.

       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:

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

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

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

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

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

       timecode_format
	   Timecode type to include in the frame and video stream metadata.
	   Must	be none, rp188vitc, rp188vitc2,	rp188ltc, rp188any, vitc,
	   vitc2, or serial. Defaults to none (not included).

       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.

       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.

       Examples

       o   Print the list of DirectShow	supported devices and exit:

		   $ ffmpeg -list_devices true -f dshow	-i dummy

       o   Open	video device Camera:

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

       o   Open	second video device with name Camera:

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

       o   Open	video device Camera and	audio device Microphone:

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

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

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

       o   Specify pin names to	capture	by name	or alternative name, specify
	   alternative device name:

		   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

       o   Configure a crossbar	device,	specifying crossbar pins, allow	user
	   to adjust video capture properties at startup:

		   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
			-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA	Analog Capture"

   fbdev
       Linux framebuffer input device.

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

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

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

       To record from the framebuffer device /dev/fb0 with ffmpeg:

	       ffmpeg -f fbdev -framerate 10 -i	/dev/fb0 out.avi

       You can take a single screenshot	image with the command:

	       ffmpeg -f fbdev -framerate 1 -i /dev/fb0	-frames:v 1 screenshot.jpeg

       Options

       framerate
	   Set the frame rate. Default is 25.

   gdigrab
       Win32 GDI-based screen capture device.

       This device allows you to capture a region of the display on Windows.

       There are two options for the input filename:

	       desktop

       or

	       title=<window_title>

       The first option	will capture the entire	desktop, or a fixed region of
       the desktop. The	second option will instead capture the contents	of a
       single window, regardless of its	position on the	screen.

       For example, to grab the	entire desktop using ffmpeg:

	       ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

       Grab a 640x480 region at	position "10,20":

	       ffmpeg -f gdigrab -framerate 6 -offset_x	10 -offset_y 20	-video_size vga	-i desktop out.mpg

       Grab the	contents of the	window named "Calculator"

	       ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. Use the value 0 to not
	   draw	the pointer. Default value is 1.

       framerate
	   Set the grabbing frame rate.	Default	value is "ntsc", corresponding
	   to a	frame rate of "30000/1001".

       show_region
	   Show	grabbed	region on screen.

	   If show_region is specified with 1, then the	grabbing region	will
	   be indicated	on screen. With	this option, it	is easy	to know	what
	   is being grabbed if only a portion of the screen is grabbed.

	   Note	that show_region is incompatible with grabbing the contents of
	   a single window.

	   For example:

		   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y	20 -i desktop out.mpg

       video_size
	   Set the video frame size. The default is to capture the full	screen
	   if desktop is selected, or the full window size if
	   title=window_title is selected.

       offset_x
	   When	capturing a region with	video_size, set	the distance from the
	   left	edge of	the screen or desktop.

	   Note	that the offset	calculation is from the	top left corner	of the
	   primary monitor on Windows. If you have a monitor positioned	to the
	   left	of your	primary	monitor, you will need to use a	negative
	   offset_x value to move the region to	that monitor.

       offset_y
	   When	capturing a region with	video_size, set	the distance from the
	   top edge of the screen or desktop.

	   Note	that the offset	calculation is from the	top left corner	of the
	   primary monitor on Windows. If you have a monitor positioned	above
	   your	primary	monitor, you will need to use a	negative offset_y
	   value to move the region to that monitor.

   iec61883
       FireWire	DV/HDV input device using libiec61883.

       To enable this input device, you	need libiec61883, libraw1394 and
       libavc1394 installed on your system. Use	the configure option
       "--enable-libiec61883" to compile with the device enabled.

       The iec61883 capture device supports capturing from a video device
       connected via IEEE1394 (FireWire), using	libiec61883 and	the new	Linux
       FireWire	stack (juju). This is the default DV/HDV input method in Linux
       Kernel 2.6.37 and later,	since the old FireWire stack was removed.

       Specify the FireWire port to be used as input file, or "auto" to	choose
       the first port connected.

       Options

       dvtype
	   Override autodetection of DV/HDV. This should only be used if auto
	   detection does not work, or if usage	of a different device type
	   should be prohibited. Treating a DV device as HDV (or vice versa)
	   will	not work and result in undefined behavior.  The	values auto,
	   dv and hdv are supported.

       dvbuffer
	   Set maximum size of buffer for incoming data, in frames. For	DV,
	   this	is an exact value. For HDV, it is not frame exact, since HDV
	   does	not have a fixed frame size.

       dvguid
	   Select the capture device by	specifying its GUID. Capturing will
	   only	be performed from the specified	device and fails if no device
	   with	the given GUID is found. This is useful	to select the input if
	   multiple devices are	connected at the same time.  Look at
	   /sys/bus/firewire/devices to	find out the GUIDs.

       Examples

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

		   ffplay -f iec61883 -i auto

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

		   ffmpeg -f iec61883 -i auto -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.  Defaults to bgr0.

       format_modifier
	   Format modifier to signal on	output frames.	This is	necessary to
	   import correctly into some APIs, but	can't be autodetected.	See
	   the libdrm documentation for	possible values.

       crtc_id
	   KMS CRTC ID to define the capture source.  The first	active plane
	   on the given	CRTC will be used.

       plane_id
	   KMS plane ID	to define the capture source.  Defaults	to the first
	   active plane	found if neither crtc_id nor plane_id are specified.

       framerate
	   Framerate to	capture	at.  This is not synchronised to any page
	   flipping or framebuffer changes - it	just defines the interval at
	   which the framebuffer is sampled.  Sampling faster than the
	   framebuffer update rate will	generate independent frames with the
	   same	content.  Defaults to 30.

       Examples

       o   Capture from	the first active plane,	download the result to normal
	   frames and encode.  This will only work if the framebuffer is both
	   linear and mappable - if not, the result may	be scrambled or	fail
	   to download.

		   ffmpeg -f kmsgrab -i	- -vf 'hwdownload,format=bgr0' output.mp4

       o   Capture from	CRTC ID	42 at 60fps, map the result to VAAPI, convert
	   to NV12 and encode as H.264.

		   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf	'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

       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. Currently
       only video data is supported. The filtergraph is	specified through the
       option graph.

       Options

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

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

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

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

       dumpgraph
	   Dump	graph to stderr.

       Examples

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

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

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

		   ffplay -f lavfi color=c=pink

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

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

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

		   ffplay -f lavfi "amovie=test.wav"

       o   Read	an audio stream	and a video stream and play it back with
	   ffplay:

		   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

       o   Dump	decoded	frames to images and closed captions to	a file
	   (experimental):

		   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

   libcdio
       Audio-CD	input device based on libcdio.

       To enable this input device during configuration	you need libcdio
       installed on your system. It requires the configure option
       "--enable-libcdio".

       This device allows playing and grabbing from an Audio-CD.

       For example to copy with	ffmpeg the entire Audio-CD in /dev/sr0,	you
       may run the command:

	       ffmpeg -f libcdio -i /dev/sr0 cd.wav

       Options

       speed
	   Set drive reading speed. Default value is 0.

	   The speed is	specified CD-ROM speed units. The speed	is set through
	   the libcdio "cdio_cddap_speed_set" function.	On many	CD-ROM drives,
	   specifying a	value too large	will result in using the fastest
	   speed.

       paranoia_mode
	   Set paranoia	recovery mode flags. It	accepts	one of the following
	   values:

	   disable
	   verify
	   overlap
	   neverskip
	   full

	   Default value is disable.

	   For more information	about the available recovery modes, consult
	   the paranoia	project	documentation.

   libdc1394
       IIDC1394	input device, based on libdc1394 and libraw1394.

       Requires	the configure option "--enable-libdc1394".

       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
	   Specify the number of bytes per frame, by default it	is set to
	   1024.

       fragment_size
	   Specify the minimal buffering fragment in PulseAudio, it will
	   affect the audio latency. By	default	it is unset.

       wallclock
	   Set the initial PTS using the current time. Default is 1.

       Examples

       Record a	stream from default device:

	       ffmpeg -f pulse -i default /tmp/pulse.wav

   sndio
       sndio input device.

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

       The filename to provide to the input device is the device node
       representing the	sndio input device, and	is usually set to /dev/audio0.

       For example to grab from	/dev/audio0 using ffmpeg use the command:

	       ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   video4linux2, v4l2
       Video4Linux2 input video	device.

       "v4l2" can be used as alias for "video4linux2".

       If FFmpeg is built with v4l-utils support (by using the
       "--enable-libv4l2" configure option), it	is possible to use it with the
       "-use_libv4l2" input device option.

       The name	of the device to grab is a file	device node, usually Linux
       systems tend to automatically create such nodes when the	device (e.g.
       an USB webcam) is plugged into the system, and has a name of the	kind
       /dev/videoN, where N is a number	associated to the device.

       Video4Linux2 devices usually support a limited set of widthxheight
       sizes and frame rates. You can check which are supported	using
       -list_formats all for Video4Linux2 devices.  Some devices, like TV
       cards, support one or more standards. It	is possible to list all	the
       supported standards using -list_standards all.

       The time	base for the timestamps	is 1 microsecond. Depending on the
       kernel version and configuration, the timestamps	may be derived from
       the real	time clock (origin at the Unix Epoch) or the monotonic clock
       (origin usually at boot time, unaffected	by NTP or manual changes to
       the clock). The -timestamps abs or -ts abs option can be	used to	force
       conversion into the real	time clock.

       Some usage examples of the video4linux2 device with ffmpeg and ffplay:

       o   List	supported formats for a	video4linux2 device:

		   ffplay -f video4linux2 -list_formats	all /dev/video0

       o   Grab	and show the input of a	video4linux2 device:

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

       o   Grab	and record the input of	a video4linux2 device, leave the frame
	   rate	and size as previously set:

		   ffmpeg -f video4linux2 -input_format	mjpeg -i /dev/video0 out.mpeg

       For more	information about Video4Linux, check <http://linuxtv.org/>.

       Options

       standard
	   Set the standard. Must be the name of a supported standard. To get
	   a list of the supported standards, use the list_standards option.

       channel
	   Set the input channel number. Default to -1,	which means using the
	   previously selected channel.

       video_size
	   Set the video frame size. The argument must be a string in the form
	   WIDTHxHEIGHT	or a valid size	abbreviation.

       pixel_format
	   Select the pixel format (only valid for raw video input).

       input_format
	   Set the preferred pixel format (for raw video) or a codec name.
	   This	option allows one to select the	input format, when several are
	   available.

       framerate
	   Set the preferred video frame rate.

       list_formats
	   List	available formats (supported pixel formats, codecs, and	frame
	   sizes) and exit.

	   Available values are:

	   all Show all	available (compressed and non-compressed) formats.

	   raw Show only raw video (non-compressed) formats.

	   compressed
	       Show only compressed formats.

       list_standards
	   List	supported standards and	exit.

	   Available values are:

	   all Show all	supported standards.

       timestamps, ts
	   Set type of timestamps for grabbed frames.

	   Available values are:

	   default
	       Use timestamps from the kernel.

	   abs Use absolute timestamps (wall clock).

	   mono2abs
	       Force conversion	from monotonic to absolute timestamps.

	   Default value is "default".

       use_libv4l2
	   Use libv4l2 (v4l-utils) conversion functions. Default is 0.

   vfwcap
       VfW (Video for Windows) capture input device.

       The filename passed as input is the capture driver number, ranging from
       0 to 9. You may use "list" as filename to print a list of drivers. Any
       other filename will be interpreted as device number 0.

       Options

       video_size
	   Set the video frame size.

       framerate
	   Set the grabbing frame rate.	Default	value is "ntsc", corresponding
	   to a	frame rate of "30000/1001".

   x11grab
       X11 video input device.

       To enable this input device during configuration	you need libxcb
       installed on your system. It will be automatically detected during
       configuration.

       This device allows one to capture a region of an	X11 display.

       The filename passed as input has	the syntax:

	       [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

       hostname:display_number.screen_number specifies the X11 display name of
       the screen to grab from.	hostname can be	omitted, and defaults to
       "localhost". The	environment variable DISPLAY contains the default
       display name.

       x_offset	and y_offset specify the offsets of the	grabbed	area with
       respect to the top-left border of the X11 screen. They default to 0.

       Check the X11 documentation (e.g. man X)	for more detailed information.

       Use the xdpyinfo	program	for getting basic information about the
       properties of your X11 display (e.g. grep for "name" or "dimensions").

       For example to grab from	:0.0 using ffmpeg:

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

       Grab at position	"10,20":

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

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. A	value of 0 specifies
	   not to draw the pointer. Default value is 1.

       follow_mouse
	   Make	the grabbed area follow	the mouse. The argument	can be
	   "centered" or a number of pixels PIXELS.

	   When	it is specified	with "centered", the grabbing region follows
	   the mouse pointer and keeps the pointer at the center of region;
	   otherwise, the region follows only when the mouse pointer reaches
	   within PIXELS (greater than zero) to	the edge of region.

	   For example:

		   ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

	   To follow only when the mouse pointer reaches within	100 pixels to
	   edge:

		   ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i	:0.0 out.mpg

       framerate
	   Set the grabbing frame rate.	Default	value is "ntsc", corresponding
	   to a	frame rate of "30000/1001".

       show_region
	   Show	grabbed	region on screen.

	   If show_region is specified with 1, then the	grabbing region	will
	   be indicated	on screen. With	this option, it	is easy	to know	what
	   is being grabbed if only a portion of the screen is grabbed.

       region_border
	   Set the region border thickness if -show_region 1 is	used.  Range
	   is 1	to 128 and default is 3	(XCB-based x11grab only).

	   For example:

		   ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20	out.mpg

	   With	follow_mouse:

		   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

       video_size
	   Set the video frame size. Default is	the full desktop.

       grab_x
       grab_y
	   Set the grabbing region coordinates.	They are expressed as offset
	   from	the top	left corner of the X11 window and correspond to	the
	   x_offset and	y_offset parameters in the device name.	The default
	   value for both options is 0.

OUTPUT DEVICES
       Output devices are configured elements in FFmpeg	that can write
       multimedia data to an output device attached to your system.

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

       You can disable all the output devices using the	configure option
       "--disable-outdevs", and	selectively enable an output device using the
       option "--enable-outdev=OUTDEV",	or you can disable a particular	input
       device using the	option "--disable-outdev=OUTDEV".

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

       A description of	the currently available	output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture)	output device.

       Examples

       o   Play	a file on default ALSA device:

		   ffmpeg -i INPUT -f alsa default

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

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

   caca
       CACA output device.

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

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

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

       Options

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

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

       driver
	   Set display driver.

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

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

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

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

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

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

       Examples

       o   The following command shows the ffmpeg output is an CACA window,
	   forcing its size to 80x25:

		   ffmpeg -i INPUT -c:v	rawvideo -pix_fmt rgb24	-window_size 80x25 -f caca -

       o   Show	the list of available drivers and exit:

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

       o   Show	the list of available dither colors and	exit:

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

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

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

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

       Options

       list_devices
	   If set to true, print a list	of devices and exit.  Defaults to
	   false. 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 mode. Must be unset,	half or	full.
	   Defaults to unset.

       timing_offset
	   Sets	the genlock timing pixel offset	on the used output.  Defaults
	   to unset.

       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.

       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.

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

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

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

       window_x
       window_y
	   Set the position of the window on the screen.

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

       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)

       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.

       ich, in_channel_count
	   Set the number of input channels. Default value is 0. Setting this
	   value is not	mandatory if the corresponding channel layout
	   in_channel_layout is	set.

       och, out_channel_count
	   Set the number of output channels. Default value is 0. Setting this
	   value is not	mandatory if the corresponding channel layout
	   out_channel_layout is set.

       uch, used_channel_count
	   Set the number of used input	channels. Default value	is 0. This
	   option is only used for special remapping.

       isr, in_sample_rate
	   Set the input sample	rate. Default value is 0.

       osr, out_sample_rate
	   Set the output sample rate. Default value is	0.

       isf, in_sample_fmt
	   Specify the input sample format. It is set by default to "none".

       osf, out_sample_fmt
	   Specify the output sample format. It	is set by default to "none".

       tsf, internal_sample_fmt
	   Set the internal sample format. Default value is "none".  This will
	   automatically be chosen when	it is not explicitly set.

       icl, in_channel_layout
       ocl, out_channel_layout
	   Set the input/output	channel	layout.

	   See the Channel Layout section in the ffmpeg-utils(1) manual	for
	   the required	syntax.

       clev, center_mix_level
	   Set the center mix level. It	is a value expressed in	deciBel, and
	   must	be in the interval [-32,32].

       slev, surround_mix_level
	   Set the surround mix	level. It is a value expressed in deciBel, and
	   must	be in the interval [-32,32].

       lfe_mix_level
	   Set LFE mix into non	LFE level. It is used when there is a LFE
	   input but no	LFE output. It is a value expressed in deciBel,	and
	   must	be in the interval [-32,32].

       rmvol, rematrix_volume
	   Set rematrix	volume.	Default	value is 1.0.

       rematrix_maxval
	   Set maximum output value for	rematrixing.  This can be used to
	   prevent clipping vs.	preventing volume reduction.  A	value of 1.0
	   prevents clipping.

       flags, swr_flags
	   Set flags used by the converter. Default value is 0.

	   It supports the following individual	flags:

	   res force resampling, this flag forces resampling to	be used	even
	       when the	input and output sample	rates match.

       dither_scale
	   Set the dither scale. Default value is 1.

       dither_method
	   Set dither method. Default value is 0.

	   Supported values:

	   rectangular
	       select rectangular dither

	   triangular
	       select triangular dither

	   triangular_hp
	       select triangular dither	with high pass

	   lipshitz
	       select Lipshitz noise shaping dither.

	   shibata
	       select Shibata noise shaping dither.

	   low_shibata
	       select low Shibata noise	shaping	dither.

	   high_shibata
	       select high Shibata noise shaping dither.

	   f_weighted
	       select f-weighted noise shaping dither

	   modified_e_weighted
	       select modified-e-weighted noise	shaping	dither

	   improved_e_weighted
	       select improved-e-weighted noise	shaping	dither

       resampler
	   Set resampling engine. Default value	is swr.

	   Supported values:

	   swr select the native SW Resampler; filter options precision	and
	       cheby are not applicable	in this	case.

	   soxr
	       select the SoX Resampler	(where available); compensation, and
	       filter options filter_size, phase_shift,	exact_rational,
	       filter_type & kaiser_beta, are not applicable in	this case.

       filter_size
	   For swr only, set resampling	filter size, default value is 32.

       phase_shift
	   For swr only, set resampling	phase shift, default value is 10, and
	   must	be in the interval [0,30].

       linear_interp
	   Use linear interpolation when enabled (the default).	Disable	it if
	   you want to preserve	speed instead of quality when exact_rational
	   fails.

       exact_rational
	   For swr only, when enabled, try to use exact	phase_count based on
	   input and output sample rate. However, if it	is larger than "1 <<
	   phase_shift", the phase_count will be "1 << phase_shift" as
	   fallback. Default is	enabled.

       cutoff
	   Set cutoff frequency	(swr: 6dB point; soxr: 0dB point) ratio; must
	   be a	float value between 0 and 1.  Default value is 0.97 with swr,
	   and 0.91 with soxr (which, with a sample-rate of 44100, preserves
	   the entire audio band to 20kHz).

       precision
	   For soxr only, the precision	in bits	to which the resampled signal
	   will	be calculated.	The default value of 20	(which,	with suitable
	   dithering, is appropriate for a destination bit-depth of 16)	gives
	   SoX's 'High Quality'; a value of 28 gives SoX's 'Very High
	   Quality'.

       cheby
	   For soxr only, selects passband rolloff none	(Chebyshev) & higher-
	   precision approximation for 'irrational' ratios. Default value is
	   0.

       async
	   For swr only, simple	1 parameter audio sync to timestamps using
	   stretching, squeezing, filling and trimming.	Setting	this to	1 will
	   enable filling and trimming,	larger values represent	the maximum
	   amount in samples that the data may be stretched or squeezed	for
	   each	second.	 Default value is 0, thus no compensation is applied
	   to make the samples match the audio timestamps.

       first_pts
	   For swr only, assume	the first pts should be	this value. The	time
	   unit	is 1 / sample rate.  This allows for padding/trimming at the
	   start of stream. By default,	no assumption is made about the	first
	   frame's expected pts, so no padding or trimming is done. For
	   example, this could be set to 0 to pad the beginning	with silence
	   if an audio stream starts after the video stream or to trim any
	   samples with	a negative pts due to encoder delay.

       min_comp
	   For swr only, set the minimum difference between timestamps and
	   audio data (in seconds) to trigger stretching/squeezing/filling or
	   trimming of the data	to make	it match the timestamps. The default
	   is that stretching/squeezing/filling	and trimming is	disabled
	   (min_comp = "FLT_MAX").

       min_hard_comp
	   For swr only, set the minimum difference between timestamps and
	   audio data (in seconds) to trigger adding/dropping samples to make
	   it match the	timestamps.  This option effectively is	a threshold to
	   select between hard (trim/fill) and soft (squeeze/stretch)
	   compensation. Note that all compensation is by default disabled
	   through min_comp.  The default is 0.1.

       comp_duration
	   For swr only, set duration (in seconds) over	which data is
	   stretched/squeezed to make it match the timestamps. Must be a non-
	   negative double float value,	default	value is 1.0.

       max_soft_comp
	   For swr only, set maximum factor by which data is
	   stretched/squeezed to make it match the timestamps. Must be a non-
	   negative double float value,	default	value is 0.

       matrix_encoding
	   Select matrixed stereo encoding.

	   It accepts the following values:

	   none
	       select none

	   dolby
	       select Dolby

	   dplii
	       select Dolby Pro	Logic II

	   Default value is "none".

       filter_type
	   For swr only, select	resampling filter type.	This only affects
	   resampling operations.

	   It accepts the following values:

	   cubic
	       select cubic

	   blackman_nuttall
	       select Blackman Nuttall windowed	sinc

	   kaiser
	       select Kaiser windowed sinc

       kaiser_beta
	   For swr only, set Kaiser window beta	value. Must be a double	float
	   value in the	interval [2,16], default value is 9.

       output_sample_bits
	   For swr only, set number of used output sample bits for dithering.
	   Must	be an integer in the interval [0,64], default value is 0,
	   which means it's not	used.

SCALER OPTIONS
       The video scaler	supports the following named options.

       Options may be set by specifying	-option	value in the FFmpeg tools,
       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.

       The name	and arguments of the filter are	optionally preceded and
       followed	by a list of link labels.  A link label	allows one to name a
       link and	associate it to	a filter output	or input pad. The preceding
       labels in_link_1	... in_link_N, are associated to the filter input
       pads, the following labels out_link_1 ... out_link_M, are associated to
       the output pads.

       When two	link labels with the same name are found in the	filtergraph, a
       link between the	corresponding input and	output pad is created.

       If an output pad	is not labelled, it is linked by default to the	first
       unlabelled input	pad of the next	filter in the filterchain.  For
       example in the filterchain

	       nullsrc,	split[L1], [L2]overlay,	nullsink

       the split filter	instance has two output	pads, and the overlay filter
       instance	two input pads.	The first output pad of	split is labelled
       "L1", the first input pad of overlay is labelled	"L2", and the second
       output pad of split is linked to	the second input pad of	overlay, which
       are both	unlabelled.

       In a filter description,	if the input label of the first	filter is not
       specified, "in" is assumed; if the output label of the last filter is
       not specified, "out" is assumed.

       In a complete filterchain all the unlabelled filter input and output
       pads must be connected. A filtergraph is	considered valid if all	the
       filter input and	output pads of all the filterchains are	connected.

       Libavfilter will	automatically insert scale filters where format
       conversion is required. It is possible to specify swscale flags for
       those automatically inserted scalers by prepending "sws_flags=flags;"
       to the filtergraph description.

       Here is a BNF description of the	filtergraph syntax:

	       <NAME>		  ::= sequence of alphanumeric characters and '_'
	       <FILTER_NAME>	  ::= <NAME>["@"<NAME>]
	       <LINKLABEL>	  ::= "[" <NAME> "]"
	       <LINKLABELS>	  ::= <LINKLABEL> [<LINKLABELS>]
	       <FILTER_ARGUMENTS> ::= sequence of chars	(possibly quoted)
	       <FILTER>		  ::= [<LINKLABELS>] <FILTER_NAME> ["="	<FILTER_ARGUMENTS>] [<LINKLABELS>]
	       <FILTERCHAIN>	  ::= <FILTER> [,<FILTERCHAIN>]
	       <FILTERGRAPH>	  ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

   Notes on filtergraph	escaping
       Filtergraph description composition entails several levels of escaping.
       See the "Quoting	and escaping" section in the ffmpeg-utils(1) manual
       for more	information about the employed escaping	procedure.

       A first level escaping affects the content of each filter option	value,
       which may contain the special character ":" used	to separate values, or
       one of the escaping characters "\'".

       A second	level escaping affects the whole filter	description, which may
       contain the escaping characters "\'" or the special characters "[],;"
       used by the filtergraph description.

       Finally,	when you specify a filtergraph on a shell commandline, you
       need to perform a third level escaping for the shell special characters
       contained within	it.

       For example, consider the following string to be	embedded in the
       drawtext	filter description text	value:

	       this is a 'string': may contain one, or more, special characters

       This string contains the	"'" special escaping character,	and the	":"
       special character, so it	needs to be escaped in this way:

	       text=this is a \'string\'\: may contain one, or more, special characters

       A second	level of escaping is required when embedding the filter
       description in a	filtergraph description, in order to escape all	the
       filtergraph special characters. Thus the	example	above becomes:

	       drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

       (note that in addition to the "\'" escaping special characters, also
       "," needs to be escaped).

       Finally an additional level of escaping is needed when writing the
       filtergraph description in a shell command, which depends on the
       escaping	rules of the adopted shell. For	example, assuming that "\" is
       special and needs to be escaped with another "\", the previous string
       will finally result in:

	       -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING
       Some filters support a generic enable option. For the filters
       supporting timeline editing, this option	can be set to an expression
       which is	evaluated before sending a frame to the	filter.	If the
       evaluation is non-zero, the filter will be enabled, otherwise the frame
       will be sent unchanged to the next filter in the	filtergraph.

       The expression accepts the following values:

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

       n   sequential number of	the input frame, starting from 0

       pos the position	in the file of the input frame,	NAN if unknown

       w
       h   width and height of the input frame if video

       Additionally, these filters support an enable command that can be used
       to re-define the	expression.

       Like any	other filtering	option,	the enable option follows the same
       rules.

       For example, to enable a	blur filter (smartblur)	from 10	seconds	to 3
       minutes,	and a curves filter starting at	3 seconds:

	       smartblur = enable='between(t,10,3*60)',
	       curves	 = enable='gte(t,3)' : preset=cross_process

       See "ffmpeg -filters" to	view which filters have	timeline support.

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.

AUDIO FILTERS
       When you	configure your FFmpeg build, you can disable any of the
       existing	filters	using "--disable-filters".  The	configure output will
       show the	audio filters included in your build.

       Below is	a description of the currently available audio filters.

   acompressor
       A compressor is mainly used to reduce the dynamic range of a signal.
       Especially modern music is mostly compressed at a high ratio to improve
       the overall loudness. It's done to get the highest attention of a
       listener, "fatten" the sound and	bring more "power" to the track.  If a
       signal is compressed too	much it	may sound dull or "dead" afterwards or
       it may start to "pump" (which could be a	powerful effect	but can	also
       destroy a track completely).  The right compression is the key to reach
       a professional sound and	is the high art	of mixing and mastering.
       Because of its complex settings it may take a long time to get the
       right feeling for this kind of effect.

       Compression is done by detecting	the volume above a chosen level
       "threshold" and dividing	it by the factor set with "ratio".  So if you
       set the threshold to -12dB and your signal reaches -6dB a ratio of 2:1
       will result in a	signal at -9dB.	Because	an exact manipulation of the
       signal would cause distortion of	the waveform the reduction can be
       levelled	over the time. This is done by setting "Attack"	and "Release".
       "attack"	determines how long the	signal has to rise above the threshold
       before any reduction will occur and "release" sets the time the signal
       has to fall below the threshold to reduce the reduction again. Shorter
       signals than the	chosen attack time will	be left	untouched.  The
       overall reduction of the	signal can be made up afterwards with the
       "makeup"	setting. So compressing	the peaks of a signal about 6dB	and
       raising the makeup to this level	results	in a signal twice as loud than
       the source. To gain a softer entry in the compression the "knee"
       flattens	the hard edge at the threshold in the range of the chosen
       decibels.

       The filter accepts the following	options:

       level_in
	   Set input gain. Default is 1. Range is between 0.015625 and 64.

       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, can be 2nd, 4th or	8th.  Default is 4th.

   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.

   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.

       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.

       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.

       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.

       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.

       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.

       m   Set overlap method.

	   It accepts the following values:

	   a   Select overlap-add method. Even not interpolated	samples	are
	       slightly	changed	with this method.

	   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.

       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.

       o   Set window overlap, in percentage of	window size. Allowed range is
	   from	50 to 95. Default value	is 75 percent.

       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.

       t   Set threshold value.	Allowed	range is from 1	to 100.	 Default value
	   is 10. Higher values	make clip detection less aggressive.

       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.

       m   Set overlap method.

	   It accepts the following values:

	   a   Select overlap-add method. Even not interpolated	samples	are
	       slightly	changed	with this method.

	   s   Select overlap-save method. Not interpolated samples remain
	       unchanged.

	   Default value is "a".

   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

   aderivative,	aintegral
       Compute derivative/integral of audio stream.

       Applying	both filters one after another produces	original audio.

   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).

   aeval
       Modify an audio signal according	to the specified expressions.

       This filter accepts one or more expressions (one	for each channel),
       which are evaluated and used to modify a	corresponding audio signal.

       It accepts the following	parameters:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   If the number of input channels is greater than the number of
	   expressions,	the last specified expression is used for the
	   remaining output channels.

       channel_layout, c
	   Set output channel layout. If not specified,	the channel layout is
	   specified by	the number of expressions. If set to same, it will use
	   by default the same input channel layout.

       Each expression in exprs	can contain the	following constants and
       functions:

       ch  channel number of the current expression

       n   number of the evaluated sample, starting from 0

       s   sample rate

       t   time	of the evaluated sample	expressed in seconds

       nb_in_channels
       nb_out_channels
	   input and output number of channels

       val(CH)
	   the value of	input channel with number CH

       Note: this filter is slow. For faster processing	you should use a
       dedicated filter.

       Examples

       o   Half	volume:

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

       o   Invert phase	of the second channel:

		   aeval=val(0)|-val(1)

   afade
       Apply fade-in/out effect	to input audio.

       A description of	the accepted parameters	follows.

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

       start_sample, ss
	   Specify the number of the start sample for starting to apply	the
	   fade	effect.	Default	is 0.

       nb_samples, ns
	   Specify the number of samples for which the fade effect has to
	   last. At the	end of the fade-in effect the output audio will	have
	   the same volume as the input	audio, at the end of the fade-out
	   transition the output audio will be silence.	Default	is 44100.

       start_time, st
	   Specify the start time of the fade effect. Default is 0.  The value
	   must	be specified as	a time duration; see the Time duration section
	   in the ffmpeg-utils(1) manual for the accepted syntax.  If set this
	   option is used instead of start_sample.

       duration, d
	   Specify the duration	of the fade effect. See	the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.  At
	   the end of the fade-in effect the output audio will have the	same
	   volume as the input audio, at the end of the	fade-out transition
	   the output audio will be silence.  By default the duration is
	   determined by nb_samples.  If set this option is used instead of
	   nb_samples.

       curve
	   Set curve for fade transition.

	   It accepts the following values:

	   tri select triangular, linear slope (default)

	   qsin
	       select quarter of sine wave

	   hsin
	       select half of sine wave

	   esin
	       select exponential sine wave

	   log select logarithmic

	   ipar
	       select inverted parabola

	   qua select quadratic

	   cub select cubic

	   squ select square root

	   cbr select cubic root

	   par select parabola

	   exp select exponential

	   iqsin
	       select inverted quarter of sine wave

	   ihsin
	       select inverted half of sine wave

	   dese
	       select double-exponential seat

	   desi
	       select double-exponential sigmoid

	   losi
	       select logistic sigmoid

	   nofade
	       no fade applied

       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.

       nr  Set the noise reduction in dB, allowed range	is 0.01	to 97.
	   Default value is 12 dB.

       nf  Set the noise floor in dB, allowed range is -80 to -20.  Default
	   value is -50	dB.

       nt  Set the noise type.

	   It accepts the following values:

	   w   Select white noise.

	   v   Select vinyl noise.

	   s   Select shellac noise.

	   c   Select custom noise, defined in "bn" option.

	       Default value is	white noise.

       bn  Set custom band noise for every one of 15 bands.  Bands are
	   separated by	' ' or '|'.

       rf  Set the residual floor in dB, allowed range is -80 to -20.  Default
	   value is -38	dB.

       tn  Enable noise	tracking. By default is	disabled.  With	this enabled,
	   noise floor is automatically	adjusted.

       tr  Enable residual tracking. By	default	is disabled.

       om  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.

       Commands

       This filter supports the	following commands:

       sample_noise, sn
	   Start or stop measuring noise profile.  Syntax for the command is :
	   "start" or "stop" string.  After measuring noise profile is stopped
	   it will be automatically applied in filtering.

       noise_reduction,	nr
	   Change noise	reduction. Argument is single float number.  Syntax
	   for the command is :	"noise_reduction"

       noise_floor, nf
	   Change noise	floor. Argument	is single float	number.	 Syntax	for
	   the command is : "noise_floor"

       output_mode, om
	   Change output mode operation.  Syntax for the command is : "i", "o"
	   or "n" string.

   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.	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"

   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
	   Enable applying gain	measured from power of IR.

	   Set which approach to use for auto gain measurement.

	   none
	       Do not apply any	gain.

	   peak
	       select peak gain, very conservative approach. This is default
	       value.

	   dc  select DC gain, limited application.

	   gn  select gain to noise approach, this is most popular one.

       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	gtype 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
	   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.

       rate
	   Set video stream frame rate.	This option is used only when response
	   is enabled.

       minp
	   Set minimal partition size used for convolution. Default is 8192.
	   Allowed range is from 1 to 32768.  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 32768.  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.

       Examples

       o   Apply reverb	to stream using	mono IR	file as	second input, complete
	   command using ffmpeg:

		   ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

   aformat
       Set output format constraints for the input audio. The framework	will
       negotiate the most appropriate format to	minimize conversions.

       It accepts the following	parameters:

       sample_fmts, 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

   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".

   aiir
       Apply an	arbitrary Infinite Impulse Response filter.

       It accepts the following	parameters:

       zeros, z
	   Set numerator/zeros coefficients.

       poles, p
	   Set denominator/poles coefficients.

       gains, k
	   Set channels	gains.

       dry_gain
	   Set input gain.

       wet_gain
	   Set output gain.

       format, f
	   Set coefficients format.

	   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 kind of processing.  Can	be "d" - direct	or "s" - serial
	   cascading. Default is "s".

       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" 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

   alimiter
       The limiter prevents an input signal from rising	over a desired
       threshold.  This	limiter	uses lookahead technology to prevent your
       signal from distorting.	It means that there is a small delay after the
       signal is processed. Keep in mind that the delay	it produces is the
       attack time you set.

       The filter accepts the following	options:

       level_in
	   Set input gain. Default is 1.

       level_out
	   Set output gain. Default is 1.

       limit
	   Don't let signals above this	level pass the limiter.	Default	is 1.

       attack
	   The limiter will reach its attenuation level	in this	amount of time
	   in milliseconds. Default is 5 milliseconds.

       release
	   Come	back from limiting to attenuation 1.0 in this amount of
	   milliseconds.  Default is 50	milliseconds.

       asc When	gain reduction is always needed	ASC takes care of releasing to
	   an average reduction	level rather than reaching a reduction of 0 in
	   the release time.

       asc_level
	   Select how much the release time is affected	by ASC,	0 means	nearly
	   no changes in release time while 1 produces higher release times.

       level
	   Auto	level output signal. Default is	enabled.  This normalizes
	   audio back to 0dB if	enabled.

       Depending on picked setting it is recommended to	upsample input 2x or
       4x times	with aresample before applying this filter.

   allpass
       Apply a two-pole	all-pass filter	with central frequency (in Hz)
       frequency, and filter-width width.  An all-pass filter changes the
       audio's frequency to phase relationship without changing	its frequency
       to amplitude relationship.

       The filter accepts the following	options:

       frequency, f
	   Set frequency in Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   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.

       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.

   amerge
       Merge two or more audio streams into a single multi-channel stream.

       The filter accepts the following	options:

       inputs
	   Set the number of inputs. Default is	2.

       If the channel layouts of the inputs are	disjoint, and therefore
       compatible, the channel layout of the output will be set	accordingly
       and the channels	will be	reordered as necessary.	If the channel layouts
       of the inputs are not disjoint, the output will have all	the channels
       of the first input then all the channels	of the second input, in	that
       order, and the channel layout of	the output will	be the default value
       corresponding to	the total number of channels.

       For example, if the first input is in 2.1 (FL+FR+LF) and	the second
       input is	FC+BL+BR, then the output will be in 5.1, with the channels in
       the following order: a1,	a2, b1,	a3, b2,	b3 (a1 is the first channel of
       the first input,	b1 is the first	channel	of the second input).

       On the other hand, if both input	are in stereo, the output channels
       will be in the default order: a1, a2, b1, b2, and the channel layout
       will be arbitrarily set to 4.0, which may or may	not be the expected
       value.

       All inputs must have the	same sample rate, and format.

       If inputs do not	have the same duration,	the output will	stop with the
       shortest.

       Examples

       o   Merge two mono files	into a stereo stream:

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

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

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

   amix
       Mixes multiple audio inputs into	a single output.

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

       For example

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

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

       It accepts the following	parameters:

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

       duration
	   How to determine the	end-of-stream.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

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

       weights
	   Specify weight of each input	audio stream as	sequence.  Each	weight
	   is separated	by space. By default all inputs	have same weight.

       Commands

       This filter supports the	following commands:

       weights
	   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 herz.  gain	set new	gain parameter
	   in dB.

	   Full	filter invocation with asendcmd	may look like this:
	   asendcmd=c='4.0 anequalizer change
	   0|f=200|w=50|g=1',anequalizer=...

   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:

       s   Set denoising strength. Allowed range is from 0.00001 to 10.
	   Default value is 0.00001.

       p   Set patch radius duration. Allowed range is from 1 to 100
	   milliseconds.  Default value	is 2 milliseconds.

       r   Set research	radius duration. Allowed range is from 2 to 300
	   milliseconds.  Default value	is 6 milliseconds.

       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.

       m   Set smooth factor. Default value is 11. Allowed range is from 1 to
	   15.

       Commands

       This filter supports the	following commands:

       s   Change denoise strength. Argument is	single float number.  Syntax
	   for the command is :	"s"

       o   Change output mode.	Syntax for the command is : "i", "o" or	"n"
	   string.

   anlms
       Apply Normalized	Least-Mean-Squares 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 filtered samples.

	   n   Pass difference between desired and filtered samples.

	       Default value is	o.

       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-zero	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-zero 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.

       Examples

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

		   apad=pad_len=1024

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

		   apad=whole_len=10000

       o   Use ffmpeg to pad the audio input with silence, so that the video
	   stream will always result the shortest and will be converted	until
	   the end in the output file when using the shortest option:

		   ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad"	-shortest OUTPUT

   aphaser
       Add a phasing effect to the input audio.

       A phaser	filter creates series of peaks and troughs in the frequency
       spectrum.  The position of the peaks and	troughs	are modulated so that
       they vary over time, creating a sweeping	effect.

       A description of	the accepted parameters	follows.

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.74

       delay
	   Set delay in	milliseconds. Default is 3.0.

       decay
	   Set decay. Default is 0.4.

       speed
	   Set modulation speed	in Hz. Default is 0.5.

       type
	   Set modulation type.	Default	is triangular.

	   It accepts the following values:

	   triangular, t
	   sinusoidal, s

   apulsator
       Audio pulsator is something between an autopanner and a tremolo.	 But
       it can produce funny stereo effects as well. Pulsator changes the
       volume of the left and right channel based on a LFO (low	frequency
       oscillator) with	different waveforms and	shifted	phases.	 This filter
       have the	ability	to define an offset between left and right channel. An
       offset of 0 means that both LFO shapes match each other.	 The left and
       right channel are altered equally - a conventional tremolo.  An offset
       of 50% means that the shape of the right	channel	is exactly shifted in
       phase (or moved backwards about half of the frequency) -	pulsator acts
       as an autopanner. At 1 both curves match	again. Every setting in
       between moves the phase shift gapless between all stages	and produces
       some "bypassing"	sounds with sine and triangle waveforms. The more you
       set the offset near 1 (starting from the	0.5) the faster	the signal
       passes from the left to the right speaker.

       The filter accepts the following	options:

       level_in
	   Set input gain. By default it is 1. Range is	[0.015625 - 64].

       level_out
	   Set output gain. By default it is 1.	Range is [0.015625 - 64].

       mode
	   Set waveform	shape the LFO will use.	Can be one of: sine, triangle,
	   square, sawup or sawdown. Default is	sine.

       amount
	   Set modulation. Define how much of original signal is affected by
	   the LFO.

       offset_l
	   Set left channel offset. Default is 0. Allowed range	is [0 -	1].

       offset_r
	   Set right channel offset. Default is	0.5. Allowed range is [0 - 1].

       width
	   Set pulse width. Default is 1. Allowed range	is [0 -	2].

       timing
	   Set possible	timing mode. Can be one	of: bpm, ms or hz. Default is
	   hz.

       bpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if
	   timing is set to bpm.

       ms  Set ms. Default is 500. Allowed range is [10	- 2000]. Only used if
	   timing is set to ms.

       hz  Set frequency in Hz.	Default	is 2. Allowed range is [0.01 - 100].
	   Only	used if	timing is set to hz.

   aresample
       Resample	the input audio	to the specified parameters, using the
       libswresample library. If none are specified then the filter will
       automatically convert between its input and output.

       This filter is also able	to stretch/squeeze the audio data to make it
       match the timestamps or to inject silence / cut out audio to make it
       match the timestamps, do	a combination of both or do neither.

       The filter accepts the syntax [sample_rate:]resampler_options, where
       sample_rate expresses a sample rate and resampler_options is a list of
       key=value pairs,	separated by ":". See the "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

   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.

   asetnsamples
       Set the number of samples per each output audio frame.

       The last	output packet may contain a different number of	samples, as
       the filter will flush all the remaining samples when the	input audio
       signals its end.

       The filter accepts the following	options:

       nb_out_samples, n
	   Set the number of frames per	each output audio frame. The number is
	   intended as the number of samples per each channel.	Default	value
	   is 1024.

       pad, p
	   If set to 1,	the filter will	pad the	last audio frame with zeroes,
	   so that the last frame will contain the same	number of samples as
	   the previous	ones. Default value is 1.

       For example, to set the number of per-frame samples to 1234 and disable
       padding for the last frame, use:

	       asetnsamples=n=1234:p=0

   asetrate
       Set the sample rate without altering the	PCM data.  This	will result in
       a change	of speed and pitch.

       The filter accepts the following	options:

       sample_rate, r
	   Set the output sample rate. Default is 44100	Hz.

   ashowinfo
       Show a line containing various information for each input audio frame.
       The input audio is not modified.

       The shown line contains a sequence of key/value pairs of	the form
       key:value.

       The following values are	shown in the output:

       n   The (sequential) number of the input	frame, starting	from 0.

       pts The presentation timestamp of the input frame, in time base units;
	   the time base depends on the	filter input pad, and is usually
	   1/sample_rate.

       pts_time
	   The presentation timestamp of the input frame in seconds.

       pos position of the frame in the	input stream, -1 if this information
	   in unavailable and/or meaningless (for example in case of synthetic
	   audio)

       fmt The sample format.

       chlayout
	   The channel layout.

       rate
	   The sample rate for the audio frame.

       nb_samples
	   The number of samples (per channel) in the frame.

       checksum
	   The Adler-32	checksum (printed in hexadecimal) of the audio data.
	   For planar audio, the data is treated as if all the planes were
	   concatenated.

       plane_checksums
	   A list of Adler-32 checksums	for each data plane.

   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:

	   tanh
	   atan
	   cubic
	   exp
	   alg
	   quintic
	   sin
       param
	   Set additional parameter which controls sigmoid function.

       Commands

       This filter supports the	all above options as commands.

   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
	   "[0.01 - 10]".

       metadata
	   Set metadata	injection. All the metadata keys are prefixed with
	   "lavfi.astats.X", where "X" is channel number starting from 1 or
	   string "Overall". Default is	disabled.

	   Available keys for each channel are:	DC_offset Min_level Max_level
	   Min_difference Max_difference Mean_difference RMS_difference
	   Peak_level RMS_peak RMS_trough Crest_factor Flat_factor Peak_count
	   Noise_floor Noise_floor_count Bit_depth Dynamic_range
	   Zero_crossings Zero_crossings_rate Number_of_NaNs Number_of_Infs
	   Number_of_denormals

	   and for Overall: DC_offset Min_level	Max_level Min_difference
	   Max_difference Mean_difference RMS_difference Peak_level RMS_level
	   RMS_peak RMS_trough Flat_factor Peak_count Noise_floor
	   Noise_floor_count Bit_depth Number_of_samples Number_of_NaNs
	   Number_of_Infs Number_of_denormals

	   For example full key	look like this "lavfi.astats.1.DC_offset" or
	   this	"lavfi.astats.Overall.Peak_count".

	   For description what	each key means read below.

       reset
	   Set number of frame after which stats are going to be recalculated.
	   Default is disabled.

       measure_perchannel
	   Select the entries which need to be 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 entries which need to be measured	overall. The metadata
	   keys	can be used as flags, default is all which measures
	   everything.	none disables all overall measurement.

       A description of	each shown parameter follows:

       DC offset
	   Mean	amplitude displacement from zero.

       Min level
	   Minimal sample level.

       Max level
	   Maximal sample level.

       Min difference
	   Minimal difference between two consecutive samples.

       Max difference
	   Maximal difference between two consecutive samples.

       Mean difference
	   Mean	difference between two consecutive samples.  The average of
	   each	difference between two consecutive samples.

       RMS difference
	   Root	Mean Square difference between two consecutive samples.

       Peak level dB
       RMS level dB
	   Standard peak and RMS level measured	in dBFS.

       RMS peak	dB
       RMS trough dB
	   Peak	and trough values for RMS level	measured over a	short window.

       Crest factor
	   Standard ratio of peak to RMS level (note: not in dB).

       Flat factor
	   Flatness (i.e. consecutive samples with the same value) of the
	   signal at its peak levels (i.e. either Min level or Max level).

       Peak count
	   Number of occasions (not the	number of samples) that	the signal
	   attained either Min level or	Max level.

       Noise floor dB
	   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.

       Bit depth
	   Overall bit depth of	audio. Number of bits used for each sample.

       Dynamic range
	   Measured dynamic range of audio in dB.

       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 0.5.

       wet Set wet gain, how much of filtered signal is	kept. Allowed range is
	   from	0 to 1.	 Default value is 0.8.

       decay
	   Set delay line decay	gain value. Allowed range is from 0 to 1.
	   Default value is 0.7.

       feedback
	   Set delay line feedback gain	value. Allowed range is	from 0 to 1.
	   Default value is 0.5.

       cutoff
	   Set cutoff frequency	in herz. 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.

       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"

   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 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".
	   Default is "slow". 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.

       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.

       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.

       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.

       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.

   bs2b
       Bauer stereo to binaural	transformation,	which improves headphone
       listening of stereo audio records.

       To enable compilation of	this filter you	need to	configure FFmpeg with
       "--enable-libbs2b".

       It accepts the following	parameters:

       profile
	   Pre-defined crossfeed level.

	   default
	       Default level (fcut=700,	feed=50).

	   cmoy
	       Chu Moy circuit (fcut=700, feed=60).

	   jmeier
	       Jan Meier circuit (fcut=650, feed=95).

       fcut
	   Cut frequency (in Hz).

       feed
	   Feed	level (in Hz).

   channelmap
       Remap input channels to new locations.

       It accepts the following	parameters:

       map Map channels	from input to output. The argument is a	'|'-separated
	   list	of mappings, each in the "in_channel-out_channel" or
	   in_channel form. in_channel can be either the name of the input
	   channel (e.g. FL for	front left) or its index in the	input channel
	   layout.  out_channel	is the name of the output channel or its index
	   in the output channel layout. If out_channel	is not given then it
	   is implicitly an index, starting with zero and increasing by	one
	   for each mapping.

       channel_layout
	   The channel layout of the output stream.

       If no mapping is	present, the filter will implicitly map	input channels
       to output channels, preserving indices.

       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.

   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.

       Commands

       This filter supports the	all above options as commands.

   crystalizer
       Simple algorithm	to expand audio	dynamic	range.

       The filter accepts the following	options:

       i   Sets	the intensity of effect	(default: 2.0).	Must be	in range
	   between 0.0 (unchanged sound) to 10.0 (maximum effect).

       c   Enable clipping. By default is enabled.

       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.

   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.

       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.

       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.

       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.

       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 - 20.0. Default value is
	   7.0.

       TP, tp
	   Set maximum true peak.  Range is -9.0 - +0.0. Default value is
	   -2.0.

       measured_I, measured_i
	   Measured IL of input	file.  Range is	-99.0 -	+0.0.

       measured_LRA, measured_lra
	   Measured LRA	of input file.	Range is  0.0 -	99.0.

       measured_TP, measured_tp
	   Measured true peak of input file.  Range is	-99.0 -	+99.0.

       measured_thresh
	   Measured threshold of input file.  Range is -99.0 - +0.0.

       offset
	   Set offset gain. Gain is applied before the true-peak limiter.
	   Range is  -99.0 - +99.0. Default is +0.0.

       linear
	   Normalize 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.

       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

   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".

   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.

   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 in start of	multi-channel
	   audio.  Can be any or all. Default is any.  With any, any sample
	   that	is detected as non-silence will	cause stopped trimming of
	   silence.  With all, only if all channels are	detected as non-
	   silence will	cause stopped trimming of silence.

       stop_periods
	   Set the count for trimming silence from the end of audio.  To
	   remove silence from the middle of a file, specify a stop_periods
	   that	is negative. This value	is then	treated	as a positive value
	   and is used to indicate the effect should restart processing	as
	   specified by	start_periods, making it suitable for removing periods
	   of silence in the middle of the audio.  Default value is 0.

       stop_duration
	   Specify a duration of silence that must exist before	audio is not
	   copied any more. By specifying a higher duration, silence that is
	   wanted can be left in the audio.  Default value is 0.

       stop_threshold
	   This	is the same as start_threshold but for trimming	silence	from
	   the end of audio.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude	ratio. Default value is	0.

       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 in end of	multi-channel
	   audio.  Can be any or all. Default is any.  With any, any sample
	   that	is detected as non-silence will	cause stopped trimming of
	   silence.  With all, only if all channels are	detected as non-
	   silence will	cause stopped trimming of silence.

       detection
	   Set how is silence detected.	Can be "rms" or	"peak".	Second is
	   faster and works better with	digital	silence	which is exactly 0.
	   Default value is "rms".

       window
	   Set duration	in number of seconds used to calculate size of window
	   in number of	samples	for detecting silence.	Default	value is 0.02.
	   Allowed range is from 0 to 10.

       Examples

       o   The following example shows how this	filter can be used to start a
	   recording that does not contain the delay at	the start which
	   usually occurs between pressing the record button and the start of
	   the performance:

		   silenceremove=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

   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"

   stereotools
       This filter has some handy utilities to manage stereo signals, for
       converting M/S stereo recordings	to L/R signal while having control
       over the	parameters or spreading	the stereo image of master track.

       The filter accepts the following	options:

       level_in
	   Set input level before filtering for	both channels. Defaults	is 1.
	   Allowed range is from 0.015625 to 64.

       level_out
	   Set output level after filtering for	both channels. Defaults	is 1.
	   Allowed range is from 0.015625 to 64.

       balance_in
	   Set input balance between both channels. Default is 0.  Allowed
	   range is from -1 to 1.

       balance_out
	   Set output balance between both channels. Default is	0.  Allowed
	   range is from -1 to 1.

       softclip
	   Enable softclipping.	Results	in analog distortion instead of	harsh
	   digital 0dB clipping. Disabled by default.

       mutel
	   Mute	the left channel. Disabled by default.

       muter
	   Mute	the right channel. Disabled by default.

       phasel
	   Change the phase of the left	channel. Disabled by default.

       phaser
	   Change the phase of the right channel. Disabled by default.

       mode
	   Set stereo mode. Available values are:

	   lr>lr
	       Left/Right to Left/Right, this is default.

	   lr>ms
	       Left/Right to Mid/Side.

	   ms>lr
	       Mid/Side	to Left/Right.

	   lr>ll
	       Left/Right to Left/Left.

	   lr>rr
	       Left/Right to Right/Right.

	   lr>l+r
	       Left/Right to Left + Right.

	   lr>rl
	       Left/Right to Right/Left.

	   ms>ll
	       Mid/Side	to Left/Left.

	   ms>rr
	       Mid/Side	to Right/Right.

       slev
	   Set level of	side signal. Default is	1.  Allowed range is from
	   0.015625 to 64.

       sbal
	   Set balance of side signal. Default is 0.  Allowed range is from -1
	   to 1.

       mlev
	   Set level of	the middle signal. Default is 1.  Allowed range	is
	   from	0.015625 to 64.

       mpan
	   Set middle signal pan. Default is 0.	Allowed	range is from -1 to 1.

       base
	   Set stereo base between mono	and inversed channels. Default is 0.
	   Allowed range is from -1 to 1.

       delay
	   Set delay in	milliseconds how much to delay left from right channel
	   and vice versa. Default is 0. Allowed range is from -20 to 20.

       sclevel
	   Set S/C level. Default is 1.	Allowed	range is from 1	to 100.

       phase
	   Set the stereo phase	in degrees. Default is 0. Allowed range	is
	   from	0 to 360.

       bmode_in, bmode_out
	   Set balance mode for	balance_in/balance_out option.

	   Can be one of the following:

	   balance
	       Classic balance mode. Attenuate one channel at time.  Gain is
	       raised up to 1.

	   amplitude
	       Similar as classic mode above but gain is raised	up to 2.

	   power
	       Equal power distribution, from -6dB to +6dB range.

       Examples

       o   Apply karaoke like effect:

		   stereotools=mlev=0.015625

       o   Convert M/S signal to L/R:

		   "stereotools=mode=ms>lr"

   stereowiden
       This filter enhance the stereo effect by	suppressing signal common to
       both channels and by delaying the signal	of left	into right and vice
       versa, thereby widening the stereo effect.

       The filter accepts the following	options:

       delay
	   Time	in milliseconds	of the delay of	left signal into right and
	   vice	versa.	Default	is 20 milliseconds.

       feedback
	   Amount of gain in delayed signal into right and vice	versa. Gives a
	   delay effect	of left	signal in right	output and vice	versa which
	   gives widening effect. Default is 0.3.

       crossfeed
	   Cross feed of left into right with inverted phase. This helps in
	   suppressing the mono. If the	value is 1 it will cancel all the
	   signal common to both channels. Default is 0.3.

       drymix
	   Set level of	input signal of	original channel. Default is 0.8.

       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.

       angle
	   Set angle of	stereo surround	transform, Allowed range is from 0 to
	   360.	 Default is 90.

       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.

       ally
	   Set spread usage of stereo image across Y axis for all channels.

       fcx, flx, frx, blx, brx,	slx, srx, bcx
	   Set spread usage of stereo image across X axis for each channel.

       fcy, fly, fry, bly, bry,	sly, sry, bcy
	   Set spread usage of stereo image across Y axis for each channel.

       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

	   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.

   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.

       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.

       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.

   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

       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. The input is not modified.	Statistics
       about the volume	will be	printed	in the log when	the input stream end
       is reached.

       In particular it	will show the mean volume (root	mean square), maximum
       volume (on a per-sample basis), and the beginning of a histogram	of the
       registered volume values	(from the maximum value	to a cumulated 1/1000
       of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

	       [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
	       [Parsed_volumedetect_0  0xa23120] max_volume: -4	dB
	       [Parsed_volumedetect_0  0xa23120] histogram_4db:	6
	       [Parsed_volumedetect_0  0xa23120] histogram_5db:	62
	       [Parsed_volumedetect_0  0xa23120] histogram_6db:	286
	       [Parsed_volumedetect_0  0xa23120] histogram_7db:	1042
	       [Parsed_volumedetect_0  0xa23120] histogram_8db:	2551
	       [Parsed_volumedetect_0  0xa23120] histogram_9db:	4609
	       [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means	that:

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

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

       o   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6	dB, etc.

       In other	words, raising the volume by +4	dB does	not cause any
       clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
       Below is	a description of the currently available audio sources.

   abuffer
       Buffer audio frames, and	make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in	libavfilter/asrc_abuffer.h.

       It accepts the following	parameters:

       time_base
	   The timebase	which will be used for timestamps of submitted frames.
	   It must be either a floating-point number or	in
	   numerator/denominator form.

       sample_rate
	   The sample rate of the incoming audio buffers.

       sample_fmt
	   The sample format of	the incoming audio buffers.  Either a sample
	   format name or its corresponding integer representation from	the
	   enum	AVSampleFormat in libavutil/samplefmt.h

       channel_layout
	   The channel layout of the incoming audio buffers.  Either a channel
	   layout name from channel_layout_map in libavutil/channel_layout.c
	   or its corresponding	integer	representation from the	AV_CH_LAYOUT_*
	   macros in libavutil/channel_layout.h

       channels
	   The number of channels of the incoming audio	buffers.  If both
	   channels and	channel_layout are specified, then they	must be
	   consistent.

       Examples

	       abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

       will instruct the source	to accept planar 16bit signed stereo at
       44100Hz.	 Since the sample format with name "s16p" corresponds to the
       number 6	and the	"stereo" channel layout	corresponds to the value 0x3,
       this is equivalent to:

	       abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

   aevalsrc
       Generate	an audio signal	specified by an	expression.

       This source accepts in input one	or more	expressions (one for each
       channel), which are evaluated and used to generate a corresponding
       audio signal.

       This source accepts the following options:

       exprs
	   Set the '|'-separated expressions list for each separate channel.
	   In case the channel_layout option is	not specified, the selected
	   channel layout depends on the number	of provided expressions.
	   Otherwise the last specified	expression is applied to the remaining
	   output channels.

       channel_layout, c
	   Set the channel layout. The number of channels in the specified
	   layout must be equal	to the number of specified expressions.

       duration, d
	   Set the minimum duration of the sourced audio. See the Time
	   duration section in the ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater than the
	   specified duration, as the generated	audio is always	cut at the end
	   of a	complete frame.

	   If not specified, or	the expressed duration is negative, the	audio
	   is supposed to be generated forever.

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default to 1024.

       sample_rate, s
	   Specify the sample rate, default to 44100.

       Each expression in exprs	can contain the	following constants:

       n   number of the evaluated sample, starting from 0

       t   time	of the evaluated sample	expressed in seconds, starting from 0

       s   sample rate

       Examples

       o   Generate silence:

		   aevalsrc=0

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

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

       o   Generate a two channels signal, specify the channel layout (Front
	   Center + Back Center) explicitly:

		   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       o   Generate white noise:

		   aevalsrc="-2+random(0)"

       o   Generate an amplitude modulated signal:

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

       o   Generate 2.5	Hz binaural beats on a 360 Hz carrier:

		   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

   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	coefficents 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.

       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.

       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]

       Since this filter is designed for reconstruction, it operates on	frame
       sequences without considering timestamps, and terminates	when either
       input reaches end of stream. This will cause problems if	your encoding
       pipeline	drops frames. If you're	trying to apply	an image as an overlay
       to a video stream, consider the overlay filter instead.

   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 bellow thresholds.

       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.

   bbox
       Compute the bounding box	for the	non-black pixels in the	input frame
       luminance plane.

       This filter computes the	bounding box containing	all the	pixels with a
       luminance value greater than the	minimum	allowed	value.	The parameters
       describing the bounding box are printed on the filter log.

       The filter accepts the following	option:

       min_val
	   Set the minimal luminance value. Default is 16.

   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 10. 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.

   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 luminance value for which
	   a pixel is considered "black". The provided value is	scaled
	   according to	the following equation:

		   <absolute_threshold>	= <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

	   luminance_range_size	and luminance_minimum_value depend on the
	   input video format, the range is [0-255] for	YUV full-range formats
	   and [16-235]	for YUV	non full-range formats.

	   Default value is 0.10.

       The following example sets the maximum pixel threshold to the minimum
       value, and detects only black intervals of 2 or more seconds:

	       blackdetect=d=2:pix_th=0.00

   blackframe
       Detect frames that are (almost) completely black. Can be	useful to
       detect chapter transitions or commercials. Output lines consist of the
       frame number of the detected frame, the percentage of blackness,	the
       position	in the file if known or	-1 and the timestamp in	seconds.

       In order	to display the output lines, you need to set the loglevel at
       least to	the AV_LOG_INFO	value.

       This filter exports frame metadata "lavfi.blackframe.pblack".  The
       value represents	the percentage of pixels in the	picture	that are below
       the threshold value.

       It accepts the following	parameters:

       amount
	   The percentage of the pixels	that have to be	below the threshold;
	   it defaults to 98.

       threshold, thresh
	   The threshold below which a pixel value is considered black;	it
	   defaults to 32.

   blend
       Blend two video frames into each	other.

       The "blend" filter takes	two input streams and outputs one stream, the
       first input is the "top"	layer and second input is "bottom" layer.  By
       default,	the output terminates when the longest input terminates.

       The "tblend" (time blend) filter	takes two consecutive frames from one
       single stream, and outputs the result obtained by blending the new
       frame on	top of the old frame.

       A description of	the accepted options follows.

       c0_mode
       c1_mode
       c2_mode
       c3_mode
       all_mode
	   Set blend mode for specific pixel component or all pixel components
	   in case of all_mode.	Default	value is "normal".

	   Available values for	component modes	are:

	   addition
	   grainmerge
	   and
	   average
	   burn
	   darken
	   difference
	   grainextract
	   divide
	   dodge
	   freeze
	   exclusion
	   extremity
	   glow
	   hardlight
	   hardmix
	   heat
	   lighten
	   linearlight
	   multiply
	   multiply128
	   negation
	   normal
	   or
	   overlay
	   phoenix
	   pinlight
	   reflect
	   screen
	   softlight
	   subtract
	   vividlight
	   xor
       c0_opacity
       c1_opacity
       c2_opacity
       c3_opacity
       all_opacity
	   Set blend opacity for specific pixel	component or all pixel
	   components in case of all_opacity. Only used	in combination with
	   pixel component blend modes.

       c0_expr
       c1_expr
       c2_expr
       c3_expr
       all_expr
	   Set blend expression	for specific pixel component or	all pixel
	   components in case of all_expr. Note	that related mode options will
	   be ignored if those are set.

	   The expressions can use the following variables:

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

	   X
	   Y   the coordinates of the current sample

	   W
	   H   the width and height of currently filtered plane

	   SW
	   SH  Width and height	scale 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

   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".

   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.

   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

   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.

   codecview
       Visualize information exported by some codecs.

       Some codecs can export information through frames using side-data or
       other means. For	example, some MPEG based codecs	export motion vectors
       through the export_mvs flag in the codec	flags2 option.

       The filter accepts the following	option:

       mv  Set motion vectors to visualize.

	   Available flags for mv are:

	   pf  forward predicted MVs of	P-frames

	   bf  forward predicted MVs of	B-frames

	   bb  backward	predicted MVs of B-frames

       qp  Display quantization	parameters using the chroma planes.

       mv_type,	mvt
	   Set motion vectors type to visualize. Includes MVs from all frames
	   unless specified by frame_type option.

	   Available flags for mv_type are:

	   fp  forward predicted MVs

	   bp  backward	predicted MVs

       frame_type, ft
	   Set frame type to visualize motion vectors of.

	   Available flags for frame_type are:

	   if  intra-coded frames (I-frames)

	   pf  predicted frames	(P-frames)

	   bf  bi-directionally	predicted frames (B-frames)

       Examples

       o   Visualize forward predicted MVs of all frames using ffplay:

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

       o   Visualize multi-directionals	MVs of P and B-Frames using ffplay:

		   ffplay -flags2 +export_mvs input.mp4	-vf codecview=mv=pf+bf+bb

   colorbalance
       Modify intensity	of primary colors (red,	green and blue)	of input
       frames.

       The filter allows an input frame	to be adjusted in the shadows,
       midtones	or highlights regions for the red-cyan,	green-magenta or blue-
       yellow balance.

       A positive adjustment value shifts the balance towards the primary
       color, a	negative value towards the complementary color.

       The filter accepts the following	options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

	   Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

       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.

   colorchannelmixer
       Adjust video input frames by re-mixing color channels.

       This filter modifies a color channel by adding the values associated to
       the other channels of the same pixels. For example if the value to
       modify is red, the output value will be:

	       <red>=<red>*<rr>	+ <blue>*<rb> +	<green>*<rg> + <alpha>*<ra>

       The filter accepts the following	options:

       rr
       rg
       rb
       ra  Adjust contribution of input	red, green, blue and alpha channels
	   for output red channel.  Default is 1 for rr, and 0 for rg, rb and
	   ra.

       gr
       gg
       gb
       ga  Adjust contribution of input	red, green, blue and alpha channels
	   for output green channel.  Default is 1 for gg, and 0 for gr, gb
	   and ga.

       br
       bg
       bb
       ba  Adjust contribution of input	red, green, blue and alpha channels
	   for output blue channel.  Default is	1 for bb, and 0	for br,	bg and
	   ba.

       ar
       ag
       ab
       aa  Adjust contribution of input	red, green, blue and alpha channels
	   for output alpha channel.  Default is 1 for aa, and 0 for ar, ag
	   and ab.

	   Allowed ranges for options are "[-2.0, 2.0]".

       Examples

       o   Convert source to grayscale:

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

       o   Simulate sepia tones:

		   colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

       Commands

       This filter supports the	all above options as commands.

   colorkey
       RGB colorspace color keying.

       The filter accepts the following	options:

       color
	   The color which will	be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01	matches	only the exact key color, while	1.0 matches
	   everything.

       blend
	   Blend percentage.

	   0.0 makes pixels either fully transparent, or not transparent at
	   all.

	   Higher values result	in semi-transparent pixels, with a higher
	   transparency	the more similar the pixels color is to	the key	color.

       Examples

       o   Make	every green pixel in the input image transparent:

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

       o   Overlay a greenscreen-video on top of a static background image.

		   ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

       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.

       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.

   colormatrix
       Convert color matrix.

       The filter accepts the following	options:

       src
       dst Specify the source and destination color matrix. Both values	must
	   be specified.

	   The accepted	values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt601
	       BT.601

	   bt470
	       BT.470

	   bt470bg
	       BT.470BG

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       For example to convert from BT.601 to SMPTE-240M, use the command:

	       colormatrix=bt601:smpte240m

   colorspace
       Convert colorspace, transfer characteristics or color primaries.	 Input
       video needs to have an even size.

       The filter accepts the following	options:

       all Specify all color properties	at once.

	   The accepted	values are:

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   bt601-6-525
	       BT.601-6	525

	   bt601-6-625
	       BT.601-6	625

	   bt709
	       BT.709

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       space
	   Specify output colorspace.

	   The accepted	values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470BG	or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   ycgco
	       YCgCo

	   bt2020ncl
	       BT.2020 with non-constant luminance

       trc Specify output transfer characteristics.

	   The accepted	values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   gamma22
	       Constant	gamma of 2.2

	   gamma28
	       Constant	gamma of 2.8

	   smpte170m
	       SMPTE-170M, BT.601-6 625	or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   srgb
	       SRGB

	   iec61966-2-1
	       iec61966-2-1

	   iec61966-2-4
	       iec61966-2-4

	   xvycc
	       xvycc

	   bt2020-10
	       BT.2020 for 10-bits content

	   bt2020-12
	       BT.2020 for 12-bits content

       primaries
	   Specify output color	primaries.

	   The accepted	values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG	or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   film
	       film

	   smpte431
	       SMPTE-431

	   smpte432
	       SMPTE-432

	   bt2020
	       BT.2020

	   jedec-p22
	       JEDEC P22 phosphors

       range
	   Specify output color	range.

	   The accepted	values are:

	   tv  TV (restricted) range

	   mpeg
	       MPEG (restricted) range

	   pc  PC (full) range

	   jpeg
	       JPEG (full) range

       format
	   Specify output color	format.

	   The accepted	values are:

	   yuv420p
	       YUV 4:2:0 planar	8-bits

	   yuv420p10
	       YUV 4:2:0 planar	10-bits

	   yuv420p12
	       YUV 4:2:0 planar	12-bits

	   yuv422p
	       YUV 4:2:2 planar	8-bits

	   yuv422p10
	       YUV 4:2:2 planar	10-bits

	   yuv422p12
	       YUV 4:2:2 planar	12-bits

	   yuv444p
	       YUV 4:4:4 planar	8-bits

	   yuv444p10
	       YUV 4:4:4 planar	10-bits

	   yuv444p12
	       YUV 4:4:4 planar	12-bits

       fast
	   Do a	fast conversion, which skips gamma/primary correction. This
	   will	take significantly less	CPU, but will be mathematically
	   incorrect. To get output compatible with that produced by the
	   colormatrix filter, use fast=1.

       dither
	   Specify dithering mode.

	   The accepted	values are:

	   none
	       No dithering

	   fsb Floyd-Steinberg dithering

       wpadapt
	   Whitepoint adaptation mode.

	   The accepted	values are:

	   bradford
	       Bradford	whitepoint adaptation

	   vonkries
	       von Kries whitepoint adaptation

	   identity
	       identity	whitepoint adaptation (i.e. no whitepoint adaptation)

       iall
	   Override all	input properties at once. Same accepted	values as all.

       ispace
	   Override input colorspace. Same accepted values as space.

       iprimaries
	   Override input color	primaries. Same	accepted values	as primaries.

       itrc
	   Override input transfer characteristics. Same accepted values as
	   trc.

       irange
	   Override input color	range. Same accepted values as range.

       The filter converts the transfer	characteristics, color space and color
       primaries to the	specified user values. The output value, if not
       specified, is set to a default value based on the "all" property. If
       that property is	also not specified, the	filter will log	an error. The
       output color range and format default to	the same value as the input
       color range and format. The input transfer characteristics, color
       space, color primaries and color	range should be	set on the input data.
       If any of these are missing, the	filter will log	an error and no
       conversion will take place.

       For example to convert the input	to SMPTE-240M, use the command:

	       colorspace=smpte240m

   convolution
       Apply convolution 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 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.

       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

   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

       t   The timestamp expressed in seconds. It's NAN	if the input timestamp
	   is unknown.

       The expression for out_w	may depend on the value	of out_h, and the
       expression for out_h may	depend on out_w, but they cannot depend	on x
       and y, as x and y are evaluated after out_w and out_h.

       The x and y parameters specify the expressions for the position of the
       top-left	corner of the output (non-cropped) area. They are evaluated
       for each	frame. If the evaluated	value is not valid, it is approximated
       to the nearest valid value.

       The expression for x may	depend on y, and the expression	for y may
       depend on x.

       Examples

       o   Crop	area with size 100x100 at position (12,34).

		   crop=100:100:12:34

	   Using named options,	the example above becomes:

		   crop=w=100:h=100:x=12:y=34

       o   Crop	the central input area with size 100x100:

		   crop=100:100

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

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

       o   Crop	the input video	central	square:

		   crop=out_w=in_h
		   crop=in_h

       o   Delimit the rectangle with the top-left corner placed at position
	   100:100 and the right-bottom	corner corresponding to	the right-
	   bottom corner of the	input image.

		   crop=in_w-100:in_h-100:100:100

       o   Crop	10 pixels from the left	and right borders, and 20 pixels from
	   the top and bottom borders

		   crop=in_w-2*10:in_h-2*20

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

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

       o   Crop	height for getting Greek harmony:

		   crop=in_w:1/PHI*in_w

       o   Apply trembling effect:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

       o   Apply erratic camera	effect depending on timestamp:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"

       o   Set x depending on the value	of y:

		   crop=in_w/2:in_h/2:y:10+10*sin(n/10)

       Commands

       This filter supports the	following commands:

       w, out_w
       h, out_h
       x
       y   Set width/height of the output video	and the	horizontal/vertical
	   position in the input video.	 The command accepts the same syntax
	   of the corresponding	option.

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

   cropdetect
       Auto-detect the crop size.

       It calculates the necessary cropping parameters and prints the
       recommended parameters via the logging system. The detected dimensions
       correspond to the non-black area	of the input video.

       It accepts the following	parameters:

       limit
	   Set higher black value threshold, which can be optionally specified
	   from	nothing	(0) to everything (255 for 8-bit based formats). An
	   intensity value greater to the set value is considered non-black.
	   It defaults to 24.  You can also specify a value between 0.0	and
	   1.0 which will be scaled depending on the bitdepth of the pixel
	   format.

       round
	   The value which the width/height should be divisible	by. It
	   defaults to 16. The offset is automatically adjusted	to center the
	   video. Use 2	to get only even dimensions (needed for	4:2:2 video).
	   16 is best when encoding to most video codecs.

       reset_count, reset
	   Set the counter that	determines after how many frames cropdetect
	   will	reset the previously detected largest video area and start
	   over	to detect the current optimal crop area. Default value is 0.

	   This	can be useful when channel logos distort the video area. 0
	   indicates 'never reset', and	returns	the largest area encountered
	   during playback.

   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 (using	a natural cubic	spline interpolation) will be define
       to pass smoothly	through	all these new coordinates. The new defined
       points needs to be strictly increasing over the x-axis, and their x and
       y values	must be	in the [0;1] interval.	If the computed	curves
       happened	to go outside the vector spaces, the values will be clipped
       accordingly.

       The filter accepts the following	options:

       preset
	   Select one of the available color presets. This option can be used
	   in addition to the r, g, b parameters; in this case,	the later
	   options takes priority on the preset	values.	 Available presets
	   are:

	   none
	   color_negative
	   cross_process
	   darker
	   increase_contrast
	   lighter
	   linear_contrast
	   medium_contrast
	   negative
	   strong_contrast
	   vintage

	   Default is "none".

       master, m
	   Set the master key points. These points will	define a second	pass
	   mapping. It is sometimes called a "luminance" or "value" mapping.
	   It can be used with r, g, b or all since it acts like a post-
	   processing LUT.

       red, r
	   Set the key points for the red component.

       green, g
	   Set the key points for the green component.

       blue, b
	   Set the key points for the blue component.

       all Set the key points for all components (not including	master).  Can
	   be used in addition to the other key	points component options. In
	   this	case, the unset	component(s) will fallback on this all
	   setting.

       psfile
	   Specify a Photoshop curves file (".acv") to import the settings
	   from.

       plot
	   Save	Gnuplot	script of the curves in	specified file.

       To avoid	some filtergraph syntax	conflicts, each	key points list	need
       to be defined using the following syntax: "x0/y0	x1/y1 x2/y2 ...".

       Examples

       o   Increase slightly the middle	level of blue:

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

       o   Vintage effect:

		   curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'

	   Here	we obtain the following	coordinates for	each components:

	   red "(0;0.11) (0.42;0.51) (1;0.95)"

	   green
	       "(0;0) (0.50;0.48) (1;1)"

	   blue
	       "(0;0.22) (0.49;0.44) (1;0.80)"

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

		   curves=preset=vintage

       o   Or simply:

		   curves=vintage

       o   Use a Photoshop preset and redefine the points of the green
	   component:

		   curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

       o   Check out the curves	of the "cross_process" profile using ffmpeg
	   and gnuplot:

		   ffmpeg -f lavfi -i color -vf	curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
		   gnuplot -p /tmp/curves.plt

   datascope
       Video data analysis filter.

       This filter shows hexadecimal pixel values of part of video.

       The filter accepts the following	options:

       size, s
	   Set output video size.

       x   Set x offset	from where to pick pixels.

       y   Set y offset	from where to pick pixels.

       mode
	   Set scope mode, can be one of the following:

	   mono
	       Draw hexadecimal	pixel values with white	color on black
	       background.

	   color
	       Draw hexadecimal	pixel values with input	video pixel color on
	       black background.

	   color2
	       Draw hexadecimal	pixel values on	color background picked	from
	       input video, the	text color is picked in	such way so its	always
	       visible.

       axis
	   Draw	rows and columns numbers on left and top of video.

       opacity
	   Set background opacity.

       format
	   Set display number format. Can be "hex", or "dec". Default is
	   "hex".

   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.

   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

   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.

   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.

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

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

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

       Examples

       o   Set a rectangle covering the	area with top left corner coordinates
	   0,0 and size	100x77,	and a band of size 10:

		   delogo=x=0:y=0:w=100:h=77:band=10

   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>.

       Native model files (.model) can be generated from TensorFlow model
       files (.pb) by using tools/python/convert.py

       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:

	   native
	       Native implementation of	DNN loading and	execution.

	   tensorflow
	       TensorFlow backend. To enable this backend you need to install
	       the TensorFlow for C library (see
	       <https://www.tensorflow.org/install/install_c>) and configure
	       FFmpeg with "--enable-libtensorflow"

	   Default value is native.

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats. TensorFlow and native backend can load files for only its
	   format.

       It can also be finished with 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.

   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_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:

	   native
	       Native implementation of	DNN loading and	execution.

	   tensorflow
	       TensorFlow backend. To enable this backend you need to install
	       the TensorFlow for C library (see
	       <https://www.tensorflow.org/install/install_c>) and configure
	       FFmpeg with "--enable-libtensorflow"

	   Default value is native.

       model
	   Set path to model file specifying network architecture and its
	   parameters.	Note that different backends use different file
	   formats. TensorFlow and native backend can load files for only its
	   format.

	   Native model	file (.model) can be generated from TensorFlow model
	   file	(.pb) by using tools/python/convert.py

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       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   Halve the pixel value of the	frame with format gray32f:

		   ffmpeg -i input.jpg -vf format=grayf32,dnn_processing=model=halve_gray_float.model:input=dnn_in:output=dnn_out:dnn_backend=native -y	out.native.png

       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):

		   ./ffmpeg -i 480p.jpg	-vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y -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.

       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".	To enable default font fallback	and the	font
       option you need to configure FFmpeg with	"--enable-libfontconfig".  To
       enable the text_shaping option, you need	to configure FFmpeg with
       "--enable-libfribidi".

       Syntax

       It accepts the following	parameters:

       box Used	to draw	a box around text using	the background color.  The
	   value must be either	1 (enable) or 0	(disable).  The	default	value
	   of box is 0.

       boxborderw
	   Set the width of the	border to be drawn around the box using
	   boxcolor.  The default value	of boxborderw is 0.

       boxcolor
	   The color to	be used	for drawing box	around text. For the syntax of
	   this	option,	check the "Color" section in the ffmpeg-utils manual.

	   The default value of	boxcolor is "white".

       line_spacing
	   Set the line	spacing	in pixels of the border	to be drawn around the
	   box using box.  The default value of	line_spacing is	0.

       borderw
	   Set the width of the	border to be drawn around the text using
	   bordercolor.	 The default value of borderw is 0.

       bordercolor
	   Set the color to be used for	drawing	border around text. For	the
	   syntax of this option, check	the "Color" section in the ffmpeg-
	   utils manual.

	   The default value of	bordercolor is "black".

       expansion
	   Select how the text is expanded. Can	be either "none", "strftime"
	   (deprecated)	or "normal" (default). See the drawtext_expansion,
	   Text	expansion section below	for details.

       basetime
	   Set a start time for	the count. Value is in microseconds. Only
	   applied in the deprecated strftime expansion	mode. To emulate in
	   normal expansion mode use the "pts" function, supplying the start
	   time	(in seconds) as	the second argument.

       fix_bounds
	   If true, check and fix text coords to avoid clipping.

       fontcolor
	   The color to	be used	for drawing fonts. For the syntax of this
	   option, check the "Color" section in	the ffmpeg-utils manual.

	   The default value of	fontcolor is "black".

       fontcolor_expr
	   String which	is expanded the	same way as text to obtain dynamic
	   fontcolor value. By default this option has empty value and is not
	   processed. When this	option is set, it overrides fontcolor option.

       font
	   The font family to be used for drawing text.	By default Sans.

       fontfile
	   The font file to be used for	drawing	text. The path must be
	   included.  This parameter is	mandatory if the fontconfig support is
	   disabled.

       alpha
	   Draw	the text applying alpha	blending. The value can	be a number
	   between 0.0 and 1.0.	 The expression	accepts	the same variables x,
	   y as	well.  The default value is 1.	Please see fontcolor_expr.

       fontsize
	   The font size to be used for	drawing	text.  The default value of
	   fontsize is 16.

       text_shaping
	   If set to 1,	attempt	to shape the text (for example,	reverse	the
	   order of right-to-left text and join	Arabic characters) before
	   drawing it.	Otherwise, just	draw the text exactly as given.	 By
	   default 1 (if supported).

       ft_load_flags
	   The flags to	be used	for loading the	fonts.

	   The flags map the corresponding flags supported by libfreetype, and
	   are a combination of	the following values:

	   default
	   no_scale
	   no_hinting
	   render
	   no_bitmap
	   vertical_layout
	   force_autohint
	   crop_bitmap
	   pedantic
	   ignore_global_advance_width
	   no_recurse
	   ignore_transform
	   monochrome
	   linear_design
	   no_autohint

	   Default value is "default".

	   For more information	consult	the documentation for the FT_LOAD_*
	   libfreetype flags.

       shadowcolor
	   The color to	be used	for drawing a shadow behind the	drawn text.
	   For the syntax of this option, check	the "Color" section in the
	   ffmpeg-utils	manual.

	   The default value of	shadowcolor is "black".

       shadowx
       shadowy
	   The x and y offsets for the text shadow position with respect to
	   the position	of the text. They can be either	positive or negative
	   values. The default value for both is "0".

       start_number
	   The starting	frame number for the n/frame_num variable. The default
	   value is "0".

       tabsize
	   The size in number of spaces	to use for rendering the tab.  Default
	   value is 4.

       timecode
	   Set the initial timecode representation in "hh:mm:ss[:;.]ff"
	   format. It can be used with or without text parameter.
	   timecode_rate option	must be	specified.

       timecode_rate, rate, r
	   Set the timecode frame rate (timecode only).	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.

       reload
	   If set to 1,	the textfile will be reloaded before each frame.  Be
	   sure	to update it atomically, or it may be read partially, or even
	   fail.

       x
       y   The expressions which specify the offsets where text	will be	drawn
	   within the video frame. They	are relative to	the top/left border of
	   the output image.

	   The default value of	x and y	is "0".

	   See below for the list of accepted constants	and functions.

       The parameters for x and	y are expressions containing the following
       constants and functions:

       dar input display aspect	ratio, it is the same as (w / h) * sar

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For	example	for
	   the pixel format "yuv422p" hsub is 2	and vsub is 1.

       line_h, lh
	   the height of each text line

       main_h, h, H
	   the input height

       main_w, w, W
	   the input width

       max_glyph_a, ascent
	   the maximum distance	from the baseline to the highest/upper grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  It	is a positive value, due to the	grid's
	   orientation with the	Y axis upwards.

       max_glyph_d, descent
	   the maximum distance	from the baseline to the lowest	grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  This is a negative	value, due to the grid's
	   orientation,	with the Y axis	upwards.

       max_glyph_h
	   maximum glyph height, that is the maximum height for	all the	glyphs
	   contained in	the rendered text, it is equivalent to ascent -
	   descent.

       max_glyph_w
	   maximum glyph width,	that is	the maximum width for all the glyphs
	   contained in	the rendered text

       n   the number of input frame, starting from 0

       rand(min, max)
	   return a random number included between min and max

       sar The input sample aspect ratio.

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

       text_h, th
	   the height of the rendered text

       text_w, tw
	   the width of	the rendered text

       x
       y   the x and y offset coordinates where	the text is drawn.

	   These parameters allow the x	and y expressions to refer 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.

       pkt_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	strftime()
       sequences in the	provided text and expands them accordingly. Check the
       documentation of	strftime(). This feature is deprecated.

       If expansion is set to "none", the text is printed verbatim.

       If expansion is set to "normal" (which is the default), the following
       expansion mechanism is used.

       The backslash character \, followed by any character, always expands to
       the second character.

       Sequences of the	form "%{...}" are expanded. The	text between the
       braces is a function name, possibly followed by arguments separated by
       ':'.  If	the arguments contain special characters or delimiters (':' or
       '}'), they should be escaped.

       Note that they probably must also be escaped as the value for the text
       option in the filter argument string and	as the filter argument in the
       filtergraph description,	and possibly also for the shell, that makes up
       to four levels of escaping; using a text	file avoids these problems.

       The following functions are available:

       expr, e
	   The expression evaluation result.

	   It must take	one argument specifying	the expression to be
	   evaluated, which accepts the	same constants and functions as	the x
	   and y values. Note that not all constants should be used, for
	   example the text size is not	known when evaluating the expression,
	   so the constants text_w and text_h will have	an undefined value.

       expr_int_format,	eif
	   Evaluate the	expression's value and output as formatted integer.

	   The first argument is the expression	to be evaluated, just as for
	   the expr function.  The second argument specifies the output
	   format. Allowed values are x, X, d and u. They are treated exactly
	   as in the "printf" function.	 The third parameter is	optional and
	   sets	the number of positions	taken by the output.  It can be	used
	   to add padding with zeros from the left.

       gmtime
	   The time at which the filter	is running, expressed in UTC.  It can
	   accept an argument: a strftime() format string.

       localtime
	   The time at which the filter	is running, expressed in the local
	   time	zone.  It can accept an	argument: a strftime() format string.

       metadata
	   Frame metadata. Takes one or	two arguments.

	   The first argument is mandatory and specifies the metadata key.

	   The second argument is optional and specifies a default value, used
	   when	the metadata key is not	found or empty.

	   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() 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.

       Examples

       o   Draw	"Test Text" with font FreeSerif, using the default values for
	   the optional	parameters.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:	text='Test Text'"

       o   Draw	'Test Text' with font FreeSerif	of size	24 at position x=100
	   and y=50 (counting from the top-left	corner of the screen), text is
	   yellow with a red box around	it. Both the text and the box have an
	   opacity of 20%.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf:	text='Test Text':\
			     x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

	   Note	that the double	quotes are not necessary if spaces are not
	   used	within the parameter list.

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

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

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

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

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

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

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

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

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

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

       o   Show	text for 1 second every	3 seconds:

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

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

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

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

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

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

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

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

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

       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/>.

   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.

   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.

   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

       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.

   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

   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, Default is 4, can	be 3, 4, 5 or 6.  Actual size
	   of block in pixels is 2 to power of block, so by default block size
	   in pixels is	2^4 which is 16.

       overlap
	   Set block overlap. Default is 0.5. Allowed range is from 0.2	to
	   0.8.

       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.

       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". Default is "absolute".

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

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

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

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

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

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

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

       The filter accepts the following	options:

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

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

	   bff Assume bottom field first.

	   tff Assume top field	first.

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

	   Default value is auto.

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

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

	   Available values are:

	   pc  2-way matching (p/c)

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

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

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

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

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

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

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

	   Default value is pc_n.

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

	   Default value is 0 (disabled).

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

	   auto
	       Automatic (same value as	order).

	   bottom
	       Match from the bottom field.

	   top Match from the top field.

	   Default value is auto.

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

	   Default value is 1.

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

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

	   Default value is 12.0.

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

	   none
	       No final	matching based on combed scores.

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

	   full
	       Use combed scores all the time.

	   Default is sc.

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

	   none
	       No forced calculation.

	   pcn Force p/c/n calculations.

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

	   Default value is none.

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

	   Default value is 9.

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

	   Default value is 0.

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

	   Default value is 16.

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

	   Default value is 80.

       p/c/n/u/b meaning

       p/c/n

       We assume the following telecined stream:

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

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

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

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

	       Matches:		     c c n n c

	       Output stream:
			       T     1 2 3 4 4
			       B     1 2 3 4 4

       As a result of the field	matching, we can see that some frames get
       duplicated.  To perform a complete inverse telecine, you	need to	rely
       on a decimation filter after this operation. See	for instance the
       decimate	filter.

       The same	operation now matching from top	fields (field=top) looks like
       this:

	       Input stream:
			       T     1 2 2 3 4	 <-- matching reference
			       B     1 2 3 4 4

	       Matches:		     c c p p c

	       Output stream:
			       T     1 2 2 3 4
			       B     1 2 2 3 4

       In these	examples, we can see what p, c and n mean; basically, they
       refer to	the frame and field of the opposite parity:

       *<p matches the field of	the opposite parity in the previous frame>
       *<c matches the field of	the opposite parity in the current frame>
       *<n matches the field of	the opposite parity in the next	frame>

       u/b

       The u and b matching are	a bit special in the sense that	they match
       from the	opposite parity	flag. In the following examples, we assume
       that we are currently matching the 2nd frame (Top:2, bottom:2).
       According to the	match, a 'x' is	placed above and below each matched
       fields.

       With bottom matching (field=bottom):

	       Match:		c	  p	      n		 b	    u

				x	x		x	 x	    x
		 Top	      1	2 2	1 2 2	    1 2	2      1 2 2	  1 2 2
		 Bottom	      1	2 3	1 2 3	    1 2	3      1 2 3	  1 2 3
				x	  x	      x	       x	      x

	       Output frames:
				2	   1	      2		 2	    2
				2	   2	      2		 1	    3

       With top	matching (field=top):

	       Match:		c	  p	      n		 b	    u

				x	  x	      x	       x	      x
		 Top	      1	2 2	1 2 2	    1 2	2      1 2 2	  1 2 2
		 Bottom	      1	2 3	1 2 3	    1 2	3      1 2 3	  1 2 3
				x	x		x	 x	    x

	       Output frames:
				2	   2	      2		 1	    2
				2	   1	      3		 2	    2

       Examples

       Simple IVTC of a	top field first	telecined stream:

	       fieldmatch=order=tff:combmatch=none, decimate

       Advanced	IVTC, with fallback on yadif for still combed frames:

	       fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

   fieldorder
       Transform the field order of the	input video.

       It accepts the following	parameters:

       order
	   The output field order. Valid values	are tff	for top	field first or
	   bff for bottom field	first.

       The default value is tff.

       The transformation is done by shifting the picture content up or	down
       by one line, and	filling	the remaining line with	appropriate picture
       content.	 This method is	consistent with	most broadcast field order
       converters.

       If the input video is not flagged as being interlaced, or it is already
       flagged as being	of the required	output field order, then this filter
       does not	alter the incoming video.

       It is very useful when converting to or from PAL	DV material, which is
       bottom field first.

       For example:

	       ffmpeg -i in.vob	-vf "fieldorder=bff" out.dv

   fifo, afifo
       Buffer input images and send them when they are requested.

       It is mainly useful when	auto-inserted by the libavfilter framework.

       It does not take	parameters.

   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

	   fixed
	       fill pixels with	constant value

	   Default is smear.

       color
	   Set color for pixels	in fixed 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

       It accepts the following	options:

       object
	   Filepath of the object image, needs to be in	gray8.

       threshold
	   Detection threshold,	default	is 0.5.

       mipmaps
	   Number of mipmaps, default is 3.

       xmin, ymin, xmax, ymax
	   Specifies the rectangle in which to search.

       Examples

       o   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

   floodfill
       Flood area with values of same pixel components with another values.

       It accepts the following	options:

       x   Set pixel x coordinate.

       y   Set pixel y coordinate.

       s0  Set source #0 component value.

       s1  Set source #1 component value.

       s2  Set source #2 component value.

       s3  Set source #3 component value.

       d0  Set destination #0 component	value.

       d1  Set destination #1 component	value.

       d2  Set destination #2 component	value.

       d3  Set destination #3 component	value.

   format
       Convert the input video to one of the specified pixel formats.
       Libavfilter will	try to pick one	that is	suitable as input to the next
       filter.

       It accepts the following	parameters:

       pix_fmts
	   A '|'-separated list	of pixel format	names, such as
	   "pix_fmts=yuv420p|monow|rgb24".

       Examples

       o   Convert the input video to the yuv420p format

		   format=pix_fmts=yuv420p

	   Convert the input video to any of the formats in the	list

		   format=pix_fmts=yuv420p|yuv444p|yuv410p

   fps
       Convert the video to specified constant frame rate by duplicating or
       dropping	frames as necessary.

       It accepts the following	parameters:

       fps The desired output frame rate. The default is 25.

       start_time
	   Assume the first PTS	should be the given value, in seconds. This
	   allows for padding/trimming at the start of stream. By default, no
	   assumption is made about the	first frame's expected PTS, so no
	   padding or trimming is done.	 For example, this could be set	to 0
	   to pad the beginning	with duplicates	of the first frame if a	video
	   stream starts after the audio stream	or to trim any frames with a
	   negative PTS.

       round
	   Timestamp (PTS) rounding method.

	   Possible values are:

	   zero
	       round towards 0

	   inf round away from 0

	   down
	       round towards -infinity

	   up  round towards +infinity

	   near
	       round to	nearest

	   The default is "near".

       eof_action
	   Action performed when reading the last frame.

	   Possible values are:

	   round
	       Use same	timestamp rounding method as used for other frames.

	   pass
	       Pass through last frame if input	duration has not been reached
	       yet.

	   The default is "round".

       Alternatively, the options can be specified as a	flat string:
       fps[:start_time[:round]].

       See also	the setpts filter.

       Examples

       o   A typical usage in order to set the fps to 25:

		   fps=fps=25

       o   Sets	the fps	to 24, using abbreviation and rounding method to round
	   to nearest:

		   fps=fps=film:round=near

   framepack
       Pack two	different video	streams	into a stereoscopic video, setting
       proper metadata on supported codecs. The	two views should have the same
       size and	framerate and processing will stop when	the shorter video
       ends. Please note that you may conveniently adjust view properties with
       the scale and fps filters.

       It accepts the following	parameters:

       format
	   The desired packing format. Supported values	are:

	   sbs The views are next to each other	(default).

	   tab The views are on	top of each other.

	   lines
	       The views are packed by line.

	   columns
	       The views are packed by column.

	   frameseq
	       The views are temporally	interleaved.

       Some examples:

	       # Convert left and right	views into a frame-sequential video
	       ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

	       # Convert views into a side-by-side video with the same output resolution as the	input
	       ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

   framerate
       Change the frame	rate by	interpolating new video	output frames from the
       source frames.

       This filter is not designed to function correctly with interlaced
       media. If you wish to change the	frame rate of interlaced media then
       you are required	to deinterlace before this filter and re-interlace
       after this filter.

       A description of	the accepted options follows.

       fps Specify the output frames per second. This option can also be
	   specified as	a value	alone. The default is 50.

       interp_start
	   Specify the start of	a range	where the output frame will be created
	   as a	linear interpolation of	two frames. The	range is [0-255], the
	   default is 15.

       interp_end
	   Specify the end of a	range where the	output frame will be created
	   as a	linear interpolation of	two frames. The	range is [0-255], the
	   default is 240.

       scene
	   Specify the level at	which a	scene change is	detected as a value
	   between 0 and 100 to	indicate a new scene; a	low value reflects a
	   low probability for the current frame to introduce a	new scene,
	   while a higher value	means the current frame	is more	likely to be
	   one.	 The default is	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>

   fspp
       Apply fast and simple postprocessing. It	is a faster version of spp.

       It splits (I)DCT	into horizontal/vertical passes. Unlike	the simple
       post- processing	filter,	one of them is performed once per block, not
       per pixel.  This	allows for much	higher speed.

       The filter accepts the following	options:

       quality
	   Set quality.	This option defines the	number of levels for
	   averaging. It accepts an integer in the range 4-5. Default value is
	   4.

       qp  Force a constant quantization parameter. It accepts an integer in
	   range 0-63.	If not set, the	filter will use	the QP from the	video
	   stream (if available).

       strength
	   Set filter strength.	It accepts an integer in range -15 to 32.
	   Lower values	mean more details but also more	artifacts, while
	   higher values make the image	smoother but also blurrier. Default
	   value is 0 X	PSNR optimal.

       use_bframe_qp
	   Enable the use of the QP from the B-Frames if set to	1. Using this
	   option may cause flicker since the B-Frames have often larger QP.
	   Default is 0	(not enabled).

   gblur
       Apply Gaussian blur filter.

       The filter accepts the following	options:

       sigma
	   Set horizontal sigma, standard deviation of Gaussian	blur. Default
	   is 0.5.

       steps
	   Set number of steps for Gaussian approximation. 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 luminance expression.

       cb_expr,	cb
	   Set the chrominance blue expression.

       cr_expr,	cr
	   Set the chrominance red expression.

       alpha_expr, a
	   Set the alpha expression.

       red_expr, r
	   Set the red expression.

       green_expr, g
	   Set the green expression.

       blue_expr, b
	   Set the blue	expression.

       The colorspace is selected according to the specified options. If one
       of the lum_expr,	cb_expr, or cr_expr options is specified, the filter
       will automatically select a YCbCr colorspace. If	one of the red_expr,
       green_expr, or blue_expr	options	is specified, it will select an	RGB
       colorspace.

       If one of the chrominance expression is not defined, it falls back on
       the other one. If no alpha expression is	specified it will evaluate to
       opaque value.  If none of chrominance expressions are specified,	they
       will evaluate to	the luminance expression.

       The expressions can use the following variables and functions:

       N   The sequential number of the	filtered frame,	starting from 0.

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height	of the image.

       SW
       SH  Width and height scale depending on the currently filtered plane.
	   It is the ratio between the corresponding luma plane	number of
	   pixels and the current plane	ones. E.g. for YUV4:2:0	the values are
	   "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

       T   Time	of the current frame, expressed	in seconds.

       p(x, y)
	   Return the value of the pixel at location (x,y) of the current
	   plane.

       lum(x, y)
	   Return the value of the pixel at location (x,y) of the luminance
	   plane.

       cb(x, y)
	   Return the value of the pixel at location (x,y) of the blue-
	   difference chroma plane. Return 0 if	there is no such plane.

       cr(x, y)
	   Return the value of the pixel at location (x,y) of the red-
	   difference chroma plane. Return 0 if	there is no such plane.

       r(x, y)
       g(x, y)
       b(x, y)
	   Return the value of the pixel at location (x,y) of the
	   red/green/blue component. Return 0 if there is no such component.

       alpha(x,	y)
	   Return the value of the pixel at location (x,y) of the alpha	plane.
	   Return 0 if there is	no such	plane.

       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, can	be fulll or compact.  In compact mode only
	   filters with	some queued frames have	displayed stats.

       flags, f
	   Set flags which enable which	stats are shown	in video.

	   Available values for	flags are:

	   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.

	   pts Display current filtered	frame pts.

	   time
	       Display current filtered	frame time.

	   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.

       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.

   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

   haldclut
       Apply a Hald CLUT to a video stream.

       First input is the video	stream to process, and second one is the Hald
       CLUT.  The Hald CLUT input can be a simple picture or a complete	video
       stream.

       The filter accepts the following	options:

       shortest
	   Force termination when the shortest input terminates. Default is 0.

       repeatlast
	   Continue applying the last CLUT after the end of the	stream.	A
	   value of 0 disable the filter after the last	frame of the CLUT is
	   reached.  Default is	1.

       "haldclut" also has the same interpolation options as lut3d (both
       filters share the same internals).

       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>.

       Workflow	examples

       Hald CLUT video stream

       Generate	an identity Hald CLUT stream altered with various effects:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

       Note: make sure you use a lossless codec.

       Then use	it with	"haldclut" to apply it on some random stream:

	       ffmpeg -f lavfi -i mandelbrot -i	clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

       The Hald	CLUT will be applied to	the 10 first seconds (duration of
       clut.nut), then the latest picture of that CLUT stream will be applied
       to the remaining	frames of the "mandelbrot" stream.

       Hald CLUT with preview

       A Hald CLUT is supposed to be a squared image of	"Level*Level*Level" by
       "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select
       the biggest possible square starting at the top left of the picture.
       The remaining padding pixels (bottom or right) will be ignored. This
       area can	be used	to add a preview of the	Hald CLUT.

       Typically, the following	generated Hald CLUT will be supported by the
       "haldclut" filter:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
		  pad=iw+320 [padded_clut];
		  smptebars=s=320x256, split [a][b];
		  [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
		  [main][b] overlay=W-320" -frames:v 1 clut.png

       It contains the original	and a preview of the effect of the CLUT: SMPTE
       color bars are displayed	on the right-top, and below the	same color
       bars processed by the color changes.

       Then, the effect	of this	Hald CLUT can be visualized with:

	       ffplay input.mkv	-vf "movie=clut.png, [in] haldclut"

   hflip
       Flip the	input video horizontally.

       For example, to horizontally flip the input video with ffmpeg:

	       ffmpeg -i in.avi	-vf "hflip" out.avi

   histeq
       This filter applies a global color histogram equalization on a per-
       frame basis.

       It can be used to correct video that has	a compressed range of pixel
       intensities.  The filter	redistributes the pixel	intensities to
       equalize	their distribution across the intensity	range. It may be
       viewed as an "automatically adjusting contrast filter". This filter is
       useful only for correcting degraded or poorly captured source video.

       The filter accepts the following	options:

       strength
	   Determine the amount	of equalization	to be applied.	As the
	   strength is reduced,	the distribution of pixel intensities more-
	   and-more approaches that of the input frame.	The value must be a
	   float number	in the range [0,1] and defaults	to 0.200.

       intensity
	   Set the maximum intensity that can generated	and scale the output
	   values appropriately.  The strength should be set as	desired	and
	   then	the intensity can be limited if	needed to avoid	washing-out.
	   The value must be a float number in the range [0,1] and defaults to
	   0.210.

       antibanding
	   Set the antibanding level. If enabled the filter will randomly vary
	   the luminance of output pixels by a small amount to avoid banding
	   of the histogram. Possible values are "none", "weak"	or "strong".
	   It defaults to "none".

   histogram
       Compute and draw	a color	distribution histogram for the input video.

       The computed histogram is a representation of the color component
       distribution in an image.

       Standard	histogram displays the color components	distribution in	an
       image.  Displays	color graph for	each color component. Shows
       distribution of the Y, U, V, A or R, G, B components, depending on
       input format, in	the current frame. Below each graph a color component
       scale meter is shown.

       The filter accepts the following	options:

       level_height
	   Set height of level.	Default	value is 200.  Allowed range is	[50,
	   2048].

       scale_height
	   Set height of color scale. Default value is 12.  Allowed range is
	   [0, 40].

       display_mode
	   Set display mode.  It accepts the following values:

	   stack
	       Per color component graphs are placed below each	other.

	   parade
	       Per color component graphs are placed side by side.

	   overlay
	       Presents	information identical to that in the "parade", except
	       that the	graphs representing color components are superimposed
	       directly	over one another.

	   Default is "stack".

       levels_mode
	   Set mode. Can be either "linear", or	"logarithmic".	Default	is
	   "linear".

       components
	   Set what color components to	display.  Default is 7.

       fgopacity
	   Set foreground opacity. Default is 0.7.

       bgopacity
	   Set background opacity. Default is 0.5.

       Examples

       o   Calculate and draw histogram:

		   ffplay -i input -vf histogram

   hqdn3d
       This is a high precision/quality	3d denoise filter. It aims to reduce
       image noise, producing smooth images and	making still images really
       still. It should	enhance	compressibility.

       It accepts the following	optional parameters:

       luma_spatial
	   A non-negative floating point number	which specifies	spatial	luma
	   strength.  It defaults to 4.0.

       chroma_spatial
	   A non-negative floating point number	which specifies	spatial	chroma
	   strength.  It defaults to 3.0*luma_spatial/4.0.

       luma_tmp
	   A floating point number which specifies luma	temporal strength. It
	   defaults to 6.0*luma_spatial/4.0.

       chroma_tmp
	   A floating point number which specifies chroma temporal strength.
	   It defaults to luma_tmp*chroma_spatial/luma_spatial.

       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.

   hue
       Modify the hue and/or the saturation of the input.

       It accepts the following	parameters:

       h   Specify the hue angle as a number of	degrees. It accepts an
	   expression, and defaults to "0".

       s   Specify the saturation in the [-10,10] range. It accepts an
	   expression and defaults to "1".

       H   Specify the hue angle as a number of	radians. It accepts an
	   expression, and defaults to "0".

       b   Specify the brightness in the [-10,10] range. It accepts an
	   expression and defaults to "0".

       h and H are mutually exclusive, and can't be specified at the same
       time.

       The b, h, H and s option	values are expressions containing the
       following constants:

       n   frame count of the input frame starting from	0

       pts presentation	timestamp of the input frame expressed in time base
	   units

       r   frame rate of the input video, NAN if the input frame rate is
	   unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is
	   unknown

       tb  time	base of	the input video

       Examples

       o   Set the hue to 90 degrees and the saturation	to 1.0:

		   hue=h=90:s=1

       o   Same	command	but expressing the hue in radians:

		   hue=H=PI/2:s=1

       o   Rotate hue and make the saturation swing between 0 and 2 over a
	   period of 1 second:

		   hue="H=2*PI*t: s=sin(2*PI*t)+1"

       o   Apply a 3 seconds saturation	fade-in	effect starting	at 0:

		   hue="s=min(t/3\,1)"

	   The general fade-in expression can be written as:

		   hue="s=min(0\, max((t-START)/DURATION\, 1))"

       o   Apply a 3 seconds saturation	fade-out effect	starting at 5 seconds:

		   hue="s=max(0\, min(1\, (8-t)/3))"

	   The general fade-out	expression can be written as:

		   hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

       Commands

       This filter supports the	following commands:

       b
       s
       h
       H   Modify the hue and/or the saturation	and/or brightness of the input
	   video.  The command accepts the same	syntax of the corresponding
	   option.

	   If the specified expression is not valid, it	is kept	at its current
	   value.

   hysteresis
       Grow first stream into second stream by connecting components.  This
       makes it	possible to build more robust edge masks.

       This filter accepts the following options:

       planes
	   Set which planes will be processed as bitmap, unprocessed planes
	   will	be copied from first stream.  By default value 0xf, all	planes
	   will	be processed.

       threshold
	   Set threshold which is used in filtering. If	pixel component	value
	   is higher than this value filter algorithm for connecting
	   components is activated.  By	default	value is 0.

       The "hysteresis"	filter also supports the framesync options.

   idet
       Detect video interlacing	type.

       This filter tries to detect if the input	frames are interlaced,
       progressive, top	or bottom field	first. It will also try	to detect
       fields that are repeated	between	adjacent frames	(a sign	of telecine).

       Single frame detection considers	only immediately adjacent frames when
       classifying each	frame.	Multiple frame detection incorporates the
       classification history of previous frames.

       The filter will log these metadata values:

       single.current_frame
	   Detected type of current frame using	single-frame detection.	One
	   of: ``tff'' (top field first), ``bff'' (bottom field	first),
	   ``progressive'', or ``undetermined''

       single.tff
	   Cumulative number of	frames detected	as top field first using
	   single-frame	detection.

       multiple.tff
	   Cumulative number of	frames detected	as top field first using
	   multiple-frame detection.

       single.bff
	   Cumulative number of	frames detected	as bottom field	first using
	   single-frame	detection.

       multiple.current_frame
	   Detected type of current frame using	multiple-frame detection. One
	   of: ``tff'' (top field first), ``bff'' (bottom field	first),
	   ``progressive'', or ``undetermined''

       multiple.bff
	   Cumulative number of	frames detected	as bottom field	first using
	   multiple-frame detection.

       single.progressive
	   Cumulative number of	frames detected	as progressive using single-
	   frame detection.

       multiple.progressive
	   Cumulative number of	frames detected	as progressive using multiple-
	   frame detection.

       single.undetermined
	   Cumulative number of	frames that could not be classified using
	   single-frame	detection.

       multiple.undetermined
	   Cumulative number of	frames that could not be classified using
	   multiple-frame detection.

       repeated.current_frame
	   Which field in the current frame is repeated	from the last. One of
	   ``neither'',	``top'', or ``bottom''.

       repeated.neither
	   Cumulative number of	frames with no repeated	field.

       repeated.top
	   Cumulative number of	frames with the	top field repeated from	the
	   previous frame's top	field.

       repeated.bottom
	   Cumulative number of	frames with the	bottom field repeated from the
	   previous frame's bottom field.

       The filter accepts the following	options:

       intl_thres
	   Set interlacing threshold.

       prog_thres
	   Set progressive threshold.

       rep_thres
	   Threshold for repeated field	detection.

       half_life
	   Number of frames after which	a given	frame's	contribution to	the
	   statistics is halved	(i.e., it contributes only 0.5 to its
	   classification). The	default	of 0 means that	all frames seen	are
	   given full weight of	1.0 forever.

       analyze_interlaced_flag
	   When	this is	not 0 then idet	will use the specified number of
	   frames to determine if the interlaced flag is accurate, it will not
	   count undetermined frames.  If the flag is found to be accurate it
	   will	be used	without	any further computations, if it	is found to be
	   inaccurate it will be cleared without any further computations.
	   This	allows inserting the idet filter as a low computational	method
	   to clean up the interlaced flag

   il
       Deinterleave or interleave fields.

       This filter allows one to process interlaced images fields without
       deinterlacing them. Deinterleaving splits the input frame into 2	fields
       (so called half pictures). Odd lines are	moved to the top half of the
       output image, even lines	to the bottom half.  You can process (filter)
       them independently and then re-interleave them.

       The filter accepts the following	options:

       luma_mode, l
       chroma_mode, c
       alpha_mode, a
	   Available values for	luma_mode, chroma_mode and alpha_mode are:

	   none
	       Do nothing.

	   deinterleave, d
	       Deinterleave fields, placing one	above the other.

	   interleave, i
	       Interleave fields. Reverse the effect of	deinterleaving.

	   Default value is "none".

       luma_swap, ls
       chroma_swap, cs
       alpha_swap, as
	   Swap	luma/chroma/alpha fields. Exchange even	& odd lines. Default
	   value is 0.

       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

   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.

   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.

       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.

   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.

       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.

       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

   libvmaf
       Obtain the VMAF (Video Multi-Method Assessment Fusion) score between
       two input videos.

       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	--enable-version3".  If	no model path is specified it
       uses the	default	model: "vmaf_v0.6.1.pkl".

       The filter has following	options:

       model_path
	   Set the model path which is to be used for SVM.  Default value:
	   "/usr/local/share/model/vmaf_v0.6.1.pkl"

       log_path
	   Set the file	path to	be used	to store logs.

       log_fmt
	   Set the format of the log file (xml or json).

       enable_transform
	   This	option can enable/disable the "score_transform"	applied	to the
	   final predicted VMAF	score, if you have specified score_transform
	   option in the input parameter file passed to	"run_vmaf_training.py"
	   Default value: "false"

       phone_model
	   Invokes the phone model which will generate VMAF scores higher than
	   in the regular model, which is more suitable	for laptop, TV,	etc.
	   viewing conditions.	Default	value: "false"

       psnr
	   Enables computing psnr along	with vmaf.  Default value: "false"

       ssim
	   Enables computing ssim along	with vmaf.  Default value: "false"

       ms_ssim
	   Enables computing ms_ssim along with	vmaf.  Default value: "false"

       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 computing vmaf.  Default
	   value: 0, which makes use of	all available logical processors.

       n_subsample
	   Set interval	for frame subsampling used when	computing vmaf.
	   Default value: 1

       enable_conf_interval
	   Enables confidence interval.	 Default value:	"false"

       This filter also	supports the framesync options.

       Examples

       o   On the below	examples the input file	main.mpg being processed is
	   compared with the reference file ref.mpg.

		   ffmpeg -i main.mpg -i ref.mpg -lavfi	libvmaf	-f null	-

       o   Example with	options:

		   ffmpeg -i main.mpg -i ref.mpg -lavfi	libvmaf="psnr=1:log_fmt=json" -f null -

       o   Example with	options	and 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]libvmaf=psnr=1:log_fmt=json" -f null -

   limiter
       Limits the pixel	components values to the specified range [min, max].

       The filter accepts the following	options:

       min Lower bound.	Defaults to the	lowest allowed value for the input.

       max Upper bound.	Defaults to the	highest	allowed	value for the input.

       planes
	   Specify which planes	will be	processed. Defaults to all available.

   loop
       Loop video frames.

       The filter accepts the following	options:

       loop
	   Set the number of loops. 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.

       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.

   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.

   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/luminance component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

       Each of them specifies the expression to	use for	computing the lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c*	options	depends	on the
       format in input.

       The lut filter requires either YUV or RGB pixel formats in input,
       lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       clipval
	   The input value, clipped to the minval-maxval range.

       maxval
	   The maximum value for the pixel component.

       minval
	   The minimum value for the pixel component.

       negval
	   The negated value for the pixel component value, clipped to the
	   minval-maxval range;	it corresponds to the expression
	   "maxval-clipval+minval".

       clip(val)
	   The computed	value in val, clipped to the minval-maxval range.

       gammaval(gamma)
	   The computed	gamma correction value of the pixel component value,
	   clipped to the minval-maxval	range. It corresponds to the
	   expression
	   "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "val".

       Examples

       o   Negate input	video:

		   lutrgb="r=maxval+minval-val:g=