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

NAME
       ffmpeg - ffmpeg media converter

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

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

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

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

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

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

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

       Some simple examples follow.

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

		   ffmpeg -i input.avi output.mp4

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

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

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

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

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

		   ffmpeg -r 1 -i input.m2v -r 24 output.mp4

       The format option may be needed for raw input files.

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

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

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

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

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

       Simple filtergraphs

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

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

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

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

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

       Complex filtergraphs

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

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

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

       The -lavfi option is equivalent to -filter_complex.

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

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

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

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

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:

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

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

       •   for subtitles, it is the first subtitle stream 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 named  filter_name.
	       Use the -filters option to get a list of all filters.

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

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

       -version
	   Show version.

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

       -formats
	   Show available formats (including devices).

       -demuxers
	   Show available demuxers.

       -muxers
	   Show available muxers.

       -devices
	   Show available devices.

       -codecs
	   Show all codecs known to libavcodec.

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

       -decoders
	   Show available decoders.

       -encoders
	   Show all available encoders.

       -bsfs
	   Show available bitstream filters.

       -protocols
	   Show available protocols.

       -filters
	   Show available libavfilter filters.

       -pix_fmts
	   Show available pixel formats.

       -sample_fmts
	   Show available sample formats.

       -layouts
	   Show channel names and standard channel layouts.

       -dispositions
	   Show stream dispositions.

       -colors
	   Show recognized color names.

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

		   ffmpeg -sources pulse,server=192.168.0.4

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

		   ffmpeg -sinks pulse,server=192.168.0.4

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

	   The optional flags prefix can consist of the following values:

	   repeat
	       Indicates that repeated log output should not be compressed  to
	       the  first  line	 and  the "Last message repeated n times" line
	       will be omitted.

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

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

	   loglevel is a string or a number containing one  of	the  following
	   values:

	   quiet, -8
	       Show nothing at all; be silent.

	   panic, 0
	       Only  show  fatal errors which could lead the process to crash,
	       such as an assertion failure. This is not  currently  used  for
	       anything.

	   fatal, 8
	       Only  show  fatal  errors.  These  are  errors  after which the
	       process absolutely cannot continue.

	   error, 16
	       Show all errors, including ones which can be recovered from.

	   warning, 24
	       Show all warnings and errors. Any message related  to  possibly
	       incorrect or unexpected events will be shown.

	   info, 32
	       Show   informative  messages  during  processing.  This	is  in
	       addition to warnings and errors. This is the default value.

	   verbose, 40
	       Same as "info", except more verbose.

	   debug, 48
	       Show everything, including debugging information.

	   trace, 56

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

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

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

		   ffmpeg [...] -loglevel +repeat

	   By default the program logs to stderr. If coloring is supported  by
	   the	terminal,  colors  are	used  to mark errors and warnings. Log
	   coloring  can  be  disabled	setting	  the	environment   variable
	   AV_LOG_FORCE_NOCOLOR,  or  can  be  forced  setting the environment
	   variable AV_LOG_FORCE_COLOR.

       -report
	   Dump	 full  command	line  and  log	output	 to   a	  file	 named
	   "program-YYYYMMDD-HHMMSS.log"  in the current directory.  This file
	   can be useful for bug reports.  It also implies "-loglevel debug".

	   Setting the environment variable FFREPORT to any value has the same
	   effect. If the value is a ':'-separated key=value  sequence,	 these
	   options  will  affect  the report; option values must be escaped if
	   they contain special characters or the options delimiter  ':'  (see
	   the ``Quoting and escaping'' section in the ffmpeg-utils manual).

	   The following options are recognized:

	   file
	       set  the file name to use for the report; %p is expanded to the
	       name of the program, %t is expanded to  a  timestamp,  "%%"  is
	       expanded to a plain "%"

	   level
	       set  the	 log  verbosity	 level	using  a  numerical value (see
	       "-loglevel").

	   For example, to output a report to a file named ffreport.log	 using
	   a log level of 32 (alias for log level "info"):

		   FFREPORT=file=ffreport.log:level=32 ffmpeg -i input output

	   Errors  in parsing the environment variable are not fatal, and will
	   not appear in the report.

       -hide_banner
	   Suppress printing banner.

	   All FFmpeg tools will  normally  show  a  copyright	notice,	 build
	   options  and	 library versions. This option can be used to suppress
	   printing this information.

       -cpuflags flags (global)
	   Allows setting and clearing cpu flags. This option is intended  for
	   testing. Do not use it unless you know what you're doing.

		   ffmpeg -cpuflags -sse+mmx ...
		   ffmpeg -cpuflags mmx ...
		   ffmpeg -cpuflags 0 ...

	   Possible flags for this option are:

	   x86
	       mmx
	       mmxext
	       sse
	       sse2
	       sse2slow
	       sse3
	       sse3slow
	       ssse3
	       atom
	       sse4.1
	       sse4.2
	       avx
	       avx2
	       xop
	       fma3
	       fma4
	       3dnow
	       3dnowext
	       bmi1
	       bmi2
	       cmov
	   ARM
	       armv5te
	       armv6
	       armv6t2
	       vfp
	       vfpv3
	       neon
	       setend
	   AArch64
	       armv8
	       vfp
	       neon
	   PowerPC
	       altivec
	   Specific Processors
	       pentium2
	       pentium3
	       pentium4
	       k6
	       k62
	       athlon
	       athlonxp
	       k8
       -cpucount count (global)
	   Override  detection	of  CPU	 count.	 This  option  is intended for
	   testing. Do not use it unless you know what you're doing.

		   ffmpeg -cpucount 2

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

   AVOptions
       These options are provided directly by the libavformat, libavdevice and
       libavcodec  libraries.  To see the list of available AVOptions, use the
       -help option. They are separated into two categories:

       generic
	   These options can be	 set  for  any	container,  codec  or  device.
	   Generic  options  are  listed  under	 AVFormatContext  options  for
	   containers/devices and under AVCodecContext options for codecs.

       private
	   These options are specific to the given container, device or codec.
	   Private   options   are   listed    under	their	 corresponding
	   containers/devices/codecs.

       For  example to write an ID3v2.3 header instead of a default ID3v2.4 to
       an MP3 file, use the id3v2_version private option of the MP3 muxer:

	       ffmpeg -i input.flac -id3v2_version 3 out.mp3

       All codec AVOptions are per-stream, and thus a stream specifier	should
       be attached to them:

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

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

       Note: the -nooption syntax cannot be used for  boolean  AVOptions,  use
       -option 0/-option 1.

       Note:  the  old	undocumented way of specifying per-stream AVOptions by
       prepending v/a/s to the options	name  is  now  obsolete	 and  will  be
       removed soon.

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

       -i url (input)
	   input file url

       -y (global)
	   Overwrite output files without asking.

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

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

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

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

	   For example

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	   Default value is -1.

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

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

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

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

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

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

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

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

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

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

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

	   To set the language of the first audio stream:

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

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

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

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

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

	   The "-dispositions" option lists the known dispositions.

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

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

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

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

	   To add an embedded cover/thumbnail:

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

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

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

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

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

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

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

	   The parameters set for each target are as follows.

	   VCD

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

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

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

	   SVCD

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

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

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

	   DVD

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

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

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

	   DV

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

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

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

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

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

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

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

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

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

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

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

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

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

       -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.

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

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

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

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

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

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

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

	   The update period is set using "-stats_period".

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

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

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

	   See also the option "-fdebug ts".

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

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

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

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

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

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

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

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

		   ffmpeg -dump_attachment:t "" -i INPUT

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

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

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

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

	   As an output option:

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	   See the "-display_rotation" option for more details.

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

	   See the "-display_rotation" option for more details.

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

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

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

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

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

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

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

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

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

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

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

       -sws_flags flags (input/output)
	   Set	default flags for the libswscale library. These flags are used
	   by automatically inserted "scale" filters and those	within	simple
	   filtergraphs, if not overridden within the filtergraph definition.

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

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

       -psnr
	   Calculate  PSNR  of	compressed  frames. This option is deprecated,
	   pass the PSNR flag to the encoder instead, using "-flags +psnr".

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

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

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

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

       -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.	In cases where this particular source frame has to  be
	       dropped, enforce the next available frame to become a key frame
	       instead.

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

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

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

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

	   cuda
	       device is the number of the CUDA device.

	       The following options are recognized:

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

	       Examples:

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

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

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

	   d3d11va
	       device is the number of the Direct3D 11 display adapter.

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

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

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

	       auto
	       sw
	       hw
	       auto_any
	       hw_any
	       hw2
	       hw3
	       hw4

	       If  not	specified,  auto_any  is  used.	  (Note that it may be
	       easier to achieve the desired result for QSV  by	 creating  the
	       platform-appropriate  subdevice (dxva2 or d3d11va or vaapi) and
	       then deriving a QSV device from that.)

	       Alternatively,  child_device_type  helps	 to  choose  platform-
	       appropriate  subdevice  type.   On  Windows  d3d11va is used as
	       default subdevice type.

	       Examples:

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

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

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

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

	       The strings usable as filters are:

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

	       The indices and filters must together uniquely select a device.

	       Examples:

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

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

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

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

	       The following options are recognized:

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

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

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

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

	       Examples:

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

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

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

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

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

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

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

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

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

	   auto
	       Automatically select the hardware acceleration method.

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

	   dxva2
	       Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

	   d3d11va
	       Use D3D11VA (DirectX Video Acceleration) hardware acceleration.

	   vaapi
	       Use VAAPI (Video Acceleration API) hardware acceleration.

	   qsv Use  the	 Intel	QuickSync   Video   acceleration   for	 video
	       transcoding.

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

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

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

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

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

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

       -hwaccels
	   List all hardware acceleration components enabled in this build  of
	   ffmpeg.   Actual  runtime  availability depends on the hardware and
	   its suitable driver being installed.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

	   Examples:

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

		       ffmpeg -i INPUT -map 0 output

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

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

	       will map the second input  stream  in  INPUT  to	 the  (single)
	       output stream in out.wav.

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

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

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

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

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

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

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

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

	   map by language
	       To pick the English audio stream:

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

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

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

       -map_channel
       [input_file_id.stream_specifier.channel_id|-1][?][:output_file_id.stream_specifier]
	   This option is deprecated and will be removed. It can  be  replaced
	   by  the  pan	 filter.  In  some  cases it may be easier to use some
	   combination of the channelsplit, channelmap, or amerge filters.

	   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.

       -readrate speed (input)
	   Limit input read speed.

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

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

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

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

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

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

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

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

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

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

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

	   auto (-1)
	       Chooses	between	 cfr  and vfr depending on muxer capabilities.
	       This is the default method.

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

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

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

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

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

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

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

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

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

	   1   Use the demuxer timebase.

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

	   0   Use the decoder timebase.

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

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

	   Default value is -1.

       -enc_time_base[:stream_specifier] timebase (output,per-stream)
	   Set	the encoder timebase. timebase can assume one of the following
	   values:

	   0   Assign a default value according to the media type.

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

	   demux
	       Use the timebase from the demuxer.

	   filter
	       Use the timebase from the filtergraph.

	   a positive number
	       Use the provided number as the timebase.

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

	   Default value is 0.

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

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

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

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

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

	   The default value is 10 seconds.

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

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

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

	   The default value is 10.

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

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

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

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

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

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

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

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

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

       -bsf[:stream_specifier] bitstream_filters (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

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

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

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

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

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

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

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

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

       -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/output)
	   For	input,	this  option sets the maximum number of queued packets
	   when reading from the file or device. With low latency / high  rate
	   live	 streams,  packets  may be discarded if they are not read in a
	   timely manner; setting  this	 value	can  force  ffmpeg  to	use  a
	   separate  input  thread and read packets as soon as they arrive. By
	   default ffmpeg only does this if multiple inputs are specified.

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

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

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

	   none
	       Discard no frame.

	   default
	       Default, which discards no frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   all Discard all frames.

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

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

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

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

       -xerror (global)
	   Stop and exit on error

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

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

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

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

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

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

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

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

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

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

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

	   fidx
	       Index of the output file.

	   sidx
	       Index of the output stream in the file.

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

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

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

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

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

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

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

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

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

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

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

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

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

	   br (packet)
	       Current bitrate in bits per second. Post-encoding only.

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

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

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

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

	   The default format strings are:

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

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

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

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

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

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

       ffpreset files

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

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

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

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

       avpreset files

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

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

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

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

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

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

       With version 1 the format is:

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

       With version 2 the format is:

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

       The value corresponding to each key is described below:

       avg_br
	   average bitrate expressed in Kbits/s

       br  bitrate expressed in Kbits/s

       frame
	   number of encoded frame

       out out file index

       PSNR
	   Peak Signal to Noise Ratio

       q   quality of the frame

       f_size
	   encoded packet size expressed as number of bytes

       s_size
	   stream size expressed in KiB

       st  out file stream index

       time
	   time of the packet

       type
	   picture type

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

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

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

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

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

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

   X11 grabbing
       Grab the X11 display with ffmpeg via

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

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

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

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

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

       Examples:

       •   You can use YUV files as input:

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

	   It will use the files:

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

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

       •   You can input from a raw YUV420P file:

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

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

       •   You can output to a raw YUV420P file:

		   ffmpeg -i mydivx.avi hugefile.yuv

       •   You can set several input files and output files:

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

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

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

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

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

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

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

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

       •   You can transcode decrypted VOBs:

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

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

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

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

	   For extracting images from a video:

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

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

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

	   For creating a video from many images:

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

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

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

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

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

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

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

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

       •   To force CBR video output:

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

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

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

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

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

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

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

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

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

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

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

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

       Examples

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

		   Crime d\'Amour

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

		   'Crime d'\''Amour'

       •   Include leading or trailing whitespaces using quoting:

		   '  this string starts and ends with whitespaces  '

       •   Escaping and quoting can be mixed together:

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

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

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

   Date
       The accepted syntax is:

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

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

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

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

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

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

       or

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

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

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

       0.2 0.2 seconds

       200ms
	   200 milliseconds, that's 0.2s

       200000us
	   200000 microseconds, that's 0.2s

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

       23.189
	   23.189 seconds

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

       The following abbreviations are recognized:

       ntsc
	   720x480

       pal 720x576

       qntsc
	   352x240

       qpal
	   352x288

       sntsc
	   640x480

       spal
	   768x576

       film
	   352x240

       ntsc-film
	   352x240

       sqcif
	   128x96

       qcif
	   176x144

       cif 352x288

       4cif
	   704x576

       16cif
	   1408x1152

       qqvga
	   160x120

       qvga
	   320x240

       vga 640x480

       svga
	   800x600

       xga 1024x768

       uxga
	   1600x1200

       qxga
	   2048x1536

       sxga
	   1280x1024

       qsxga
	   2560x2048

       hsxga
	   5120x4096

       wvga
	   852x480

       wxga
	   1366x768

       wsxga
	   1600x1024

       wuxga
	   1920x1200

       woxga
	   2560x1600

       wqsxga
	   3200x2048

       wquxga
	   3840x2400

       whsxga
	   6400x4096

       whuxga
	   7680x4800

       cga 320x200

       ega 640x350

       hd480
	   852x480

       hd720
	   1280x720

       hd1080
	   1920x1080

       2k  2048x1080

       2kflat
	   1998x1080

       2kscope
	   2048x858

       4k  4096x2160

       4kflat
	   3996x2160

       4kscope
	   4096x1716

       nhd 640x360

       hqvga
	   240x160

       wqvga
	   400x240

       fwqvga
	   432x240

       hvga
	   480x320

       qhd 960x540

       2kdci
	   2048x1080

       4kdci
	   4096x2160

       uhd2160
	   3840x2160

       uhd4320
	   7680x4320

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

       The following abbreviations are recognized:

       ntsc
	   30000/1001

       pal 25/1

       qntsc
	   30000/1001

       qpal
	   25/1

       sntsc
	   30000/1001

       spal
	   25/1

       film
	   24/1

       ntsc-film
	   24000/1001

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

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

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

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

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

       The string random will result in a random color.

       The following names of colors are recognized:

       AliceBlue
	   0xF0F8FF

       AntiqueWhite
	   0xFAEBD7

       Aqua
	   0x00FFFF

       Aquamarine
	   0x7FFFD4

       Azure
	   0xF0FFFF

       Beige
	   0xF5F5DC

       Bisque
	   0xFFE4C4

       Black
	   0x000000

       BlanchedAlmond
	   0xFFEBCD

       Blue
	   0x0000FF

       BlueViolet
	   0x8A2BE2

       Brown
	   0xA52A2A

       BurlyWood
	   0xDEB887

       CadetBlue
	   0x5F9EA0

       Chartreuse
	   0x7FFF00

       Chocolate
	   0xD2691E

       Coral
	   0xFF7F50

       CornflowerBlue
	   0x6495ED

       Cornsilk
	   0xFFF8DC

       Crimson
	   0xDC143C

       Cyan
	   0x00FFFF

       DarkBlue
	   0x00008B

       DarkCyan
	   0x008B8B

       DarkGoldenRod
	   0xB8860B

       DarkGray
	   0xA9A9A9

       DarkGreen
	   0x006400

       DarkKhaki
	   0xBDB76B

       DarkMagenta
	   0x8B008B

       DarkOliveGreen
	   0x556B2F

       Darkorange
	   0xFF8C00

       DarkOrchid
	   0x9932CC

       DarkRed
	   0x8B0000

       DarkSalmon
	   0xE9967A

       DarkSeaGreen
	   0x8FBC8F

       DarkSlateBlue
	   0x483D8B

       DarkSlateGray
	   0x2F4F4F

       DarkTurquoise
	   0x00CED1

       DarkViolet
	   0x9400D3

       DeepPink
	   0xFF1493

       DeepSkyBlue
	   0x00BFFF

       DimGray
	   0x696969

       DodgerBlue
	   0x1E90FF

       FireBrick
	   0xB22222

       FloralWhite
	   0xFFFAF0

       ForestGreen
	   0x228B22

       Fuchsia
	   0xFF00FF

       Gainsboro
	   0xDCDCDC

       GhostWhite
	   0xF8F8FF

       Gold
	   0xFFD700

       GoldenRod
	   0xDAA520

       Gray
	   0x808080

       Green
	   0x008000

       GreenYellow
	   0xADFF2F

       HoneyDew
	   0xF0FFF0

       HotPink
	   0xFF69B4

       IndianRed
	   0xCD5C5C

       Indigo
	   0x4B0082

       Ivory
	   0xFFFFF0

       Khaki
	   0xF0E68C

       Lavender
	   0xE6E6FA

       LavenderBlush
	   0xFFF0F5

       LawnGreen
	   0x7CFC00

       LemonChiffon
	   0xFFFACD

       LightBlue
	   0xADD8E6

       LightCoral
	   0xF08080

       LightCyan
	   0xE0FFFF

       LightGoldenRodYellow
	   0xFAFAD2

       LightGreen
	   0x90EE90

       LightGrey
	   0xD3D3D3

       LightPink
	   0xFFB6C1

       LightSalmon
	   0xFFA07A

       LightSeaGreen
	   0x20B2AA

       LightSkyBlue
	   0x87CEFA

       LightSlateGray
	   0x778899

       LightSteelBlue
	   0xB0C4DE

       LightYellow
	   0xFFFFE0

       Lime
	   0x00FF00

       LimeGreen
	   0x32CD32

       Linen
	   0xFAF0E6

       Magenta
	   0xFF00FF

       Maroon
	   0x800000

       MediumAquaMarine
	   0x66CDAA

       MediumBlue
	   0x0000CD

       MediumOrchid
	   0xBA55D3

       MediumPurple
	   0x9370D8

       MediumSeaGreen
	   0x3CB371

       MediumSlateBlue
	   0x7B68EE

       MediumSpringGreen
	   0x00FA9A

       MediumTurquoise
	   0x48D1CC

       MediumVioletRed
	   0xC71585

       MidnightBlue
	   0x191970

       MintCream
	   0xF5FFFA

       MistyRose
	   0xFFE4E1

       Moccasin
	   0xFFE4B5

       NavajoWhite
	   0xFFDEAD

       Navy
	   0x000080

       OldLace
	   0xFDF5E6

       Olive
	   0x808000

       OliveDrab
	   0x6B8E23

       Orange
	   0xFFA500

       OrangeRed
	   0xFF4500

       Orchid
	   0xDA70D6

       PaleGoldenRod
	   0xEEE8AA

       PaleGreen
	   0x98FB98

       PaleTurquoise
	   0xAFEEEE

       PaleVioletRed
	   0xD87093

       PapayaWhip
	   0xFFEFD5

       PeachPuff
	   0xFFDAB9

       Peru
	   0xCD853F

       Pink
	   0xFFC0CB

       Plum
	   0xDDA0DD

       PowderBlue
	   0xB0E0E6

       Purple
	   0x800080

       Red 0xFF0000

       RosyBrown
	   0xBC8F8F

       RoyalBlue
	   0x4169E1

       SaddleBrown
	   0x8B4513

       Salmon
	   0xFA8072

       SandyBrown
	   0xF4A460

       SeaGreen
	   0x2E8B57

       SeaShell
	   0xFFF5EE

       Sienna
	   0xA0522D

       Silver
	   0xC0C0C0

       SkyBlue
	   0x87CEEB

       SlateBlue
	   0x6A5ACD

       SlateGray
	   0x708090

       Snow
	   0xFFFAFA

       SpringGreen
	   0x00FF7F

       SteelBlue
	   0x4682B4

       Tan 0xD2B48C

       Teal
	   0x008080

       Thistle
	   0xD8BFD8

       Tomato
	   0xFF6347

       Turquoise
	   0x40E0D0

       Violet
	   0xEE82EE

       Wheat
	   0xF5DEB3

       White
	   0xFFFFFF

       WhiteSmoke
	   0xF5F5F5

       Yellow
	   0xFFFF00

       YellowGreen
	   0x9ACD32

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

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

       FL  front left

       FR  front right

       FC  front center

       LFE low frequency

       BL  back left

       BR  back right

       FLC front left-of-center

       FRC front right-of-center

       BC  back center

       SL  side left

       SR  side right

       TC  top center

       TFL top front left

       TFC top front center

       TFR top front right

       TBL top back left

       TBC top back center

       TBR top back right

       DL  downmix left

       DR  downmix right

       WL  wide left

       WR  wide right

       SDL surround direct left

       SDR surround direct right

       LFE2
	   low frequency 2

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

       mono
	   FC

       stereo
	   FL+FR

       2.1 FL+FR+LFE

       3.0 FL+FR+FC

       3.0(back)
	   FL+FR+BC

       4.0 FL+FR+FC+BC

       quad
	   FL+FR+BL+BR

       quad(side)
	   FL+FR+SL+SR

       3.1 FL+FR+FC+LFE

       5.0 FL+FR+FC+BL+BR

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

       4.1 FL+FR+FC+LFE+BC

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       downmix
	   DL+DR

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

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

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

       A standard channel layout can be specified by the following:

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

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

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

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

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

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

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

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

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

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

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

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

       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 / 2^50

       E   10^18 / 2^60

       Z   10^21 / 2^70

       Y   10^24 / 2^80

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

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

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

       The list of supported options follow:

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

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

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

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

	   Possible values:

	   mv4 Use four motion vector by macroblock (mpeg4).

	   qpel
	       Use 1/4 pel motion compensation.

	   loop
	       Use loop filter.

	   qscale
	       Use fixed qscale.

	   pass1
	       Use internal 2pass ratecontrol in first pass mode.

	   pass2
	       Use internal 2pass ratecontrol in second pass mode.

	   gray
	       Only decode/encode grayscale.

	   psnr
	       Set error[?] variables during encoding.

	   truncated
	       Input bitstream might be randomly truncated.

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

	   ildct
	       Use interlaced DCT.

	   low_delay
	       Force low delay.

	   global_header
	       Place global headers in extradata instead of every keyframe.

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

	   aic Apply H263 advanced intra coding / mpeg4 ac prediction.

	   ilme
	       Apply interlaced motion estimation.

	   cgop
	       Use closed gop.

	   output_corrupt
	       Output even potentially corrupted frames.

       time_base rational number
	   Set codec time base.

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

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

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

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

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

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

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

       frame_number integer
	   Set the frame number.

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

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

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

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

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

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

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

	   Default value is 0.

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

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

	   Possible values:

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

	   ump4
	       (autodetected if fourcc==UMP4)

	   no_padding
	       padding bug (autodetected)

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

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

	   edge
	       edge padding bug (autodetected per fourcc/version)

	   hpel_chroma
	   dc_clip
	   ms  Workaround various bugs in microsoft broken decoders.

	   trunc
	       trancated frames

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

	   Possible values:

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

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

	   normal
	   unofficial
	       allow unofficial extensions

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

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

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

	   Possible values:

	   crccheck
	       verify embedded CRCs

	   bitstream
	       detect bitstream specification deviations

	   buffer
	       detect improper bitstream length

	   explode
	       abort decoding on minor error detection

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

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

	   compliant
	       consider all spec non compliancies as errors

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

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

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

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

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

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

       dct integer (encoding,video)
	   Set DCT algorithm.

	   Possible values:

	   auto
	       autoselect a good one (default)

	   fastint
	       fast integer

	   int accurate integer

	   mmx
	   altivec
	   faan
	       floating point AAN DCT

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

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

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

       p_mask float (encoding,video)
	   Set inter masking.

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

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

	   Possible values:

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

	   arm
	   altivec
	   sh4
	   simplearm
	   simplearmv5te
	   simplearmv6
	   simpleneon
	   xvid
	   faani
	       floating point AAN IDCT

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

	   Possible values:

	   guess_mvs
	       iterative motion vector (MV) search (slow)

	   deblock
	       use strong deblock filter for damaged MBs

	   favor_inter
	       favor predicting from the previous frame instead of the current

       bits_per_coded_sample integer
       aspect rational number (encoding,video)
	   Set sample aspect ratio.

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

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

	   Possible values:

	   pict
	       picture info

	   rc  rate control

	   bitstream
	   mb_type
	       macroblock (MB) type

	   qp  per-block quantization parameter (QP)

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

	   skip
	   startcode
	   er  error recognition

	   mmco
	       memory management control operations (H.264)

	   bugs
	   buffers
	       picture buffer allocations

	   thread_ops
	       threading operations

	   nomc
	       skip motion compensation

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   (1024, INT_MAX)
	       full motion estimation(slowest)

	   (768, 1024]
	       umh motion estimation

	   (512, 768]
	       hex motion estimation

	   (256, 512]
	       l2s diamond motion estimation

	   [2,256]
	       var diamond motion estimation

	   (-1,	 2)
	       small diamond motion estimation

	   -1  funny diamond motion estimation

	   (INT_MIN, -1)
	       sab diamond motion estimation

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

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

	   Possible values:

	   simple
	       use mbcmp (default)

	   bits
	       use fewest bits

	   rd  use best rate distortion

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

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

	   fast
	       Allow non spec compliant speedup tricks.

	   noout
	       Skip bitstream encoding.

	   ignorecrop
	       Ignore cropping information from sps.

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

	   chunks
	       Frame data might be split into multiple chunks.

	   showall
	       Show all frames before the first keyframe.

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

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

	   ass_ro_flush_noop
	       Do not reset ASS ReadOrder field on flush.

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

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

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

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

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

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

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

	   Possible values:

	   auto, 0
	       automatically select the number of threads to set

	   Default value is auto.

       dc integer (encoding,video)
	   Set intra_dc_precision.

       nssew integer (encoding,video)
	   Set nsse weight.

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

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

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

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

	   Possible values:

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

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

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

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

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

	   Possible values:

	   none
	       Discard no frame.

	   default
	       Discard useless frames like 0-sized frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   nointra
	       Discard all frames except I frames.

	   all Discard all frames.

	   Default value is default.

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

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

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

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

       mv0_threshold integer (encoding,video)
       compression_level integer (encoding,audio,video)
       bits_per_raw_sample integer
       channel_layout integer (decoding/encoding,audio)
	   Possible values:

       request_channel_layout integer (decoding,audio)
	   Possible values:

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

	   bt709
	       BT.709

	   bt470m
	       BT.470 M

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   film
	       Film

	   bt2020
	       BT.2020

	   smpte428
	   smpte428_1
	       SMPTE ST 428-1

	   smpte431
	       SMPTE 431-2

	   smpte432
	       SMPTE 432-1

	   jedec-p22
	       JEDEC P22

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

	   bt709
	       BT.709

	   gamma22
	       BT.470 M

	   gamma28
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   linear
	       Linear

	   log
	   log100
	       Log

	   log_sqrt
	   log316
	       Log square root

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

	   bt1361
	   bt1361e
	       BT.1361

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

	   bt2020_10
	   bt2020_10bit
	       BT.2020 - 10 bit

	   bt2020_12
	   bt2020_12bit
	       BT.2020 - 12 bit

	   smpte2084
	       SMPTE ST 2084

	   smpte428
	   smpte428_1
	       SMPTE ST 428-1

	   arib-std-b67
	       ARIB STD-B67

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

	   rgb RGB

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470 BG

	   smpte170m
	       SMPTE 170 M

	   smpte240m
	       SMPTE 240 M

	   ycocg
	       YCOCG

	   bt2020nc
	   bt2020_ncl
	       BT.2020 NCL

	   bt2020c
	   bt2020_cl
	       BT.2020 CL

	   smpte2085
	       SMPTE 2085

	   chroma-derived-nc
	       Chroma-derived NCL

	   chroma-derived-c
	       Chroma-derived CL

	   ictcp
	       ICtCp

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

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

	   pc
	   jpeg
	   full
	       JPEG (2^n-1)

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

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

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

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

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

	   Possible values:

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

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

	   frame
	       Decode more than one frame at once.

	   Default value is slice+frame.

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

	   Possible values:

	   ma  Main Audio Service

	   ef  Effects

	   vi  Visually Impaired

	   hi  Hearing Impaired

	   di  Dialogue

	   co  Commentary

	   em  Emergency

	   vo  Voice Over

	   ka  Karaoke

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

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

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

	   progressive
	       Progressive video

	   tt  Interlaced video, top field coded and displayed first

	   bb  Interlaced video, bottom field coded and displayed first

	   tb  Interlaced video, top coded first, bottom displayed first

	   bt  Interlaced video, bottom coded first, top displayed first

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

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

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

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

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

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

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

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

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

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

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

   av1
       AOMedia Video 1 (AV1) decoder.

       Options

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

   rawvideo
       Raw video decoder.

       This decoder decodes rawvideo streams.

       Options

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

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

	   0   bottom-field-first is assumed

	   1   top-field-first is assumed

   libdav1d
       dav1d AV1 decoder.

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

       Options

       The following options are supported by the libdav1d wrapper.

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

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

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

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

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

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

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

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

   libuavs3d
       AVS3-P2/IEEE1857.10 video decoder.

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

       Options

       The following option is supported by the libuavs3d wrapper.

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

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

       Common Options

       The following options are supported by all qsv decoders.

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

       gpu_copy
	   A GPU-accelerated copy between video and system memory

	   default
	   on
	   off

       HEVC Options

       Extra options for hevc_qsv.

       load_plugin
	   A user plugin to load in an internal session

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

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

       Options

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

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

   ac3
       AC-3 audio decoder.

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

       AC-3 Decoder Options

       -drc_scale value
	   Dynamic Range Scale Factor. The factor to apply  to	dynamic	 range
	   values  from the AC-3 stream. This factor is applied exponentially.
	   The default value is 1.  There are 3 notable scale factor ranges:

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

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

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

   flac
       FLAC audio decoder.

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

       FLAC Decoder options

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

   ffwavesynth
       Internal wave synthesizer.

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

   libcelt
       libcelt decoder wrapper.

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

   libgsm
       libgsm decoder wrapper.

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

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

   libilbc
       libilbc decoder wrapper.

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

       Options

       The following option is supported by the libilbc wrapper.

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

   libopencore-amrnb
       libopencore-amrnb decoder wrapper.

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

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

   libopencore-amrwb
       libopencore-amrwb decoder wrapper.

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

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

   libopus
       libopus decoder wrapper.

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

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

SUBTITLES DECODERS
   libaribb24
       ARIB STD-B24 caption decoder.

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

       libaribb24 Decoder Options

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

	   Unset by default.

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

	   Enabled by default.

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

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

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

       libaribcaption Decoder Options

       -sub_type subtitle_type
	   Specifies the format of the decoded subtitles.

	   bitmap
	       Graphical image.

	   ass ASS formatted text.

	   text
	       Simple text based output without formatting.

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

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

	   auto
	       Automatically detect text encoding (default).

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

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

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

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

	   If not specified, use internaly defined default font family.

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

	   The default is false.

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

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

	   The default is false.

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

	   The default is 1.5.

       -ignore_background boolean
	   Specify whether to ignore background color rendering.

	   The default is false.

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

	   The default is false.

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

	   The default is true.

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

	   The default is true.

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

	   The default is true.

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

	   The default is true.

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

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

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

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

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

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

       libaribcaption decoder usage examples

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

	       ffplay -sub_type bitmap MPEG.TS

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

	       ffplay -sub_type bitmap -canvas_size 1920x1080 MPEG.TS

       Embed ARIB subtitle in transcoded video:

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

   dvbsub
       Options

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

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

	   0   Never compute CLUT

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

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

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

       Options

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

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

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

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

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

       Options

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

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

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

       txt_format
	   Specifies the format of the decoded subtitles.

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

	   text
	       Simple text based output without formatting.

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

       txt_left
	   X offset of generated bitmaps, default is 0.

       txt_top
	   Y offset of generated bitmaps, default is 0.

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

       txt_duration
	   Sets	 the  display  duration	 of  the  decoded  teletext  pages  or
	   subtitles in milliseconds. Default value is -1 which means infinity
	   or until the next subtitle event comes.

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

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

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

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

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

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

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

   aac
       Advanced Audio Coding (AAC) encoder.

       This encoder is the default  AAC	 encoder,  natively  implemented  into
       FFmpeg.

       Options

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

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

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

       aac_coder
	   Set AAC encoder coding method. Possible values:

	   twoloop
	       Two loop searching (TLS) method. This is the default method.

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

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

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

	   fast
	       Constant quantizer method.

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

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

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

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

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

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

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

       profile
	   Sets the encoding profile, possible values:

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

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

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

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

	   If this option is unspecified it is set to aac_low.

   ac3 and ac3_fixed
       AC-3 audio encoders.

       These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, 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 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.

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

       Examples

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

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

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

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

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

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

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

       Options

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

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

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

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

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

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

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

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

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

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

   libopencore-amrnb
       OpenCORE Adaptive Multi-Rate Narrowband encoder.

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

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

       Options

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

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

   libopus
       libopus Opus Interactive Audio Codec encoder wrapper.

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

       Option Mapping

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

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

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

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

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

	   constrained (cvbr)
	       Use constrained variable bit rate encoding.

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

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

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

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

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

	   voip
	       Favor improved speech intelligibility.

	   audio
	       Favor faithfulness to the input (the default).

	   lowdelay
	       Restrict to only the lowest delay  modes	 by  disabling	voice-
	       optimized modes.

       cutoff (N.A.)
	   Set cutoff bandwidth in Hz. The argument must be exactly one of the
	   following:  4000,  6000,  8000,  12000,  or 20000, corresponding to
	   narrowband, mediumband,  wideband,  super  wideband,	 and  fullband
	   respectively. The default is 0 (cutoff disabled). Note that libopus
	   forces  a  wideband cutoff for bitrates < 15 kbps, unless CELT-only
	   (application set to lowdelay) mode is used.

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

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

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

   libshine
       Shine Fixed-Point MP3 encoder wrapper.

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

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

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

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

       See also libmp3lame.

       Options

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

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

   libtwolame
       TwoLAME MP2 encoder wrapper.

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

       Options

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

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

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

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

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

	   stereo
	       Stereo

	   joint_stereo
	       Joint stereo

	   dual_channel
	       Dual channel

	   mono
	       Mono

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

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

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

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

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

   libvo-amrwbenc
       VisualOn Adaptive Multi-Rate Wideband encoder.

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

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

       Options

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

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

   libvorbis
       libvorbis encoder wrapper.

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

       Options

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

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

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

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

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

       cutoff (--advanced-encode-option lowpass_frequency=N)
	   Set	cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's
	   related option is expressed in kHz. The default value is 0  (cutoff
	   disabled).

       minrate (-m)
	   Set	minimum bitrate expressed in bits/s. oggenc -m is expressed in
	   kilobits/s.

       maxrate (-M)
	   Set maximum bitrate expressed in bits/s. oggenc -M is expressed  in
	   kilobits/s. This only has effect on ABR mode.

       iblock (--advanced-encode-option impulse_noisetune=N)
	   Set	noise  floor  bias  for	 impulse  blocks. The value is a float
	   number from -15.0 to 0.0. A negative bias instructs the encoder  to
	   pay special attention to the crispness of transients in the encoded
	   audio.  The	tradeoff  for  better  transient  response is a higher
	   bitrate.

   mjpeg
       Motion JPEG encoder.

       Options

       huffman
	   Set the huffman encoding strategy. Possible values:

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

	   optimal
	       Compute and use optimal huffman tables.

   wavpack
       WavPack lossless audio encoder.

       Options

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

       Shared options

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

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

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

       compression_level (-f, -h, -hh, and -x)

       Private options

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

	   on (1)
	       Force mid/side audio encoding.

	   off (0)
	       Force left/right audio encoding.

	   auto
	       Let the encoder decide automatically.

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

	   on  enabled

	   off disabled

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

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

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

       Options

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

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

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

       skip_empty_cb boolean
	   Avoid wasting bytes, ignore vintage MacOS decoder.

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

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

   GIF
       GIF image/animation encoder.

       Options

       gifflags integer
	   Sets the flags used for GIF encoding.

	   offsetting
	       Enables picture offsetting.

	       Default is enabled.

	   transdiff
	       Enables transparency detection between frames.

	       Default is enabled.

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

	   Default value is 0.

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

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

	   Default value is 1.

   Hap
       Vidvox Hap video encoder.

       Options

       format integer
	   Specifies the Hap format to encode.

	   hap
	   hap_alpha
	   hap_q

	   Default value is hap.

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

	   Default value is 1.

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

	   none
	   snappy

	   Default value is snappy.

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

       Options

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

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

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

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

	   dwt97int (Lossy)
	   dwt53 (Lossless)

	   Default is "dwt97int"

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

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

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

	   lrcp
	   rlcp
	   rpcl
	   pcrl
	   cprl

	   Set to "lrcp" by default.

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

	   Example usage:

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

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

   librav1e
       rav1e AV1 encoder wrapper.

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

       Options

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

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

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

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

       tiles
	   Selects how many tiles to encode with.

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

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

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

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

		   ffmpeg -i input -c:v librav1e -b:v 500K -rav1e-params speed=5:low_latency=true output.mp4

   libaom-av1
       libaom AV1 encoder wrapper.

       Requires	 the  presence	of  the	 libaom	 headers  and  library	during
       configuration.	You  need  to  explicitly  configure  the  build  with
       "--enable-libaom".

       Options

       The wrapper supports the following standard libavcodec options:

       b   Set bitrate target  in  bits/second.	  By  default  this  will  use
	   variable-bitrate  mode.  If maxrate and minrate are also set to the
	   same value then it will use constant-bitrate mode, otherwise if crf
	   is set as well then it will use constrained-quality mode.

       g keyint_min
	   Set key frame placement.  The GOP size sets	the  maximum  distance
	   between  key	 frames; if zero the output stream will be intra-only.
	   The minimum distance is ignored unless it is the same  as  the  GOP
	   size,  in  which  case  key	frames	will  always appear at a fixed
	   interval.  Not set by default, so without this option  the  library
	   has completely free choice about where to place key frames.

       qmin qmax
	   Set	minimum/maximum quantisation values.  Valid range is from 0 to
	   63 (warning: this does not match the quantiser values actually used
	   by AV1 - divide by four  to	map  real  quantiser  values  to  this
	   range).  Defaults to min/max (no constraint).

       minrate maxrate bufsize rc_init_occupancy
	   Set	rate  control  buffering  parameters.	Not  used if not set -
	   defaults to unconstrained variable bitrate.

       threads
	   Set the number of threads to use while encoding.  This may  require
	   the	tiles  or  row-mt  options  to also be set to actually use the
	   specified number of	threads	 fully.	 Defaults  to  the  number  of
	   hardware threads supported by the host machine.

       profile
	   Set	the  encoding  profile.	  Defaults  to using the profile which
	   matches the bit depth and chroma subsampling of the input.

       The wrapper also has some specific options:

       cpu-used
	   Set the quality/encoding speed tradeoff.  Valid range is from 0  to
	   8,  higher numbers indicating greater speed and lower quality.  The
	   default value is 1, which will be slow and high quality.

       auto-alt-ref
	   Enable use of alternate reference frames.  Defaults to the internal
	   default of the library.

       arnr-max-frames (frames)
	   Set altref noise reduction max frame count. Default is -1.

       arnr-strength (strength)
	   Set altref noise reduction filter  strength.	 Range	is  -1	to  6.
	   Default is -1.

       aq-mode (aq-mode)
	   Set adaptive quantization mode. Possible values:

	   none (0)
	       Disabled.

	   variance (1)
	       Variance-based.

	   complexity (2)
	       Complexity-based.

	   cyclic (3)
	       Cyclic refresh.

       tune (tune)
	   Set	the  distortion	 metric	 the encoder is tuned with. Default is
	   "psnr".

	   psnr (0)
	   ssim (1)
       lag-in-frames
	   Set the maximum number of frames which  the	encoder	 may  keep  in
	   flight  at  any  one	 time for lookahead purposes.  Defaults to the
	   internal default of the library.

       error-resilience
	   Enable error resilience features:

	   default
	       Improve resilience against losses of whole frames.

	   Not enabled by default.

       crf Set the quality/size	 tradeoff  for	constant-quality  (no  bitrate
	   target)  and	 constrained-quality  (with  maximum  bitrate  target)
	   modes. Valid range is 0 to  63,  higher  numbers  indicating	 lower
	   quality and smaller output size.  Only used if set; by default only
	   the bitrate target is used.

       static-thresh
	   Set	a  change threshold on blocks below which they will be skipped
	   by the encoder.   Defined  in  arbitrary  units  as	a  nonnegative
	   integer, defaulting to zero (no blocks are skipped).

       drop-threshold
	   Set	a  threshold  for  dropping  frames when close to rate control
	   bounds.  Defined as a percentage of the target buffer  -  when  the
	   rate	 control  buffer  falls	 below this percentage, frames will be
	   dropped until it has refilled above	the  threshold.	  Defaults  to
	   zero (no frames are dropped).

       denoise-noise-level (level)
	   Amount  of noise to be removed for grain synthesis. Grain synthesis
	   is disabled if this option is not set or set to 0.

       denoise-block-size (pixels)
	   Block size used for denoising for grain synthesis. If not set,  AV1
	   codec uses the default value of 32.

       undershoot-pct (pct)
	   Set	datarate  undershoot  (min)  percentage of the target bitrate.
	   Range is -1 to 100.	Default is -1.

       overshoot-pct (pct)
	   Set datarate overshoot (max)	 percentage  of	 the  target  bitrate.
	   Range is -1 to 1000.	 Default is -1.

       minsection-pct (pct)
	   Minimum  percentage	variation  of  the GOP bitrate from the target
	   bitrate. If	minsection-pct	is  not	 set,  the  libaomenc  wrapper
	   computes  it	 as follows: "(minrate * 100 / bitrate)".  Range is -1
	   to 100. Default is -1 (unset).

       maxsection-pct (pct)
	   Maximum percentage variation of the GOP  bitrate  from  the	target
	   bitrate.  If	 maxsection-pct	 is  not  set,	the  libaomenc wrapper
	   computes it as follows: "(maxrate * 100 / bitrate)".	 Range	is  -1
	   to 5000. Default is -1 (unset).

       frame-parallel (boolean)
	   Enable frame parallel decodability features. Default is true.

       tiles
	   Set	the number of tiles to encode the input video with, as columns
	   x rows.  Larger numbers allow greater parallelism in both  encoding
	   and	decoding, but may decrease coding efficiency.  Defaults to the
	   minimum number of tiles required by the size	 of  the  input	 video
	   (this is 1x1 (that is, a single tile) for sizes up to and including
	   4K).

       tile-columns tile-rows
	   Set	the  number  of	 tiles	as log2 of the number of tile rows and
	   columns.  Provided for compatibility with libvpx/VP9.

       row-mt (Requires libaom >= 1.0.0-759-g90a15f4f2)
	   Enable row based multi-threading. Disabled by default.

       enable-cdef (boolean)
	   Enable Constrained Directional Enhancement Filter.  The  libaom-av1
	   encoder enables CDEF by default.

       enable-restoration (boolean)
	   Enable Loop Restoration Filter. Default is true for libaom-av1.

       enable-global-motion (boolean)
	   Enable  the	use  of global motion for block prediction. Default is
	   true.

       enable-intrabc (boolean)
	   Enable block copy mode for intra block  prediction.	This  mode  is
	   useful for screen content. Default is true.

       enable-rect-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable rectangular partitions. Default is true.

       enable-1to4-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable 1:4/4:1 partitions. Default is true.

       enable-ab-partitions (boolean) (Requires libaom >= v2.0.0)
	   Enable AB shape partitions. Default is true.

       enable-angle-delta (boolean) (Requires libaom >= v2.0.0)
	   Enable angle delta intra prediction. Default is true.

       enable-cfl-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable  chroma  predicted  from  luma  intra prediction. Default is
	   true.

       enable-filter-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable filter intra predictor. Default is true.

       enable-intra-edge-filter (boolean) (Requires libaom >= v2.0.0)
	   Enable intra edge filter. Default is true.

       enable-smooth-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable smooth intra prediction mode. Default is true.

       enable-paeth-intra (boolean) (Requires libaom >= v2.0.0)
	   Enable paeth predictor in intra prediction. Default is true.

       enable-palette (boolean) (Requires libaom >= v2.0.0)
	   Enable palette prediction mode. Default is true.

       enable-flip-idtx (boolean) (Requires libaom >= v2.0.0)
	   Enable   extended   transform   type,    including	 FLIPADST_DCT,
	   DCT_FLIPADST,   FLIPADST_FLIPADST,	ADST_FLIPADST,	FLIPADST_ADST,
	   IDTX, V_DCT, H_DCT, V_ADST, H_ADST, V_FLIPADST, H_FLIPADST. Default
	   is true.

       enable-tx64 (boolean) (Requires libaom >= v2.0.0)
	   Enable 64-pt transform. Default is true.

       reduced-tx-type-set (boolean) (Requires libaom >= v2.0.0)
	   Use reduced set of transform types. Default is false.

       use-intra-dct-only (boolean) (Requires libaom >= v2.0.0)
	   Use DCT only for INTRA modes. Default is false.

       use-inter-dct-only (boolean) (Requires libaom >= v2.0.0)
	   Use DCT only for INTER modes. Default is false.

       use-intra-default-tx-only (boolean) (Requires libaom >= v2.0.0)
	   Use Default-transform only for INTRA modes. Default is false.

       enable-ref-frame-mvs (boolean) (Requires libaom >= v2.0.0)
	   Enable temporal mv prediction. Default is true.

       enable-reduced-reference-set (boolean) (Requires libaom >= v2.0.0)
	   Use reduced set of  single  and  compound  references.  Default  is
	   false.

       enable-obmc (boolean) (Requires libaom >= v2.0.0)
	   Enable obmc. Default is true.

       enable-dual-filter (boolean) (Requires libaom >= v2.0.0)
	   Enable dual filter. Default is true.

       enable-diff-wtd-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable difference-weighted compound. Default is true.

       enable-dist-wtd-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable distance-weighted compound. Default is true.

       enable-onesided-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable one sided compound. Default is true.

       enable-interinter-wedge (boolean) (Requires libaom >= v2.0.0)
	   Enable interinter wedge compound. Default is true.

       enable-interintra-wedge (boolean) (Requires libaom >= v2.0.0)
	   Enable interintra wedge compound. Default is true.

       enable-masked-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable masked compound. Default is true.

       enable-interintra-comp (boolean) (Requires libaom >= v2.0.0)
	   Enable interintra compound. Default is true.

       enable-smooth-interintra (boolean) (Requires libaom >= v2.0.0)
	   Enable smooth interintra mode. Default is true.

       aom-params
	   Set	libaom	options	 using	a list of key=value pairs separated by
	   ":". For a list of supported options, see aomenc --help  under  the
	   section "AV1 Specific Options".

	   For example to specify libaom encoding options with -aom-params:

		   ffmpeg -i input -c:v libaom-av1 -b:v 500K -aom-params tune=psnr:enable-tpl-model=1 output.mp4

   libsvtav1
       SVT-AV1 encoder wrapper.

       Requires	 the  presence	of  the	 SVT-AV1  headers  and	library during
       configuration.	You  need  to  explicitly  configure  the  build  with
       "--enable-libsvtav1".

       Options

       profile
	   Set the encoding profile.

	   main
	   high
	   professional
       level
	   Set the operating point level. For example: '4.0'

       hielevel
	   Set the Hierarchical prediction levels.

	   3level
	   4level
	       This is the default.

       tier
	   Set the operating point tier.

	   main
	       This is the default.

	   high
       qmax
	   Set the maximum quantizer to use when using a bitrate mode.

       qmin
	   Set the minimum quantizer to use when using a bitrate mode.

       crf Constant rate factor value used in crf rate control mode (0-63).

       qp  Set the quantizer used in cqp rate control mode (0-63).

       sc_detection
	   Enable scene change detection.

       la_depth
	   Set number of frames to look ahead (0-120).

       preset
	   Set	the  quality-speed  tradeoff,  in  the	range 0 to 13.	Higher
	   values are faster but lower quality.

       tile_rows
	   Set log2 of the number of rows of tiles to use (0-6).

       tile_columns
	   Set log2 of the number of columns of tiles to use (0-4).

       svtav1-params
	   Set SVT-AV1 options using a list of key=value  pairs	 separated  by
	   ":".	 See  the  SVT-AV1  encoder  user guide for a list of accepted
	   parameters.

   libjxl
       libjxl JPEG XL encoder wrapper.

       Requires	 the  presence	of  the	 libjxl	 headers  and  library	during
       configuration.	You  need  to  explicitly  configure  the  build  with
       "--enable-libjxl".

       Options

       The libjxl wrapper supports the following options:

       distance
	   Set the target Butteraugli distance. This  is  a  quality  setting:
	   lower  distance  yields  higher  quality, with distance=1.0 roughly
	   comparable to libjpeg Quality 90 for photographic content.  Setting
	   distance=0.0	 yields	 true  lossless	 encoding.  Valid values range
	   between 0.0 and 15.0, and sane values rarely	 exceed	 5.0.  Setting
	   distance=0.1	 usually  attains  transparency	 for  most  input. The
	   default is 1.0.

       effort
	   Set the encoding effort used. Higher	 effort	 values	 produce  more
	   consistent quality and usually produces a better quality/bpp curve,
	   at the cost of more CPU time required. Valid values range from 1 to
	   9, and the default is 7.

       modular
	   Force   the	encoder	 to  use  Modular  mode	 instead  of  choosing
	   automatically. The default is to use VarDCT for lossy encoding  and
	   Modular  for	 lossless. VarDCT is generally superior to Modular for
	   lossy encoding but does not support lossless encoding.

   libkvazaar
       Kvazaar H.265/HEVC encoder.

       Requires the presence of the  libkvazaar	 headers  and  library	during
       configuration.	You  need  to  explicitly  configure  the  build  with
       --enable-libkvazaar.

       Options

       b   Set target video bitrate in bit/s and enable rate control.

       kvazaar-params
	   Set kvazaar parameters as a list of name=value pairs	 separated  by
	   commas (,). See kvazaar documentation for a list of options.

   libopenh264
       Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

       This  encoder  requires	the  presence  of  the libopenh264 headers and
       library during configuration. You  need	to  explicitly	configure  the
       build  with  "--enable-libopenh264". The library is detected using pkg-
       config.

       For more information about the library see <http://www.openh264.org>.

       Options

       The following FFmpeg global options affect the  configurations  of  the
       libopenh264 encoder.

       b   Set the bitrate (as a number of bits per second).

       g   Set the GOP size.

       maxrate
	   Set the max bitrate (as a number of bits per second).

       flags +global_header
	   Set global header in the bitstream.

       slices
	   Set	the  number  of slices, used in parallelized encoding. Default
	   value is 0. This is only used when slice_mode is set to fixed.

       loopfilter
	   Enable loop filter, if set to 1 (automatically enabled). To disable
	   set a value of 0.

       profile
	   Set profile restrictions. If set to the value of main enable	 CABAC
	   (set the "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

       max_nal_size
	   Set maximum NAL size in bytes.

       allow_skip_frames
	   Allow skipping frames to hit the target bitrate if set to 1.

   libtheora
       libtheora Theora encoder wrapper.

       Requires	 the  presence	of  the	 libtheora  headers and library during
       configuration.  You  need  to  explicitly  configure  the  build	  with
       "--enable-libtheora".

       For    more    information    about    the    libtheora	 project   see
       <http://www.theora.org/>.

       Options

       The following global options are mapped to internal  libtheora  options
       which affect the quality and the bitrate of the encoded stream.

       b   Set	the  video  bitrate in bit/s for CBR (Constant Bit Rate) mode.
	   In case VBR (Variable Bit Rate) mode	 is  enabled  this  option  is
	   ignored.

       flags
	   Used	 to  enable  constant  quality mode (VBR) encoding through the
	   qscale flag, and to enable the "pass1" and "pass2" modes.

       g   Set the GOP size.

       global_quality
	   Set the global quality as an integer in lambda units.

	   Only relevant when VBR mode is enabled with	"flags	+qscale".  The
	   value  is  converted	 to QP units by dividing it by "FF_QP2LAMBDA",
	   clipped in the [0 - 10] range, and then multiplied by 6.3 to get  a
	   value  in  the  native  libtheora  range  [0-63].  A	 higher	 value
	   corresponds to a higher quality.

       q   Enable VBR mode when set to a non-negative value, and set  constant
	   quality value as a double floating point value in QP units.

	   The	value  is  clipped in the [0-10] range, and then multiplied by
	   6.3 to get a value in the native libtheora range [0-63].

	   This option is valid only using the ffmpeg command-line  tool.  For
	   library interface users, use global_quality.

       Examples

       •   Set maximum constant quality (VBR) encoding with ffmpeg:

		   ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg

       •   Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

		   ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

   libvpx
       VP8/VP9 format supported through libvpx.

       Requires	 the  presence	of  the	 libvpx	 headers  and  library	during
       configuration.	You  need  to  explicitly  configure  the  build  with
       "--enable-libvpx".

       Options

       The  following  options	are  supported	by  the	 libvpx	 wrapper.  The
       vpxenc-equivalent options or values are listed in parentheses for  easy
       migration.

       To  reduce  the	duplication of documentation, only the private options
       and some others requiring special attention are	documented  here.  For
       the  documentation  of  the undocumented generic options, see the Codec
       Options chapter.

       To get more documentation of the libvpx	options,  invoke  the  command
       ffmpeg  -h  encoder=libvpx,  ffmpeg  -h	encoder=libvpx-vp9  or	vpxenc
       --help.	Further	 information  is   available   in   the	  libvpx   API
       documentation.

       b (target-bitrate)
	   Set	bitrate in bits/s. Note that FFmpeg's b option is expressed in
	   bits/s, while vpxenc's target-bitrate is in kilobits/s.

       g (kf-max-dist)
       keyint_min (kf-min-dist)
       qmin (min-q)
	   Minimum (Best Quality) Quantizer.

       qmax (max-q)
	   Maximum (Worst Quality) Quantizer.  Can be changed per-frame.

       bufsize (buf-sz, buf-optimal-sz)
	   Set ratecontrol buffer size (in bits). Note	vpxenc's  options  are
	   specified  in  milliseconds, the libvpx wrapper converts this value
	   as follows: "buf-sz = bufsize * 1000 / bitrate", "buf-optimal-sz  =
	   bufsize * 1000 / bitrate * 5 / 6".

       rc_init_occupancy (buf-initial-sz)
	   Set number of bits which should be loaded into the rc buffer before
	   decoding starts. Note vpxenc's option is specified in milliseconds,
	   the	  libvpx    wrapper    converts	  this	 value	 as   follows:
	   "rc_init_occupancy * 1000 / bitrate".

       undershoot-pct
	   Set datarate undershoot (min) percentage of the target bitrate.

       overshoot-pct
	   Set datarate overshoot (max) percentage of the target bitrate.

       skip_threshold (drop-frame)
       qcomp (bias-pct)
       maxrate (maxsection-pct)
	   Set GOP max bitrate in bits/s. Note vpxenc's option is specified as
	   a percentage of the target bitrate,	the  libvpx  wrapper  converts
	   this value as follows: "(maxrate * 100 / bitrate)".

       minrate (minsection-pct)
	   Set GOP min bitrate in bits/s. Note vpxenc's option is specified as
	   a  percentage  of  the  target bitrate, the libvpx wrapper converts
	   this value as follows: "(minrate * 100 / bitrate)".

       minrate, maxrate, b end-usage=cbr
	   "(minrate == maxrate == bitrate)".

       crf (end-usage=cq, cq-level)
       tune (tune)
	   psnr (psnr)
	   ssim (ssim)
       quality, deadline (deadline)
	   best
	       Use best quality deadline. Poorly named and  quite  slow,  this
	       option  should  be  avoided as it may give worse quality output
	       than good.

	   good
	       Use good quality deadline. This is  a  good  trade-off  between
	       speed and quality when used with the cpu-used option.

	   realtime
	       Use realtime quality deadline.

       speed, cpu-used (cpu-used)
	   Set quality/speed ratio modifier. Higher values speed up the encode
	   at the cost of quality.

       nr (noise-sensitivity)
       static-thresh
	   Set	a  change threshold on blocks below which they will be skipped
	   by the encoder.

       slices (token-parts)
	   Note	 that  FFmpeg's	 slices	 option	 gives	the  total  number  of
	   partitions,	  while	   vpxenc's    token-parts    is    given   as
	   log2(partitions).

       max-intra-rate
	   Set maximum I-frame bitrate as a percentage of the target  bitrate.
	   A value of 0 means unlimited.

       force_key_frames
	   "VPX_EFLAG_FORCE_KF"

       Alternate reference frame related
	   auto-alt-ref
	       Enable use of alternate reference frames (2-pass only).	Values
	       greater	than  1	 enable multi-layer alternate reference frames
	       (VP9 only).

	   arnr-maxframes
	       Set altref noise reduction max frame count.

	   arnr-type
	       Set altref noise	 reduction  filter  type:  backward,  forward,
	       centered.

	   arnr-strength
	       Set altref noise reduction filter strength.

	   rc-lookahead, lag-in-frames (lag-in-frames)
	       Set   number   of  frames  to  look  ahead  for	frametype  and
	       ratecontrol.

	   min-gf-interval
	       Set minimum  golden/alternate  reference	 frame	interval  (VP9
	       only).

       error-resilient
	   Enable error resiliency features.

       sharpness integer
	   Increase  sharpness	at the expense of lower PSNR.  The valid range
	   is [0, 7].

       ts-parameters
	   Sets the temporal scalability  configuration	 using	a  :-separated
	   list	  of   key=value  pairs.  For  example,	 to  specify  temporal
	   scalability parameters with "ffmpeg":

		   ffmpeg -i INPUT -c:v libvpx -ts-parameters ts_number_layers=3:\
		   ts_target_bitrate=250,500,1000:ts_rate_decimator=4,2,1:\
		   ts_periodicity=4:ts_layer_id=0,2,1,2:ts_layering_mode=3 OUTPUT

	   Below is a brief explanation of  each  of  the  parameters,	please
	   refer to "struct vpx_codec_enc_cfg" in "vpx/vpx_encoder.h" for more
	   details.

	   ts_number_layers
	       Number of temporal coding layers.

	   ts_target_bitrate
	       Target  bitrate	for  each  temporal layer (in kbps).  (bitrate
	       should be inclusive of the lower temporal layer).

	   ts_rate_decimator
	       Frame rate decimation factor for each temporal layer.

	   ts_periodicity
	       Length  of  the	sequence   defining   frame   temporal	 layer
	       membership.

	   ts_layer_id
	       Template defining the membership of frames to temporal layers.

	   ts_layering_mode
	       (optional)  Selecting the temporal structure from a set of pre-
	       defined	temporal  layering  modes.   Currently	supports   the
	       following options.

	       0   No  temporal layering flags are provided internally, relies
		   on  flags  being  passed  in	 using	"metadata"  field   in
		   "AVFrame" with following keys.

		   vp8-flags
		       Sets  the flags passed into the encoder to indicate the
		       referencing scheme for the  current  frame.   Refer  to
		       function	 "vpx_codec_encode" in "vpx/vpx_encoder.h" for
		       more details.

		   temporal_id
		       Explicitly sets the temporal id of the current frame to
		       encode.

	       2   Two temporal layers. 0-1...

	       3   Three temporal layers. 0-2-1-2...;  with  single  reference
		   frame.

	       4   Same	 as  option  "3", except there is a dependency between
		   the two temporal layer 2 frames within the temporal period.

       VP9-specific options
	   lossless
	       Enable lossless mode.

	   tile-columns
	       Set number of tile columns  to  use.  Note  this	 is  given  as
	       log2(tile_columns).  For	 example,  8  tile  columns  would  be
	       requested by setting the tile-columns option to 3.

	   tile-rows
	       Set number  of  tile  rows  to  use.  Note  this	 is  given  as
	       log2(tile_rows).	  For  example, 4 tile rows would be requested
	       by setting the tile-rows option to 2.

	   frame-parallel
	       Enable frame parallel decodability features.

	   aq-mode
	       Set adaptive quantization mode (0: off (default),  1:  variance
	       2: complexity, 3: cyclic refresh, 4: equator360).

	   colorspace color-space
	       Set input color space. The VP9 bitstream supports signaling the
	       following colorspaces:

	       rgb sRGB
	       bt709 bt709
	       unspecified unknown
	       bt470bg bt601
	       smpte170m smpte170
	       smpte240m smpte240
	       bt2020_ncl bt2020
	   row-mt boolean
	       Enable row based multi-threading.

	   tune-content
	       Set content type: default (0), screen (1), film (2).

	   corpus-complexity
	       Corpus  VBR  mode  is  a	 variant  of  standard	VBR  where the
	       complexity distribution	midpoint  is  passed  in  rather  than
	       calculated for a specific clip or chunk.

	       The valid range is [0, 10000]. 0 (default) uses standard VBR.

	   enable-tpl boolean
	       Enable temporal dependency model.

	   ref-frame-config
	       Using   per-frame   metadata,  set  members  of	the  structure
	       "vpx_svc_ref_frame_config_t" in "vpx/vp8cx.h"  to  fine-control
	       referencing   schemes  and  frame  buffer  management.	Use  a
	       :-separated list of key=value pairs.  For example,

		       av_dict_set(&av_frame->metadata, "ref-frame-config", \
		       "rfc_update_buffer_slot=7:rfc_lst_fb_idx=0:rfc_gld_fb_idx=1:rfc_alt_fb_idx=2:rfc_reference_last=0:rfc_reference_golden=0:rfc_reference_alt_ref=0");

	       rfc_update_buffer_slot
		   Indicates the buffer slot number to update

	       rfc_update_last
		   Indicates whether to update the LAST frame

	       rfc_update_golden
		   Indicates whether to update GOLDEN frame

	       rfc_update_alt_ref
		   Indicates whether to update ALT_REF frame

	       rfc_lst_fb_idx
		   LAST frame buffer index

	       rfc_gld_fb_idx
		   GOLDEN frame buffer index

	       rfc_alt_fb_idx
		   ALT_REF frame buffer index

	       rfc_reference_last
		   Indicates whether to reference LAST frame

	       rfc_reference_golden
		   Indicates whether to reference GOLDEN frame

	       rfc_reference_alt_ref
		   Indicates whether to reference ALT_REF frame

	       rfc_reference_duration
		   Indicates frame duration

       For more information about libvpx see: <http://www.webmproject.org/>

   libwebp
       libwebp WebP Image encoder wrapper

       libwebp is Google's official encoder for WebP images. It can encode  in
       either  lossy  or lossless mode. Lossy images are essentially a wrapper
       around a VP8 frame. Lossless images are a separate codec	 developed  by
       Google.

       Pixel Format

       Currently,  libwebp only supports YUV420 for lossy and RGB for lossless
       due to limitations of the format and libwebp. Alpha  is	supported  for
       either  mode.   Because	of  API	 limitations, if RGB is passed in when
       encoding lossy or YUV is passed in for  encoding	 lossless,  the	 pixel
       format  will  automatically  be converted using functions from libwebp.
       This is not ideal and is done only for convenience.

       Options

       -lossless boolean
	   Enables/Disables use of lossless mode. Default is 0.

       -compression_level integer
	   For lossy, this is a quality/speed  tradeoff.  Higher  values  give
	   better  quality  for a given size at the cost of increased encoding
	   time. For lossless, this is a size/speed  tradeoff.	Higher	values
	   give	 smaller  size	at  the	 cost of increased encoding time. More
	   specifically, it  controls  the  number  of	extra  algorithms  and
	   compression	tools used, and varies the combination of these tools.
	   This maps to the method option in libwebp. The valid range is 0  to
	   6.  Default is 4.

       -quality float
	   For	lossy  encoding,  this	controls  image	 quality. For lossless
	   encoding, this controls the effort and time spent  in  compression.
	   Range is 0 to 100. Default is 75.

       -preset type
	   Configuration  preset.  This	 does some automatic settings based on
	   the general type of the image.

	   none
	       Do not use a preset.

	   default
	       Use the encoder default.

	   picture
	       Digital picture, like portrait, inner shot

	   photo
	       Outdoor photograph, with natural lighting

	   drawing
	       Hand or line drawing, with high-contrast details

	   icon
	       Small-sized colorful images

	   text
	       Text-like

   libx264, libx264rgb
       x264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libx264 headers  and  library
       during  configuration.  You need to explicitly configure the build with
       "--enable-libx264".

       libx264 supports an impressive number of features,  including  8x8  and
       4x4 adaptive spatial transform, adaptive B-frame placement, CAVLC/CABAC
       entropy	coding,	 interlacing (MBAFF), lossless mode, psy optimizations
       for detail retention (adaptive quantization, psy-RD, psy-trellis).

       Many libx264 encoder options are mapped to FFmpeg global codec options,
       while unique encoder options  are  provided  through  private  options.
       Additionally the x264opts and x264-params private options allows one to
       pass   a	  list	 of  key=value	tuples	as  accepted  by  the  libx264
       "x264_param_parse" function.

       The	   x264		project		website		is	    at
       <http://www.videolan.org/developers/x264.html>.

       The libx264rgb encoder is the same as libx264, except it accepts packed
       RGB pixel formats as input instead of YUV.

       Supported Pixel Formats

       x264  supports  8-  to  10-bit  color  spaces.  The  exact bit depth is
       controlled at x264's configure time.

       Options

       The following  options  are  supported  by  the	libx264	 wrapper.  The
       x264-equivalent	options	 or  values are listed in parentheses for easy
       migration.

       To reduce the duplication of documentation, only	 the  private  options
       and  some  others  requiring special attention are documented here. For
       the documentation of the undocumented generic options,  see  the	 Codec
       Options chapter.

       To  get	a  more	 accurate  and	extensive documentation of the libx264
       options, invoke the command x264	 --fullhelp  or	 consult  the  libx264
       documentation.

       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.

       level (level)
	   Set	the  "x264_param_t.i_level_idc"	 value	in  case  the value is
	   positive, it is ignored otherwise.

	   This value can be set  using	 the  "AVCodecContext"	API  (e.g.  by
	   setting  the	 "AVCodecContext" value directly), and is specified as
	   an integer mapped on a corresponding level (e.g. the value 31  maps
	   to  H.264  level IDC "3.1", as defined in the "x264_levels" table).
	   It is ignored when set to a non positive value.

	   Alternatively it can be set as a  private  option,  overriding  the
	   value  set  in "AVCodecContext", and in this case must be specified
	   as the level IDC identifier (e.g. "3.1"), as defined by H.264 Annex
	   A.

       sc_threshold (scenecut)
	   Sets the threshold for the scene change detection.

       trellis (trellis)
	   Performs Trellis quantization to increase  efficiency.  Enabled  by
	   default.

       nr  (nr)
       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).

       udu_sei boolean
	   Import user data unregistered SEI if available into output. Default
	   is 0 (off).

       mb_info boolean
	   Set	mb_info	 data  through	AVFrameSideData, only useful when used
	   from the API. Default is 0 (off).

       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.

       udu_sei boolean
	   Import user data unregistered SEI if available into output. Default
	   is 0 (off).

       x265-params
	   Set	x265  options  using  a list of key=value couples separated by
	   ":". See x265 --help for a list of options.

	   For example to specify libx265 encoding options with -x265-params:

		   ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

   libxavs2
       xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

       This encoder requires the presence of the libxavs2 headers and  library
       during  configuration.  You need to explicitly configure the build with
       --enable-libxavs2.

       The following standard libavcodec options are used:

       •   b / bit_rate

       •   g / gop_size

       •   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_quality
	   Set	motion estimation quality level. Possible values in decreasing
	   order of speed and increasing order of quality:

	   0   Use no motion estimation (default).

	   1, 2
	       Enable advanced diamond zonal search for 16x16 blocks and half-
	       pixel refinement for 16x16 blocks.

	   3, 4
	       Enable all of the things described above, plus advanced diamond
	       zonal search for 8x8 blocks and half-pixel refinement  for  8x8
	       blocks,	also  enable  motion estimation on chroma planes for P
	       and B-frames.

	   5, 6
	       Enable all of the things described above, plus  extended	 16x16
	       and 8x8 blocks search.

       mbd Set	 macroblock   decision	 algorithm.  Possible  values  in  the
	   increasing order of quality:

	   simple
	       Use macroblock comparing function algorithm (default).

	   bits
	       Enable rate  distortion-based  half  pixel  and	quarter	 pixel
	       refinement for 16x16 blocks.

	   rd  Enable all of the things described above, plus rate distortion-
	       based  half  pixel and quarter pixel refinement for 8x8 blocks,
	       and rate distortion-based search using square pattern.

       lumi_aq
	   Enable lumi masking adaptive quantization when set to 1. Default is
	   0 (disabled).

       variance_aq
	   Enable variance adaptive quantization when set to 1. Default	 is  0
	   (disabled).

	   When	 combined  with	 lumi_aq,  the	resulting  quality will not be
	   better than any of the two specified individually. In other	words,
	   the resulting quality will be the worse one of the two effects.

       ssim
	   Set	 structural  similarity	 (SSIM)	 displaying  method.  Possible
	   values:

	   off Disable displaying of SSIM information.

	   avg Output average SSIM at the  end	of  encoding  to  stdout.  The
	       format of showing the average SSIM is:

		       Average SSIM: %f

	       For users who are not familiar with C, %f means a float number,
	       or a decimal (e.g. 0.939232).

	   frame
	       Output  both  per-frame	SSIM  data during encoding and average
	       SSIM at the end of encoding to stdout. The format of  per-frame
	       information is:

			      SSIM: avg: %1.3f min: %1.3f max: %1.3f

	       For  users  who	are  not  familiar with C, %1.3f means a float
	       number rounded to 3 digits after the dot (e.g. 0.932).

       ssim_acc
	   Set SSIM accuracy. Valid options are integers within the  range  of
	   0-4,	 while	0  gives  the  most accurate result and 4 computes the
	   fastest.

   MediaFoundation
       This provides wrappers to  encoders  (both  audio  and  video)  in  the
       MediaFoundation	framework.  It	can  access  both  SW and HW encoders.
       Video encoders can take input in either of nv12 or yuv420p  form	 (some
       encoders	 support both, some support only either - in practice, nv12 is
       the safer choice, especially among HW encoders).

   Microsoft RLE
       Microsoft RLE aka MSRLE encoder.	 Only 8-bit  palette  mode  supported.
       Compatible with Windows 3.1 and Windows 95.

       Options

       g integer
	   Keyframe  interval.	 A  keyframe  is  inserted at least every "-g"
	   frames, sometimes sooner.

   mpeg2
       MPEG-2 video encoder.

       Options

       profile
	   Select the mpeg2 profile to encode:

	   422
	   high
	   ss  Spatially Scalable

	   snr SNR Scalable

	   main
	   simple
       level
	   Select the mpeg2 level to encode:

	   high
	   high1440
	   main
	   low
       seq_disp_ext integer
	   Specifies if the encoder should write a  sequence_display_extension
	   to the output.

	   -1
	   auto
	       Decide  automatically  to write it or not (this is the default)
	       by checking if the data to be written  is  different  from  the
	       default or unspecified values.

	   0
	   never
	       Never write it.

	   1
	   always
	       Always write it.

       video_format integer
	   Specifies  the  video_format	 written  into	the  sequence  display
	   extension indicating the source of the video pictures. The  default
	   is  unspecified,  can  be  component, pal, ntsc, secam or mac.  For
	   maximum compatibility, use component.

       a53cc boolean
	   Import closed captions (which must be ATSC compatible format)  into
	   output.  Default is 1 (on).

   png
       PNG image encoder.

       Private options

       dpi integer
	   Set physical density of pixels, in dots per inch, unset by default

       dpm integer
	   Set physical density of pixels, in dots per meter, unset by default

   ProRes
       Apple ProRes encoder.

       FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.
       The used encoder can be chosen with the "-vcodec" option.

       Private Options for prores-ks

       profile integer
	   Select the ProRes profile to encode

	   proxy
	   lt
	   standard
	   hq
	   4444
	   4444xq
       quant_mat integer
	   Select quantization matrix.

	   auto
	   default
	   proxy
	   lt
	   standard
	   hq

	   If set to auto, the matrix matching the profile will be picked.  If
	   not set, the matrix providing the highest quality, default, will be
	   picked.

       bits_per_mb integer
	   How	many  bits  to	allot  for  coding  one	 macroblock. Different
	   profiles use between 200 and 2400 bits per macroblock, the  maximum
	   is 8000.

       mbs_per_slice integer
	   Number  of  macroblocks  in each slice (1-8); the default value (8)
	   should be good in almost all situations.

       vendor string
	   Override the 4-byte vendor ID.  A custom vendor ID like apl0	 would
	   claim the stream was produced by the Apple encoder.

       alpha_bits integer
	   Specify number of bits for alpha component.	Possible values are 0,
	   8 and 16.  Use 0 to disable alpha plane coding.

       Speed considerations

       In  the	default	 mode  of  operation  the  encoder  has to honor frame
       constraints (i.e. not produce frames with size bigger  than  requested)
       while  still  making  output  picture  as  good	as  possible.  A frame
       containing a lot of small details is harder to compress and the encoder
       would spend more time searching for  appropriate	 quantizers  for  each
       slice.

       Setting a higher bits_per_mb limit will improve the speed.

       For  the	 fastest  encoding  speed  set	the qscale parameter (4 is the
       recommended value) and do not set a size constraint.

   QSV Encoders
       The family of Intel QuickSync  Video  encoders  (MPEG-2,	 H.264,	 HEVC,
       JPEG/MJPEG, VP9, AV1)

       Ratecontrol Method

       The ratecontrol method is selected as follows:

       •   When	 global_quality	 is  specified,	 a quality-based mode is used.
	   Specifically this means either

	   -   CQP - constant quantizer scale, when the qscale codec  flag  is
	       also set (the -qscale ffmpeg option).

	   -   LA_ICQ  - intelligent constant quality with lookahead, when the
	       look_ahead option is also set.

	   -   ICQ -- intelligent constant  quality  otherwise.	 For  the  ICQ
	       modes,  global  quality range is 1 to 51, with 1 being the best
	       quality.

       •   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, both
	       avbr_accuracy and avbr_convergence are set  to  non-zero.  This
	       mode is available for H264 and HEVC on Windows.

       Note  that  depending on your system, a different mode than the one you
       specified may be selected by the encoder. Set the  verbosity  level  to
       verbose or higher to see the actual settings used by the QSV runtime.

       Global Options -> MSDK Options

       Additional  libavcodec  global  options	are  mapped to MSDK options as
       follows:

       •   g/gop_size -> GopPicSize

       •   bf/max_b_frames+1 -> GopRefDist

       •   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       •   slices -> NumSlice

       •   refs -> NumRefFrame

       •   b_strategy/b_frame_strategy -> BRefType

       •   cgop/CLOSED_GOP codec flag -> GopOptFlag

       •   For the CQP mode, the i_qfactor/i_qoffset  and  b_qfactor/b_qoffset
	   set	 the   difference  between  QPP	 and  QPI,  and	 QPP  and  QPB
	   respectively.

       •   Setting the coder option to the  value  vlc	will  make  the	 H.264
	   encoder use CAVLC instead of CABAC.

       Common Options

       Following options are used by all qsv encoders.

       async_depth
	   Specifies  how many asynchronous operations an application performs
	   before the application explicitly synchronizes the result. If zero,
	   the value is not specified.

       preset
	   This option itemizes a range of choices from veryfast (best	speed)
	   to veryslow (best quality).

	   veryfast
	   faster
	   fast
	   medium
	   slow
	   slower
	   veryslow
       forced_idr
	   Forcing I frames as IDR frames.

       low_power
	   For	encoders  set  this flag to ON to reduce power consumption and
	   GPU usage.

       Runtime Options

       Following options can be used durning qsv encoding.

       global_quality
       i_quant_factor
       i_quant_offset
       b_quant_factor
       b_quant_offset
	   Supported in h264_qsv and hevc_qsv.	Change these  value  to	 reset
	   qsv codec's qp configuration.

       max_frame_size
	   Supported in h264_qsv and hevc_qsv.	Change this value to reset qsv
	   codec's MaxFrameSize configuration.

       gop_size
	   Change this value to reset qsv codec's gop configuration.

       int_ref_type
       int_ref_cycle_size
       int_ref_qp_delta
       int_ref_cycle_dist
	   Supported  in  h264_qsv  and hevc_qsv.  Change these value to reset
	   qsv codec's Intra Refresh configuration.

       qmax
       qmin
       max_qp_i
       min_qp_i
       max_qp_p
       min_qp_p
       max_qp_b
       min_qp_b
	   Supported in h264_qsv.  Change these value  to  reset  qsv  codec's
	   max/min qp configuration.

       low_delay_brc
	   Supported  in h264_qsv, hevc_qsv and av1_qsv.  Change this value to
	   reset qsv codec's low_delay_brc configuration.

       framerate
	   Change this value to reset qsv codec's framerate configuration.

       bit_rate
       rc_buffer_size
       rc_initial_buffer_occupancy
       rc_max_rate
	   Change  these  value	 to  reset   qsv   codec's   bitrate   control
	   configuration.

       pic_timing_sei
	   Supported in h264_qsv and hevc_qsv.	Change this value to reset qsv
	   codec's pic_timing_sei configuration.

       H264 options

       These options are used by h264_qsv

       extbrc
	   Extended bitrate control.

       recovery_point_sei
	   Set	this  flag  to	insert	the  recovery point SEI message at the
	   beginning of every intra refresh cycle.

       rdo Enable rate distortion optimization.

       max_frame_size
	   Maximum encoded frame size in bytes.

       max_frame_size_i
	   Maximum encoded frame size for I frames in bytes. If this value  is
	   set	as  larger  than  zero,	 then  for  I  frames the value set by
	   max_frame_size is ignored.

       max_frame_size_p
	   Maximum encoded frame size for P frames in bytes. If this value  is
	   set	as  larger  than  zero,	 then  for  P  frames the value set by
	   max_frame_size is ignored.

       max_slice_size
	   Maximum encoded slice size in bytes.

       bitrate_limit
	   Toggle bitrate limitations.	Modifies bitrate to be	in  the	 range
	   imposed  by	the  QSV  encoder.  Setting  this flag off may lead to
	   violation of HRD conformance. Mind that  specifying	bitrate	 below
	   the	QSV  encoder  range  might significantly affect quality. If on
	   this option takes effect in non CQP modes: if bitrate is not in the
	   range imposed by the QSV encoder, it will be changed to be  in  the
	   range.

       mbbrc
	   Setting  this  flag	enables	 macroblock level bitrate control that
	   generally improves subjective visual quality.  Enabling  this  flag
	   may	have  negative	impact	on  performance	 and  objective visual
	   quality metric.

       low_delay_brc
	   Setting this flag turns  on	or  off	 LowDelayBRC  feautre  in  qsv
	   plugin,  which  provides  more accurate bitrate control to minimize
	   the variance of bitstream size frame by  frame.  Value:  -1-default
	   0-off 1-on

       adaptive_i
	   This	 flag  controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       p_strategy
	   Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to
	   0).

       b_strategy
	   This option controls usage of B frames as reference.

       dblk_idc
	   This option disable deblocking. It has value in range 0~2.

       cavlc
	   If set, CAVLC is used; if unset, CABAC is used for encoding.

       vcm Video conferencing mode, please see ratecontrol method.

       idr_interval
	   Distance (in I-frames) between IDR frames.

       pic_timing_sei
	   Insert picture timing SEI with pic_struct_syntax element.

       single_sei_nal_unit
	   Put all the SEI messages into one NALU.

       max_dec_frame_buffering
	   Maximum number of frames buffered in the DPB.

       look_ahead
	   Use VBR algorithm with look ahead.

       look_ahead_depth
	   Depth of look ahead in number frames.

       look_ahead_downsampling
	   Downscaling factor for the frames saved for the lookahead analysis.

	   unknown
	   auto
	   off
	   2x
	   4x
       int_ref_type
	   Specifies intra refresh type. The major goal of  intra  refresh  is
	   improvement	of  error  resilience  without	significant  impact on
	   encoded bitstream size caused by I frames. The SDK encoder achieves
	   this by encoding part of each frame in refresh  cycle  using	 intra
	   MBs.	 none  means  no  refresh. vertical means vertical refresh, by
	   column of MBs. horizontal means horizontal refresh, by rows of MBs.
	   slice means horizontal refresh by slices  without  overlapping.  In
	   case	 of  slice,  in_ref_cycle_size	is  ignored.  To  enable intra
	   refresh, B frame should be set to 0.

       int_ref_cycle_size
	   Specifies number of pictures within refresh cycle starting from  2.
	   0 and 1 are invalid values.

       int_ref_qp_delta
	   Specifies  QP  difference  for  inserted  intra MBs. This is signed
	   value in [-51, 51] range if	target	encoding  bit-depth  for  luma
	   samples  is 8 and this range is [-63, 63] for 10 bit-depth or [-75,
	   75] for 12 bit-depth respectively.

       int_ref_cycle_dist
	   Distance between the beginnings  of	the  intra-refresh  cycles  in
	   frames.

       profile
	   unknown
	   baseline
	   main
	   high
       a53cc
	   Use A53 Closed Captions (if available).

       aud Insert the Access Unit Delimiter NAL.

       mfmode
	   Multi-Frame Mode.

	   off
	   auto
       repeat_pps
	   Repeat pps for every frame.

       max_qp_i
	   Maximum video quantizer scale for I frame.

       min_qp_i
	   Minimum video quantizer scale for I frame.

       max_qp_p
	   Maximum video quantizer scale for P frame.

       min_qp_p
	   Minimum video quantizer scale for P frame.

       max_qp_b
	   Maximum video quantizer scale for B frame.

       min_qp_b
	   Minimum video quantizer scale for B frame.

       scenario
	   Provides  a	hint  to  encoder  about the scenario for the encoding
	   session.

	   unknown
	   displayremoting
	   videoconference
	   archive
	   livestreaming
	   cameracapture
	   videosurveillance
	   gamestreaming
	   remotegaming
       avbr_accuracy
	   Accuracy of the AVBR ratecontrol (unit of tenth of percent).

       avbr_convergence
	   Convergence of the AVBR ratecontrol (unit of 100 frames)

	   The parameters  avbr_accuracy  and  avbr_convergence	 are  for  the
	   average  variable  bitrate control (AVBR) algorithm.	 The algorithm
	   focuses on overall encoding quality	while  meeting	the  specified
	   bitrate,  target_bitrate,  within the accuracy range avbr_accuracy,
	   after a avbr_Convergence period. This method does  not  follow  HRD
	   and the instant bitrate is not capped or padded.

       skip_frame
	   Use	 per-frame   metadata  "qsv_skip_frame"	 to  skip  frame  when
	   encoding. This option defines the usage of this metadata.

	   no_skip
	       Frame skipping is disabled.

	   insert_dummy
	       Encoder inserts into bitstream frame where all macroblocks  are
	       encoded as skipped.

	   insert_nothing
	       Similar	to  insert_dummy,  but	encoder	 inserts  nothing into
	       bitstream. The skipped  frames  are  still  used	 in  brc.  For
	       example, gop still include skipped frames, and the frames after
	       skipped frames will be larger in size.

	   brc_only
	       skip_frame  metadata  indicates	the  number  of	 missed frames
	       before the current frame.

       HEVC Options

       These options are used by hevc_qsv

       extbrc
	   Extended bitrate control.

       recovery_point_sei
	   Set this flag to insert the	recovery  point	 SEI  message  at  the
	   beginning of every intra refresh cycle.

       rdo Enable rate distortion optimization.

       max_frame_size
	   Maximum encoded frame size in bytes.

       max_frame_size_i
	   Maximum  encoded frame size for I frames in bytes. If this value is
	   set as larger than zero,  then  for	I  frames  the	value  set  by
	   max_frame_size is ignored.

       max_frame_size_p
	   Maximum  encoded frame size for P frames in bytes. If this value is
	   set as larger than zero,  then  for	P  frames  the	value  set  by
	   max_frame_size is ignored.

       max_slice_size
	   Maximum encoded slice size in bytes.

       mbbrc
	   Setting  this  flag	enables	 macroblock level bitrate control that
	   generally improves subjective visual quality.  Enabling  this  flag
	   may	have  negative	impact	on  performance	 and  objective visual
	   quality metric.

       low_delay_brc
	   Setting this flag turns  on	or  off	 LowDelayBRC  feautre  in  qsv
	   plugin,  which  provides  more accurate bitrate control to minimize
	   the variance of bitstream size frame by  frame.  Value:  -1-default
	   0-off 1-on

       adaptive_i
	   This	 flag  controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       p_strategy
	   Enable P-pyramid: 0-default 1-simple 2-pyramid(bf need to be set to
	   0).

       b_strategy
	   This option controls usage of B frames as reference.

       dblk_idc
	   This option disable deblocking. It has value in range 0~2.

       idr_interval
	   Distance (in I-frames) between IDR frames.

	   begin_only
	       Output an IDR-frame only at the beginning of the stream.

       load_plugin
	   A user plugin to load in an internal session.

	   none
	   hevc_sw
	   hevc_hw
       load_plugins
	   A :-separate list of hexadecimal plugin UIDs to load in an internal
	   session.

       look_ahead_depth
	   Depth of look ahead in number frames, available when extbrc	option
	   is enabled.

       profile
	   Set the encoding profile (scc requires libmfx >= 1.32).

	   unknown
	   main
	   main10
	   mainsp
	   rext
	   scc
       tier
	   Set	the  encoding  tier  (only  level >= 4 can support high tier).
	   This option only takes effect when the level option is specified.

	   main
	   high
       gpb 1: GPB (generalized P/B frame)

	   0: regular P frame.

       tile_cols
	   Number of columns for tiled encoding.

       tile_rows
	   Number of rows for tiled encoding.

       aud Insert the Access Unit Delimiter NAL.

       pic_timing_sei
	   Insert picture timing SEI with pic_struct_syntax element.

       transform_skip
	   Turn this option ON to enable transformskip.	 It  is	 supported  on
	   platform equal or newer than ICL.

       int_ref_type
	   Specifies  intra  refresh  type. The major goal of intra refresh is
	   improvement of  error  resilience  without  significant  impact  on
	   encoded bitstream size caused by I frames. The SDK encoder achieves
	   this	 by  encoding  part of each frame in refresh cycle using intra
	   MBs. none means no refresh. vertical	 means	vertical  refresh,  by
	   column of MBs. horizontal means horizontal refresh, by rows of MBs.
	   slice  means	 horizontal  refresh by slices without overlapping. In
	   case of  slice,  in_ref_cycle_size  is  ignored.  To	 enable	 intra
	   refresh, B frame should be set to 0.

       int_ref_cycle_size
	   Specifies  number of pictures within refresh cycle starting from 2.
	   0 and 1 are invalid values.

       int_ref_qp_delta
	   Specifies QP difference for inserted	 intra	MBs.  This  is	signed
	   value  in  [-51,  51]  range	 if target encoding bit-depth for luma
	   samples is 8 and this range is [-63, 63] for 10 bit-depth or	 [-75,
	   75] for 12 bit-depth respectively.

       int_ref_cycle_dist
	   Distance  between  the  beginnings  of  the intra-refresh cycles in
	   frames.

       max_qp_i
	   Maximum video quantizer scale for I frame.

       min_qp_i
	   Minimum video quantizer scale for I frame.

       max_qp_p
	   Maximum video quantizer scale for P frame.

       min_qp_p
	   Minimum video quantizer scale for P frame.

       max_qp_b
	   Maximum video quantizer scale for B frame.

       min_qp_b
	   Minimum video quantizer scale for B frame.

       scenario
	   Provides a hint to encoder about  the  scenario  for	 the  encoding
	   session.

	   unknown
	   displayremoting
	   videoconference
	   archive
	   livestreaming
	   cameracapture
	   videosurveillance
	   gamestreaming
	   remotegaming
       avbr_accuracy
	   Accuracy of the AVBR ratecontrol (unit of tenth of percent).

       avbr_convergence
	   Convergence of the AVBR ratecontrol (unit of 100 frames)

	   The	parameters  avbr_accuracy  and	avbr_convergence  are  for the
	   average variable bitrate control (AVBR) algorithm.	The  algorithm
	   focuses  on	overall	 encoding  quality while meeting the specified
	   bitrate, target_bitrate, within the accuracy	 range	avbr_accuracy,
	   after  a  avbr_Convergence  period. This method does not follow HRD
	   and the instant bitrate is not capped or padded.

       skip_frame
	   Use	per-frame  metadata  "qsv_skip_frame"  to  skip	  frame	  when
	   encoding. This option defines the usage of this metadata.

	   no_skip
	       Frame skipping is disabled.

	   insert_dummy
	       Encoder	inserts into bitstream frame where all macroblocks are
	       encoded as skipped.

	   insert_nothing
	       Similar to  insert_dummy,  but  encoder	inserts	 nothing  into
	       bitstream.  The	skipped	 frames	 are  still  used  in brc. For
	       example, gop still include skipped frames, and the frames after
	       skipped frames will be larger in size.

	   brc_only
	       skip_frame metadata  indicates  the  number  of	missed	frames
	       before the current frame.

       MPEG2 Options

       These options are used by mpeg2_qsv

       profile
	   unknown
	   simple
	   main
	   high

       VP9 Options

       These options are used by vp9_qsv

       profile
	   unknown
	   profile0
	   profile1
	   profile2
	   profile3
       tile_cols
	   Number of columns for tiled encoding (requires libmfx >= 1.29).

       tile_rows
	   Number of rows for tiled encoding (requires libmfx  >= 1.29).

       AV1 Options

       These options are used by av1_qsv (requires libvpl).

       profile
	   unknown
	   main
       tile_cols
	   Number of columns for tiled encoding.

       tile_rows
	   Number of rows for tiled encoding.

       adaptive_i
	   This	 flag  controls insertion of I frames by the QSV encoder. Turn
	   ON this flag to allow changing of frame type from P and B to I.

       adaptive_b
	   This flag controls changing of frame type from B to P.

       b_strategy
	   This option controls usage of B frames as reference.

       extbrc
	   Extended bitrate control.

       look_ahead_depth
	   Depth of look ahead in number frames, available when extbrc	option
	   is enabled.

       low_delay_brc
	   Setting  this  flag	turns  on  or  off  LowDelayBRC feautre in qsv
	   plugin, which provides more accurate bitrate	 control  to  minimize
	   the	variance  of  bitstream size frame by frame. Value: -1-default
	   0-off 1-on

       max_frame_size
	   Set the allowed max size in bytes for each frame. If the frame size
	   exceeds the limitation, encoder will adjust the QP value to control
	   the frame size.  Invalid in CQP rate control mode.

   snow
       Options

       iterative_dia_size
	   dia size for the iterative motion estimation

   VAAPI encoders
       Wrappers for hardware encoders accessible via VAAPI.

       These encoders only accept input in VAAPI hardware  surfaces.   If  you
       have  input  in software frames, use the hwupload filter to upload them
       to the GPU.

       The following standard libavcodec options are used:

       •   g / gop_size

       •   bf / max_b_frames

       •   profile

	   If not set, this will be determined automatically from  the	format
	   of the input frames and the profiles supported by the driver.

       •   level

       •   b / bit_rate

       •   maxrate / rc_max_rate

       •   bufsize / rc_buffer_size

       •   rc_init_occupancy / rc_initial_buffer_occupancy

       •   compression_level

	   Speed / quality tradeoff: higher values are faster / worse quality.

       •   q / global_quality

	   Size / quality tradeoff: higher values are smaller / worse quality.

       •   qmin

       •   qmax

       •   i_qfactor / i_quant_factor

       •   i_qoffset / i_quant_offset

       •   b_qfactor / b_quant_factor

       •   b_qoffset / b_quant_offset

       •   slices

       All encoders support the following options:

       low_power
	   Some	 drivers/platforms  offer  a  second  encoder  for some codecs
	   intended to use less power than the default encoder;	 setting  this
	   option  will attempt to use that encoder.  Note that it may support
	   a reduced feature set, so some other options may not	 be  available
	   in this mode.

       idr_interval
	   Set	the  number  of normal intra frames between full-refresh (IDR)
	   frames in open-GOP mode.  The intra frames  are  still  IRAPs,  but
	   will	 not include global headers and may have non-decodable leading
	   pictures.

       b_depth
	   Set the B-frame reference depth.  When set to  one  (the  default),
	   all	B-frames  will	refer  only  to	 P-  or I-frames.  When set to
	   greater values multiple layers of B-frames will be present,	frames
	   in each layer only referring to frames in higher layers.

       async_depth
	   Maximum  processing	parallelism.  Increase	this to improve single
	   channel performance. This option doesn't  work  if  driver  doesn't
	   implement  vaSyncBuffer function. Please make sure there are enough
	   hw_frames allocated if a large number of async_depth is used.

       max_frame_size
	   Set the allowed max size in bytes for each frame. If the frame size
	   exceeds the limitation, encoder will adjust the QP value to control
	   the frame size.  Invalid in CQP rate control mode.

       rc_mode
	   Set the rate control mode to use.  A given driver may only  support
	   a subset of modes.

	   Possible modes:

	   auto
	       Choose  the  mode automatically based on driver support and the
	       other options.  This is the default.

	   CQP Constant-quality.

	   CBR Constant-bitrate.

	   VBR Variable-bitrate.

	   ICQ Intelligent constant-quality.

	   QVBR
	       Quality-defined variable-bitrate.

	   AVBR
	       Average variable bitrate.

       Each encoder also has its own specific options:

       av1_vaapi
	   profile sets the value of seq_profile.   tier  sets	the  value  of
	   seq_tier.  level sets the value of seq_level_idx.

	   tiles
	       Set  the	 number	 of  tiles  to encode the input video with, as
	       columns x rows.	(default is auto, which means use minimal tile
	       column/row number).

	   tile_groups
	       Set tile groups number. All the tiles will  be  distributed  as
	       evenly as possible to each tile group. (default is 1).

       h264_vaapi
	   profile     sets    the    value    of    profile_idc    and	   the
	   constraint_set*_flags.  level sets the value of level_idc.

	   coder
	       Set entropy encoder (default is cabac).	Possible values:

	       ac
	       cabac
		   Use CABAC.

	       vlc
	       cavlc
		   Use CAVLC.

	   aud Include access unit delimiters in the stream (not  included  by
	       default).

	   sei Set  SEI	 message  types	 to  include.  Some combination of the
	       following values:

	       identifier
		   Include   a	 user_data_unregistered	  message   containing
		   information about the encoder.

	       timing
		   Include  picture  timing  parameters	 (buffering_period and
		   pic_timing messages).

	       recovery_point
		   Include recovery points where  appropriate  (recovery_point
		   messages).

       hevc_vaapi
	   profile  and	 level	set  the  values  of  general_profile_idc  and
	   general_level_idc respectively.

	   aud Include access unit delimiters in the stream (not  included  by
	       default).

	   tier
	       Set  general_tier_flag.	 This  may affect the level chosen for
	       the stream if it is not explicitly specified.

	   sei Set SEI message types to	 include.   Some  combination  of  the
	       following values:

	       hdr Include   HDR   metadata   if  the  input  frames  have  it
		   (mastering_display_colour_volume  and   content_light_level
		   messages).

	   tiles
	       Set  the	 number	 of  tiles  to encode the input video with, as
	       columns x rows.	Larger numbers allow  greater  parallelism  in
	       both encoding and decoding, but may decrease coding efficiency.

       mjpeg_vaapi
	   Only	 baseline  DCT encoding is supported.  The encoder always uses
	   the standard	 quantisation  and  huffman  tables  -	global_quality
	   scales the standard quantisation table (range 1-100).

	   For	YUV,  4:2:0,  4:2:2 and 4:4:4 subsampling modes are supported.
	   RGB is also supported, and will create an RGB JPEG.

	   jfif
	       Include JFIF header in each frame (not included by default).

	   huffman
	       Include standard huffman tables (on by default).	 Turning  this
	       off will save a few hundred bytes in each output frame, but may
	       lose  compatibility  with  some JPEG decoders which don't fully
	       handle MJPEG.

       mpeg2_vaapi
	   profile and level set the value of profile_and_level_indication.

       vp8_vaapi
	   B-frames are not supported.

	   global_quality sets	the  q_idx  used  for  non-key	frames	(range
	   0-127).

	   loop_filter_level
	   loop_filter_sharpness
	       Manually set the loop filter parameters.

       vp9_vaapi
	   global_quality sets the q_idx used for P-frames (range 0-255).

	   loop_filter_level
	   loop_filter_sharpness
	       Manually set the loop filter parameters.

	   B-frames  are  supported, but the output stream is always in encode
	   order rather than display order.  If B-frames are enabled,  it  may
	   be  necessary to use the vp9_raw_reorder bitstream filter to modify
	   the output stream to display frames in the correct order.

	   Only normal frames are  produced  -	the  vp9_superframe  bitstream
	   filter  may	be  required  to  produce  a  stream  usable  with all
	   decoders.

   vbn
       Vizrt Binary Image encoder.

       This format is used by the broadcast vendor  Vizrt  for	quick  texture
       streaming.   Advanced features of the format such as LZW compression of
       texture data or generation of mipmaps are not supported.

       Options

       format string
	   Sets the texture compression used by the VBN	 file.	Can  be	 dxt1,
	   dxt5 or raw. Default is dxt5.

   vc2
       SMPTE  VC-2  (previously BBC Dirac Pro). This codec was primarily aimed
       at professional broadcasting but since it supports yuv420,  yuv422  and
       yuv444 at 8 (limited range or full range), 10 or 12 bits, this makes it
       suitable for other tasks which require low overhead and low compression
       (like screen recording).

       Options

       b   Sets	 target	 video	bitrate.  Usually  that's  around  1:6	of the
	   uncompressed video bitrate  (e.g.  for  1920x1080  50fps  yuv422p10
	   that's  around  400Mbps).  Higher values (close to the uncompressed
	   bitrate) turn on lossless compression mode.

       field_order
	   Enables field coding when set (e.g. to tt - top  field  first)  for
	   interlaced  inputs.	Should	increase  compression  with interlaced
	   content as it splits the fields and encodes each separately.

       wavelet_depth
	   Sets the total amount of wavelet transforms to apply, between 1 and
	   5 (default).	 Lower values reduce  compression  and	quality.  Less
	   capable  decoders may not be able to handle values of wavelet_depth
	   over 3.

       wavelet_type
	   Sets the transform  type.  Currently	 only  5_3  (LeGall)  and  9_7
	   (Deslauriers-Dubuc)	are  implemented,  with 9_7 being the one with
	   better compression and thus is the default.

       slice_width
       slice_height
	   Sets the slice size for each slice. Larger values result in	better
	   compression.	  For  compatibility  with other more limited decoders
	   use slice_width of 32 and slice_height of 8.

       tolerance
	   Sets the  undershoot	 tolerance  of	the  rate  control  system  in
	   percent. This is to prevent an expensive search from being run.

       qm  Sets	 the  quantization  matrix  preset  to	use by default or when
	   wavelet_depth is set to 5

	   -   default	Uses  the  default  quantization   matrix   from   the
	       specifications,	extended with values for the fifth level. This
	       provides a good balance between	keeping	 detail	 and  omitting
	       artifacts.

	   -   flat  Use  a  completely	 zeroed	 out quantization matrix. This
	       increases PSNR  but  might  reduce  perception.	Use  in	 bogus
	       benchmarks.

	   -   color   Reduces	detail	but  attempts  to  preserve  color  at
	       extremely low bitrates.

SUBTITLES ENCODERS
   dvdsub
       This codec encodes the bitmap subtitle format that  is  used  in	 DVDs.
       Typically  they	are  stored  in VOBSUB file pairs (*.idx + *.sub), and
       they can also be used in Matroska files.

       Options

       palette
	   Specify the global palette used by the bitmaps.

	   The format for this	option	is  a  string  containing  16  24-bits
	   hexadecimal	numbers	 (without  0x prefix) separated by commas, for
	   example "0d00ee, ee450d, 101010, eaeaea,  0ce60b,  ec14ed,  ebff0b,
	   0d617a,  7b7b7b,  d1d1d1,  7b2a0e,  0d950c, 0f007b, cf0dec, cfa80c,
	   7c127b".

       even_rows_fix
	   When set to 1, enable a work-around that makes the number of	 pixel
	   rows even in all subtitles.	This fixes a problem with some players
	   that	 cut off the bottom row if the number is odd.  The work-around
	   just adds a fully transparent row if needed.	 The overhead is  low,
	   typically one byte per subtitle on average.

	   By default, this work-around is disabled.

BITSTREAM FILTERS
       When  you  configure  your  FFmpeg  build,  all the supported bitstream
       filters are enabled by default. You can list all available  ones	 using
       the configure option "--list-bsfs".

       You  can	 disable  all the bitstream filters using the configure option
       "--disable-bsfs", and selectively enable any bitstream filter using the
       option "--enable-bsf=BSF", or you can disable  a	 particular  bitstream
       filter using the option "--disable-bsf=BSF".

       The  option  "-bsfs"  of the ff* tools will display the list of all the
       supported bitstream filters included in your build.

       The ff* tools have a -bsf option applied per stream,  taking  a	comma-
       separated  list	of  filters,  whose  parameters follow the filter name
       after a '='.

	       ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

       Below is a description of the currently	available  bitstream  filters,
       with their parameters, if any.

   aac_adtstoasc
       Convert	MPEG-2/4  AAC  ADTS  to an MPEG-4 Audio Specific Configuration
       bitstream.

       This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS
       header and removes the ADTS header.

       This filter is required for example when copying an AAC stream  from  a
       raw  ADTS  AAC or an MPEG-TS container to MP4A-LATM, to an FLV file, or
       to MOV/MP4 files and related formats such as 3GP or  M4A.  Please  note
       that it is auto-inserted for MP4A-LATM and MOV/MP4 and related formats.

   av1_metadata
       Modify metadata embedded in an AV1 stream.

       td  Insert  or  remove temporal delimiter OBUs in all temporal units of
	   the stream.

	   insert
	       Insert a TD at the beginning of every TU which does not already
	       have one.

	   remove
	       Remove the TD from the beginning of every TU which has one.

       color_primaries
       transfer_characteristics
       matrix_coefficients
	   Set the color description fields in the  stream  (see  AV1  section
	   6.4.2).

       color_range
	   Set the color range in the stream (see AV1 section 6.4.2; note that
	   this	 cannot	 be  set  for  streams	using  BT.709  primaries, sRGB
	   transfer characteristic and identity (RGB) matrix coefficients).

	   tv  Limited range.

	   pc  Full range.

       chroma_sample_position
	   Set the chroma sample location  in  the  stream  (see  AV1  section
	   6.4.2).  This can only be set for 4:2:0 streams.

	   vertical
	       Left position (matching the default in MPEG-2 and H.264).

	   colocated
	       Top-left position.

       tick_rate
	   Set	the  tick rate (time_scale / num_units_in_display_tick) in the
	   timing info in the sequence header.

       num_ticks_per_picture
	   Set the number of ticks in  each  picture,  to  indicate  that  the
	   stream  has	a  fixed  framerate.  Ignored if tick_rate is not also
	   set.

       delete_padding
	   Deletes Padding OBUs.

   chomp
       Remove zero padding at the end of a packet.

   dca_core
       Extract the core from a DCA/DTS stream,	dropping  extensions  such  as
       DTS-HD.

   dump_extra
       Add extradata to the beginning of the filtered packets except when said
       packets already exactly begin with the extradata that is intended to be
       added.

       freq
	   The additional argument specifies which packets should be filtered.
	   It accepts the values:

	   k
	   keyframe
	       add extradata to all key packets

	   e
	   all add extradata to all packets

       If not specified it is assumed k.

       For  example  the following ffmpeg command forces a global header (thus
       disabling individual packet headers) in the H.264 packets generated  by
       the "libx264" encoder, but corrects them by adding the header stored in
       extradata to the key packets:

	       ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

   dv_error_marker
       Blocks  in DV which are marked as damaged are replaced by blocks of the
       specified color.

       color
	   The color to replace damaged blocks by

       sta A 16 bit mask which specifies which of the 16 possible error status
	   values are to be replaced by colored blocks. 0xFFFE is the  default
	   which replaces all non 0 error status values.

	   ok  No error, no concealment

	   err Error, No concealment

	   res Reserved

	   notok
	       Error or concealment

	   notres
	       Not reserved

	   Aa, Ba, Ca, Ab, Bb, Cb, A, B, C, a, b, erri, erru
	       The specific error status code

	   see	     page	44-46	    or	     section	   5.5	    of
	   <http://web.archive.org/web/20060927044735/http://www.smpte.org/smpte_store/standards/pdf/s314m.pdf>

   eac3_core
       Extract the core from a E-AC-3 stream, dropping extra channels.

   extract_extradata
       Extract the in-band extradata.

       Certain codecs  allow  the  long-term  headers  (e.g.  MPEG-2  sequence
       headers,	 or  H.264/HEVC	 (VPS/)SPS/PPS)	 to be transmitted either "in-
       band" (i.e. as a part of the bitstream containing the coded frames)  or
       "out of band" (e.g. on the container level). This latter form is called
       "extradata" in FFmpeg terminology.

       This  bitstream	filter	detects	 the  in-band  headers	and makes them
       available as extradata.

       remove
	   When this option is enabled, the long-term headers are removed from
	   the bitstream after extraction.

   filter_units
       Remove units with types in or not in a given set from the stream.

       pass_types
	   List of unit types or ranges of unit types to  pass	through	 while
	   removing  all others.  This is specified as a '|'-separated list of
	   unit type values or ranges of values with '-'.

       remove_types
	   Identical to pass_types, except the units in the given set  removed
	   and all others passed through.

       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.

	   pass
	   insert
	   remove

	   Default is pass.

       sample_aspect_ratio
	   Set the sample aspect ratio of the stream in	 the  VUI  parameters.
	   See H.264 table E-1.

       overscan_appropriate_flag
	   Set	whether	 the  stream is suitable for display using overscan or
	   not (see H.264 section E.2.1).

       video_format
       video_full_range_flag
	   Set the video format in the stream (see  H.264  section  E.2.1  and
	   table E-2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
	   Set	the  colour description in the stream (see H.264 section E.2.1
	   and tables E-3, E-4 and E-5).

       chroma_sample_loc_type
	   Set the chroma sample location in the  stream  (see	H.264  section
	   E.2.1 and figure E-1).

       tick_rate
	   Set	the  tick  rate	 (time_scale  /	 num_units_in_tick) in the VUI
	   parameters.	This is the smallest time unit	representable  in  the
	   stream,  and	 in many cases represents the field rate of the stream
	   (double the frame rate).

       fixed_frame_rate_flag
	   Set whether	the  stream  has  fixed	 framerate  -  typically  this
	   indicates that the framerate is exactly half the tick rate, but the
	   exact meaning is dependent on interlacing and the picture structure
	   (see H.264 section E.2.1 and table E-6).

       zero_new_constraint_set_flags
	   Zero	 constraint_set4_flag  and  constraint_set5_flag  in  the SPS.
	   These bits were reserved in a previous version of the  H.264	 spec,
	   and	thus  some  hardware  decoders	require	 these to be zero. The
	   result of zeroing this is still a valid bitstream.

       crop_left
       crop_right
       crop_top
       crop_bottom
	   Set the frame cropping offsets  in  the  SPS.   These  values  will
	   replace the current ones if the stream is already cropped.

	   These  fields  are  set in pixels.  Note that some sizes may not be
	   representable  if  the  chroma  is  subsampled  or  the  stream  is
	   interlaced (see H.264 section 7.4.2.1.1).

       sei_user_data
	   Insert  a  string as SEI unregistered user data.  The argument must
	   be of the form  UUID+string,	 where	the  UUID  is  as  hex	digits
	   possibly separated by hyphens, and the string can be anything.

	   For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert
	   the string ``hello'' associated with the given UUID.

       delete_filler
	   Deletes both filler NAL units and filler SEI messages.

       display_orientation
	   Insert,  extract  or	 remove Display orientation SEI messages.  See
	   H.264 section D.1.27 and D.2.27 for syntax and semantics.

	   pass
	   insert
	   remove
	   extract

	   Default is pass.

	   Insert mode works in conjunction with "rotate" and "flip"  options.
	   Any	pre-existing  Display  orientation messages will be removed in
	   insert or remove mode.  Extract mode attaches the display matrix to
	   the packet as side data.

       rotate
	   Set rotation in display orientation	SEI  (anticlockwise  angle  in
	   degrees).  Range is -360 to +360. Default is NaN.

       flip
	   Set flip in display orientation SEI.

	   horizontal
	   vertical

	   Default is unset.

       level
	   Set	the  level  in the SPS.	 Refer to H.264 section A.3 and tables
	   A-1 to A-5.

	   The argument must be the name of a  level  (for  example,  4.2),  a
	   level_idc  value  (for  example,  42),  or  the  special  name auto
	   indicating that the filter should attempt to guess the  level  from
	   the input stream properties.

   h264_mp4toannexb
       Convert	an  H.264  bitstream  from  length prefixed mode to start code
       prefixed	 mode  (as  defined  in	 the  Annex  B	of  the	 ITU-T	 H.264
       specification).

       This  is	 required  by  some  streaming	formats,  typically the MPEG-2
       transport stream format (muxer "mpegts").

       For example to remux an MP4 file containing an H.264 stream  to	mpegts
       format with ffmpeg, you can use the command:

	       ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

       Please  note  that  this	 filter	 is  auto-inserted  for MPEG-TS (muxer
       "mpegts") and raw H.264 (muxer "h264") output formats.

   h264_redundant_pps
       This applies a specific fixup to some  Blu-ray  streams	which  contain
       redundant  PPSs	modifying  irrelevant  parameters  of the stream which
       confuse other transformations which require correct extradata.

   hevc_metadata
       Modify metadata embedded in an HEVC stream.

       aud Insert or remove AUD NAL units in all access units of the stream.

	   insert
	   remove
       sample_aspect_ratio
	   Set the sample aspect ratio in the stream in the VUI parameters.

       video_format
       video_full_range_flag
	   Set the video format in the stream (see  H.265  section  E.3.1  and
	   table E.2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
	   Set	the  colour description in the stream (see H.265 section E.3.1
	   and tables E.3, E.4 and E.5).

       chroma_sample_loc_type
	   Set the chroma sample location in the  stream  (see	H.265  section
	   E.3.1 and figure E.1).

       tick_rate
	   Set	the  tick  rate	 in  the  VPS and VUI parameters (time_scale /
	   num_units_in_tick). Combined with num_ticks_poc_diff_one, this  can
	   set	a constant framerate in the stream.  Note that it is likely to
	   be overridden by container parameters  when	the  stream  is	 in  a
	   container.

       num_ticks_poc_diff_one
	   Set	poc_proportional_to_timing_flag	 in  VPS  and VUI and use this
	   value to  set  num_ticks_poc_diff_one_minus1	 (see  H.265  sections
	   7.4.3.1 and E.3.1).	Ignored if tick_rate is not also set.

       crop_left
       crop_right
       crop_top
       crop_bottom
	   Set	the  conformance  window  cropping  offsets in the SPS.	 These
	   values will replace the current  ones  if  the  stream  is  already
	   cropped.

	   These  fields  are  set in pixels.  Note that some sizes may not be
	   representable  if  the  chroma   is	 subsampled   (H.265   section
	   7.4.3.2.1).

       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
	   Accepts an expression whose evaluation  per-packet  determines  how
	   often  bytes	 in that packet will be modified. A value below 0 will
	   result in a variable frequency.  Default is 0 which results	in  no
	   modification.  However,  if	neither	 amount nor drop is specified,
	   amount will be set to -1. See below for accepted variables.

       drop
	   Accepts an expression evaluated per-packet whose  value  determines
	   whether  that  packet  is  dropped.	Evaluation to a positive value
	   results in the packet being dropped. Evaluation to a negative value
	   results in a variable chance of it being dropped,  roughly  inverse
	   in  proportion  to  the  magnitude of the value. Default is 0 which
	   results in no drops. See below for accepted variables.

       dropamount
	   Accepts a non-negative integer, which assigns a variable chance  of
	   it  being  dropped,	roughly	 inverse  in  proportion to the value.
	   Default is 0 which results in no drops. This	 option	 is  kept  for
	   backwards  compatibility  and  is  equivalent  to setting drop to a
	   negative value with the same magnitude i.e. "dropamount=4"  is  the
	   same as "drop=-4". Ignored if drop is also specified.

       Both  "amount"  and  "drop" accept expressions containing the following
       variables:

       n   The index of the packet, starting from zero.

       tb  The timebase for packet timestamps.

       pts Packet presentation timestamp.

       dts Packet decoding timestamp.

       nopts
	   Constant representing AV_NOPTS_VALUE.

       startpts
	   First non-AV_NOPTS_VALUE PTS seen in the stream.

       startdts
	   First non-AV_NOPTS_VALUE DTS seen in the stream.

       duration
       d   Packet duration, in timebase units.

       pos Packet position in input; may be -1 when unknown or not set.

       size
	   Packet size, in bytes.

       key Whether packet is marked as a keyframe.

       state
	   A pseudo random integer, primarily  derived	from  the  content  of
	   packet payload.

       Examples

       Apply modification to every byte but don't drop any packets.

	       ffmpeg -i INPUT -c copy -bsf noise=1 output.mkv

       Drop  every  video  packet not marked as a keyframe after timestamp 30s
       but do not modify any of the remaining packets.

	       ffmpeg -i INPUT -c copy -bsf:v noise=drop='gt(t\,30)*not(key)' output.mkv

       Drop one second of audio every 10 seconds and add some random noise  to
       the rest.

	       ffmpeg -i INPUT -c copy -bsf:a noise=amount=-1:drop='between(mod(t\,10)\,9\,10)' output.mkv

   null
       This bitstream filter passes the packets through unchanged.

   pcm_rechunk
       Repacketize  PCM	 audio	to  a  fixed number of samples per packet or a
       fixed packet rate per second. This is similar to the asetnsamples audio
       filter but works on audio packets instead of audio frames.

       nb_out_samples, n
	   Set the number of samples per each output audio packet. The	number
	   is  intended	 as  the  number  of samples per each channel. Default
	   value is 1024.

       pad, p
	   If set to 1, the  filter  will  pad	the  last  audio  packet  with
	   silence,  so	 that  it  will contain the same number of samples (or
	   roughly the same number of samples, see frame_rate) as the previous
	   ones. Default value is 1.

       frame_rate, r
	   This option makes the filter output a fixed number of  packets  per
	   second  instead  of	a  fixed  number of samples per packet. If the
	   audio sample rate is not divisible  by  the	frame  rate  then  the
	   number  of  samples	will not be constant but will vary slightly so
	   that each packet will start as  close  to  the  frame  boundary  as
	   possible. Using this option has precedence over nb_out_samples.

       You  can	 generate  the	well known 1602-1601-1602-1601-1602 pattern of
       48kHz audio for NTSC frame rate using the frame_rate option.

	       ffmpeg -f lavfi -i sine=r=48000:d=1 -c pcm_s16le -bsf pcm_rechunk=r=30000/1001 -f framecrc -

   pgs_frame_merge
       Merge a sequence of PGS	Subtitle  segments  ending  with  an  "end  of
       display set" segment into a single packet.

       This  is	 required by some containers that support PGS subtitles (muxer
       "matroska").

   prores_metadata
       Modify color property metadata embedded in prores stream.

       color_primaries
	   Set the color primaries.  Available values are:

	   auto
	       Keep the same color primaries property (default).

	   unknown
	   bt709
	   bt470bg
	       BT601 625

	   smpte170m
	       BT601 525

	   bt2020
	   smpte431
	       DCI P3

	   smpte432
	       P3 D65

       transfer_characteristics
	   Set the color transfer.  Available values are:

	   auto
	       Keep the same transfer characteristics property (default).

	   unknown
	   bt709
	       BT 601, BT 709, BT 2020

	   smpte2084
	       SMPTE ST 2084

	   arib-std-b67
	       ARIB STD-B67

       matrix_coefficients
	   Set the matrix coefficient.	Available values are:

	   auto
	       Keep the same colorspace property (default).

	   unknown
	   bt709
	   smpte170m
	       BT 601

	   bt2020nc

       Set Rec709 colorspace for each frame of the file

	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt709:color_trc=bt709:colorspace=bt709 output.mov

       Set Hybrid Log-Gamma parameters for each frame of the file

	       ffmpeg -i INPUT -c copy -bsf:v prores_metadata=color_primaries=bt2020:color_trc=arib-std-b67:colorspace=bt2020nc output.mov

   remove_extra
       Remove extradata from packets.

       It accepts the following parameter:

       freq
	   Set which frame types to remove extradata from.

	   k   Remove extradata from non-keyframes only.

	   keyframe
	       Remove extradata from keyframes only.

	   e, all
	       Remove extradata from all frames.

   setts
       Set PTS and DTS in packets.

       It accepts the following parameters:

       ts
       pts
       dts Set expressions for PTS, DTS or both.

       duration
	   Set expression for duration.

       time_base
	   Set output time base.

       The expressions are evaluated through the eval API and can contain  the
       following constants:

       N   The count of the input packet. Starting from 0.

       TS  The	demux  timestamp  in  input in case of "ts" or "dts" option or
	   presentation timestamp in case of "pts" option.

       POS The original position in the file of the packet,  or	 undefined  if
	   undefined for the current packet

       DTS The demux timestamp in input.

       PTS The presentation timestamp in input.

       DURATION
	   The duration in input.

       STARTDTS
	   The DTS of the first packet.

       STARTPTS
	   The PTS of the first packet.

       PREV_INDTS
	   The previous input DTS.

       PREV_INPTS
	   The previous input PTS.

       PREV_INDURATION
	   The previous input duration.

       PREV_OUTDTS
	   The previous output DTS.

       PREV_OUTPTS
	   The previous output PTS.

       PREV_OUTDURATION
	   The previous output duration.

       NEXT_DTS
	   The next input DTS.

       NEXT_PTS
	   The next input PTS.

       NEXT_DURATION
	   The next input duration.

       TB  The timebase of stream packet belongs.

       TB_OUT
	   The output timebase.

       SR  The sample rate of stream packet belongs.

       NOPTS
	   The AV_NOPTS_VALUE constant.

   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.

	   nobuffer
	       Reduce the latency introduced by buffering during initial input
	       streams analysis.

	   nofillin
	       Do  not	fill  in  missing  values in packet fields that can be
	       exactly calculated.

	   noparse
	       Disable AVParsers, this needs "+nofillin" too.

	   sortdts
	       Try to interleave output packets by DTS. At present,  available
	       only for AVIs with an index.

	   Possible values for output files:

	   autobsf
	       Automatically apply bitstream filters as required by the output
	       format. Enabled by default.

	   bitexact
	       Only  write  platform-, build- and time-independent data.  This
	       ensures that file and data checksums are reproducible and match
	       between platforms. Its primary use is for regression testing.

	   flush_packets
	       Write out packets immediately.

	   shortest
	       Stop muxing at the end of  the  shortest	 stream.   It  may  be
	       needed  to  increase max_interleave_delta to avoid flushing the
	       longer streams before EOF.

       seek2any integer (input)
	   Allow seeking to non-keyframes on demuxer level when	 supported  if
	   set to 1.  Default is 0.

       analyzeduration integer (input)
	   Specify  how	 many  microseconds are analyzed to probe the input. A
	   higher value will enable detecting more accurate  information,  but
	   will	 increase  latency.  It defaults to 5,000,000 microseconds = 5
	   seconds.

       cryptokey hexadecimal string (input)
	   Set decryption key.

       indexmem integer (input)
	   Set max memory used for timestamp index (per stream).

       rtbufsize integer (input)
	   Set max memory used for buffering real-time frames.

       fdebug flags (input/output)
	   Print specific debug info.

	   Possible values:

	   ts
       max_delay integer (input/output)
	   Set maximum muxing or demuxing delay in microseconds.

       fpsprobesize integer (input)
	   Set number of frames used to probe fps.

       audio_preload integer (output)
	   Set microseconds by	which  audio  packets  should  be  interleaved
	   earlier.

       chunk_duration integer (output)
	   Set microseconds for each chunk.

       chunk_size integer (output)
	   Set size in bytes for each chunk.

       err_detect, f_err_detect flags (input)
	   Set	error detection flags. "f_err_detect" is deprecated and should
	   be used only via the ffmpeg tool.

	   Possible values:

	   crccheck
	       Verify embedded CRCs.

	   bitstream
	       Detect bitstream specification deviations.

	   buffer
	       Detect improper bitstream length.

	   explode
	       Abort decoding on minor error detection.

	   careful
	       Consider things that violate the spec and have not been seen in
	       the wild as errors.

	   compliant
	       Consider all spec non compliancies as errors.

	   aggressive
	       Consider things that a sane encoder should not do as an error.

       max_interleave_delta integer (output)
	   Set maximum buffering duration for interleaving.  The  duration  is
	   expressed in microseconds, and defaults to 10000000 (10 seconds).

	   To  ensure  all  the streams are interleaved correctly, libavformat
	   will wait until it has at least one packet for each	stream	before
	   actually  writing any packets to the output file. When some streams
	   are	"sparse"  (i.e.	 there	are  large  gaps  between   successive
	   packets), this can result in excessive buffering.

	   This	 field specifies the maximum difference between the timestamps
	   of the first and the last packet in the muxing queue,  above	 which
	   libavformat	will  output  a	 packet	 regardless  of whether it has
	   queued a packet for all the streams.

	   If set to 0, libavformat will continue buffering packets  until  it
	   has	a  packet for each stream, regardless of the maximum timestamp
	   difference between the buffered packets.

       use_wallclock_as_timestamps integer (input)
	   Use wallclock as timestamps if set to 1. Default is 0.

       avoid_negative_ts integer (output)
	   Possible values:

	   make_non_negative
	       Shift timestamps to make them  non-negative.   Also  note  that
	       this  affects  only  leading  negative timestamps, and not non-
	       monotonic negative timestamps.

	   make_zero
	       Shift timestamps so that the first timestamp is 0.

	   auto (default)
	       Enables shifting when required by the target format.

	   disabled
	       Disables shifting of timestamp.

	   When shifting is enabled, all output timestamps are shifted by  the
	   same	 amount.  Audio,  video, and subtitles desynching and relative
	   timestamp differences are preserved compared to how they would have
	   been without shifting.

       skip_initial_bytes integer (input)
	   Set number of bytes to skip before reading header and frames if set
	   to 1.  Default is 0.

       correct_ts_overflow integer (input)
	   Correct single timestamp overflows if set to 1. Default is 1.

       flush_packets integer (output)
	   Flush the underlying I/O stream after each packet.  Default	is  -1
	   (auto),  which  means  that	the underlying protocol will decide, 1
	   enables it, and has the effect of reducing the latency, 0  disables
	   it and may increase IO throughput in some cases.

       output_ts_offset offset (output)
	   Set the output time offset.

	   offset must be a time duration specification, see the Time duration
	   section in the ffmpeg-utils(1) manual.

	   The offset is added by the muxer to the output timestamps.

	   Specifying  a  positive offset means that the corresponding streams
	   are delayed bt the time duration specified in offset. Default value
	   is 0 (meaning that no offset is applied).

       format_whitelist list (input)
	   "," separated list of allowed demuxers. By default all are allowed.

       dump_separator string (input)
	   Separator used to separate the fields printed on the	 command  line
	   about  the  Stream parameters.  For example, to separate the fields
	   with newlines and indentation:

		   ffprobe -dump_separator "
					     "	-i ~/videos/matrixbench_mpeg2.mpg

       max_streams integer (input)
	   Specifies the maximum number of streams. This can be used to reject
	   files that would require too many resources due to a	 large	number
	   of streams.

       skip_estimate_duration_from_pts bool (input)
	   Skip	 estimation  of	 input duration 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.

   aac
       Raw Audio Data Transport Stream AAC demuxer.

       This  demuxer  is  used	to demux an ADTS input containing a single AAC
       stream alongwith any ID3v1/2 or APE tags in it.

   apng
       Animated Portable Network Graphics demuxer.

       This demuxer is used to demux APNG files.  All  headers,	 but  the  PNG
       signature,  up  to  (but	 not  including)  the  first  fcTL  chunk  are
       transmitted as extradata.  Frames are  then  split  as  being  all  the
       chunks between two fcTL ones, or between the last fcTL and IEND chunks.

       -ignore_loop bool
	   Ignore the loop variable in the file if set. Default is enabled.

       -max_fps int
	   Maximum  framerate  in  frames  per second. Default of 0 imposes no
	   limit.

       -default_fps int
	   Default framerate in frames per second when none  is	 specified  in
	   the file (0 meaning as fast as possible). Default is 15.

   asf
       Advanced Systems Format demuxer.

       This demuxer is used to demux ASF files and MMS network streams.

       -no_resync_search bool
	   Do not try to resynchronize by looking for a certain optional start
	   code.

   concat
       Virtual concatenation script demuxer.

       This  demuxer  reads  a	list of files and other directives from a text
       file and demuxes them one after the other, as if all their packets  had
       been muxed together.

       The  timestamps in the files are adjusted so that the first file starts
       at 0 and each next file starts where the previous  one  finishes.  Note
       that  it is done globally and may cause gaps if all streams do not have
       exactly the same length.

       All files must have the same streams  (same  codecs,  same  time	 base,
       etc.).

       The  duration of each file is used to adjust the timestamps of the next
       file: if the duration is incorrect (because it was computed  using  the
       bit-rate	 or  because the file is truncated, for example), it can cause
       artifacts. The  "duration"  directive  can  be  used  to	 override  the
       duration stored in each file.

       Syntax

       The  script  is	a  text file in extended-ASCII, with one directive per
       line.  Empty lines, leading spaces and  lines  starting	with  '#'  are
       ignored. The following directive is recognized:

       "file path"
	   Path	 to  a	file  to  read;	 special characters and spaces must be
	   escaped with backslash or single quotes.

	   All subsequent file-related directives apply to that file.

       "ffconcat version 1.0"
	   Identify the script type and version.

	   To make FFmpeg recognize the format automatically,  this  directive
	   must	 appear	 exactly  as is (no extra space or byte-order-mark) on
	   the very first line of the script.

       "duration dur"
	   Duration of the file. This information can be  specified  from  the
	   file;  specifying  it  here	may  be	 more efficient or help if the
	   information from the file is not available or accurate.

	   If the duration is set for all files, then it is possible  to  seek
	   in the whole concatenated video.

       "inpoint timestamp"
	   In  point of the file. When the demuxer opens the file it instantly
	   seeks to the specified timestamp.  Seeking  is  done	 so  that  all
	   streams can be presented successfully at In point.

	   This directive works best with intra frame codecs, because for non-
	   intra  frame	 ones  you  will  usually get extra packets before the
	   actual In point and the decoded content will	 most  likely  contain
	   frames before In point too.

	   For	each  file,  packets  before  the  file	 In  point  will  have
	   timestamps less than the calculated start  timestamp	 of  the  file
	   (negative in case of the first file), and the duration of the files
	   (if	not  specified	by  the	 "duration" directive) will be reduced
	   based on their specified In point.

	   Because of potential packets before the specified In point,	packet
	   timestamps may overlap between two concatenated files.

       "outpoint timestamp"
	   Out	point  of  the	file.  When  the demuxer reaches the specified
	   decoding timestamp in any of the streams, it handles it as  an  end
	   of  file  condition	and  skips  the	 current and all the remaining
	   packets from all streams.

	   Out point is exclusive, which  means	 that  the  demuxer  will  not
	   output  packets  with  a decoding timestamp greater or equal to Out
	   point.

	   This directive works best with intra frame codecs and formats where
	   all streams are tightly interleaved. For non-intra frame codecs you
	   will usually get additional	packets	 with  presentation  timestamp
	   after  Out  point  therefore	 the  decoded content will most likely
	   contain frames after Out point too. If your streams are not tightly
	   interleaved you may not get all the packets from all streams before
	   Out point and you may only will be  able  to	 decode	 the  earliest
	   stream until Out point.

	   The	duration  of  the  files  (if  not specified by the "duration"
	   directive) will be reduced based on their specified Out point.

       "file_packet_metadata key=value"
	   Metadata of the packets of the file. The specified metadata will be
	   set for each file packet. You can specify this  directive  multiple
	   times   to  add  multiple  metadata	entries.   This	 directive  is
	   deprecated, use "file_packet_meta" instead.

       "file_packet_meta key value"
	   Metadata of the packets of the file. The specified metadata will be
	   set for each file packet. You can specify this  directive  multiple
	   times to add multiple metadata entries.

       "option key value"
	   Option to access, open and probe the file.  Can be present multiple
	   times.

       "stream"
	   Introduce  a	 stream	 in  the virtual file.	All subsequent stream-
	   related directives apply  to	 the  last  introduced	stream.	  Some
	   streams  properties	must  be set in order to allow identifying the
	   matching streams in the subfiles.  If no streams are defined in the
	   script, the streams from the first file are copied.

       "exact_stream_id id"
	   Set the id of the stream.  If this directive is given,  the	string
	   with	 the  corresponding  id in the subfiles will be used.  This is
	   especially useful for MPEG-PS (VOB) files, where the order  of  the
	   streams is not reliable.

       "stream_meta key value"
	   Metadata for the stream.  Can be present multiple times.

       "stream_codec value"
	   Codec for the stream.

       "stream_extradata hex_string"
	   Extradata for the string, encoded in hexadecimal.

       "chapter id start end"
	   Add	a  chapter.  id	 is  an	 unique identifier, possibly small and
	   consecutive.

       Options

       This demuxer accepts the following option:

       safe
	   If set to 1, reject unsafe file paths and directives.  A file  path
	   is  considered safe if it does not contain a protocol specification
	   and is relative and all components only contain characters from the
	   portable character set (letters,  digits,  period,  underscore  and
	   hyphen) and have no period at the beginning of a component.

	   If set to 0, any file name is accepted.

	   The default is 1.

       auto_convert
	   If set to 1, try to perform automatic conversions on packet data to
	   make the streams concatenable.  The default is 1.

	   Currently,  the  only  conversion  is  adding  the h264_mp4toannexb
	   bitstream filter to H.264 streams in MP4 format. This is  necessary
	   in particular if there are resolution changes.

       segment_time_metadata
	   If  set  to 1, every packet will contain the lavf.concat.start_time
	   and the lavf.concat.duration packet metadata values which  are  the
	   start_time  and the duration of the respective file segments in the
	   concatenated	 output	 expressed  in	microseconds.	The   duration
	   metadata  is only set if it is known based on the concat file.  The
	   default is 0.

       Examples

       •   Use absolute filenames and include some comments:

		   # my first filename
		   file /mnt/share/file-1.wav
		   # my second filename including whitespace
		   file '/mnt/share/file 2.wav'
		   # my third filename including whitespace plus single quote
		   file '/mnt/share/file 3'\''.wav'

       •   Allow for input format auto-probing, use safe filenames and set the
	   duration of the first file:

		   ffconcat version 1.0

		   file file-1.wav
		   duration 20.0

		   file subdir/file-2.wav

   dash
       Dynamic Adaptive Streaming over HTTP demuxer.

       This demuxer presents all AVStreams found in the manifest.  By  setting
       the  discard  flags on AVStreams the caller can decide which streams to
       actually	 receive.   Each  stream  mirrors  the	"id"  and  "bandwidth"
       properties  from the "<Representation>" as metadata keys named "id" and
       "variant_bitrate" respectively.

       Options

       This demuxer accepts the following option:

       cenc_decryption_key
	   16-byte key, in hex, to decrypt files encrypted  using  ISO	Common
	   Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

   ea
       Electronic Arts Multimedia format demuxer.

       This format is used by various Electronic Arts games.

       Options

       merge_alpha bool
	   Normally  the  VP6  alpha  channel  (if  exists)  is	 returned as a
	   secondary video stream, by setting this option  you	can  make  the
	   demuxer  return  a  single  video  stream  which contains the alpha
	   channel in addition to the ordinary video.

   imf
       Interoperable Master Format demuxer.

       This  demuxer  presents	audio  and  video  streams  found  in  an  IMF
       Composition,		   as		    specified		    in
       <https://doi.org/10.5594/SMPTE.ST2067-2.2020>.

	       ffmpeg [-assetmaps <path of ASSETMAP1>,<path of ASSETMAP2>,...] -i <path of CPL> ...

       If "-assetmaps" is not specified, the demuxer looks for a  file	called
       ASSETMAP.xml in the same directory as the CPL.

   flv, live_flv, kux
       Adobe Flash Video Format demuxer.

       This  demuxer  is  used to demux FLV files and RTMP network streams. In
       case of live network streams, if you force format, you may use live_flv
       option instead of flv to survive timestamp discontinuities.  KUX	 is  a
       flv variant used on the Youku platform.

	       ffmpeg -f flv -i myfile.flv ...
	       ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....

       -flv_metadata bool
	   Allocate the streams according to the onMetaData array content.

       -flv_ignore_prevtag bool
	   Ignore the size of previous tag value.

       -flv_full_metadata bool
	   Output all context of the onMetadata.

   gif
       Animated GIF demuxer.

       It accepts the following options:

       min_delay
	   Set	the  minimum  valid  delay  between  frames  in	 hundredths of
	   seconds.  Range is 0 to 6000. Default value is 2.

       max_gif_delay
	   Set the maximum valid delay between frames in hundredth of seconds.
	   Range is  0	to  65535.  Default  value  is	65535  (nearly	eleven
	   minutes), the maximum value allowed by the specification.

       default_delay
	   Set	the  default  delay  between  frames in hundredths of seconds.
	   Range is 0 to 6000. Default value is 10.

       ignore_loop
	   GIF files can contain information to loop a certain number of times
	   (or infinitely). If ignore_loop is set to 1, then the loop  setting
	   from	 the  input will be ignored and looping will not occur. If set
	   to 0, then looping will occur and will cycle the  number  of	 times
	   according to the GIF. Default value is 1.

       For  example,  with the overlay filter, place an infinitely looping GIF
       over another video:

	       ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

       Note that in the above example the shortest option for  overlay	filter
       is  used	 to  end  the output video at the length of the shortest input
       file, which in this case is input.mp4 as the GIF in this example	 loops
       infinitely.

   hls
       HLS demuxer

       Apple HTTP Live Streaming demuxer.

       This  demuxer  presents all AVStreams from all variant streams.	The id
       field is set to the  bitrate  variant  index  number.  By  setting  the
       discard	flags  on  AVStreams  (by  pressing 'a' or 'v' in ffplay), the
       caller can decide which variant streams to actually receive.  The total
       bitrate of the variant that the stream belongs to  is  available	 in  a
       metadata key named "variant_bitrate".

       It accepts the following options:

       live_start_index
	   segment  index  to  start live streams at (negative values are from
	   the end).

       prefer_x_start
	   prefer  to  use  #EXT-X-START  if  it's  in	playlist  instead   of
	   live_start_index.

       allowed_extensions
	   ','	separated  list	 of  file  extensions  that  hls is allowed to
	   access.

       max_reload
	   Maximum number of times a insufficient  list	 is  attempted	to  be
	   reloaded.  Default value is 1000.

       m3u8_hold_counters
	   The	maximum number of times to load m3u8 when it refreshes without
	   new segments.  Default value is 1000.

       http_persistent
	   Use persistent HTTP connections. Applicable only for HTTP  streams.
	   Enabled by default.

       http_multiple
	   Use	multiple  HTTP	connections  for  downloading  HTTP  segments.
	   Enabled by default for HTTP/1.1 servers.

       http_seekable
	   Use HTTP partial requests  for  downloading	HTTP  segments.	  0  =
	   disable, 1 = enable, -1 = auto, Default is auto.

       seg_format_options
	   Set	options	 for  the  demuxer  of	media segments using a list of
	   key=value pairs separated by ":".

       seg_max_retry
	   Maximum number of times to reload a segment on error,  useful  when
	   segment skip on network error is not desired.  Default value is 0.

   image2
       Image file demuxer.

       This  demuxer  reads from a list of image files specified by a pattern.
       The syntax and meaning of  the  pattern	is  specified  by  the	option
       pattern_type.

       The  pattern  may  contain  a  suffix  which  is	 used to automatically
       determine the format of the images contained in the files.

       The size, the pixel format, and the format of each image	 must  be  the
       same for all the files in the sequence.

       This demuxer accepts the following options:

       framerate
	   Set the frame rate for the video stream. It defaults to 25.

       loop
	   If set to 1, loop over the input. Default value is 0.

       pattern_type
	   Select the pattern type used to interpret the provided filename.

	   pattern_type accepts one of the following values.

	   none
	       Disable pattern matching, therefore the video will only contain
	       the  specified  image. You should use this option if you do not
	       want  to	 create	 sequences  from  multiple  images  and	  your
	       filenames may contain special pattern characters.

	   sequence
	       Select  a  sequence pattern type, used to specify a sequence of
	       files indexed by sequential numbers.

	       A sequence pattern may contain the string "%d" or "%0Nd", which
	       specifies  the  position	 of  the  characters  representing   a
	       sequential  number  in each filename matched by the pattern. If
	       the form "%d0Nd" is used, the string representing the number in
	       each filename is 0-padded and N is the total number of 0-padded
	       digits representing the number. The literal character  '%'  can
	       be specified in the pattern with the string "%%".

	       If  the	sequence  pattern  contains  "%d" or "%0Nd", the first
	       filename of the file list specified by the pattern must contain
	       a  number  inclusively  contained  between   start_number   and
	       start_number+start_number_range-1,   and	  all	the  following
	       numbers must be sequential.

	       For example the pattern "img-%03d.bmp" will match a sequence of
	       filenames  of   the   form   img-001.bmp,   img-002.bmp,	  ...,
	       img-010.bmp,  etc.;  the	 pattern "i%%m%%g-%d.jpg" will match a
	       sequence of filenames of	 the  form  i%m%g-1.jpg,  i%m%g-2.jpg,
	       ..., i%m%g-10.jpg, etc.

	       Note  that  the	pattern	 must  not necessarily contain "%d" or
	       "%0Nd", for example to convert a single image file img.jpeg you
	       can employ the command:

		       ffmpeg -i img.jpeg img.png

	   glob
	       Select a glob wildcard pattern type.

	       The pattern is interpreted like a glob() pattern. This is  only
	       selectable if libavformat was compiled with globbing support.

	   glob_sequence (deprecated, will be removed)
	       Select a mixed glob wildcard/sequence pattern.

	       If  your	 version  of  libavformat  was	compiled with globbing
	       support, and the provided pattern contains at  least  one  glob
	       meta character among "%*?[]{}" that is preceded by an unescaped
	       "%",   the  pattern  is	interpreted  like  a  glob()  pattern,
	       otherwise it is interpreted like a sequence pattern.

	       All glob special characters "%*?[]{}"  must  be	prefixed  with
	       "%". To escape a literal "%" you shall use "%%".

	       For  example  the  pattern  "foo-%*.jpeg"  will	match  all the
	       filenames prefixed by "foo-" and terminating with ".jpeg",  and
	       "foo-%?%?%?.jpeg"  will	match  all the filenames prefixed with
	       "foo-",	followed  by  a	 sequence  of  three  characters,  and
	       terminating with ".jpeg".

	       This pattern type is deprecated in favor of glob and sequence.

	   Default value is glob_sequence.

       pixel_format
	   Set	the  pixel  format of the images to read. If not specified the
	   pixel format is guessed from the first image file in the sequence.

       start_number
	   Set the index of the file matched by	 the  image  file  pattern  to
	   start to read from. Default value is 0.

       start_number_range
	   Set	the  index  interval range to check when looking for the first
	   image file in the sequence,	starting  from	start_number.  Default
	   value is 5.

       ts_from_file
	   If set to 1, will set frame timestamp to modification time of image
	   file. Note that monotonity of timestamps is not provided: images go
	   in  the  same order as without this option. Default value is 0.  If
	   set to 2, will set frame timestamp to the modification time of  the
	   image file in nanosecond precision.

       video_size
	   Set	the  video  size  of  the images to read. If not specified the
	   video size is guessed from the first image file in the sequence.

       export_path_metadata
	   If set to 1, will add two extra fields to  the  metadata  found  in
	   input,  making  them also available for other filters (see drawtext
	   filter for examples). Default value is  0.  The  extra  fields  are
	   described below:

	   lavf.image2dec.source_path
	       Corresponds to the full path to the input file being read.

	   lavf.image2dec.source_basename
	       Corresponds to the name of the file being read.

       Examples

       •   Use	ffmpeg	for  creating  a  video	 from  the  images in the file
	   sequence img-001.jpeg, img-002.jpeg, ..., assuming an  input	 frame
	   rate of 10 frames per second:

		   ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

       •   As  above,  but  start by reading from a file with index 100 in the
	   sequence:

		   ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

       •   Read images matching the "*.png" glob pattern ,  that  is  all  the
	   files terminating with the ".png" suffix:

		   ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv

   libgme
       The  Game  Music	 Emu  library is a collection of video game music file
       emulators.

       See  <https://bitbucket.org/mpyne/game-music-emu/overview>   for	  more
       information.

       It accepts the following options:

       track_index
	   Set	the index of which track to demux. The demuxer can only export
	   one track.  Track indexes start at 0. Default is to pick the	 first
	   track. Number of tracks is exported as tracks metadata entry.

       sample_rate
	   Set	the  sampling  rate  of	 the  exported track. Range is 1000 to
	   999999. Default is 44100.

       max_size (bytes)
	   The demuxer buffers the entire file into memory. Adjust this	 value
	   to  set  the	 maximum buffer size, which in turn, acts as a ceiling
	   for the size of files that can be read.  Default is 50 MiB.

   libmodplug
       ModPlug based module demuxer

       See <https://github.com/Konstanty/libmodplug>

       It will export one 2-channel 16-bit 44.1 kHz audio stream.  Optionally,
       a "pal8" 16-color video stream can be exported with or without  printed
       metadata.

       It accepts the following options:

       noise_reduction
	   Apply  a  simple low-pass filter. Can be 1 (on) or 0 (off). Default
	   is 0.

       reverb_depth
	   Set amount of reverb. Range 0-100. Default is 0.

       reverb_delay
	   Set delay in ms, clamped to 40-250 ms. Default is 0.

       bass_amount
	   Apply bass expansion a.k.a. XBass or megabass. Range is  0  (quiet)
	   to 100 (loud). Default is 0.

       bass_range
	   Set	cutoff	i.e. upper-bound for bass frequencies. Range is 10-100
	   Hz. Default is 0.

       surround_depth
	   Apply a Dolby Pro-Logic surround effect. Range is 0 (quiet) to  100
	   (heavy). Default is 0.

       surround_delay
	   Set surround delay in ms, clamped to 5-40 ms. Default is 0.

       max_size
	   The	demuxer buffers the entire file into memory. Adjust this value
	   to set the maximum buffer size, which in turn, acts	as  a  ceiling
	   for	the  size of files that can be read. Range is 0 to 100 MiB.  0
	   removes buffer size limit (not recommended). Default is 5 MiB.

       video_stream_expr
	   String which is evaluated using the eval API to  assign  colors  to
	   the	generated  video stream.  Variables which can be used are "x",
	   "y", "w", "h", "t", "speed", "tempo", "order", "pattern" and "row".

       video_stream
	   Generate video stream. Can be 1 (on) or 0 (off). Default is 0.

       video_stream_w
	   Set video frame width in 'chars' where one char indicates 8 pixels.
	   Range is 20-512. Default is 30.

       video_stream_h
	   Set video frame height  in  'chars'	where  one  char  indicates  8
	   pixels. Range is 20-512. Default is 30.

       video_stream_ptxt
	   Print metadata on video stream. Includes "speed", "tempo", "order",
	   "pattern",  "row"  and "ts" (time in ms). Can be 1 (on) or 0 (off).
	   Default is 1.

   libopenmpt
       libopenmpt based module demuxer

       See <https://lib.openmpt.org/libopenmpt/> for more information.

       Some files have multiple subsongs (tracks) this can  be	set  with  the
       subsong option.

       It accepts the following options:

       subsong
	   Set	the  subsong  index. This can be either	 'all', 'auto', or the
	   index of the subsong. Subsong indexes start at 0.  The  default  is
	   'auto'.

	   The default value is to let libopenmpt choose.

       layout
	   Set	the  channel  layout.  Valid  values  are  1, 2, and 4 channel
	   layouts.  The default value is STEREO.

       sample_rate
	   Set the sample rate for libopenmpt to output.  Range is  from  1000
	   to INT_MAX. The value default is 48000.

   mov/mp4/3gp
       Demuxer	for  Quicktime	File  Format  & ISO/IEC Base Media File Format
       (ISO/IEC 14496-12 or MPEG-4 Part 12, ISO/IEC 15444-12 or JPEG 2000 Part
       12).

       Registered extensions: mov, mp4, m4a, 3gp, 3g2,	mj2,  psp,  m4b,  ism,
       ismv, isma, f4v

       Options

       This demuxer accepts the following options:

       enable_drefs
	   Enable  loading  of external tracks, disabled by default.  Enabling
	   this can theoretically leak information in some use cases.

       use_absolute_path
	   Allows loading of external tracks via absolute paths,  disabled  by
	   default.   Enabling	this  poses a security risk. It should only be
	   enabled if the source is known to be non-malicious.

       seek_streams_individually
	   When	 seeking,  identify  the  closest   point   in	 each	stream
	   individually	 and  demux  packets  in  that	stream from identified
	   point. This can lead to a different sequence of packets compared to
	   demuxing linearly from the beginning. Default is true.

       ignore_editlist
	   Ignore any edit list atoms. The demuxer, by default,	 modifies  the
	   stream  index  to  reflect the timeline described by the edit list.
	   Default is false.

       advanced_editlist
	   Modify the stream index to reflect the timeline  described  by  the
	   edit	 list.	"ignore_editlist" must be set to false for this option
	   to be effective.  If both "ignore_editlist" and this option are set
	   to false, then only the start of the stream index  is  modified  to
	   reflect  initial  dwell time or starting timestamp described by the
	   edit list. Default is true.

       ignore_chapters
	   Don't parse chapters. This includes GoPro  'HiLight'	 tags/moments.
	   Note	 that chapters are only parsed when input is seekable. Default
	   is false.

       use_mfra_for
	   For seekable fragmented input, set  fragment's  starting  timestamp
	   from media fragment random access box, if present.

	   Following options are available:

	   auto
	       Auto-detect  whether  to	 set  mfra  timestamps	as  PTS or DTS
	       (default)

	   dts Set mfra timestamps as DTS

	   pts Set mfra timestamps as PTS

	   0   Don't use mfra box to set timestamps

       use_tfdt
	   For	fragmented  input,  set	 fragment's  starting	timestamp   to
	   "baseMediaDecodeTime"  from	the  "tfdt"  box.  Default is enabled,
	   which will prefer to use the "tfdt" box to set DTS. Disable to  use
	   the	"earliest_presentation_time"  from  the "sidx" box.  In either
	   case, the timestamp from the	 "mfra"	 box  will  be	used  if  it's
	   available and "use_mfra_for" is set to pts or dts.

       export_all
	   Export  unrecognized boxes within the udta box as metadata entries.
	   The first four characters of the box	 type  are  set	 as  the  key.
	   Default is false.

       export_xmp
	   Export  entire  contents  of XMP_ box and uuid box as a string with
	   key "xmp". Note that if "export_all" is set and this option	isn't,
	   the	contents  of  XMP_ box are still exported but with key "XMP_".
	   Default is false.

       activation_bytes
	   4-byte key required to decrypt Audible  AAX	and  AAX+  files.  See
	   Audible AAX subsection below.

       audible_fixed_key
	   Fixed  key  used  for  handling Audible AAX/AAX+ files. It has been
	   pre-set so should not be necessary to specify.

       decryption_key
	   16-byte key, in hex, to decrypt files encrypted  using  ISO	Common
	   Encryption (CENC/AES-128 CTR; ISO/IEC 23001-7).

       max_stts_delta
	   Very	  high	sample	deltas	written	 in  a	trak's	stts  box  may
	   occasionally be intended but usually they are written in  error  or
	   used	 to  store a negative value for dts correction when treated as
	   signed 32-bit integers. This option lets  the  user	set  an	 upper
	   limit,  beyond which the delta is clamped to 1. Values greater than
	   the limit if negative when cast to int32 are used to adjust	onward
	   dts.

	   Unit	 is  the  track time scale. Range is 0 to UINT_MAX. Default is
	   "UINT_MAX - 48000*10" which allows upto a 10 second dts  correction
	   for	48  kHz	 audio	streams	 while accommodating 99.9% of "uint32"
	   range.

       interleaved_read
	   Interleave packets from multiple tracks at demuxer level. For badly
	   interleaved files, this prevents playback issues  caused  by	 large
	   gaps	 between  packets  in different tracks, as MOV/MP4 do not have
	   packet placement requirements.  However, this can  cause  excessive
	   seeking  on	very  badly  interleaved files, due to seeking between
	   tracks, so disabling it may prevent I/O issues, at the  expense  of
	   playback.

       Audible AAX

       Audible AAX files are encrypted M4B files, and they can be decrypted by
       specifying a 4 byte activation secret.

	       ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

   mpegts
       MPEG-2 transport stream demuxer.

       This demuxer accepts the following options:

       resync_size
	   Set	size limit for looking up a new synchronization. Default value
	   is 65536.

       skip_unknown_pmt
	   Skip PMTs for programs not defined in the PAT. Default value is 0.

       fix_teletext_pts
	   Override teletext packet PTS and DTS	 values	 with  the  timestamps
	   calculated  from  the  PCR  of the first program which the teletext
	   stream is part of and is not discarded. Default  value  is  1,  set
	   this	 option	 to  0	if  you	 want your teletext packet PTS and DTS
	   values untouched.

       ts_packetsize
	   Output option carrying the raw packet  size	in  bytes.   Show  the
	   detected raw packet size, cannot be set by the user.

       scan_all_pmts
	   Scan	 and combine all PMTs. The value is an integer with value from
	   -1 to 1 (-1 means automatic	setting,  1  means  enabled,  0	 means
	   disabled). Default value is -1.

       merge_pmt_versions
	   Re-use  existing  streams  when  a  PMT's  version  is  updated and
	   elementary streams move to different PIDs. Default value is 0.

       max_packet_size
	   Set maximum size, in bytes,	of  packet  emitted  by	 the  demuxer.
	   Payloads  above  this size are split across multiple packets. Range
	   is 1 to INT_MAX/2. Default is 204800 bytes.

   mpjpeg
       MJPEG encapsulated in multi-part MIME demuxer.

       This demuxer allows reading of MJPEG, where each frame  is  represented
       as a part of multipart/x-mixed-replace stream.

       strict_mime_boundary
	   Default  implementation  applies  a	relaxed standard to multi-part
	   MIME	 boundary  detection,  to  prevent  regression	with  numerous
	   existing  endpoints	not  generating	 a  proper  MIME MJPEG stream.
	   Turning this option on by setting it to 1 will result in a stricter
	   check of the boundary value.

   rawvideo
       Raw video demuxer.

       This demuxer allows one to read raw  video  data.  Since	 there	is  no
       header  specifying  the assumed video parameters, the user must specify
       them in order to be able to decode the data correctly.

       This demuxer accepts the following options:

       framerate
	   Set input video frame rate. Default value is 25.

       pixel_format
	   Set the input video pixel format. Default value is "yuv420p".

       video_size
	   Set the input video size. This value must be specified explicitly.

       For example to read a rawvideo file input.raw with ffplay,  assuming  a
       pixel format of "rgb24", a video size of "320x240", and a frame rate of
       10 images per second, use the command:

	       ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

   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.

   a64
       A64  muxer  for	Commodore  64  video.  Accepts a single "a64_multi" or
       "a64_multi5" codec video stream.

   adts
       Audio Data Transport Stream muxer. It accepts a single AAC stream.

       Options

       It accepts the following options:

       write_id3v2 bool
	   Enable to write ID3v2.4 tags at the start of the stream. Default is
	   disabled.

       write_apetag bool
	   Enable to write APE tags at the  end	 of  the  stream.  Default  is
	   disabled.

       write_mpeg2 bool
	   Enable  to set MPEG version bit in the ADTS frame header to 1 which
	   indicates MPEG-2. Default is 0, which indicates MPEG-4.

   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.

   alp
       Muxer for audio of High Voltage Software's Lego Racers game. It accepts
       a single ADPCM_IMA_ALP stream with no more than 2 channels nor a sample
       rate greater than 44100 Hz.

       Extensions: tun, pcm

       Options

       It accepts the following options:

       type type
	   Set file type.

	   tun Set file type as music. Must have a sample rate of 22050 Hz.

	   pcm Set file type as sfx.

	   auto
	       Set file type as per output file extension. ".pcm"  results  in
	       type "pcm" else type "tun" is set. (default)

   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).

       flipped_raw_rgb
	   If set to true, store positive height for raw  RGB  bitmaps,	 which
	   indicates  bitmap  is  stored bottom-up. Note that this option does
	   not flip the bitmap which has to be done manually beforehand,  e.g.
	   by  using  the vflip filter.	 Default is false and indicates bitmap
	   is stored top down.

   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 -

   dash
       Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments
       and   manifest  files  according	 to  the  MPEG-DASH  standard  ISO/IEC
       23009-1:2014.

       For more information see:

       •   ISO			     DASH			Specification:
	   <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

       •   WebM			      DASH			Specification:
	   <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

       It creates a MPD manifest file and segment files for each stream.

       The segment filename might contain pre-defined  identifiers  used  with
       SegmentTemplate	as  defined  in	 section  5.3.9.4.4  of	 the standard.
       Available    identifiers	   are	  "$RepresentationID$",	   "$Number$",
       "$Bandwidth$"  and  "$Time$".  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

       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 use_timeline is  disabled  and  as  minimum  segment
	   duration for all the other use cases.

       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 hls_master_name.  One media playlist
	   file is generated for  each	stream	with  filenames	 media_0.m3u8,
	   media_1.m3u8, etc.

       hls_master_name file_name
	   HLS master playlist name. Default is "master.m3u8".

       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.   hls.js  player folks are trying to standardize an
	   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	 tries to comply with the above open spec.  It enables
	   streaming and  hls_playlist	options	 automatically.	  This	is  an
	   experimental feature.

	   Note:     This     is     not    Apple's    version	  LHLS.	   See
	   <https://datatracker.ietf.org/doc/html/draft-pantos-hls-rfc8216bis>

       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.

       update_period update_period
	    Set the mpd update period ,for dynamic content.
	    The unit is second.

   fifo
       The fifo pseudo-muxer allows the separation of encoding and  muxing  by
       using  first-in-first-out  queue	 and  running  the  actual  muxer in a
       separate thread. This is especially useful in combination with the  tee
       muxer  and  can	be  used  to  send  data  to several destinations with
       different reliability/writing speed/latency.

       API users should be aware that callback functions  (interrupt_callback,
       io_open	and  io_close) used within its AVFormatContext must be thread-
       safe.

       The behavior of the fifo muxer if the queue fills up or if  the	output
       fails is selectable,

       •   output  can	be  transparently  restarted  with  configurable delay
	   between retries based on real time or time of the processed stream.

       •   encoding can be  blocked  during  temporary	failure,  or  continue
	   transparently dropping packets in case fifo queue fills up.

       fifo_format
	   Specify  the	 format	 name. Useful if it cannot be guessed from the
	   output name suffix.

       queue_size
	   Specify size of the queue (number of packets). Default value is 60.

       format_opts
	   Specify format options for the underlying muxer. Muxer options  can
	   be specified as a list of key=value pairs separated by ':'.

       drop_pkts_on_overflow bool
	   If  set  to 1 (true), in case the fifo queue fills up, packets will
	   be dropped rather than blocking the encoder. This makes it possible
	   to continue streaming without delaying the input, at	 the  cost  of
	   omitting  part  of  the  stream. By default this option is set to 0
	   (false), so in such cases the encoder will  be  blocked  until  the
	   muxer processes some of the packets and none of them is lost.

       attempt_recovery bool
	   If	failure	 occurs,  attempt  to  recover	the  output.  This  is
	   especially useful when used with network output, since it makes  it
	   possible  to	 restart  streaming  transparently.   By  default this
	   option is set to 0 (false).

       max_recovery_attempts
	   Sets maximum number of successive  unsuccessful  recovery  attempts
	   after which the output fails permanently. By default this option is
	   set to 0 (unlimited).

       recovery_wait_time duration
	   Waiting  time  before  the  next  recovery  attempt	after previous
	   unsuccessful recovery attempt. Default value is 5 seconds.

       recovery_wait_streamtime bool
	   If set to 0 (false), the real time is used  when  waiting  for  the
	   recovery  attempt  (i.e.  the  recovery  will be attempted after at
	   least recovery_wait_time seconds).  If set to 1 (true), the time of
	   the processed stream	 is  taken  into  account  instead  (i.e.  the
	   recovery  will  be  attempted  after	 at  least  recovery_wait_time
	   seconds of the stream is omitted).  By default, this option is  set
	   to 0 (false).

       recover_any_error bool
	   If  set  to 1 (true), recovery will be attempted regardless of type
	   of the error causing the failure. By default this option is set  to
	   0  (false)  and  in	case of certain (usually permanent) errors the
	   recovery is not attempted even when attempt_recovery is set to 1.

       restart_with_keyframe bool
	   Specify whether to wait for	the  keyframe  after  recovering  from
	   queue  overflow  or	failure.  This	option	is set to 0 (false) by
	   default.

       timeshift duration
	   Buffer the specified	 amount	 of  packets  and  delay  writing  the
	   output.  Note  that	queue_size  must  be  big  enough to store the
	   packets for timeshift. At the end of the input the fifo  buffer  is
	   flushed at realtime speed.

       Examples

       •   Stream  something to rtmp server, continue processing the stream at
	   real-time rate even in case of temporary failure  (network  outage)
	   and attempt to recover streaming every second indefinitely.

		   ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
		     -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name

   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.

   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 duration
	   Set the initial target segment length. Default value is 0.

	   duration must be  a	time  duration	specification,	see  the  Time
	   duration section in the ffmpeg-utils(1) manual.

	   Segment  will  be  cut  on  the  next key frame after this time has
	   passed on the first m3u8  list.   After  the	 initial  playlist  is
	   filled ffmpeg will cut segments at duration equal to "hls_time"

       hls_time duration
	   Set the target segment length. Default value is 2.

	   duration  must  be  a  time	duration  specification,  see the Time
	   duration section in the ffmpeg-utils(1) manual.   Segment  will  be
	   cut on the next key frame after this time has passed.

       hls_list_size size
	   Set	the  maximum  number of playlist entries. If set to 0 the list
	   file will contain all the segments. Default value is 5.

       hls_delete_threshold size
	   Set the number of unreferenced segments  to	keep  on  disk	before
	   "hls_flags  delete_segments"	 deletes  them. Increase this to allow
	   continue  clients  to  download  segments   which   were   recently
	   referenced  in  the	playlist. Default value is 1, meaning segments
	   older than "hls_list_size+1" will be deleted.

       hls_start_number_source
	   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.

       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.

       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_segment_options options_list
	   Set output format options using a  :-separated  list	 of  key=value
	   parameters.	Values	containing  ":"	 special  characters  must  be
	   escaped.

       hls_key_info_file key_info_file
	   Use the information in key_info_file for  segment  encryption.  The
	   first  line	of  key_info_file specifies the key URI written to the
	   playlist. The key URL is used to access the encryption  key	during
	   playback.  The  second line specifies the path to the key file used
	   to obtain the key during the encryption process. The	 key  file  is
	   read	 as  a	single packed array of 16 octets in binary format. The
	   optional third line specifies the initialization vector (IV)	 as  a
	   hexadecimal	string	to  be	used  instead  of the segment sequence
	   number (default) for	 encryption.  Changes  to  key_info_file  will
	   result  in  segment	encryption with the new key/IV and an entry in
	   the playlist for the new key URI/IV if  "hls_flags  periodic_rekey"
	   is enabled.

	   Key info file format:

		   <key URI>
		   <key file path>
		   <IV> (optional)

	   Example key URIs:

		   http://server/file.key
		   /path/to/file.key
		   file.key

	   Example key file paths:

		   file.key
		   /path/to/file.key

	   Example IV:

		   0123456789ABCDEF0123456789ABCDEF

	   Key info file example:

		   http://server/file.key
		   /path/to/file.key
		   0123456789ABCDEF0123456789ABCDEF

	   Example shell script:

		   #!/bin/sh
		   BASE_URL=${1:-'.'}
		   openssl rand 16 > file.key
		   echo $BASE_URL/file.key > file.keyinfo
		   echo file.key >> file.keyinfo
		   echo $(openssl rand -hex 16) >> file.keyinfo
		   ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
		     -hls_key_info_file file.keyinfo out.m3u8

       -hls_enc 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
	   16-octet  key  to  encrypt  the segments, by default it is randomly
	   generated.

       -hls_enc_key_url keyurl
	   If set, keyurl is prepended instead of baseurl to the key  filename
	   in the playlist.

       -hls_enc_iv iv
	   16-octet  initialization  vector  for  every segment instead of the
	   autogenerated ones.

       hls_segment_type flags
	   Possible values:

	   mpegts
	       Output segment files in MPEG-2 Transport Stream format. This is
	       compatible with all HLS versions.

	   fmp4
	       Output segment files in fragmented MP4 format, similar to MPEG-
	       DASH.  fmp4 files may be used in HLS version 7 and above.

       hls_fmp4_init_filename filename
	   Set filename to the fragment files header file, default filename is
	   init.mp4.

	   Use "-strftime 1" on filename to expand the segment	filename  with
	   localtime.

		   ffmpeg -i in.nut  -hls_segment_type fmp4 -strftime 1 -hls_fmp4_init_filename "%s_init.mp4" out.m3u8

	   This will produce init like this 1602678741_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.  If there are
	       no other features requiring higher HLS versions be  used,  then
	       this will allow ffmpeg to output a HLS version 2 m3u8.

	   discont_start
	       Add  the "#EXT-X-DISCONTINUITY" tag to the playlist, before the
	       first segment's information.

	   omit_endlist
	       Do not append  the  "EXT-X-ENDLIST"  tag	 at  the  end  of  the
	       playlist.

	   periodic_rekey
	       The  file  specified  by	 "hls_key_info_file"  will  be checked
	       periodically and detect updates to the encryption info. Be sure
	       to replace this file atomically, including the file  containing
	       the AES encryption key.

	   independent_segments
	       Add  the	 "#EXT-X-INDEPENDENT-SEGMENTS"	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:

       •   Size cannot exceed 256 pixels in any dimension

       •   Only BMP and PNG images can be stored

       •   If  a  BMP  image  is  used,	 it must be one of the following pixel
	   formats:

		   BMP Bit Depth      FFmpeg Pixel Format
		   1bit		      pal8
		   4bit		      pal8
		   8bit		      pal8
		   16bit	      rgb555le
		   24bit	      bgr24
		   32bit	      bgra

       •   If a BMP image is used, it must use the BITMAPINFOHEADER DIB header

       •   If a PNG image is used, it must use the rgba pixel format

   image2
       Image file muxer.

       The image file muxer writes video frames to image files.

       The output filenames are specified by a pattern, which can be  used  to
       produce sequentially numbered series of files.  The pattern may contain
       the  string  "%d"  or "%0Nd", this string specifies the position of the
       characters representing a numbering  in	the  filenames.	 If  the  form
       "%0Nd"  is used, the string representing the number in each filename is
       0-padded to N digits. The literal character '%' can be specified in the
       pattern with the string "%%".

       If the pattern contains "%d" or "%0Nd", the first filename of the  file
       list  specified	will  contain  the number 1, all the following numbers
       will be sequential.

       The pattern may	contain	 a  suffix  which  is  used  to	 automatically
       determine the format of the image files to write.

       For  example  the  pattern  "img-%03d.bmp"  will	 specify a sequence of
       filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp,  etc.
       The  pattern "img%%-%d.jpg" will specify a sequence of filenames of the
       form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.

       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.

       atomic_writing
	   Write  output  to  a	 temporary  file,  which  is renamed to target
	   filename once writing is completed. Default is disabled.

       protocol_opts options_list
	   Set protocol options as a :-separated list of key=value parameters.
	   Values containing the ":" special character must be escaped.

       Examples

       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.

       cues_to_front
	   If set, the muxer will write the index at the beginning of the file
	   by  shifting	 the main data if necessary. This can be combined with
	   reserve_index_space in which case the data is only shifted  if  the
	   initially reserved space turns out to be insufficient.

	   This option is ignored if the output is unseekable.

       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 passthrough.

	   infer
	       Every track with disposition default will have the  FlagDefault
	       set.   Additionally,  for  each	type of track (audio, video or
	       subtitle), if no track with disposition default	of  this  type
	       exists,	then  the  first  track of this type will be marked as
	       default (if existing). This ensures that the  default  flag  is
	       set  in	a  sensible  way  even	if  the	 input originated from
	       containers that lack the concept of default tracks.

	   infer_no_subs
	       This mode is the same as infer except that if no subtitle track
	       with disposition default exists,	 no  subtitle  track  will  be
	       marked as default.

	   passthrough
	       In  this	 mode  the  FlagDefault	 is  set  if  and  only if the
	       AV_DISPOSITION_DEFAULT flag is set in the  disposition  of  the
	       corresponding stream.

       flipped_raw_rgb
	   If  set  to	true, store positive height for raw RGB bitmaps, which
	   indicates bitmap is stored bottom-up. Note that  this  option  does
	   not	flip the bitmap which has to be done manually beforehand, e.g.
	   by using the vflip filter.  Default is false and  indicates	bitmap
	   is stored top down.

   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.

       Fragmentation is enabled by setting one of the options that define  how
       to   cut	 the  file  into  fragments:  "-frag_duration",	 "-frag_size",
       "-min_frag_duration",   "-movflags   +frag_keyframe"   and   "-movflags
       +frag_custom".  If  more than one condition is specified, fragments are
       cut when one of the specified conditions is fulfilled. The exception to
       this is "-min_frag_duration", which has to be fulfilled for any of  the
       other conditions to apply.

       Options

       frag_duration duration
	   Create fragments that are duration microseconds long.

       frag_size size
	   Create fragments that contain up to size bytes of payload data.

       min_frag_duration duration
	   Don't  create fragments that are shorter than duration microseconds
	   long.

       movflags flags
	   Set various muxing switches. The following flags can be used:

	   frag_keyframe
	       Start a new fragment at each video keyframe.

	   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.)

	   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.

	   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.

	   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.

	   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.

	   rtphint
	       Add RTP hinting tracks to the output file.

	   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.

	   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.

	   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).

	   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.

       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.

       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).

       write_btrt bool
	   Force  or  disable  writing bitrate box inside stsd box of a track.
	   The box contains decoding buffer size (in bytes),  maximum  bitrate
	   and	average bitrate for the track. The box will be skipped if none
	   of these values can be computed.  Default is -1  or	"auto",	 which
	   will write the box only in MP4 mode.

       write_prft
	   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.

       empty_hdlr_name bool
	   Enable to skip writing the name inside a "hdlr"  box.   Default  is
	   "false".

       movie_timescale scale
	   Set	the timescale written in the movie header box ("mvhd").	 Range
	   is 1 to INT_MAX. Default is 1000.

       video_track_timescale scale
	   Set the timescale used for video tracks. Range is 0 to INT_MAX.  If
	   set to 0, the timescale is automatically set based  on  the	native
	   stream time base. Default is 0.

       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:

       •   An  ID3v2  metadata	header	at the beginning (enabled by default).
	   Versions 2.3 and 2.4 are  supported,	 the  "id3v2_version"  private
	   option controls which one is used (3 or 4). Setting "id3v2_version"
	   to 0 disables the ID3v2 header completely.

	   The	muxer  supports writing attached pictures (APIC frames) to the
	   ID3v2 header.  The pictures are supplied to the muxer in form of  a
	   video stream with a single packet. There can be any number of those
	   streams,  each  will correspond to a single APIC frame.  The stream
	   metadata tags title and comment map to APIC description and picture
	   type	 respectively.	 See   <http://id3.org/id3v2.4.0-frames>   for
	   allowed picture types.

	   Note	 that the APIC frames must be written at the beginning, so the
	   muxer will buffer the audio frames until it gets all the  pictures.
	   It is therefore advised to provide the pictures as soon as possible
	   to avoid excessive buffering.

       •   A  Xing/LAME frame right after the ID3v2 header (if present). It is
	   enabled by default, but will be  written  only  if  the  output  is
	   seekable.  The  "write_xing"	 private option can be used to disable
	   it.	The frame contains various information that may be  useful  to
	   the decoder, like the audio duration or encoder delay.

       •   A legacy ID3v1 tag at the end of the file (disabled by default). It
	   may	be  enabled  with the "write_id3v1" private option, but as its
	   capabilities are very limited, its usage is not recommended.

       Examples:

       Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

	       ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

       To attach a picture to an mp3  file  select  both  the  audio  and  the
       picture stream with "map":

	       ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
	       -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

       Write a "clean" MP3 without any extra features:

	       ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

   mpegts
       MPEG transport stream muxer.

       This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

       The recognized metadata settings in mpegts muxer are "service_provider"
       and   "service_name".   If   they   are	 not   set   the  default  for
       "service_provider" is FFmpeg and	 the  default  for  "service_name"  is
       Service01.

       Options

       The muxer options are:

       mpegts_transport_stream_id integer
	   Set	the transport_stream_id. This identifies a transponder in DVB.
	   Default is 0x0001.

       mpegts_original_network_id integer
	   Set the original_network_id. This is unique identifier of a network
	   in DVB. Its main use is in the unique identification of  a  service
	   through  the path Original_Network_ID, Transport_Stream_ID. Default
	   is 0x0001.

       mpegts_service_id integer
	   Set the service_id, also  known  as	program	 in  DVB.  Default  is
	   0x0001.

       mpegts_service_type integer
	   Set the program service_type. Default is "digital_tv".  Accepts the
	   following options:

	   hex_value
	       Any  hexadecimal value between 0x01 and 0xff as defined in ETSI
	       300 468.

	   digital_tv
	       Digital TV service.

	   digital_radio
	       Digital Radio service.

	   teletext
	       Teletext service.

	   advanced_codec_digital_radio
	       Advanced Codec Digital Radio service.

	   mpeg2_digital_hdtv
	       MPEG2 Digital HDTV service.

	   advanced_codec_digital_sdtv
	       Advanced Codec Digital SDTV service.

	   advanced_codec_digital_hdtv
	       Advanced Codec Digital HDTV service.

       mpegts_pmt_start_pid integer
	   Set the first PID for PMTs. Default is 0x1000, minimum  is  0x0020,
	   maximum is 0x1ffa. This option has no effect in m2ts mode where the
	   PMT PID is fixed 0x0100.

       mpegts_start_pid integer
	   Set	the  first  PID	 for  elementary  streams.  Default is 0x0100,
	   minimum is 0x0020, maximum is 0x1ffa. This option has no effect  in
	   m2ts mode where the elementary stream PIDs are fixed.

       mpegts_m2ts_mode boolean
	   Enable  m2ts	 mode  if set to 1. Default value is -1 which disables
	   m2ts mode.

       muxrate integer
	   Set a constant muxrate. Default is VBR.

       pes_payload_size integer
	   Set minimum PES packet payload in bytes. Default is 2930.

       mpegts_flags flags
	   Set mpegts flags. Accepts the following options:

	   resend_headers
	       Reemit PAT/PMT before writing the next packet.

	   latm
	       Use LATM packetization for AAC.

	   pat_pmt_at_frames
	       Reemit PAT and PMT at each video frame.

	   system_b
	       Conform to System B (DVB) instead of System A (ATSC).

	   initial_discontinuity
	       Mark the initial packet of each stream as discontinuity.

	   nit Emit NIT table.

	   omit_rai
	       Disable writing of random access indicator.

       mpegts_copyts boolean
	   Preserve original timestamps, if value is set to 1.	Default	 value
	   is -1, which results in shifting timestamps so that they start from
	   0.

       omit_video_pes_length boolean
	   Omit the PES packet length for video packets. Default is 1 (true).

       pcr_period integer
	   Override  the  default  PCR	retransmission	time  in milliseconds.
	   Default is -1 which means that the PCR interval will be  determined
	   automatically:  20 ms is used for CBR streams, the highest multiple
	   of the frame duration which is less than 100 ms  is	used  for  VBR
	   streams.

       pat_period duration
	   Maximum time in seconds between PAT/PMT tables. Default is 0.1.

       sdt_period duration
	   Maximum time in seconds between SDT tables. Default is 0.5.

       nit_period duration
	   Maximum time in seconds between NIT tables. Default is 0.5.

       tables_version integer
	   Set PAT, PMT, SDT and NIT version (default 0, valid values are from
	   0   to  31,	inclusively).	This  option  allows  updating	stream
	   structure so that standard consumer may detect the  change.	To  do
	   so,	reopen	output	"AVFormatContext"  (in	case  of API usage) or
	   restart ffmpeg instance, cyclically changing tables_version value:

		   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
		   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
		   ...
		   ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
		   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
		   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
		   ...

       Example

	       ffmpeg -i file.mpg -c copy \
		    -mpegts_original_network_id 0x1122 \
		    -mpegts_transport_stream_id 0x3344 \
		    -mpegts_service_id 0x5566 \
		    -mpegts_pmt_start_pid 0x1500 \
		    -mpegts_start_pid 0x150 \
		    -metadata service_provider="Some provider" \
		    -metadata service_name="Some Channel" \
		    out.ts

   mxf, mxf_d10, mxf_opatom
       MXF muxer.

       Options

       The muxer options are:

       store_user_comments bool
	   Set if user comments should be stored if available or  never.   IRT
	   D-10	 does  not  allow  user comments. The default is thus to write
	   them for mxf and mxf_opatom but not for mxf_d10

   null
       Null muxer.

       This muxer does not generate any output file, it is mainly  useful  for
       testing or benchmarking purposes.

       For example to benchmark decoding with ffmpeg you can use the command:

	       ffmpeg -benchmark -i INPUT -f null out.null

       Note  that  the above command does not read or write the out.null file,
       but specifying the output file is required by the ffmpeg syntax.

       Alternatively you can write the command as:

	       ffmpeg -benchmark -i INPUT -f null -

   nut
       -syncpoints flags
	   Change the syncpoint usage in nut:

	   default use the normal low-overhead seeking aids.
	   none do not use the syncpoints at all, reducing the overhead but
	   making the stream non-seekable;
		   Use of this option is not recommended, as the resulting files are very damage
		   sensitive and seeking is not possible. Also in general the overhead from
		   syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
		   all growing data tables, allowing to mux endless streams with limited memory
		   and without these disadvantages.

	   timestamped extend the syncpoint with a wallclock field.

	   The none and timestamped flags are experimental.

       -write_index bool
	   Write index at the end, the default is to write an index.

	       ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor

   ogg
       Ogg container muxer.

       -page_duration duration
	   Preferred page duration, in microseconds. The muxer will attempt to
	   create pages that are  approximately	 duration  microseconds	 long.
	   This	 allows	 the  user  to compromise between seek granularity and
	   container overhead. The default is 1 second. A value of 0 will fill
	   all segments, making pages as large as possible. A value of 1  will
	   effectively	use  1	packet-per-page	 in  most situations, giving a
	   small  seek	granularity  at	 the  cost  of	additional   container
	   overhead.

       -serial_offset value
	   Serial  value from which to set the streams serial number.  Setting
	   it to different and sufficiently  large  values  ensures  that  the
	   produced ogg files can be safely chained.

   raw muxers
       Raw  muxers  accept a single stream matching the designated codec. They
       do not store timestamps or metadata.  The recognized extension  is  the
       same as the muxer name unless indicated otherwise.

       ac3

       Dolby Digital, also known as AC-3, audio.

       adx

       CRI Middleware ADX audio.

       This  muxer will write out the total sample count near the start of the
       first packet when the output is seekable and the count can be stored in
       32 bits.

       aptx

       aptX (Audio Processing Technology for Bluetooth) audio.

       aptx_hd

       aptX HD (Audio Processing Technology for Bluetooth) audio.

       Extensions: aptxhd

       avs2

       AVS2-P2/IEEE1857.4 video.

       Extensions: avs, avs2

       cavsvideo

       Chinese AVS (Audio Video Standard) video.

       Extensions: cavs

       codec2raw

       Codec 2 audio.

       No extension is registered so format name has to be supplied e.g.  with
       the ffmpeg CLI tool "-f codec2raw".

       data

       Data  muxer  accepts  a	single stream with any codec of any type.  The
       input stream has to be selected using the "-map" option with the ffmpeg
       CLI tool.

       No extension is registered so format name has to be supplied e.g.  with
       the ffmpeg CLI tool "-f data".

       dirac

       BBC Dirac video. The Dirac Pro codec is a subset and is standardized as
       SMPTE VC-2.

       Extensions: drc, vc2

       dnxhd

       Avid  DNxHD  video.  It	is  standardized  as SMPTE VC-3. Accepts DNxHR
       streams.

       Extensions: dnxhd, dnxhr

       dts

       DTS Coherent Acoustics (DCA) audio.

       eac3

       Dolby Digital Plus, also known as Enhanced AC-3, audio.

       evc

       MPEG-5 Essential Video Coding (EVC) / EVC / MPEG-5 Part 1 EVC video.

       Extensions: evc

       g722

       ITU-T G.722 audio.

       g723_1

       ITU-T G.723.1 audio.

       Extensions: tco, rco

       g726

       ITU-T G.726 big-endian ("left-justified") audio.

       No extension is registered so format name has to be supplied e.g.  with
       the ffmpeg CLI tool "-f g726".

       g726le

       ITU-T G.726 little-endian ("right-justified") audio.

       No  extension is registered so format name has to be supplied e.g. with
       the ffmpeg CLI tool "-f g726le".

       gsm

       Global System for Mobile Communications audio.

       h261

       ITU-T H.261 video.

       h263

       ITU-T H.263 / H.263-1996, H.263+ / H.263-1998 / H.263 version 2 video.

       h264

       ITU-T H.264 / MPEG-4 Part 10 AVC video. Bitstream shall be converted to
       Annex B syntax if it's in length-prefixed mode.

       Extensions: h264, 264

       hevc

       ITU-T H.265 / MPEG-H Part 2 HEVC video. Bitstream shall be converted to
       Annex B syntax if it's in length-prefixed mode.

       Extensions: hevc, h265, 265

       m4v

       MPEG-4 Part 2 video.

       mjpeg

       Motion JPEG video.

       Extensions: mjpg, mjpeg

       mlp

       Meridian Lossless Packing, also known as Packed PCM, audio.

       mp2

       MPEG-1 Audio Layer II audio.

       Extensions: mp2, m2a, mpa

       mpeg1video

       MPEG-1 Part 2 video.

       Extensions: mpg, mpeg, m1v

       mpeg2video

       ITU-T H.262 / MPEG-2 Part 2 video.

       Extensions: m2v

       obu

       AV1 low overhead Open Bitstream Units muxer.  Temporal  delimiter  OBUs
       will be inserted in all temporal units of the stream.

       rawvideo

       Raw uncompressed video.

       Extensions: yuv, rgb

       sbc

       Bluetooth SIG low-complexity subband codec audio.

       Extensions: sbc, msbc

       truehd

       Dolby TrueHD audio.

       Extensions: thd

       vc1

       SMPTE 421M / VC-1 video.

   segment, stream_segment, ssegment
       Basic stream segmenter.

       This  muxer  outputs  streams  to  a number of separate files of nearly
       fixed duration. Output filename pattern can be set in a fashion similar
       to image2, or by using a "strftime" template if the strftime option  is
       enabled.

       "stream_segment"	 is  a variant of the muxer used to write to streaming
       output formats, i.e. which  do  not  require  global  headers,  and  is
       recommended  for	 outputting  e.g.  to  MPEG transport stream segments.
       "ssegment" is a shorter alias for "stream_segment".

       Every segment starts with a keyframe of the selected reference  stream,
       which is set through the reference_stream option.

       Note  that if you want accurate splitting for a video file, you need to
       make the input key frames  correspond  to  the  exact  splitting	 times
       expected	 by  the  segmenter,  or  the segment muxer will start the new
       segment with the key frame found next after the specified start time.

       The segment muxer works best with a single constant frame rate video.

       Optionally it can generate a list of the created segments,  by  setting
       the   option   segment_list.   The   list  type	is  specified  by  the
       segment_list_type option. The entry filenames in the segment  list  are
       set by default to the basename of the corresponding segment files.

       See  also  the hls muxer, which provides a more specific implementation
       for HLS segmentation.

       Options

       The segment muxer supports the following options:

       increment_tc 1|0
	   if set to 1, increment timecode between each	 segment  If  this  is
	   selected,  the  input  need	to  have a timecode in the first video
	   stream. Default value is 0.

       reference_stream specifier
	   Set the reference stream, as specified by the string specifier.  If
	   specifier is set to "auto", the reference is chosen	automatically.
	   Otherwise   it  must	 be  a	stream	specifier  (see	 the  ``Stream
	   specifiers'' chapter in the	ffmpeg	manual)	 which	specifies  the
	   reference stream. The default value is "auto".

       segment_format format
	   Override  the  inner	 container format, by default it is guessed by
	   the filename extension.

       segment_format_options options_list
	   Set output format options using a  :-separated  list	 of  key=value
	   parameters.	Values	containing  the	 ":" special character must be
	   escaped.

       segment_list name
	   Generate also a listfile named name. If not specified  no  listfile
	   is generated.

       segment_list_flags flags
	   Set flags affecting the segment list generation.

	   It currently supports the following flags:

	   cache
	       Allow caching (only affects M3U8 list files).

	   live
	       Allow live-friendly file generation.

       segment_list_size size
	   Update  the list file so that it contains at most size segments. If
	   0 the list file will contain all the segments. Default value is 0.

       segment_list_entry_prefix prefix
	   Prepend prefix to each entry. Useful to  generate  absolute	paths.
	   By default no prefix is applied.

       segment_list_type type
	   Select the listing format.

	   The following values are recognized:

	   flat
	       Generate	 a flat list for the created segments, one segment per
	       line.

	   csv, ext
	       Generate a list for the created segments, one segment per line,
	       each line matching the format (comma-separated values):

		       <segment_filename>,<segment_start_time>,<segment_end_time>

	       segment_filename is the name of the output  file	 generated  by
	       the  muxer  according  to  the  provided	 pattern. CSV escaping
	       (according to RFC4180) is applied if required.

	       segment_start_time and  segment_end_time	 specify  the  segment
	       start and end time expressed in seconds.

	       A  list	file with the suffix ".csv" or ".ext" will auto-select
	       this format.

	       ext is deprecated in favor or csv.

	   ffconcat
	       Generate	 an  ffconcat  file  for  the  created	segments.  The
	       resulting file can be read using the FFmpeg concat demuxer.

	       A  list file with the suffix ".ffcat" or ".ffconcat" will auto-
	       select this format.

	   m3u8
	       Generate an extended  M3U8  file,  version  3,  compliant  with
	       <http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

	       A  list	file  with  the	 suffix	 ".m3u8" will auto-select this
	       format.

	   If not specified the type  is  guessed  from	 the  list  file  name
	   suffix.

       segment_time time
	   Set	segment	 duration  to  time,  the  value  must	be  a duration
	   specification. Default value is "2".	 See  also  the	 segment_times
	   option.

	   Note	 that  splitting  may  not  be	accurate, unless you force the
	   reference stream key-frames at the given time. See the introductory
	   notice and the examples below.

       min_seg_duration time
	   Set minimum segment duration to time, the value must be a  duration
	   specification.  This	 prevents  the	muxer  ending  segments	 at  a
	   duration below this	value.	Only  effective	 with  "segment_time".
	   Default value is "0".

       segment_atclocktime 1|0
	   If  set  to "1" split at regular clock time intervals starting from
	   00:00 o'clock. The time value specified in segment_time is used for
	   setting the length of the splitting interval.

	   For example with segment_time set to "900" this makes  it  possible
	   to create files at 12:00 o'clock, 12:15, 12:30, etc.

	   Default value is "0".

       segment_clocktime_offset duration
	   Delay  the segment splitting times with the specified duration when
	   using segment_atclocktime.

	   For	  example    with    segment_time    set    to	  "900"	   and
	   segment_clocktime_offset  set  to  "300"  this makes it possible to
	   create files at 12:05, 12:20, 12:35, etc.

	   Default value is "0".

       segment_clocktime_wrap_duration duration
	   Force the segmenter to only start a new segment if a packet reaches
	   the muxer within the specified duration after the segmenting	 clock
	   time.  This	way  you  can  make  the  segmenter  more resilient to
	   backward local time jumps, such as leap seconds  or	transition  to
	   standard time from daylight savings time.

	   Default is the maximum possible duration which means starting a new
	   segment regardless of the elapsed time since the last clock time.

       segment_time_delta delta
	   Specify  the	 accuracy  time	 when  selecting  the start time for a
	   segment, expressed as a duration specification.  Default  value  is
	   "0".

	   When delta is specified a key-frame will start a new segment if its
	   PTS satisfies the relation:

		   PTS >= start_time - time_delta

	   This option is useful when splitting video content, which is always
	   split  at  GOP boundaries, in case a key frame is found just before
	   the specified split time.

	   In particular may be used in combination  with  the	ffmpeg	option
	   force_key_frames. The key frame times specified by force_key_frames
	   may	not  be	 set  accurately  because of rounding issues, with the
	   consequence that a key frame time may result set  just  before  the
	   specified   time.  For  constant  frame  rate  videos  a  value  of
	   1/(2*frame_rate) should address the worst case mismatch between the
	   specified time and the time set by force_key_frames.

       segment_times times
	   Specify a list of split points. times  contains  a  list  of	 comma
	   separated  duration	specifications,	 in increasing order. See also
	   the segment_time option.

       segment_frames frames
	   Specify a list of split video frame numbers. frames contains a list
	   of comma separated integer numbers, in increasing order.

	   This option specifies to start a new segment whenever  a  reference
	   stream  key frame is found and the sequential number (starting from
	   0) of the frame is greater or equal to the next value in the list.

       segment_wrap limit
	   Wrap around segment index once it reaches limit.

       segment_start_number number
	   Set the sequence number of the first segment. Defaults to 0.

       strftime 1|0
	   Use the "strftime" function to define the name of the new  segments
	   to write. If this is selected, the output segment name must contain
	   a "strftime" function template. Default value is 0.

       break_non_keyframes 1|0
	   If enabled, allow segments to start on frames other than keyframes.
	   This	 improves  behavior  on	 some  players	when  the time between
	   keyframes is inconsistent, but may make things worse on others, and
	   can cause some oddities during seeking. Defaults to 0.

       reset_timestamps 1|0
	   Reset timestamps at the beginning of each  segment,	so  that  each
	   segment  will  start with near-zero timestamps. It is meant to ease
	   the playback of the generated segments.  May	 not  work  with  some
	   combinations of muxers/codecs. It is set to 0 by default.

       initial_offset offset
	   Specify  timestamp offset to apply to the output packet timestamps.
	   The argument must be a time duration specification, and defaults to
	   0.

       write_empty_segments 1|0
	   If enabled, write an empty segment if there are no  packets	during
	   the	period	a  segment  would usually span. Otherwise, the segment
	   will be filled with the next packet written. Defaults to 0.

       Make sure to require a closed GOP when encoding and to set the GOP size
       to fit your segment time constraint.

       Examples

       •   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

       •   Segment  input  and	set  output  format  options  for  the	output
	   segments:

		   ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4

       •   Segment the input file according to the split points	 specified  by
	   the segment_times option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut

       •   Use	the  ffmpeg force_key_frames option to force key frames in the
	   input at the specified location, together with the  segment	option
	   segment_time_delta  to account for possible roundings operated when
	   setting key frame times.

		   ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
		   -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut

	   In order to force key frames on  the	 input	file,  transcoding  is
	   required.

       •   Segment the input file by splitting the input file according to the
	   frame numbers sequence specified with the segment_frames option:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut

       •   Convert  the	 in.mkv	 to  TS segments using the "libx264" and "aac"
	   encoders:

		   ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts

       •   Segment the input file, and create an M3U8 live  playlist  (can  be
	   used as live HLS source):

		   ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
		   -segment_list_flags +live -segment_time 10 out%03d.mkv

   smoothstreaming
       Smooth  Streaming  muxer	 generates  a  set of files (Manifest, chunks)
       suitable for serving with conventional web server.

       window_size
	   Specify the number of fragments kept in  the	 manifest.  Default  0
	   (keep all).

       extra_window_size
	   Specify the number of fragments kept outside of the manifest before
	   removing from disk. Default 5.

       lookahead_count
	   Specify the number of lookahead fragments. Default 2.

       min_frag_duration
	   Specify  the	 minimum  fragment duration (in microseconds). Default
	   5000000.

       remove_at_exit
	   Specify whether to remove all fragments when	 finished.  Default  0
	   (do not remove).

   streamhash
       Per stream hash testing format.

       This  muxer  computes  and prints a cryptographic hash of all the input
       frames, on a per-stream basis. This can be  used	 for  equality	checks
       without having to do a complete binary comparison.

       By  default  audio  frames are converted to signed 16-bit raw audio and
       video frames to raw video before computing the hash, but the output  of
       explicit	 conversions  to other codecs can also be used. Timestamps are
       ignored. It uses the SHA-256 cryptographic hash	function  by  default,
       but supports several other algorithms.

       The  output  of	the muxer consists of one line per stream of the form:
       streamindex,streamtype,algo=hash, where streamindex is the index of the
       mapped stream, streamtype is a single character indicating the type  of
       stream, algo is a short string representing the hash function used, and
       hash is a hexadecimal number representing the computed hash.

       hash algorithm
	   Use	the  cryptographic  hash  function  specified  by  the	string
	   algorithm.  Supported values include "MD5", "murmur3", "RIPEMD128",
	   "RIPEMD160", "RIPEMD256", "RIPEMD320", "SHA160", "SHA224", "SHA256"
	   (default), "SHA512/224", "SHA512/256", "SHA384", "SHA512",  "CRC32"
	   and "adler32".

       Examples

       To  compute  the	 SHA-256  hash of the input converted to raw audio and
       video, and store it in the file out.sha256:

	       ffmpeg -i INPUT -f streamhash out.sha256

       To print an MD5 hash to stdout use the command:

	       ffmpeg -i INPUT -f streamhash -hash md5 -

       See also the hash and framehash muxers.

   tee
       The tee muxer can be used to write the same data	 to  several  outputs,
       such  as	 files	or  streams.  It can be used, for example, to stream a
       video over a network and save it to disk at the same time.

       It is different from specifying several outputs to the ffmpeg  command-
       line tool. With the tee muxer, the audio and video data will be encoded
       only  once.   With  conventional	 multiple  outputs,  multiple encoding
       operations in parallel are initiated, which can	be  a  very  expensive
       process.	 The  tee  muxer  is not useful when using the libavformat API
       directly because it is then  possible  to  feed	the  same  packets  to
       several muxers directly.

       Since  the  tee	muxer does not represent any particular output format,
       ffmpeg cannot auto-select output streams. So all streams	 intended  for
       output must be specified using "-map". See the examples below.

       Some  encoders  may  need  different  options  depending	 on the output
       format; the auto-detection of this can not work with the tee muxer,  so
       they  need  to  be  explicitly  specified.   The	 main  example	is the
       global_header flag.

       The slave outputs are specified in the file name given  to  the	muxer,
       separated  by '|'. If any of the slave name contains the '|' separator,
       leading or trailing spaces or any  special  character,  those  must  be
       escaped	(see the "Quoting and escaping" section in the ffmpeg-utils(1)
       manual).

       Options

       use_fifo bool
	   If set to 1, slave outputs will be processed	 in  separate  threads
	   using  the  fifo  muxer.  This  allows  to compensate for different
	   speed/latency/reliability  of   outputs   and   setup   transparent
	   recovery. By default this feature is turned off.

       fifo_options
	   Options to pass to fifo pseudo-muxer instances. See fifo.

       Muxer  options  can be specified for each slave by prepending them as a
       list of key=value pairs separated by ':', between square	 brackets.  If
       the  options  values  contain a special character or the ':' separator,
       they must be escaped; note that this is a second level escaping.

       The following special options are also recognized:

       f   Specify the format name. Required if it cannot be guessed from  the
	   output URL.

       bsfs[/spec]
	   Specify  a  list  of	 bitstream  filters  to apply to the specified
	   output.

	   It is possible to specify to which streams a given bitstream filter
	   applies, by appending a stream specifier to the option separated by
	   "/".	 spec  must  be	 a  stream  specifier	(see   Format	stream
	   specifiers).

	   If  the  stream  specifier  is not specified, the bitstream filters
	   will be applied to all streams in the output. This will cause  that
	   output  operation  to  fail if the output contains streams to which
	   the bitstream filter	 cannot	 be  applied  e.g.  "h264_mp4toannexb"
	   being applied to an output containing an audio stream.

	   Options  for	 a  bitstream  filter must be specified in the form of
	   "opt=value".

	   Several bitstream filters can be specified, separated by ",".

       use_fifo bool
	   This allows to override tee muxer use_fifo  option  for  individual
	   slave muxer.

       fifo_options
	   This allows to override tee muxer fifo_options for individual slave
	   muxer.  See fifo.

       select
	   Select  the	streams	 that  should  be  mapped to the slave output,
	   specified by a stream specifier. If not specified, this defaults to
	   all the mapped streams. This will cause that	 output	 operation  to
	   fail if the output format does not accept all mapped streams.

	   You	may  use  multiple stream specifiers separated by commas (",")
	   e.g.: "a:0,v"

       onfail
	   Specify behaviour on output failure. This  can  be  set  to	either
	   "abort"  (which  is	default) or "ignore". "abort" will cause whole
	   process to fail in case of failure on this slave  output.  "ignore"
	   will	 ignore failure on this output, so other outputs will continue
	   without being affected.

       Examples

       •   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/"

       •   As above, but continue streaming even if output to local file fails
	   (for example local drive fills up):

		   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		     "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       •   Use	ffmpeg	to  encode  the	 input,	 and  send the output to three
	   different destinations. The "dump_extra" bitstream filter  is  used
	   to  add  extradata  information  to	all the output video keyframes
	   packets, as requested by the MPEG-TS format. The select  option  is
	   applied to out.aac in order to make it contain only audio packets.

		   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
			  -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

       •   As  above,  but select only stream "a:1" for the audio output. Note
	   that a second level escaping must be performed, as ":" is a special
	   character used to separate options.

		   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
			  -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

   webm_chunk
       WebM Live Chunk Muxer.

       This muxer writes out WebM headers and chunks as separate  files	 which
       can be consumed by clients that support WebM Live streams via DASH.

       Options

       This muxer supports the following options:

       chunk_start_index
	   Index of the first chunk (defaults to 0).

       header
	   Filename  of	 the  header  where  the  initialization  data will be
	   written.

       audio_chunk_duration
	   Duration of each audio chunk in milliseconds (defaults to 5000).

       Example

	       ffmpeg -f v4l2 -i /dev/video0 \
		      -f alsa -i hw:0 \
		      -map 0:0 \
		      -c:v libvpx-vp9 \
		      -s 640x360 -keyint_min 30 -g 30 \
		      -f webm_chunk \
		      -header webm_live_video_360.hdr \
		      -chunk_start_index 1 \
		      webm_live_video_360_%d.chk \
		      -map 1:0 \
		      -c:a libvorbis \
		      -b:a 128k \
		      -f webm_chunk \
		      -header webm_live_audio_128.hdr \
		      -chunk_start_index 1 \
		      -audio_chunk_duration 1000 \
		      webm_live_audio_128_%d.chk

   webm_dash_manifest
       WebM DASH Manifest muxer.

       This muxer implements the WebM DASH Manifest specification to  generate
       the  DASH  manifest  XML. It also supports manifest generation for DASH
       live streams.

       For more information see:

       •   WebM			     DASH			Specification:
	   <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

       •   ISO			     DASH			Specification:
	   <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

       Options

       This muxer supports the following options:

       adaptation_sets
	   This	 option	 has   the   following	 syntax:   "id=x,streams=a,b,c
	   id=y,streams=d,e"  where  x and y are the unique identifiers of the
	   adaptation  sets  and  a,b,c,d  and	e  are	the  indices  of   the
	   corresponding  audio	 and  video  streams. Any number of adaptation
	   sets can be added using this option.

       live
	   Set this to 1 to create a live stream DASH Manifest. Default: 0.

       chunk_start_index
	   Start index of the first chunk. This will  go  in  the  startNumber
	   attribute  of the SegmentTemplate element in the manifest. Default:
	   0.

       chunk_duration_ms
	   Duration of each  chunk  in	milliseconds.  This  will  go  in  the
	   duration  attribute of the SegmentTemplate element in the manifest.
	   Default: 1000.

       utc_timing_url
	   URL of the page that will return the UTC timestamp in  ISO  format.
	   This will go in the value attribute of the UTCTiming element in the
	   manifest.  Default: None.

       time_shift_buffer_depth
	   Smallest   time   (in   seconds)  shifting  buffer  for  which  any
	   Representation is guaranteed to be available. This will go  in  the
	   timeShiftBufferDepth attribute of the MPD element. Default: 60.

       minimum_update_period
	   Minimum update period (in seconds) of the manifest. This will go in
	   the minimumUpdatePeriod attribute of the MPD element. Default: 0.

       Example

	       ffmpeg -f webm_dash_manifest -i video1.webm \
		      -f webm_dash_manifest -i video2.webm \
		      -f webm_dash_manifest -i audio1.webm \
		      -f webm_dash_manifest -i audio2.webm \
		      -map 0 -map 1 -map 2 -map 3 \
		      -c copy \
		      -f webm_dash_manifest \
		      -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
		      manifest.xml

METADATA
       FFmpeg  is  able	 to  dump  metadata  from  media  files	 into a simple
       UTF-8-encoded INI-like text file	 and  then  load  it  back  using  the
       metadata muxer/demuxer.

       The file format is as follows:

       1.  A  file  consists of a header and a number of metadata tags divided
	   into sections, each on its own line.

       2.  The header is a ;FFMETADATA string, followed by  a  version	number
	   (now 1).

       3.  Metadata tags are of the form key=value

       4.  Immediately after header follows global metadata

       5.  After    global    metadata	  there	   may	  be   sections	  with
	   per-stream/per-chapter metadata.

       6.  A section starts with the section name in uppercase (i.e. STREAM or
	   CHAPTER) in brackets ([, ]) and ends with next section  or  end  of
	   file.

       7.  At  the  beginning  of  a  chapter section there may be an optional
	   timebase to be used for  start/end  values.	It  must  be  in  form
	   TIMEBASE=num/den,  where  num and den are integers. If the timebase
	   is missing then start/end times are assumed to be in nanoseconds.

	   Next a chapter section must contain chapter start and end times  in
	   form START=num, END=num, where num is a positive integer.

       8.  Empty lines and lines starting with ; or # are ignored.

       9.  Metadata  keys  or values containing special characters (=, ;, #, \
	   and a newline) must be escaped with a backslash \.

       10. Note that whitespace in metadata (e.g. foo = bar) is considered  to
	   be a part of the tag (in the example above key is foo , value is
	    bar).

       A ffmetadata file might look like this:

	       ;FFMETADATA1
	       title=bike\\shed
	       ;this is a comment
	       artist=FFmpeg troll team

	       [CHAPTER]
	       TIMEBASE=1/1000
	       START=0
	       #chapter ends at 0:01:00
	       END=60000
	       title=chapter \#1
	       [STREAM]
	       title=multi\
	       line

       By  using  the  ffmetadata  muxer and demuxer it is possible to extract
       metadata from an input file to an ffmetadata file, and  then  transcode
       the file into an output file with the edited ffmetadata file.

       Extracting an ffmetadata file with ffmpeg goes as follows:

	       ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

       Reinserting  edited  metadata  information from the FFMETADATAFILE file
       can be done as:

	       ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

PROTOCOL OPTIONS
       The libavformat library provides some generic global options, which can
       be set on all the protocols. In addition each protocol may support  so-
       called private options, which are specific for that component.

       Options	may be set by specifying -option value in the FFmpeg tools, or
       by setting the value explicitly in  the	"AVFormatContext"  options  or
       using the libavutil/opt.h API for programmatic use.

       The list of supported options follows:

       protocol_whitelist list (input)
	   Set	a  ","-separated  list of allowed protocols. "ALL" matches all
	   protocols. Protocols prefixed by "-" are disabled.	All  protocols
	   are	allowed	 by  default but protocols used by an another protocol
	   (nested protocols) are restricted to a per protocol subset.

PROTOCOLS
       Protocols are configured elements  in  FFmpeg  that  enable  access  to
       resources that require specific protocols.

       When  you  configure your FFmpeg build, all the supported protocols are
       enabled by default. You can list all available ones using the configure
       option "--list-protocols".

       You  can	 disable  all  the  protocols  using  the   configure	option
       "--disable-protocols",  and  selectively	 enable	 a  protocol using the
       option "--enable-protocol=PROTOCOL", or you can	disable	 a  particular
       protocol using the option "--disable-protocol=PROTOCOL".

       The  option  "-protocols"  of  the  ff*	tools will display the list of
       supported protocols.

       All protocols accept the following options:

       rw_timeout
	   Maximum  time  to  wait  for	 (network)  read/write	operations  to
	   complete, in microseconds.

       A description of the currently available protocols follows.

   amqp
       Advanced	 Message  Queueing  Protocol  (AMQP) version 0-9-1 is a broker
       based publish-subscribe communication protocol.

       FFmpeg must be compiled with --enable-librabbitmq to  support  AMQP.  A
       separate	 AMQP  broker  must  also  be run. An example open-source AMQP
       broker is RabbitMQ.

       After starting the broker, an FFmpeg client  may	 stream	 data  to  the
       broker using the command:

	       ffmpeg -re -i input -f mpegts amqp://[[user]:[password]@]hostname[:port][/vhost]

       Where hostname and port (default is 5672) is the address of the broker.
       The client may also set a user/password for authentication. The default
       for  both  fields is "guest". Name of virtual host on broker can be set
       with vhost. The default value is "/".

       Muliple subscribers may stream from the broker using the command:

	       ffplay amqp://[[user]:[password]@]hostname[:port][/vhost]

       In RabbitMQ all data published to the broker flows through  a  specific
       exchange,  and  each  subscribing  client has an assigned queue/buffer.
       When a packet arrives at an exchange, it may be copied  to  a  client's
       queue depending on the exchange and routing_key fields.

       The following options are supported:

       exchange
	   Sets	 the  exchange	to  use	 on  the  broker. RabbitMQ has several
	   predefined exchanges: "amq.direct" is the default  exchange,	 where
	   the	publisher  and	subscriber  must  have a matching routing_key;
	   "amq.fanout" is the same as a broadcast operation (i.e. the data is
	   forwarded to all queues on the fanout exchange independent  of  the
	   routing_key);  and  "amq.topic"  is	similar	 to  "amq.direct", but
	   allows for more complex pattern matching  (refer  to	 the  RabbitMQ
	   documentation).

       routing_key
	   Sets	 the routing key. The default value is "amqp". The routing key
	   is used on the "amq.direct" and  "amq.topic"	 exchanges  to	decide
	   whether packets are written to the queue of a subscriber.

       pkt_size
	   Maximum size of each packet sent/received to the broker. Default is
	   131072.   Minimum is 4096 and max is any large value (representable
	   by an int). When receiving packets, this sets  an  internal	buffer
	   size	 in  FFmpeg. It should be equal to or greater than the size of
	   the published packets to the broker. Otherwise the received message
	   may be truncated causing decoding errors.

       connection_timeout
	   The timeout in seconds during the initial connection to the broker.
	   The default value is rw_timeout, or 5 seconds if rw_timeout is  not
	   set.

       delivery_mode mode
	   Sets	 the  delivery	mode  of  each	message	 sent  to broker.  The
	   following values are accepted:

	   persistent
	       Delivery mode set to "persistent"  (2).	This  is  the  default
	       value.	Messages may be written to the broker's disk depending
	       on its setup.

	   non-persistent
	       Delivery mode set to "non-persistent" (1).  Messages will  stay
	       in broker's memory unless the broker is under memory pressure.

   async
       Asynchronous data filling wrapper for input stream.

       Fill  data in a background thread, to decouple I/O operation from demux
       thread.

	       async:<URL>
	       async:http://host/resource
	       async:cache:http://host/resource

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
	   BluRay angle

       chapter
	   Start chapter (1...N)

       playlist
	   Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

       Read longest playlist from BluRay mounted to /mnt/bluray:

	       bluray:/mnt/bluray

       Read angle 2 of playlist 4 from BluRay mounted  to  /mnt/bluray,	 start
       from chapter 2:

	       -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

   cache
       Caching wrapper for input stream.

       Cache  the input stream to temporary file. It brings seeking capability
       to live streams.

       The accepted options are:

       read_ahead_limit
	   Amount  in  bytes  that  may	 be  read  ahead  when	seeking	 isn't
	   supported.  Range  is  -1 to INT_MAX.  -1 for unlimited. Default is
	   65536.

       URL Syntax is

	       cache:<URL>

   concat
       Physical concatenation protocol.

       Read and seek from many resources in sequence as if they were a	unique
       resource.

       A URL accepted by this protocol has the syntax:

	       concat:<URL1>|<URL2>|...|<URLN>

       where  URL1,  URL2,  ...,  URLN	are  the  urls	of  the resource to be
       concatenated, each one possibly specifying a distinct protocol.

       For example to read  a  sequence	 of  files  split1.mpeg,  split2.mpeg,
       split3.mpeg with ffplay use the command:

	       ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

       Note that you may need to escape the character "|" which is special for
       many shells.

   concatf
       Physical	 concatenation	protocol  using a line break delimited list of
       resources.

       Read and seek from many resources in sequence as if they were a	unique
       resource.

       A URL accepted by this protocol has the syntax:

	       concatf:<URL>

       where  URL  is  the  url	 containing  a	line  break  delimited list of
       resources to be concatenated, each one possibly specifying  a  distinct
       protocol.  Special  characters must be escaped with backslash or single
       quotes. See the "Quoting and escaping" section in  the  ffmpeg-utils(1)
       manual.

       For  example  to	 read  a  sequence  of files split1.mpeg, split2.mpeg,
       split3.mpeg listed in separate  lines  within  a	 file  split.txt  with
       ffplay use the command:

	       ffplay concatf:split.txt

       Where split.txt contains the lines:

	       split1.mpeg
	       split2.mpeg
	       split3.mpeg

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set	the  AES  decryption  key  binary block from given hexadecimal
	   representation.

       iv  Set the AES decryption  initialization  vector  binary  block  from
	   given hexadecimal representation.

       Accepted URL formats:

	       crypto:<URL>
	       crypto+<URL>

   data
       Data	     in-line	      in	  the	      URI.	   See
       <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given inline with ffmpeg:

	       ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   fd
       File descriptor access protocol.

       The accepted syntax is:

	       fd: -fd <file_descriptor>

       If fd is not specified, by default the stdout file descriptor  will  be
       used  for  writing,  stdin  for	reading.  Unlike the pipe protocol, fd
       protocol has seek support if it corresponding to	 a  regular  file.  fd
       protocol doesn't support pass file descriptor via URL for security.

       This protocol accepts the following options:

       blocksize
	   Set	I/O  operation	maximum block size, in bytes. Default value is
	   "INT_MAX", which results in not limiting the requested block	 size.
	   Setting this value reasonably low improves user termination request
	   reaction time, which is valuable if data transmission is slow.

       fd  Set file descriptor.

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

	       file:<filename>

       where filename is the path of the file to read.

       An  URL	that  does  not have a protocol prefix will be assumed to be a
       file URL. Depending on the build, an URL that looks like a Windows path
       with the drive letter at the beginning will also be  assumed  to	 be  a
       file URL (usually not the case in builds for unix-like systems).

       For example to read from a file input.mpeg with ffmpeg use the command:

	       ffmpeg -i file:input.mpeg output.mpeg

       This protocol accepts the following options:

       truncate
	   Truncate  existing  files  on  write,  if  set  to  1. A value of 0
	   prevents truncating. Default value is 1.

       blocksize
	   Set I/O operation maximum block size, in bytes.  Default  value  is
	   "INT_MAX",  which results in not limiting the requested block size.
	   Setting this value reasonably low improves user termination request
	   reaction time, which is valuable for files on slow medium.

       follow
	   If set to 1, the protocol will retry reading	 at  the  end  of  the
	   file, allowing reading files that still are being written. In order
	   for	this  to  terminate,  you  either  need	 to use the rw_timeout
	   option, or use the interrupt callback (for API users).

       seekable
	   Controls if seekability is advertised on the	 file.	0  means  non-
	   seekable,  -1  means	 auto (seekable for normal files, non-seekable
	   for named pipes).

	   Many	 demuxers   handle   seekable	and   non-seekable   resources
	   differently,	 overriding  this might speed up opening certain files
	   at the cost of losing some features (e.g. accurate seeking).

   ftp
       FTP (File Transfer Protocol).

       Read from or write to remote resources using FTP protocol.

       Following syntax is required.

	       ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
	   Set timeout in microseconds of socket I/O operations	 used  by  the
	   underlying  low  level operation. By default it is set to -1, which
	   means that the timeout is not specified.

       ftp-user
	   Set a user to be used for authenticating to the FTP server. This is
	   overridden by the user in the FTP URL.

       ftp-password
	   Set a password to be used for authenticating	 to  the  FTP  server.
	   This	 is  overridden	 by  the  password  in the FTP URL, or by ftp-
	   anonymous-password if no user is set.

       ftp-anonymous-password
	   Password used when login as anonymous  user.	 Typically  an	e-mail
	   address should be used.

       ftp-write-seekable
	   Control  seekability of connection during encoding. If set to 1 the
	   resource is supposed to be seekable, if set to 0 it is assumed  not
	   to be seekable. Default value is 0.

       NOTE:  Protocol	can be used as output, but it is recommended to not do
       it,  unless  special  care   is	 taken	 (tests,   customized	server
       configuration  etc.).  Different	 FTP  servers  behave in different way
       during seek operation. ff* tools may produce incomplete content due  to
       server limitations.

   gopher
       Gopher protocol.

   gophers
       Gophers protocol.

       The Gopher protocol with TLS encapsulation.

   hls
       Read  Apple HTTP Live Streaming compliant segmented stream as a uniform
       one. The M3U8 playlists describing the  segments	 can  be  remote  HTTP
       resources  or  local  files, accessed using the standard file protocol.
       The nested protocol is declared by specifying "+proto"  after  the  hls
       URI scheme name, where proto is either "file" or "http".

	       hls+http://host/path/to/remote/resource.m3u8
	       hls+file://path/to/local/resource.m3u8

       Using  this  protocol is discouraged - the hls demuxer should work just
       as well (if not, please report the issues) and is  more	complete.   To
       use  the	 hls  demuxer  instead, simply use the direct URLs to the m3u8
       files.

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options:

       seekable
	   Control seekability of connection. If set  to  1  the  resource  is
	   supposed  to	 be  seekable,	if  set	 to  0 it is assumed not to be
	   seekable, if set to -1 it will try to autodetect if it is seekable.
	   Default value is -1.

       chunked_post
	   If set to 1 use chunked Transfer-Encoding for posts, default is 1.

       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>")

       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_on_network_error
	   Reconnect automatically in case of TCP/TLS errors during connect.

       reconnect_on_http_error
	   A  comma  separated	list of HTTP status codes to reconnect on. The
	   list can include specific status codes (e.g. '503') or the  strings
	   '4xx' / '5xx'.

       reconnect_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.

       auth_type
	   Set	HTTP  authentication  type.  No	 option for Digest, since this
	   method requires getting nonce parameters from the server first  and
	   can't be used straight away like Basic.

	   none
	       Choose  the HTTP authentication type automatically. This is the
	       default.

	   basic
	       Choose the HTTP basic authentication.

	       Basic  authentication  sends  a	Base64-encoded	 string	  that
	       contains a user name and password for the client. Base64 is not
	       a  form	of  encryption	and  should  be considered the same as
	       sending the user name and password in clear text (Base64	 is  a
	       reversible  encoding).	If  a  resource needs to be protected,
	       strongly consider using an  authentication  scheme  other  than
	       basic  authentication.  HTTPS/TLS  should  be  used  with basic
	       authentication.	   Without    these    additional     security
	       enhancements,  basic  authentication  should  not  be  used  to
	       protect sensitive or valuable information.

       HTTP Cookies

       Some HTTP requests will be denied unless cookie values  are  passed  in
       with  the  request.  The	 cookies  option  allows  these	 cookies to be
       specified. At the very least, each cookie must specify  a  value	 along
       with  a	path and domain.  HTTP requests that match both the domain and
       path will automatically include the cookie value	 in  the  HTTP	Cookie
       header field. Multiple cookies can be delimited by a newline.

       The required syntax to play a stream specifying a cookie is:

	       ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

   Icecast
       Icecast protocol (stream to Icecast servers)

       This protocol accepts the following options:

       ice_genre
	   Set the stream genre.

       ice_name
	   Set the stream name.

       ice_description
	   Set the stream description.

       ice_url
	   Set the stream website URL.

       ice_public
	   Set if the stream should be public.	The default is 0 (not public).

       user_agent
	   Override  the  User-Agent  header. If not specified a string of the
	   form "Lavf/<version>" will be used.

       password
	   Set the Icecast mountpoint password.

       content_type
	   Set the stream content type. This must be set if  it	 is  different
	   from audio/mpeg.

       legacy_icecast
	   This	 enables  support  for	Icecast	 versions < 2.4.0, that do not
	   support the HTTP PUT method but the SOURCE method.

       tls Establish a TLS (HTTPS) connection to Icecast.

	       icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

   ipfs
       InterPlanetary File System (IPFS)  protocol  support.  One  can	access
       files  stored on the IPFS network through so-called gateways. These are
       http(s) endpoints.  This	 protocol  wraps  the  IPFS  native  protocols
       (ipfs://	 and  ipns://)	to  be	sent to such a gateway. Users can (and
       should) host their own node which means this protocol  will  use	 one's
       local gateway to access files on the IPFS network.

       This protocol accepts the following options:

       gateway
	   Defines  the	 gateway to use. When not set, the protocol will first
	   try	locating  the  local  gateway  by  looking  at	$IPFS_GATEWAY,
	   $IPFS_PATH and "$HOME/.ipfs/", in that order.

       One can use this protocol in 2 ways. Using IPFS:

	       ffplay ipfs://<hash>

       Or the IPNS protocol (IPNS is mutable IPFS):

	       ffplay ipns://<hash>

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

	       mmsh://<server>[:<port>][/<app>][/<playpath>]

   md5
       MD5 output protocol.

       Computes	 the  MD5  hash of the data to be written, and on close writes
       this to the designated output or stdout if none is specified. It can be
       used to test muxers without writing an actual file.

       Some examples follow.

	       # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
	       ffmpeg -i input.flv -f avi -y md5:output.avi.md5

	       # Write the MD5 hash of the encoded AVI file to stdout.
	       ffmpeg -i input.flv -f avi -y md5:

       Note that some formats (typically MOV) require the output  protocol  to
       be seekable, so they will fail with the MD5 output protocol.

   pipe
       UNIX pipe access protocol.

       Read and write from UNIX pipes.

       The accepted syntax is:

	       pipe:[<number>]

       If  fd  isn't specified, number is the number corresponding to the file
       descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for  stderr).
       If  number is not specified, by default the stdout file descriptor will
       be used for writing, stdin for reading.

       For example to read from stdin with ffmpeg:

	       cat test.wav | ffmpeg -i pipe:0
	       # ...this is the same as...
	       cat test.wav | ffmpeg -i pipe:

       For writing to stdout with ffmpeg:

	       ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
	       # ...this is the same as...
	       ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       This protocol accepts the following options:

       blocksize
	   Set I/O operation maximum block size, in bytes.  Default  value  is
	   "INT_MAX",  which results in not limiting the requested block size.
	   Setting this value reasonably low improves user termination request
	   reaction time, which is valuable if data transmission is slow.

       fd  Set file descriptor.

       Note that some formats (typically MOV), require the output protocol  to
       be seekable, so they will fail with the pipe output protocol.

   prompeg
       Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

       The  Pro-MPEG  CoP#3  FEC is a 2D parity-check forward error correction
       mechanism for MPEG-2 Transport Streams sent over RTP.

       This protocol must be used in conjunction with the  "rtp_mpegts"	 muxer
       and the "rtp" protocol.

       The required syntax is:

	       -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

       The  destination UDP ports are "port + 2" for the column FEC stream and
       "port + 4" for the row FEC stream.

       This protocol accepts the following options:

       l=n The number of columns (4-20, LxD <= 100)

       d=n The number of rows (4-20, LxD <= 100)

       Example usage:

	       -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

   rist
       Reliable Internet Streaming Transport protocol

       The accepted options are:

       rist_profile
	   Supported values:

	   simple
	   main
	       This one is default.

	   advanced
       buffer_size
	   Set internal RIST buffer size in milliseconds for retransmission of
	   data.  Default value is 0 which means the librist default (1	 sec).
	   Maximum value is 30 seconds.

       fifo_size
	   Size of the librist receiver output fifo in number of packets. This
	   must	 be a power of 2.  Defaults to 8192 (vs the librist default of
	   1024).

       overrun_nonfatal=1|0
	   Survive in case of librist fifo buffer overrun. Default value is 0.

       pkt_size
	   Set maximum packet size for sending data. 1316 by default.

       log_level
	   Set loglevel for RIST logging messages. You only need to  set  this
	   if  you  explicitly	want  to enable debug level messages or packet
	   loss simulation, otherwise the regular loglevel is respected.

       secret
	   Set override of encryption secret, by default is unset.

       encryption
	   Set encryption type, by default is disabled.	 Acceptable values are
	   128 and 256.

   rtmp
       Real-Time Messaging Protocol.

       The  Real-Time  Messaging  Protocol  (RTMP)  is	used   for   streaming
       multimedia content across a TCP/IP network.

       The required syntax is:

	       rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

       The accepted parameters are:

       username
	   An optional username (mostly for publishing).

       password
	   An optional password (mostly for publishing).

       server
	   The address of the RTMP server.

       port
	   The number of the TCP port to use (by default is 1935).

       app It is the name of the application to access. It usually corresponds
	   to  the  path where the application is installed on the RTMP server
	   (e.g. /ondemand/, /flash/live/, etc.). You can override  the	 value
	   parsed from the URI through the "rtmp_app" option, too.

       playpath
	   It  is  the	path or name of the resource to play with reference to
	   the application specified in app, may be prefixed  by  "mp4:".  You
	   can	 override   the	  value	  parsed  from	the  URI  through  the
	   "rtmp_playpath" option, too.

       listen
	   Act as a server, listening for an incoming connection.

       timeout
	   Maximum time to wait for the incoming connection. Implies listen.

       Additionally, the following parameters can  be  set  via	 command  line
       options (or in code via "AVOption"s):

       rtmp_app
	   Name	 of  application  to  connect  on the RTMP server. This option
	   overrides the parameter specified in the URI.

       rtmp_buffer
	   Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
	   Extra arbitrary AMF connection parameters, parsed  from  a  string,
	   e.g.	 like  "B:1  S:authMe  O:1 NN:code:1.23 NS:flag:ok O:0".  Each
	   value is prefixed by a single character denoting the	 type,	B  for
	   Boolean,  N	for number, S for string, O for object, or Z for null,
	   followed by a colon. For Booleans the data must be either  0	 or  1
	   for	FALSE  or  TRUE,  respectively.	 Likewise for Objects the data
	   must be 0 or 1 to end or begin an object, respectively. Data	 items
	   in  subobjects  may	be  named,  by prefixing the type with 'N' and
	   specifying the name before the  value  (i.e.	 "NB:myFlag:1").  This
	   option  may	be  used  multiple  times  to  construct arbitrary AMF
	   sequences.

       rtmp_enhanced_codecs
	   Specify the list of codecs the client advertises to support	in  an
	   enhanced  RTMP  stream.  This  option  should  be  set  to  a comma
	   separated list of fourcc values, like "hvc1,av01,vp09" for multiple
	   codecs or "hvc1" for only one codec. The  specified	list  will  be
	   presented  in  the  "fourCcLive"  property  of  the Connect Command
	   Message.

       rtmp_flashver
	   Version of the Flash plugin used to run the SWF player. The default
	   is  LNX  9,0,124,2.	(When  publishing,  the	 default  is  FMLE/3.0
	   (compatible; <libavformat version>).)

       rtmp_flush_interval
	   Number  of  packets	flushed	 in the same request (RTMPT only). The
	   default is 10.

       rtmp_live
	   Specify that the media is a live stream. No resuming or seeking  in
	   live	 streams  is possible. The default value is "any", which means
	   the subscriber first tries to play the live stream specified in the
	   playpath. If a live stream of that name is not found, it plays  the
	   recorded   stream.	The  other  possible  values  are  "live"  and
	   "recorded".

       rtmp_pageurl
	   URL of the web page in which the media was embedded. By default  no
	   value will be sent.

       rtmp_playpath
	   Stream  identifier to play or to publish. This option overrides the
	   parameter specified in the URI.

       rtmp_subscribe
	   Name of live stream to subscribe to. By default no  value  will  be
	   sent.   It  is only sent if the option is specified or if rtmp_live
	   is set to live.

       rtmp_swfhash
	   SHA256 hash of the decompressed SWF file (32 bytes).

       rtmp_swfsize
	   Size of the decompressed SWF file, required for SWFVerification.

       rtmp_swfurl
	   URL of the SWF player for the media. By default no  value  will  be
	   sent.

       rtmp_swfverify
	   URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
	   URL of the target stream. Defaults to proto://host[:port]/app.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark:  Writing  to	 the  socket  is  currently  not  optimized to
	   minimize system calls  and  reduces	the  efficiency	 /  effect  of
	   TCP_NODELAY.

       For  example  to	 read with ffplay a multimedia resource named "sample"
       from the application "vod" from an RTMP server "myserver":

	       ffplay rtmp://myserver/vod/sample

       To publish to a password protected server, passing the playpath and app
       names separately:

	       ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

   rtmpe
       Encrypted Real-Time Messaging Protocol.

       The  Encrypted  Real-Time  Messaging  Protocol  (RTMPE)	is  used   for
       streaming  multimedia content within standard cryptographic primitives,
       consisting of Diffie-Hellman key exchange and HMACSHA256, generating  a
       pair of RC4 keys.

   rtmps
       Real-Time Messaging Protocol over a secure SSL connection.

       The   Real-Time	Messaging  Protocol  (RTMPS)  is  used	for  streaming
       multimedia content across an encrypted connection.

   rtmpt
       Real-Time Messaging Protocol tunneled through HTTP.

       The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is  used
       for  streaming  multimedia  content  within  HTTP  requests to traverse
       firewalls.

   rtmpte
       Encrypted Real-Time Messaging Protocol tunneled through HTTP.

       The  Encrypted  Real-Time  Messaging  Protocol  tunneled	 through  HTTP
       (RTMPTE)	 is used for streaming multimedia content within HTTP requests
       to traverse firewalls.

   rtmpts
       Real-Time Messaging Protocol tunneled through HTTPS.

       The Real-Time Messaging Protocol tunneled  through  HTTPS  (RTMPTS)  is
       used for streaming multimedia content within HTTPS requests to traverse
       firewalls.

   libsmbclient
       libsmbclient permits one to manipulate CIFS/SMB network resources.

       Following syntax is required.

	       smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

       This protocol accepts the following options.

       timeout
	   Set	timeout	 in  milliseconds of socket I/O operations used by the
	   underlying low level operation. By default it is set to  -1,	 which
	   means that the timeout is not specified.

       truncate
	   Truncate  existing  files  on  write,  if  set  to  1. A value of 0
	   prevents truncating. Default value is 1.

       workgroup
	   Set the workgroup used for making connections. By default workgroup
	   is not specified.

       For more information see: <http://www.samba.org/>.

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax is required.

	       sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
	   Set timeout of socket I/O operations used  by  the  underlying  low
	   level  operation.  By default it is set to -1, which means that the
	   timeout is not specified.

       truncate
	   Truncate existing files on write,  if  set  to  1.  A  value	 of  0
	   prevents truncating. Default value is 1.

       private_key
	   Specify  the	 path of the file containing private key to use during
	   authorization.  By default libssh searches for keys in the  ~/.ssh/
	   directory.

       Example: Play a file stored on remote server.

	       ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

   librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
       Real-Time   Messaging  Protocol	and  its  variants  supported  through
       librtmp.

       Requires the  presence  of  the	librtmp	 headers  and  library	during
       configuration.	You  need  to  explicitly  configure  the  build  with
       "--enable-librtmp". If  enabled	this  will  replace  the  native  RTMP
       protocol.

       This protocol provides most client functions and a few server functions
       needed  to  support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP
       (RTMPE), RTMP over SSL/TLS  (RTMPS)  and	 tunneled  variants  of	 these
       encrypted types (RTMPTE, RTMPTS).

       The required syntax is:

	       <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

       where  rtmp_proto  is  one  of  the  strings  "rtmp", "rtmpt", "rtmpe",
       "rtmps", "rtmpte", "rtmpts" corresponding to  each  RTMP	 variant,  and
       server,	port,  app and playpath have the same meaning as specified for
       the RTMP native protocol.  options contains a list  of  space-separated
       options of the form key=val.

       See the librtmp manual page (man 3 librtmp) for more information.

       For  example,  to  stream  a  file in real-time to an RTMP server using
       ffmpeg:

	       ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

       To play the same stream using ffplay:

	       ffplay "rtmp://myserver/live/mystream live=1"

   rtp
       Real-time Transport Protocol.

       The     required	    syntax	for	 an	 RTP	  URL	   is:
       rtp://hostname[:port][?option=val...]

       port specifies the RTP port to use.

       The following URL options are supported:

       ttl=n
	   Set the TTL (Time-To-Live) value (for multicast only).

       rtcpport=n
	   Set the remote RTCP port to n.

       localrtpport=n
	   Set the local RTP port to n.

       localrtcpport=n'
	   Set the local RTCP port to n.

       pkt_size=n
	   Set max packet size (in bytes) to n.

       buffer_size=size
	   Set the maximum UDP socket buffer size in bytes.

       connect=0|1
	   Do  a  connect()  on the UDP socket (if set to 1) or not (if set to
	   0).

       sources=ip[,ip]
	   List allowed source IP addresses.

       block=ip[,ip]
	   List disallowed (blocked) source IP addresses.

       write_to_source=0|1
	   Send packets to the source address of the  latest  received	packet
	   (if set to 1) or to a default remote address (if set to 0).

       localport=n
	   Set the local RTP port to n.

       localaddr=addr
	   Local IP address of a network interface used for sending packets or
	   joining multicast groups.

       timeout=n
	   Set timeout (in microseconds) of socket I/O operations to n.

	   This is a deprecated option. Instead, localrtpport should be used.

       Important notes:

       1.  If  rtcpport	 is  not set the RTCP port will be set to the RTP port
	   value plus 1.

       2.  If localrtpport (the local RTP port) is not set any available  port
	   will be used for the local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it will be set to
	   the local RTP port value plus 1.

   rtsp
       Real-Time Streaming Protocol.

       RTSP  is	 not  technically  a  protocol handler in libavformat, it is a
       demuxer and muxer. The demuxer supports both  normal  RTSP  (with  data
       transferred  over  RTP;	this  is used by e.g. Apple and Microsoft) and
       Real-RTSP (with data transferred over RDT).

       The muxer can be used to send a stream using RTSP ANNOUNCE to a	server
       supporting   it	 (currently   Darwin   Streaming   Server  and	Mischa
       Spiegelmock's <https://github.com/revmischa/rtsp-server>).

       The required syntax for a RTSP url is:

	       rtsp://<hostname>[:<port>]/<path>

       Options can be set on the ffmpeg/ffplay command line, or	 set  in  code
       via "AVOption"s or in "avformat_open_input".

       Muxer

       The following options are supported.

       rtsp_transport
	   Set RTSP transport protocols.

	   It accepts the following values:

	   udp Use UDP as lower transport protocol.

	   tcp Use TCP (interleaving within the RTSP control channel) as lower
	       transport protocol.

	   Default value is 0.

       rtsp_flags
	   Set RTSP flags.

	   The following values are accepted:

	   latm
	       Use MP4A-LATM packetization instead of MPEG4-GENERIC for AAC.

	   rfc2190
	       Use RFC 2190 packetization instead of RFC 4629 for H.263.

	   skip_rtcp
	       Don't send RTCP sender reports.

	   h264_mode0
	       Use mode 0 for H.264 in RTP.

	   send_bye
	       Send RTCP BYE packets when finishing.

	   Default value is 0.

       min_port
	   Set minimum local UDP port. Default value is 5000.

       max_port
	   Set maximum local UDP port. Default value is 65000.

       buffer_size
	   Set the maximum socket buffer size in bytes.

       pkt_size
	   Set max send packet size (in bytes). Default value is 1472.

       Demuxer

       The following options are supported.

       initial_pause
	   Do  not  start  playing the stream immediately if set to 1. Default
	   value is 0.

       rtsp_transport
	   Set RTSP transport protocols.

	   It accepts the following values:

	   udp Use UDP as lower transport protocol.

	   tcp Use TCP (interleaving within the RTSP control channel) as lower
	       transport protocol.

	   udp_multicast
	       Use UDP multicast as lower transport protocol.

	   http
	       Use HTTP tunneling as lower transport protocol, which is useful
	       for passing proxies.

	   https
	       Use HTTPs tunneling  as	lower  transport  protocol,  which  is
	       useful  for  passing  proxies  and  widely  used	 for  security
	       consideration.

	   Multiple lower transport protocols may be specified, in  that  case
	   they	 are  tried one at a time (if the setup of one fails, the next
	   one is tried).  For the muxer, only the tcp	and  udp  options  are
	   supported.

       rtsp_flags
	   Set RTSP flags.

	   The following values are accepted:

	   filter_src
	       Accept packets only from negotiated peer address and port.

	   listen
	       Act as a server, listening for an incoming connection.

	   prefer_tcp
	       Try  TCP	 for  RTP transport first, if TCP is available as RTSP
	       RTP transport.

	   satip_raw
	       Export raw MPEG-TS stream instead of demuxing.  The  flag  will
	       simply write out the raw stream, with the original PAT/PMT/PIDs
	       intact.

	   Default value is none.

       allowed_media_types
	   Set media types to accept from the server.

	   The following flags are accepted:

	   video
	   audio
	   data
	   subtitle

	   By default it accepts all media types.

       min_port
	   Set minimum local UDP port. Default value is 5000.

       max_port
	   Set maximum local UDP port. Default value is 65000.

       listen_timeout
	   Set	 maximum   timeout   (in  seconds)  to	establish  an  initial
	   connection. Setting listen_timeout > 0 sets rtsp_flags  to  listen.
	   Default  is	-1 which means an infinite timeout when listen mode is
	   set.

       reorder_queue_size
	   Set number of packets to buffer for handling of reordered packets.

       timeout
	   Set socket TCP I/O timeout in microseconds.

       user_agent
	   Override User-Agent header. If not specified, it  defaults  to  the
	   libavformat identifier string.

       buffer_size
	   Set the maximum socket buffer size in bytes.

       When  receiving	data  over  UDP, the demuxer tries to reorder received
       packets (since they may arrive out of order, or packets	may  get  lost
       totally). This can be disabled by setting the maximum demuxing delay to
       zero (via the "max_delay" field of AVFormatContext).

       When  watching multi-bitrate Real-RTSP streams with ffplay, the streams
       to display can be chosen with "-vst" n and "-ast" n for video and audio
       respectively, and can be switched on the fly by pressing "v" and "a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       •   Watch a stream over	UDP,  with  a  max  reordering	delay  of  0.5
	   seconds:

		   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

       •   Watch a stream tunneled over HTTP:

		   ffplay -rtsp_transport http rtsp://server/video.mp4

       •   Send a stream in realtime to a RTSP server, for others to watch:

		   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       •   Receive a stream in realtime:

		   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   sap
       Session	Announcement  Protocol	(RFC  2974). This is not technically a
       protocol handler in libavformat, it is a muxer and demuxer.  It is used
       for signalling of RTP streams, by announcing the SDP  for  the  streams
       regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

	       sap://<destination>[:<port>][?<options>]

       The  RTP	 packets are sent to destination on port port, or to port 5004
       if no  port  is	specified.   options  is  a  "&"-separated  list.  The
       following options are supported:

       announce_addr=address
	   Specify  the	 destination  IP address for sending the announcements
	   to.	If omitted, the announcements are sent to  the	commonly  used
	   SAP	announcement  multicast address 224.2.127.254 (sap.mcast.net),
	   or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
	   Specify the port to send the announcements on, defaults to 9875  if
	   not specified.

       ttl=ttl
	   Specify  the	 time  to  live	 value	for  the announcements and RTP
	   packets, defaults to 255.

       same_port=0|1
	   If set to 1, send all RTP streams on the same port  pair.  If  zero
	   (the	 default),  all	 streams  are  sent on unique ports, with each
	   stream on a port 2 numbers higher than the  previous.   VLC/Live555
	   requires  this  to  be  set to 1, to be able to receive the stream.
	   The RTP stack in libavformat for receiving requires all streams  to
	   be sent on unique ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

	       ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

	       sap://[<address>][:<port>]

       address	is  the	 multicast  address to listen for announcements on, if
       omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
       port that is listened on, 9875 if omitted.

       The demuxers listens for announcements on the given address  and	 port.
       Once  an	 announcement is received, it tries to receive that particular
       stream.

       Example command lines follow.

       To play back the first stream announced on  the	normal	SAP  multicast
       address:

	       ffplay sap://

       To  play	 back  the  first stream announced on one the default IPv6 SAP
       multicast address:

	       ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

	       sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

       listen
	   If set to any value, listen for an  incoming	 connection.  Outgoing
	   connection is done by default.

       max_streams
	   Set the maximum number of streams. By default no limit is set.

   srt
       Haivision Secure Reliable Transport Protocol via libsrt.

       The supported syntax for a SRT URL is:

	       srt://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       or

	       <options> srt://<hostname>:<port>

       options contains a list of '-key val' options.

       This protocol accepts the following options.

       connect_timeout=milliseconds
	   Connection  timeout;	 SRT  cannot  connect  for  RTT > 1500 msec (2
	   handshake exchanges) with the default connect timeout of 3 seconds.
	   This option applies to the caller and rendezvous connection	modes.
	   The	connect	 timeout  is 10 times the value set for the rendezvous
	   mode (which can be used as a workaround for this connection problem
	   with earlier versions).

       ffs=bytes
	   Flight Flag Size (Window  Size),  in	 bytes.	 FFS  is  actually  an
	   internal  parameter	and  you  should  set  it  to  not  less  than
	   recv_buffer_size and mss. The default value	is  relatively	large,
	   therefore  unless  you set a very large receiver buffer, you do not
	   need to change this option. Default value is 25600.

       inputbw=bytes/seconds
	   Sender nominal input rate, in bytes per seconds.  Used  along  with
	   oheadbw,  when  maxbw  is set to relative (0), to calculate maximum
	   sending rate when recovery packets are sent	along  with  the  main
	   media stream: inputbw * (100 + oheadbw) / 100 if inputbw is not set
	   while  maxbw	 is  set  to  relative	(0),  the actual input rate is
	   evaluated inside the library. Default value is 0.

       iptos=tos
	   IP Type of Service. Applies to sender only. Default value is 0xB8.

       ipttl=ttl
	   IP Time To Live. Applies to sender only. Default value is 64.

       latency=microseconds
	   Timestamp-based Packet Delivery Delay.  Used to  absorb  bursts  of
	   missed  packet retransmissions.  This flag sets both rcvlatency and
	   peerlatency to the same value. Note that  prior  to	version	 1.3.0
	   this	 is  the  only	flag  to  set  the  latency,  however  this is
	   effectively equivalent to setting peerlatency, when side is	sender
	   and	rcvlatency when side is receiver, and the bidirectional stream
	   sending is not supported.

       listen_timeout=microseconds
	   Set socket listen timeout.

       maxbw=bytes/seconds
	   Maximum sending bandwidth,  in  bytes  per  seconds.	  -1  infinite
	   (CSRTCC  limit is 30mbps) 0 relative to input rate (see inputbw) >0
	   absolute limit value Default value is 0 (relative)

       mode=caller|listener|rendezvous
	   Connection mode.  caller opens client connection.  listener	starts
	   server  to listen for incoming connections.	rendezvous use Rendez-
	   Vous connection mode.  Default value is caller.

       mss=bytes
	   Maximum Segment Size, in bytes. Used for buffer allocation and rate
	   calculation using a packet counter assuming fully  filled  packets.
	   The smallest MSS between the peers is used. This is 1500 by default
	   in  the  overall  internet.	 This  is  the maximum size of the UDP
	   packet and can be only decreased,  unless  you  have	 some  unusual
	   dedicated network settings. Default value is 1500.

       nakreport=1|0
	   If	set  to	 1,  Receiver  will  send  `UMSG_LOSSREPORT`  messages
	   periodically until a lost packet is retransmitted or	 intentionally
	   dropped. Default value is 1.

       oheadbw=percents
	   Recovery  bandwidth	overhead  above	 input rate, in percents.  See
	   inputbw. Default value is 25%.

       passphrase=string
	   HaiCrypt Encryption/Decryption Passphrase string, length from 10 to
	   79 characters. The passphrase is  the  shared  secret  between  the
	   sender  and the receiver. It is used to generate the Key Encrypting
	   Key using PBKDF2 (Password-Based Key Derivation  Function).	It  is
	   used	 only if pbkeylen is non-zero. It is used on the receiver only
	   if the received  data  is  encrypted.   The	configured  passphrase
	   cannot be recovered (write-only).

       enforced_encryption=1|0
	   If  true,  both  connection parties must have the same password set
	   (including empty, that is, with no  encryption).  If	 the  password
	   doesn't  match  or  only one side is unencrypted, the connection is
	   rejected. Default is true.

       kmrefreshrate=packets
	   The number of packets to be transmitted after which the  encryption
	   key	is  switched  to  a  new  key.	Default	 is -1.	 -1 means auto
	   (0x1000000 in srt library). The range for this option  is  integers
	   in the 0 - "INT_MAX".

       kmpreannounce=packets
	   The	interval  between  when	 a new encryption key is sent and when
	   switchover occurs.  This  value  also  applies  to  the  subsequent
	   interval between when switchover occurs and when the old encryption
	   key is decommissioned. Default is -1.  -1 means auto (0x1000 in srt
	   library).  The  range  for  this  option  is	 integers  in  the 0 -
	   "INT_MAX".

       snddropdelay=microseconds
	   The sender's extra delay before dropping  packets.  This  delay  is
	   added to the default drop delay time interval value.

	   Special value -1: Do not drop packets on the sender at all.

       payload_size=bytes
	   Sets	 the  maximum declared size of a packet transferred during the
	   single call to the sending function in Live mode.  Use  0  if  this
	   value  isn't	 used  (which is default in file mode).	 Default is -1
	   (automatic), which typically means MPEG-TS; if you are going to use
	   SRT to send any different kind of payload, such  as,	 for  example,
	   wrapping  a	live  stream  in very small frames, then you can use a
	   bigger maximum frame size, though not greater than 1456 bytes.

       pkt_size=bytes
	   Alias for payload_size.

       peerlatency=microseconds
	   The latency value (as described in rcvlatency) that is set  by  the
	   sender side as a minimum value for the receiver.

       pbkeylen=bytes
	   Sender  encryption key length, in bytes.  Only can be set to 0, 16,
	   24 and 32.  Enable sender encryption if not	0.   Not  required  on
	   receiver  (set  to  0),  key	 size obtained from sender in HaiCrypt
	   handshake.  Default value is 0.

       rcvlatency=microseconds
	   The time that should elapse since the moment when  the  packet  was
	   sent and the moment when it's delivered to the receiver application
	   in the receiving function.  This time should be a buffer time large
	   enough  to  cover the time spent for sending, unexpectedly extended
	   RTT time, and the time needed to retransmit the  lost  UDP  packet.
	   The	effective  latency  value will be the maximum of this options'
	   value and the value of peerlatency set by  the  peer	 side.	Before
	   version 1.3.0 this option is only available as latency.

       recv_buffer_size=bytes
	   Set UDP receive buffer size, expressed in bytes.

       send_buffer_size=bytes
	   Set UDP send buffer size, expressed in bytes.

       timeout=microseconds
	   Set	raise  error  timeouts for read, write and connect operations.
	   Note that the SRT  library  has  internal  timeouts	which  can  be
	   controlled separately, the value set here is only a cap on those.

       tlpktdrop=1|0
	   Too-late  Packet  Drop.  When enabled on receiver, it skips missing
	   packets that have not been  delivered  in  time  and	 delivers  the
	   following  packets  to  the application when their time-to-play has
	   come. It also sends a fake ACK  to  the  sender.  When  enabled  on
	   sender  and	enabled	 on  the  receiving peer, the sender drops the
	   older packets that have no chance of being delivered	 in  time.  It
	   was	automatically  enabled	in the sender if the receiver supports
	   it.

       sndbuf=bytes
	   Set send buffer size, expressed in bytes.

       rcvbuf=bytes
	   Set receive buffer size, expressed in bytes.

	   Receive buffer must not be greater than ffs.

       lossmaxttl=packets
	   The value up to which the Reorder Tolerance may grow. When  Reorder
	   Tolerance  is  >  0,	 then packet loss report is delayed until that
	   number of packets come in. Reorder Tolerance increases every time a
	   "belated" packet has come, but  it  wasn't  due  to	retransmission
	   (that  is,  when  UDP  packets tend to come out of order), with the
	   difference between the latest sequence and this packet's  sequence,
	   and	not  more  than	 the  value of this option. By default it's 0,
	   which means that this mechanism is turned off, and the loss	report
	   is always sent immediately upon experiencing a "gap" in sequences.

       minversion
	   The	minimum	 SRT  version  that  is	 required  from	 the  peer.  A
	   connection to a peer that does  not	satisfy	 the  minimum  version
	   requirement will be rejected.

	   The	version	 format in hex is 0xXXYYZZ for x.y.z in human readable
	   form.

       streamid=string
	   A string limited to 512 characters that can be set  on  the	socket
	   prior to connecting. This stream ID will be able to be retrieved by
	   the	listener side from the socket that is returned from srt_accept
	   and was connected by a socket with that set stream ID. SRT does not
	   enforce any special interpretation of the contents of this  string.
	   This option doesn’t make sense in Rendezvous connection; the result
	   might  be  that  simply  one	 side will override the value from the
	   other side and it’s the matter of luck which one would win

       srt_streamid=string
	   Alias for streamid to  avoid	 conflict  with	 ffmpeg	 command  line
	   option.

       smoother=live|file
	   The	type  of  Smoother  used for the transmission for that socket,
	   which is responsible for the transmission and  congestion  control.
	   The	Smoother  type	must  be  exactly  the same on both connecting
	   parties, otherwise the connection is rejected.

       messageapi=1|0
	   When set, this socket uses  the  Message  API,  otherwise  it  uses
	   Buffer  API.	 Note  that  in live mode (see transtype) there’s only
	   message API available. In File mode you can chose to use one of two
	   modes:

	   Stream API (default, when this option is false). In this  mode  you
	   may	send as many data as you wish with one sending instruction, or
	   even use dedicated functions that read directly from	 a  file.  The
	   internal  facility  will  take  care	 of  any  speed and congestion
	   control. When receiving, you can  also  receive  as	many  data  as
	   desired,  the data not extracted will be waiting for the next call.
	   There is no boundary between data portions in the Stream mode.

	   Message API. In this mode your single  sending  instruction	passes
	   exactly one piece of data that has boundaries (a message). Contrary
	   to Live mode, this message may span across multiple UDP packets and
	   the	only  size  limitation	is that it shall fit as a whole in the
	   sending buffer. The receiver shall use as large buffer as necessary
	   to receive the message, otherwise the message will not be given up.
	   When the message is not complete (not all packets received or there
	   was a packet loss) it will not be given up.

       transtype=live|file
	   Sets the transmission type for the socket, in  particular,  setting
	   this	 option sets multiple other parameters to their default values
	   as required for a particular transmission type.

	   live: Set options as for  live  transmission.  In  this  mode,  you
	   should  send	 by one sending instruction only so many data that fit
	   in one UDP packet, and  limited  to	the  value  defined  first  in
	   payload_size	 (1316	is  default  in	 this mode). There is no speed
	   control in this mode, only the bandwidth control, if configured, in
	   order to not exceed the bandwidth with  the	overhead  transmission
	   (retransmitted and control packets).

	   file:  Set options as for non-live transmission. See messageapi for
	   further explanations

       linger=seconds
	   The number of seconds that the socket waits for  unsent  data  when
	   closing.   Default is -1. -1 means auto (off with 0 seconds in live
	   mode, on with 180 seconds in file mode). The range for this	option
	   is integers in the 0 - "INT_MAX".

       tsbpd=1|0
	   When	 true,	use  Timestamp-based Packet Delivery mode. The default
	   behavior depends on the transmission type: enabled  in  live	 mode,
	   disabled in file mode.

       For more information see: <https://github.com/Haivision/srt>.

   srtp
       Secure Real-time Transport Protocol.

       The accepted options are:

       srtp_in_suite
       srtp_out_suite
	   Select input and output encoding suites.

	   Supported values:

	   AES_CM_128_HMAC_SHA1_80
	   SRTP_AES128_CM_HMAC_SHA1_80
	   AES_CM_128_HMAC_SHA1_32
	   SRTP_AES128_CM_HMAC_SHA1_32
       srtp_in_params
       srtp_out_params
	   Set	input and output encoding parameters, which are expressed by a
	   base64-encoded representation of a binary block. The first 16 bytes
	   of this binary block are used as master key, the following 14 bytes
	   are used as master salt.

   subfile
       Virtually  extract  a  segment  of  a  file  or	another	 stream.   The
       underlying stream must be seekable.

       Accepted options:

       start
	   Start offset of the extracted segment, in bytes.

       end End	offset	of  the	 extracted  segment,  in  bytes.  If set to 0,
	   extract till end of file.

       Examples:

       Extract a chapter from a DVD VOB file (start and end  sectors  obtained
       externally and multiplied by 2048):

	       subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

       Play an AVI file directly from a TAR archive:

	       subfile,,start,183241728,end,366490624,,:archive.tar

       Play a MPEG-TS file from start offset till end:

	       subfile,,start,32815239,end,0,,:video.ts

   tee
       Writes  the  output  to	multiple protocols. The individual outputs are
       separated by |

	       tee:file://path/to/local/this.avi|file://path/to/local/that.avi

   tcp
       Transmission Control Protocol.

       The required syntax for a TCP url is:

	       tcp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       The list of supported options follows.

       listen=2|1|0
	   Listen for an incoming connection. 0	 disables  listen,  1  enables
	   listen  in  single  client  mode,  2 enables listen in multi-client
	   mode. Default value is 0.

       local_addr=addr
	   Local IP address  of	 a  network  interface	used  for  tcp	socket
	   connect.

       local_port=port
	   Local port used for tcp socket connect.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This	 option	 is  only relevant in read mode: if no data arrived in
	   more than this time interval, raise error.

       listen_timeout=milliseconds
	   Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
	   Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
	   Set send buffer size, expressed bytes.

       tcp_nodelay=1|0
	   Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

	   Remark: Writing  to	the  socket  is	 currently  not	 optimized  to
	   minimize  system  calls  and	 reduces  the  efficiency  / effect of
	   TCP_NODELAY.

       tcp_mss=bytes
	   Set maximum segment size for outgoing  TCP  packets,	 expressed  in
	   bytes.

       The  following  example	shows  how to setup a listening TCP connection
       with ffmpeg, which is then accessed with ffplay:

	       ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
	       ffplay tcp://<hostname>:<port>

   tls
       Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

       The required syntax for a TLS/SSL url is:

	       tls://<hostname>:<port>[?<options>]

       The following parameters can be set via command	line  options  (or  in
       code via "AVOption"s):

       ca_file, cafile=filename
	   A  file  containing certificate authority (CA) root certificates to
	   treat as trusted. If the linked TLS library contains a default this
	   might not need to be specified for verification to  work,  but  not
	   all	libraries and setups have defaults built in.  The file must be
	   in OpenSSL PEM format.

       tls_verify=1|0
	   If enabled, try to verify the peer that we are communicating	 with.
	   Note,  if  using  OpenSSL,  this currently only makes sure that the
	   peer certificate is signed by one of the root certificates  in  the
	   CA database, but it does not validate that the certificate actually
	   matches  the	 host  name  we	 are trying to connect to. (With other
	   backends, the host name is validated as well.)

	   This is disabled by default since it requires a CA database	to  be
	   provided by the caller in many cases.

       cert_file, cert=filename
	   A  file  containing	a certificate to use in the handshake with the
	   peer.  (When operating as server, in	 listen	 mode,	this  is  more
	   often  required  by	the  peer,  while client certificates only are
	   mandated in certain setups.)

       key_file, key=filename
	   A file containing the private key for the certificate.

       listen=1|0
	   If enabled, listen for connections on the provided port, and assume
	   the server role in the handshake instead of the client role.

       http_proxy
	   The HTTP proxy to tunnel through,  e.g.  "http://example.com:1234".
	   The proxy must support the CONNECT method.

       Example command lines:

       To create a TLS/SSL server that serves an input stream.

	       ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>

       To play back a stream from the TLS/SSL server using ffplay:

	       ffplay tls://<hostname>:<port>

   udp
       User Datagram Protocol.

       The required syntax for an UDP URL is:

	       udp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       In  case	 threading is enabled on the system, a circular buffer is used
       to store the incoming data, which allows one to reduce loss of data due
       to UDP socket  buffer  overruns.	 The  fifo_size	 and  overrun_nonfatal
       options are related to this buffer.

       The list of supported options follows.

       buffer_size=size
	   Set	the  UDP  maximum socket buffer size in bytes. This is used to
	   set either the receive or send buffer size, depending on  what  the
	   socket is used for.	Default is 32 KB for output, 384 KB for input.
	   See also fifo_size.

       bitrate=bitrate
	   If  set  to	nonzero,  the  output will have the specified constant
	   bitrate if the input has enough packets to sustain it.

       burst_bits=bits
	   When using bitrate this specifies the maximum  number  of  bits  in
	   packet bursts.

       localport=port
	   Override the local UDP port to bind with.

       localaddr=addr
	   Local IP address of a network interface used for sending packets or
	   joining multicast groups.

       pkt_size=size
	   Set the size in bytes of UDP packets.

       reuse=1|0
	   Explicitly allow or disallow reusing UDP sockets.

       ttl=ttl
	   Set the time to live value (for multicast only).

       connect=1|0
	   Initialize  the  UDP	 socket	 with  connect().  In  this  case, the
	   destination address can't  be  changed  with	 ff_udp_set_remote_url
	   later.   If	the destination address isn't known at the start, this
	   option can be specified in ff_udp_set_remote_url, too.  This allows
	   finding out the source address for the  packets  with  getsockname,
	   and	makes writes return with AVERROR(ECONNREFUSED) if "destination
	   unreachable" is received.  For receiving, this gives the benefit of
	   only receiving packets from the specified peer address/port.

       sources=address[,address]
	   Only receive packets sent from the specified addresses. In case  of
	   multicast,  also  subscribe	to multicast traffic coming from these
	   addresses only.

       block=address[,address]
	   Ignore packets sent	from  the  specified  addresses.  In  case  of
	   multicast,  also  exclude  the  source  addresses  in the multicast
	   subscription.

       fifo_size=units
	   Set the UDP receiving circular buffer size, expressed as  a	number
	   of  packets	with  size  of 188 bytes. If not specified defaults to
	   7*4096.

       overrun_nonfatal=1|0
	   Survive in case of UDP receiving circular buffer  overrun.  Default
	   value is 0.

       timeout=microseconds
	   Set raise error timeout, expressed in microseconds.

	   This	 option	 is  only relevant in read mode: if no data arrived in
	   more than this time interval, raise error.

       broadcast=1|0
	   Explicitly allow or disallow UDP broadcasting.

	   Note that broadcasting may not work properly on networks  having  a
	   broadcast storm protection.

       Examples

       •   Use ffmpeg to stream over UDP to a remote endpoint:

		   ffmpeg -i <input> -f <format> udp://<hostname>:<port>

       •   Use	ffmpeg to stream in mpegts format over UDP using 188 sized UDP
	   packets, using a large input buffer:

		   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

       •   Use ffmpeg to receive over UDP from a remote endpoint:

		   ffmpeg -i udp://[<multicast-address>]:<port> ...

   unix
       Unix local socket

       The required syntax for a Unix socket URL is:

	       unix://<filepath>

       The following parameters can be set via command	line  options  (or  in
       code via "AVOption"s):

       timeout
	   Timeout in ms.

       listen
	   Create the Unix socket in listening mode.

   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

       •   Print the list of AVFoundation supported devices and exit:

		   $ ffmpeg -f avfoundation -list_devices true -i ""

       •   Record video from video device 0 and audio from audio device 0 into
	   out.avi:

		   $ ffmpeg -f avfoundation -i "0:0" out.avi

       •   Record video from video device 2 and audio from audio device 1 into
	   out.avi:

		   $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

       •   Record  video  from the system default video device using the pixel
	   format bgr0 and do not record any audio into out.avi:

		   $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

       •   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:

	   auto
	       This is the default which means 8-bit YUV 422 or 8-bit ARGB  if
	       format autodetection is used, 8-bit YUV 422 otherwise.

	   uyvy422
	       8-bit YUV 422.

	   yuv422p10
	       10-bit YUV 422.

	   argb
	       8-bit RGB.

	   bgra
	       8-bit RGB.

	   rgb10
	       10-bit RGB.

       teletext_lines
	   If  set  to nonzero, an additional teletext stream will be captured
	   from the vertical ancillary data. Both SD PAL (576i) and HD	(1080i
	   or  1080p)  sources	are  supported.	 In  case  of HD sources, OP47
	   packets are decoded.

	   This option is  a  bitmask  of  the	SD  PAL	 VBI  lines  captured,
	   specifically lines 6 to 22, and lines 318 to 335. Line 6 is the LSB
	   in	the  mask.  Selected  lines  which  do	not  contain  teletext
	   information will be ignored. You can use the special	 all  constant
	   to  select all possible lines, or standard to skip lines 6, 318 and
	   319, which are not compatible with all receivers.

	   For	 SD   sources,	 ffmpeg	  needs	  to	be    compiled	  with
	   "--enable-libzvbi". For HD sources, on older (pre-4K) DeckLink card
	   models you have to capture in 10 bit mode.

       channels
	   Defines  number  of	audio channels to capture. Must be 2, 8 or 16.
	   Defaults to 2.

       duplex_mode
	   Sets the decklink device duplex/profile mode. Must be unset,	 half,
	   full,	   one_sub_device_full,		  one_sub_device_half,
	   two_sub_device_full, four_sub_device_half Defaults to unset.

	   Note: DeckLink SDK 11.0 have replaced  the  duplex  property	 by  a
	   profile  property.	For  the DeckLink Duo 2 and DeckLink Quad 2, a
	   profile is shared between any 2 sub-devices that utilize  the  same
	   connectors.	For  the  DeckLink 8K Pro, a profile is shared between
	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.

	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
	   one_sub_device_full,	  one_sub_device_half,	  two_sub_device_full,
	   four_sub_device_half

	   Valid  profile  modes for DeckLink Quad 2 and DeckLink Duo 2: half,
	   full

       timecode_format
	   Timecode type to include in the frame and  video  stream  metadata.
	   Must	 be none, rp188vitc, rp188vitc2, rp188ltc, rp188hfr, rp188any,
	   vitc, vitc2, or serial.  Defaults to none (not included).

	   In order to properly support 50/60 fps timecodes, the  ordering  of
	   the	queried	 timecode  types for rp188any is HFR, VITC1, VITC2 and
	   LTC for >30 fps content. Note that this is  slightly	 different  to
	   the	ordering  used	by the DeckLink API, which is HFR, VITC1, LTC,
	   VITC2.

       video_input
	   Sets the video input source. Must be unset, sdi, hdmi, optical_sdi,
	   component, composite or s_video.  Defaults to unset.

       audio_input
	   Sets the audio input source.	 Must  be  unset,  embedded,  aes_ebu,
	   analog, analog_xlr, analog_rca or microphone. Defaults to unset.

       video_pts
	   Sets	 the  video  packet  timestamp	source.	 Must be video, audio,
	   reference, wallclock or abs_wallclock.  Defaults to video.

       audio_pts
	   Sets the audio packet  timestamp  source.  Must  be	video,	audio,
	   reference, wallclock or abs_wallclock.  Defaults to audio.

       draw_bars
	   If set to true, color bars are drawn in the event of a signal loss.
	   Defaults to true.

       queue_size
	   Sets	 maximum  input buffer size in bytes. If the buffering reaches
	   this	 value,	 incoming  frames  will	 be  dropped.	 Defaults   to
	   1073741824.

       audio_depth
	   Sets the audio sample bit depth. Must be 16 or 32.  Defaults to 16.

       decklink_copyts
	   If  set  to	true,  timestamps  are	forwarded  as they are without
	   removing the initial offset.	 Defaults to false.

       timestamp_align
	   Capture start time alignment in seconds. If set to  nonzero,	 input
	   frames are dropped till the system timestamp aligns with configured
	   value.   Alignment  difference  of  up  to  one  frame  duration is
	   tolerated.  This is useful for  maintaining	input  synchronization
	   across   N	different   hardware   devices	deployed  for  'N-way'
	   redundancy. The system time of different hardware devices should be
	   synchronized with protocols such as NTP or PTP, before  using  this
	   option.   Note  that	 this  method is not foolproof. In some border
	   cases input synchronization may not happen due to thread scheduling
	   jitters in the OS.  Either sync could go wrong by 1 frame or	 in  a
	   rarer case timestamp_align seconds.	Defaults to 0.

       wait_for_tc (bool)
	   Drop	 frames	 till  a  frame	 with  timecode is received. Sometimes
	   serial timecode isn't received with the first input frame. If  that
	   happens,  the  stored  stream  timecode will be inaccurate. If this
	   option is set to true, input frames are dropped till a  frame  with
	   timecode  is	 received.   Option timecode_format must be specified.
	   Defaults to false.

       enable_klv(bool)
	   If set to true,  extracts  KLV  data	 from  VANC  and  outputs  KLV
	   packets.   KLV  VANC packets are joined based on MID and PSC fields
	   and aggregated into one KLV packet.	Defaults to false.

       Examples

       •   List input devices:

		   ffmpeg -sources decklink

       •   List supported formats:

		   ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'

       •   Capture video clip at 1080i50:

		   ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi

       •   Capture video clip at 1080i50 10 bit:

		   ffmpeg -raw_format yuv422p10 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

       •   Capture video clip at 1080i50 with 16 audio channels:

		   ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

   dshow
       Windows DirectShow input device.

       DirectShow support is enabled when FFmpeg is built with	the  mingw-w64
       project.	 Currently only audio and video devices are supported.

       Multiple devices may be opened as separate inputs, but they may also be
       opened  on  the	same  input,  which should improve synchronism between
       them.

       The input name should be in the format:

	       <TYPE>=<NAME>[:<TYPE>=<NAME>]

       where TYPE can be either audio or video, and NAME is the device's  name
       or alternative name..

       Options

       If  no  options	are specified, the device's defaults are used.	If the
       device does not support the requested options, it will fail to open.

       video_size
	   Set the video size in the captured video.

       framerate
	   Set the frame rate in the captured video.

       sample_rate
	   Set the sample rate (in Hz) of the captured audio.

       sample_size
	   Set the sample size (in bits) of the captured audio.

       channels
	   Set the number of channels in the captured audio.

       list_devices
	   If set to true, print a list of devices and exit.

       list_options
	   If set to true, print a list of selected device's options and exit.

       video_device_number
	   Set video device number for devices with the same name  (starts  at
	   0, defaults to 0).

       audio_device_number
	   Set	audio  device number for devices with the same name (starts at
	   0, defaults to 0).

       pixel_format
	   Select pixel format to be used by DirectShow. This may only be  set
	   when the video codec is not set or set to rawvideo.

       audio_buffer_size
	   Set	audio  device  buffer size in milliseconds (which can directly
	   impact latency, depending on the device).  Defaults	to  using  the
	   audio  device's  default  buffer  size  (typically some multiple of
	   500ms).  Setting this value too low can degrade  performance.   See
	   also
	   <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>

       video_pin_name
	   Select video capture pin to use by name or alternative name.

       audio_pin_name
	   Select audio capture pin to use by name or alternative name.

       crossbar_video_input_pin_number
	   Select  video  input	 pin  number for crossbar device. This will be
	   routed to the crossbar device's Video  Decoder  output  pin.	  Note
	   that	 changing this value can affect future invocations (sets a new
	   default) until system reboot occurs.

       crossbar_audio_input_pin_number
	   Select audio input pin number for crossbar  device.	This  will  be
	   routed  to  the  crossbar  device's Audio Decoder output pin.  Note
	   that changing this value can affect future invocations (sets a  new
	   default) until system reboot occurs.

       show_video_device_dialog
	   If  set  to	true, before capture starts, popup a display dialog to
	   the end user, allowing them to change video filter  properties  and
	   configurations manually.  Note that for crossbar devices, adjusting
	   values  in this dialog may be needed at times to toggle between PAL
	   (25 fps) and NTSC (29.97) input frame  rates,  sizes,  interlacing,
	   etc.	  Changing  these values can enable different scan rates/frame
	   rates and avoiding green bars at the bottom, flickering scan lines,
	   etc.	 Note that with some devices, changing	these  properties  can
	   also	 affect	 future	 invocations  (sets new defaults) until system
	   reboot occurs.

       show_audio_device_dialog
	   If set to true, before capture starts, popup a  display  dialog  to
	   the	end  user, allowing them to change audio filter properties and
	   configurations manually.

       show_video_crossbar_connection_dialog
	   If set to true, before capture starts, popup a  display  dialog  to
	   the	end  user,  allowing  them  to	manually  modify  crossbar pin
	   routings, when it opens a video device.

       show_audio_crossbar_connection_dialog
	   If set to true, before capture starts, popup a  display  dialog  to
	   the	end  user,  allowing  them  to	manually  modify  crossbar pin
	   routings, when it opens an audio device.

       show_analog_tv_tuner_dialog
	   If set to true, before capture starts, popup a  display  dialog  to
	   the	end  user,  allowing  them  to manually modify TV channels and
	   frequencies.

       show_analog_tv_tuner_audio_dialog
	   If set to true, before capture starts, popup a  display  dialog  to
	   the	end user, allowing them to manually modify TV audio (like mono
	   vs. stereo, Language A,B or C).

       audio_device_load
	   Load an audio capture filter device from file instead of  searching
	   it  by  name.  It may load additional parameters too, if the filter
	   supports the serialization of its properties to.  To	 use  this  an
	   audio  capture  source  has to be specified, but it can be anything
	   even fake one.

       audio_device_save
	   Save the  currently	used  audio  capture  filter  device  and  its
	   parameters  (if  the filter supports it) to a file.	If a file with
	   the same name exists it will be overwritten.

       video_device_load
	   Load a video capture filter device from file instead	 of  searching
	   it  by  name.  It may load additional parameters too, if the filter
	   supports the serialization of its properties to.   To  use  this  a
	   video  capture  source  has to be specified, but it can be anything
	   even fake one.

       video_device_save
	   Save the  currently	used  video  capture  filter  device  and  its
	   parameters  (if  the filter supports it) to a file.	If a file with
	   the same name exists it will be overwritten.

       use_video_device_timestamps
	   If set to false, the timestamp for video  frames  will  be  derived
	   from the wallclock instead of the timestamp provided by the capture
	   device.  This allows working around devices that provide unreliable
	   timestamps.

       Examples

       •   Print the list of DirectShow supported devices and exit:

		   $ ffmpeg -list_devices true -f dshow -i dummy

       •   Open video device Camera:

		   $ ffmpeg -f dshow -i video="Camera"

       •   Open second video device with name Camera:

		   $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"

       •   Open video device Camera and audio device Microphone:

		   $ ffmpeg -f dshow -i video="Camera":audio="Microphone"

       •   Print the list of supported options in selected device and exit:

		   $ ffmpeg -list_options true -f dshow -i video="Camera"

       •   Specify pin names to capture by name or alternative	name,  specify
	   alternative device name:

		   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

       •   Configure  a	 crossbar device, specifying crossbar pins, allow user
	   to adjust video capture properties at startup:

		   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
			-crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

   fbdev
       Linux framebuffer input device.

       The Linux framebuffer is	 a  graphic  hardware-independent  abstraction
       layer to show graphics on a computer monitor, typically on the console.
       It is accessed through a file device node, usually /dev/fb0.

       For	more	  detailed	information	 read	  the	  file
       Documentation/fb/framebuffer.txt included in the Linux source tree.

       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

       To record from the framebuffer device /dev/fb0 with ffmpeg:

	       ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

       You can take a single screenshot image with the command:

	       ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

       Options

       framerate
	   Set the frame rate. Default is 25.

   gdigrab
       Win32 GDI-based screen capture device.

       This device allows you to capture a region of the display on Windows.

       There are two options for the input filename:

	       desktop

       or

	       title=<window_title>

       The first option will capture the entire desktop, or a fixed region  of
       the  desktop.  The second option will instead capture the contents of a
       single window, regardless of its position on the screen.

       For example, to grab the entire desktop using ffmpeg:

	       ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

       Grab a 640x480 region at position "10,20":

	       ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

       Grab the contents of the window named "Calculator"

	       ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. Use the value 0  to  not
	   draw the pointer. Default value is 1.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

       show_region
	   Show grabbed region on screen.

	   If  show_region  is specified with 1, then the grabbing region will
	   be indicated on screen. With this option, it is easy to  know  what
	   is being grabbed if only a portion of the screen is grabbed.

	   Note that show_region is incompatible with grabbing the contents of
	   a single window.

	   For example:

		   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

       video_size
	   Set the video frame size. The default is to capture the full screen
	   if	desktop	  is   selected,   or	the   full   window   size  if
	   title=window_title is selected.

       offset_x
	   When capturing a region with video_size, set the distance from  the
	   left edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary monitor on Windows. If you have a monitor positioned to the
	   left	 of  your  primary  monitor,  you  will need to use a negative
	   offset_x value to move the region to that monitor.

       offset_y
	   When capturing a region with video_size, set the distance from  the
	   top edge of the screen or desktop.

	   Note that the offset calculation is from the top left corner of the
	   primary  monitor on Windows. If you have a monitor positioned above
	   your primary monitor, you will need	to  use	 a  negative  offset_y
	   value to move the region to that monitor.

   iec61883
       FireWire DV/HDV input device using libiec61883.

       To  enable  this	 input	device,	 you  need libiec61883, libraw1394 and
       libavc1394  installed  on  your	system.	 Use  the   configure	option
       "--enable-libiec61883" to compile with the device enabled.

       The  iec61883  capture  device  supports	 capturing from a video device
       connected via IEEE1394 (FireWire), using libiec61883 and the new	 Linux
       FireWire stack (juju). This is the default DV/HDV input method in Linux
       Kernel 2.6.37 and later, since the old FireWire stack was removed.

       Specify the FireWire port to be used as input file, or "auto" to choose
       the first port connected.

       Options

       dvtype
	   Override  autodetection of DV/HDV. This should only be used if auto
	   detection does not work, or if usage of  a  different  device  type
	   should  be  prohibited. Treating a DV device as HDV (or vice versa)
	   will not work and result in undefined behavior.  The	 values	 auto,
	   dv and hdv are supported.

       dvbuffer
	   Set	maximum	 size  of buffer for incoming data, in frames. For DV,
	   this is an exact value. For HDV, it is not frame exact,  since  HDV
	   does not have a fixed frame size.

       dvguid
	   Select  the	capture	 device by specifying its GUID. Capturing will
	   only be performed from the specified device and fails if no	device
	   with the given GUID is found. This is useful to select the input if
	   multiple   devices  are  connected  at  the	same  time.   Look  at
	   /sys/bus/firewire/devices to find out the GUIDs.

       Examples

       •   Grab and show the input of a FireWire DV/HDV device.

		   ffplay -f iec61883 -i auto

       •   Grab and record the input of a  FireWire  DV/HDV  device,  using  a
	   packet buffer of 100000 packets if the source is HDV.

		   ffmpeg -f iec61883 -i auto -dvbuffer 100000 out.mpg

   jack
       JACK input device.

       To  enable  this	 input	device	during	configuration you need libjack
       installed on your system.

       A JACK input device creates one or more JACK writable clients, one  for
       each audio channel, with name client_name:input_N, where client_name is
       the  name  provided  by	the  application,  and	N  is  a  number which
       identifies the channel.	Each writable client will  send	 the  acquired
       data to the FFmpeg input device.

       Once  you  have	created one or more JACK readable clients, you need to
       connect them to one or more JACK writable clients.

       To connect or disconnect JACK clients you can use the jack_connect  and
       jack_disconnect	programs,  or do it through a graphical interface, for
       example with qjackctl.

       To list the JACK clients	 and  their  properties	 you  can  invoke  the
       command jack_lsp.

       Follows	an  example  which shows how to capture a JACK readable client
       with ffmpeg.

	       # Create a JACK writable client with name "ffmpeg".
	       $ ffmpeg -f jack -i ffmpeg -y out.wav

	       # Start the sample jack_metro readable client.
	       $ jack_metro -b 120 -d 0.2 -f 4000

	       # List the current JACK clients.
	       $ jack_lsp -c
	       system:capture_1
	       system:capture_2
	       system:playback_1
	       system:playback_2
	       ffmpeg:input_1
	       metro:120_bpm

	       # Connect metro to the ffmpeg writable client.
	       $ jack_connect metro:120_bpm ffmpeg:input_1

       For more information read: <http://jackaudio.org/>

       Options

       channels
	   Set the number of channels. Default is 2.

   kmsgrab
       KMS video input device.

       Captures the KMS scanout framebuffer associated with a  specified  CRTC
       or  plane  as  a	 DRM  object  that  can	 be  passed  to other hardware
       functions.

       Requires either DRM master or CAP_SYS_ADMIN to run.

       If you don't understand what all of that means, you probably don't want
       this.  Look at x11grab instead.

       Options

       device
	   DRM device to capture on.  Defaults to /dev/dri/card0.

       format
	   Pixel format of the framebuffer.  This can be autodetected  if  you
	   are	running	 Linux	5.7  or	 later,	 but  needs to be provided for
	   earlier versions.  Defaults to  bgr0,  which	 is  the  most	common
	   format used by the Linux console and Xorg X server.

       format_modifier
	   Format  modifier  to signal on output frames.  This is necessary to
	   import correctly into some APIs.  It can be autodetected if you are
	   running Linux 5.7 or later, but will need to be provided explicitly
	   when needed in earlier versions.  See the libdrm documentation  for
	   possible values.

       crtc_id
	   KMS	CRTC  ID to define the capture source.	The first active plane
	   on the given CRTC will be used.

       plane_id
	   KMS plane ID to define the capture source.  Defaults to  the	 first
	   active plane found if neither crtc_id nor plane_id are specified.

       framerate
	   Framerate  to  capture  at.	 This  is not synchronised to any page
	   flipping or framebuffer changes - it just defines the  interval  at
	   which  the  framebuffer  is	sampled.   Sampling  faster  than  the
	   framebuffer update rate will generate independent frames  with  the
	   same content.  Defaults to 30.

       Examples

       •   Capture  from the first active plane, download the result to normal
	   frames and encode.  This will only work if the framebuffer is  both
	   linear  and	mappable - if not, the result may be scrambled or fail
	   to download.

		   ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4

       •   Capture from CRTC ID 42 at 60fps, map the result to VAAPI,  convert
	   to NV12 and encode as H.264.

		   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

       •   To  capture	only  part of a plane the output can be cropped - this
	   can be used to capture a single window, as long as it has  a	 known
	   absolute position and size.	For example, to capture and encode the
	   middle quarter of a 1920x1080 plane:

		   ffmpeg -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,crop=960:540:480:270,scale_vaapi=960:540:nv12' -c:v h264_vaapi output.mp4

   lavfi
       Libavfilter input virtual device.

       This input device reads data from the open output pads of a libavfilter
       filtergraph.

       For  each  filtergraph  open  output,  the  input  device will create a
       corresponding stream which is mapped  to	 the  generated	 output.   The
       filtergraph is specified through the option graph.

       Options

       graph
	   Specify  the	 filtergraph  to  use as input. Each video open output
	   must be labelled by a unique string of the form "outN", where N  is
	   a  number  starting from 0 corresponding to the mapped input stream
	   generated  by  the  device.	 The  first   unlabelled   output   is
	   automatically assigned to the "out0" label, but all the others need
	   to be specified explicitly.

	   The	suffix	"+subcc" can be appended to the output label to create
	   an extra stream with the closed captions packets attached  to  that
	   output  (experimental;  only	 for  EIA-608 / CEA-708 for now).  The
	   subcc streams are created after all	the  normal  streams,  in  the
	   order  of  the  corresponding  stream.   For	 example,  if there is
	   "out19+subcc", "out7+subcc" and up to "out42", the  stream  #43  is
	   subcc for stream #7 and stream #44 is subcc for stream #19.

	   If  not  specified defaults to the filename specified for the input
	   device.

       graph_file
	   Set the filename of the filtergraph to be  read  and	 sent  to  the
	   other  filters.  Syntax  of	the filtergraph is the same as the one
	   specified by the option graph.

       dumpgraph
	   Dump graph to stderr.

       Examples

       •   Create a color video stream and play it back with ffplay:

		   ffplay -f lavfi -graph "color=c=pink [out0]" dummy

       •   As the previous example, but use filename for specifying the	 graph
	   description, and omit the "out0" label:

		   ffplay -f lavfi color=c=pink

       •   Create three different video test filtered sources and play them:

		   ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

       •   Read	 an  audio stream from a file using the amovie source and play
	   it back with ffplay:

		   ffplay -f lavfi "amovie=test.wav"

       •   Read an audio stream and a video  stream  and  play	it  back  with
	   ffplay:

		   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

       •   Dump	 decoded  frames  to  images  and  closed  captions  to a file
	   (experimental):

		   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

   libcdio
       Audio-CD input device based on libcdio.

       To enable this input  device  during  configuration  you	 need  libcdio
       installed   on	your   system.	 It   requires	the  configure	option
       "--enable-libcdio".

       This device allows playing and grabbing from an Audio-CD.

       For example to copy with ffmpeg the entire Audio-CD  in	/dev/sr0,  you
       may run the command:

	       ffmpeg -f libcdio -i /dev/sr0 cd.wav

       Options

       speed
	   Set drive reading speed. Default value is 0.

	   The speed is specified CD-ROM speed units. The speed is set through
	   the libcdio "cdio_cddap_speed_set" function. On many CD-ROM drives,
	   specifying  a  value	 too  large  will  result in using the fastest
	   speed.

       paranoia_mode
	   Set paranoia recovery mode flags. It accepts one of	the  following
	   values:

	   disable
	   verify
	   overlap
	   neverskip
	   full

	   Default value is disable.

	   For	more  information  about the available recovery modes, consult
	   the paranoia project documentation.

   libdc1394
       IIDC1394 input device, based on libdc1394 and libraw1394.

       Requires the configure option "--enable-libdc1394".

       Options

       framerate
	   Set the frame rate. Default is "ntsc",  corresponding  to  a	 frame
	   rate of "30000/1001".

       pixel_format
	   Select the pixel format. Default is "uyvy422".

       video_size
	   Set	the video size given as a string such as "640x480" or "hd720".
	   Default is "qvga".

   openal
       The OpenAL input device provides audio capture on all  systems  with  a
       working OpenAL 1.1 implementation.

       To  enable  this	 input	device	during	configuration, you need OpenAL
       headers and libraries installed on your system, and need	 to  configure
       FFmpeg with "--enable-openal".

       OpenAL  headers and libraries should be provided as part of your OpenAL
       implementation, or as an additional download  (an  SDK).	 Depending  on
       your  installation  you	may  need  to specify additional flags via the
       "--extra-cflags" and "--extra-ldflags" for allowing the build system to
       locate the OpenAL headers and libraries.

       An incomplete list of OpenAL implementations follows:

       Creative
	   The	 official   Windows   implementation,	 providing    hardware
	   acceleration	 with  supported  devices  and software fallback.  See
	   <http://openal.org/>.

       OpenAL Soft
	   Portable, open  source  (LGPL)  software  implementation.  Includes
	   backends  for  the  most  common  sound APIs on the Windows, Linux,
	   Solaris,	 and	  BSD	   operating	   systems.	   See
	   <http://kcat.strangesoft.net/openal.html>.

       Apple
	   OpenAL  is  part  of	 Core  Audio,  the  official  Mac  OS  X Audio
	   interface.							   See
	   <http://developer.apple.com/technologies/mac/audio-and-video.html>

       This  device  allows  one to capture from an audio input device handled
       through OpenAL.

       You need to specify the name of the device to capture in	 the  provided
       filename.   If	the   empty   string  is  provided,  the  device  will
       automatically select the default device. You can get the	 list  of  the
       supported devices by using the option list_devices.

       Options

       channels
	   Set the number of channels in the captured audio. Only the values 1
	   (monaural) and 2 (stereo) are currently supported.  Defaults to 2.

       sample_size
	   Set	the  sample  size  (in	bits)  of the captured audio. Only the
	   values 8 and 16 are currently supported. Defaults to 16.

       sample_rate
	   Set the sample rate (in Hz) of the  captured	 audio.	  Defaults  to
	   44.1k.

       list_devices
	   If  set  to	true,  print  a list of devices and exit.  Defaults to
	   false.

       Examples

       Print the list of OpenAL supported devices and exit:

	       $ ffmpeg -list_devices true -f openal -i dummy out.ogg

       Capture from the OpenAL device DR-BT101 via PulseAudio:

	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

       Capture from the default device (note the empty string '' as filename):

	       $ ffmpeg -f openal -i '' out.ogg

       Capture from two	 devices  simultaneously,  writing  to	two  different
       files, within the same ffmpeg command:

	       $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

       Note:  not  all	OpenAL	implementations	 support multiple simultaneous
       capture - try the latest OpenAL Soft if the above does not work.

   oss
       Open Sound System input device.

       The filename to	provide	 to  the  input	 device	 is  the  device  node
       representing the OSS input device, and is usually set to /dev/dsp.

       For example to grab from /dev/dsp using ffmpeg use the command:

	       ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

       For	   more	       information	  about	       OSS	  see:
       <http://manuals.opensound.com/usersguide/dsp.html>

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   pulse
       PulseAudio input device.

       To enable  this	output	device	you  need  to  configure  FFmpeg  with
       "--enable-libpulse".

       The  filename  to provide to the input device is a source device or the
       string "default"

       To list the PulseAudio source devices  and  their  properties  you  can
       invoke the command pactl list sources.

       More	information    about	PulseAudio    can    be	   found    on
       <http://www.pulseaudio.org>.

       Options

       server
	   Connect to  a  specific  PulseAudio	server,	 specified  by	an  IP
	   address.  Default server is used when not provided.

       name
	   Specify  the	 application  name  PulseAudio	will  use when showing
	   active clients, by default it is the "LIBAVFORMAT_IDENT" string.

       stream_name
	   Specify the stream name PulseAudio will  use	 when  showing	active
	   streams, by default it is "record".

       sample_rate
	   Specify the samplerate in Hz, by default 48kHz is used.

       channels
	   Specify the channels in use, by default 2 (stereo) is set.

       frame_size
	   This option does nothing and is deprecated.

       fragment_size
	   Specify  the	 size  in  bytes  of the minimal buffering fragment in
	   PulseAudio, it will affect the audio latency. By default it is  set
	   to 50 ms amount of data.

       wallclock
	   Set the initial PTS using the current time. Default is 1.

       Examples

       Record a stream from default device:

	       ffmpeg -f pulse -i default /tmp/pulse.wav

   sndio
       sndio input device.

       To  enable  this	 input	device	during configuration you need libsndio
       installed on your system.

       The filename to	provide	 to  the  input	 device	 is  the  device  node
       representing the sndio input device, and is usually set to /dev/audio0.

       For example to grab from /dev/audio0 using ffmpeg use the command:

	       ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

       Options

       sample_rate
	   Set the sample rate in Hz. Default is 48000.

       channels
	   Set the number of channels. Default is 2.

   video4linux2, v4l2
       Video4Linux2 input video device.

       "v4l2" can be used as alias for "video4linux2".

       If   FFmpeg   is	  built	  with	 v4l-utils   support   (by  using  the
       "--enable-libv4l2" configure option), it is possible to use it with the
       "-use_libv4l2" input device option.

       The name of the device to grab is a file	 device	 node,	usually	 Linux
       systems	tend  to automatically create such nodes when the device (e.g.
       an USB webcam) is plugged into the system, and has a name of  the  kind
       /dev/videoN, where N is a number associated to the device.

       Video4Linux2  devices  usually  support	a  limited set of widthxheight
       sizes and  frame	 rates.	 You  can  check  which	 are  supported	 using
       -list_formats  all  for	Video4Linux2  devices.	 Some devices, like TV
       cards, support one or more standards. It is possible to	list  all  the
       supported standards using -list_standards all.

       The  time  base	for  the timestamps is 1 microsecond. Depending on the
       kernel version and configuration, the timestamps may  be	 derived  from
       the  real  time clock (origin at the Unix Epoch) or the monotonic clock
       (origin usually at boot time, unaffected by NTP or  manual  changes  to
       the  clock). The -timestamps abs or -ts abs option can be used to force
       conversion into the real time clock.

       Some usage examples of the video4linux2 device with ffmpeg and ffplay:

       •   List supported formats for a video4linux2 device:

		   ffplay -f video4linux2 -list_formats all /dev/video0

       •   Grab and show the input of a video4linux2 device:

		   ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

       •   Grab and record the input of a video4linux2 device, leave the frame
	   rate and size as previously set:

		   ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

       For more information about Video4Linux, check <http://linuxtv.org/>.

       Options

       standard
	   Set the standard. Must be the name of a supported standard. To  get
	   a list of the supported standards, use the list_standards option.

       channel
	   Set	the input channel number. Default to -1, which means using the
	   previously selected channel.

       video_size
	   Set the video frame size. The argument must be a string in the form
	   WIDTHxHEIGHT or a valid size abbreviation.

       pixel_format
	   Select the pixel format (only valid for raw video input).

       input_format
	   Set the preferred pixel format (for raw video)  or  a  codec	 name.
	   This option allows one to select the input format, when several are
	   available.

       framerate
	   Set the preferred video frame rate.

       list_formats
	   List	 available formats (supported pixel formats, codecs, and frame
	   sizes) and exit.

	   Available values are:

	   all Show all available (compressed and non-compressed) formats.

	   raw Show only raw video (non-compressed) formats.

	   compressed
	       Show only compressed formats.

       list_standards
	   List supported standards and exit.

	   Available values are:

	   all Show all supported standards.

       timestamps, ts
	   Set type of timestamps for grabbed frames.

	   Available values are:

	   default
	       Use timestamps from the kernel.

	   abs Use absolute timestamps (wall clock).

	   mono2abs
	       Force conversion from monotonic to absolute timestamps.

	   Default value is "default".

       use_libv4l2
	   Use libv4l2 (v4l-utils) conversion functions. Default is 0.

   vfwcap
       VfW (Video for Windows) capture input device.

       The filename passed as input is the capture driver number, ranging from
       0 to 9. You may use "list" as filename to print a list of drivers.  Any
       other filename will be interpreted as device number 0.

       Options

       video_size
	   Set the video frame size.

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

   x11grab
       X11 video input device.

       To  enable  this	 input	device	during	configuration  you need libxcb
       installed on your system. It  will  be  automatically  detected	during
       configuration.

       This device allows one to capture a region of an X11 display.

       The filename passed as input has the syntax:

	       [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

       hostname:display_number.screen_number specifies the X11 display name of
       the  screen  to	grab  from.  hostname  can be omitted, and defaults to
       "localhost". The environment  variable  DISPLAY	contains  the  default
       display name.

       x_offset	 and  y_offset	specify	 the  offsets of the grabbed area with
       respect to the top-left border of the X11 screen. They default to 0.

       Check the X11 documentation (e.g. man X) for more detailed information.

       Use the xdpyinfo	 program  for  getting	basic  information  about  the
       properties of your X11 display (e.g. grep for "name" or "dimensions").

       For example to grab from :0.0 using ffmpeg:

	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg

       Grab at position "10,20":

	       ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

       Options

       select_region
	   Specify  whether  to select the grabbing area graphically using the
	   pointer.  A value of 1 prompts the user to select the grabbing area
	   graphically by clicking  and	 dragging.  A  single  click  with  no
	   dragging  will select the whole screen. A region with zero width or
	   height will also select the whole screen.  This  option  overwrites
	   the video_size, grab_x, and grab_y options. Default value is 0.

       draw_mouse
	   Specify  whether  to draw the mouse pointer. A value of 0 specifies
	   not to draw the pointer. Default value is 1.

       follow_mouse
	   Make the grabbed  area  follow  the	mouse.	The  argument  can  be
	   "centered" or a number of pixels PIXELS.

	   When	 it  is specified with "centered", the grabbing region follows
	   the mouse pointer and keeps the pointer at the  center  of  region;
	   otherwise,  the  region follows only when the mouse pointer reaches
	   within PIXELS (greater than zero) to the edge of region.

	   For example:

		   ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

	   To follow only when the mouse pointer reaches within 100 pixels  to
	   edge:

		   ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

       framerate
	   Set the grabbing frame rate. Default value is "ntsc", corresponding
	   to a frame rate of "30000/1001".

       show_region
	   Show grabbed region on screen.

	   If  show_region  is specified with 1, then the grabbing region will
	   be indicated on screen. With this option, it is easy to  know  what
	   is being grabbed if only a portion of the screen is grabbed.

       region_border
	   Set	the  region border thickness if -show_region 1 is used.	 Range
	   is 1 to 128 and default is 3 (XCB-based x11grab only).

	   For example:

		   ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

	   With follow_mouse:

		   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

       window_id
	   Grab this window, instead of the whole screen. Default value is  0,
	   which maps to the whole screen (root window).

	   The	id  of	a  window  can	be  found  using the xwininfo program,
	   possibly with options -tree and -root.

	   If the window is later enlarged, the	 new  area  is	not  recorded.
	   Video ends when the window is closed, unmapped (i.e., iconified) or
	   shrunk  beyond the video size (which defaults to the initial window
	   size).

	   This option disables options follow_mouse and select_region.

       video_size
	   Set the video frame size. Default is the full desktop or window.

       grab_x
       grab_y
	   Set the grabbing region coordinates. They are expressed  as	offset
	   from	 the  top  left corner of the X11 window and correspond to the
	   x_offset and y_offset parameters in the device  name.  The  default
	   value for both options is 0.

OUTPUT DEVICES
       Output  devices	are  configured	 elements  in  FFmpeg  that  can write
       multimedia data to an output device attached to your system.

       When you configure your FFmpeg build, all the supported output  devices
       are  enabled  by	 default.  You	can  list all available ones using the
       configure option "--list-outdevs".

       You can disable all the	output	devices	 using	the  configure	option
       "--disable-outdevs",  and selectively enable an output device using the
       option "--enable-outdev=OUTDEV", or you can disable a particular	 input
       device using the option "--disable-outdev=OUTDEV".

       The option "-devices" of the ff* tools will display the list of enabled
       output devices.

       A description of the currently available output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) output device.

       Examples

       •   Play a file on default ALSA device:

		   ffmpeg -i INPUT -f alsa default

       •   Play a file on soundcard 1, audio device 7:

		   ffmpeg -i INPUT -f alsa hw:1,7

   AudioToolbox
       AudioToolbox output device.

       Allows native output to CoreAudio devices on OSX.

       The  output  filename  can  be  empty  (or "-") to refer to the default
       system output device or a number that refers to	the  device  index  as
       shown using: "-list_devices true".

       Alternatively, the audio input device can be chosen by index using the

	   B<-audio_device_index E<lt>INDEXE<gt>>

       , overriding any device name or index given in the input filename.

       All  available  devices	can be enumerated by using -list_devices true,
       listing all device names, UIDs and corresponding indices.

       Options

       AudioToolbox supports the following options:

       -audio_device_index <INDEX>
	   Specify the audio device by its index. Overrides anything given  in
	   the output filename.

       Examples

       •   Print  the  list of supported devices and output a sine wave to the
	   default device:

		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -list_devices true -

       •   Output a sine wave to the device with the index 2,  overriding  any
	   output filename:

		   $ ffmpeg -f lavfi -i sine=r=44100 -f audiotoolbox -audio_device_index 2 -

   caca
       CACA output device.

       This  output  device  allows one to show a video stream in CACA window.
       Only one CACA window is allowed per application, so you can  have  only
       one instance of this output device in an application.

       To  enable  this	 output	 device	 you  need  to	configure  FFmpeg with
       "--enable-libcaca".  libcaca is a graphics library  that	 outputs  text
       instead of pixels.

       For	 more	    information	      about	  libcaca,	check:
       <http://caca.zoy.org/wiki/libcaca>

       Options

       window_title
	   Set the CACA window title, if not specified default to the filename
	   specified for the output device.

       window_size
	   Set the CACA window size, can be a string of the form  widthxheight
	   or  a video size abbreviation.  If not specified it defaults to the
	   size of the input video.

       driver
	   Set display driver.

       algorithm
	   Set dithering algorithm. Dithering is necessary because the picture
	   being rendered has usually far  more	 colours  than	the  available
	   palette.    The  accepted  values  are  listed  with	 "-list_dither
	   algorithms".

       antialias
	   Set antialias method. Antialiasing smoothens the rendered image and
	   avoids the commonly seen staircase effect.  The accepted values are
	   listed with "-list_dither antialiases".

       charset
	   Set which characters are going to be used when rendering text.  The
	   accepted values are listed with "-list_dither charsets".

       color
	   Set color to be used when rendering text.  The accepted values  are
	   listed with "-list_dither colors".

       list_drivers
	   If set to true, print a list of available drivers and exit.

       list_dither
	   List	 available  dither  options  related  to  the  argument.   The
	   argument must be one of  "algorithms",  "antialiases",  "charsets",
	   "colors".

       Examples

       •   The	following  command  shows the ffmpeg output is an CACA window,
	   forcing its size to 80x25:

		   ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

       •   Show the list of available drivers and exit:

		   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -

       •   Show the list of available dither colors and exit:

		   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -

   decklink
       The  decklink  output  device  provides	 playback   capabilities   for
       Blackmagic DeckLink devices.

       To  enable this output device, you need the Blackmagic DeckLink SDK and
       you  need  to  configure	 with  the  appropriate	 "--extra-cflags"  and
       "--extra-ldflags".   On	Windows, you need to run the IDL files through
       widl.

       DeckLink is very picky about the formats it supports. Pixel  format  is
       always	uyvy422,  framerate,  field  order  and	 video	size  must  be
       determined for your device with -list_formats 1. Audio sample  rate  is
       always 48 kHz.

       Options

       list_devices
	   If  set  to	true,  print  a list of devices and exit.  Defaults to
	   false. This option is deprecated, please use the "-sinks" option of
	   ffmpeg to list the available output devices.

       list_formats
	   If set to true,  print  a  list  of	supported  formats  and	 exit.
	   Defaults to false.

       preroll
	   Amount of time to preroll video in seconds.	Defaults to 0.5.

       duplex_mode
	   Sets	 the decklink device duplex/profile mode. Must be unset, half,
	   full,	   one_sub_device_full,		  one_sub_device_half,
	   two_sub_device_full, four_sub_device_half Defaults to unset.

	   Note:  DeckLink  SDK	 11.0  have  replaced the duplex property by a
	   profile property.  For the DeckLink Duo 2 and DeckLink  Quad	 2,  a
	   profile  is	shared between any 2 sub-devices that utilize the same
	   connectors. For the DeckLink 8K Pro, a profile  is  shared  between
	   all 4 sub-devices. So DeckLink 8K Pro support four profiles.

	   Valid profile modes for DeckLink 8K Pro(with DeckLink SDK >= 11.0):
	   one_sub_device_full,	   one_sub_device_half,	  two_sub_device_full,
	   four_sub_device_half

	   Valid profile modes for DeckLink Quad 2 and DeckLink Duo  2:	 half,
	   full

       timing_offset
	   Sets	 the genlock timing pixel offset on the used output.  Defaults
	   to unset.

       link
	   Sets the SDI video link configuration on the used output.  Must  be
	   unset,  single  link SDI, dual link SDI or quad link SDI.  Defaults
	   to unset.

       sqd Enable Square Division Quad Split mode for  Quad-link  SDI  output.
	   Must be unset, true or false.  Defaults to unset.

       level_a
	   Enable  SMPTE Level A mode on the used output.  Must be unset, true
	   or false.  Defaults to unset.

       vanc_queue_size
	   Sets maximum output buffer size in bytes  for  VANC	data.  If  the
	   buffering  reaches  this value, outgoing VANC data will be dropped.
	   Defaults to 1048576.

       Examples

       •   List output devices:

		   ffmpeg -sinks decklink

       •   List supported formats:

		   ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'

       •   Play video clip:

		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'

       •   Play video clip with non-standard framerate or video size:

		   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

   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_borderless
	   Set SDL window border off.	Default	 value	is  0  (enable	window
	   border).

       window_enable_quit
	   Enable  quit action (using window button or keyboard key) when non-
	   zero value is provided.  Default value is 1 (enable quit action).

       window_fullscreen
	   Set fullscreen mode when non-zero value is provided.	 Default value
	   is zero.

       window_size
	   Set the SDL window size, can be a string of the  form  widthxheight
	   or  a video size abbreviation.  If not specified it defaults to the
	   size of the input video, downscaled according to the aspect ratio.

       window_title
	   Set the SDL window title, if not specified default to the  filename
	   specified for the output device.

       window_x
       window_y
	   Set the position of the window on the screen.

       Interactive commands

       The  window  created  by	 the  device  can  be  controlled  through the
       following interactive commands.

       q, ESC
	   Quit the device immediately.

       Examples

       The following command shows the ffmpeg output is an SDL window, forcing
       its size to the qcif format:

	       ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

   sndio
       sndio audio output device.

   v4l2
       Video4Linux2 output device.

   xv
       XV (XVideo) output device.

       This output device allows one to show a video  stream  in  a  X	Window
       System window.

       Options

       display_name
	   Specify the hardware display name, which determines the display and
	   communications domain to be used.

	   The display name or DISPLAY environment variable can be a string in
	   the format hostname[:number[.screen_number]].

	   hostname  specifies	the  name  of  the  host  machine on which the
	   display is physically attached. number specifies the number of  the
	   display  server  on	that host machine. screen_number specifies the
	   screen to be used on that server.

	   If unspecified, it defaults to the value of the DISPLAY environment
	   variable.

	   For example, "dual-headed:0.1" would specify screen 1 of display  0
	   on the machine named ``dual-headed''.

	   Check the X11 specification for more detailed information about the
	   display name format.

       window_id
	   When	 set  to non-zero value then device doesn't create new window,
	   but uses existing one with  provided	 window_id.  By	 default  this
	   options is set to zero and device creates its own window.

       window_size
	   Set	the  created  window  size,  can  be  a	 string	 of  the  form
	   widthxheight or a video size	 abbreviation.	If  not	 specified  it
	   defaults to the size of the input video.  Ignored when window_id is
	   set.

       window_x
       window_y
	   Set	the  X	and  Y window offsets for the created window. They are
	   both set to 0 by default. The values may be ignored by  the	window
	   manager.  Ignored when window_id is set.

       window_title
	   Set	the  window  title,  if	 not specified default to the filename
	   specified for the output device. Ignored when window_id is set.

       For more information about XVideo see <http://www.x.org/>.

       Examples

       •   Decode, display and encode video input  with	 ffmpeg	 at  the  same
	   time:

		   ffmpeg -i INPUT OUTPUT -f xv display

       •   Decode and display the input video to multiple X11 windows:

		   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
       The audio resampler supports the following named options.

       Options	may  be	 set  by specifying -option value in the FFmpeg tools,
       option=value for the aresample filter, by setting the value  explicitly
       in  the	"SwrContext"  options  or  using  the  libavutil/opt.h API for
       programmatic use.

       uchl, used_chlayout
	   Set used input channel layout. Default is  unset.  This  option  is
	   only used for special remapping.

       isr, in_sample_rate
	   Set the input sample rate. Default value is 0.

       osr, out_sample_rate
	   Set the output sample rate. Default value is 0.

       isf, in_sample_fmt
	   Specify the input sample format. It is set by default to "none".

       osf, out_sample_fmt
	   Specify the output sample format. It is set by default to "none".

       tsf, internal_sample_fmt
	   Set the internal sample format. Default value is "none".  This will
	   automatically be chosen when it is not explicitly set.

       ichl, in_chlayout
       ochl, out_chlayout
	   Set the input/output channel layout.

	   See	the  Channel  Layout section in the ffmpeg-utils(1) manual for
	   the required syntax.

       clev, center_mix_level
	   Set the center mix level. It is a value expressed in	 deciBel,  and
	   must be in the interval [-32,32].

       slev, surround_mix_level
	   Set the surround mix level. It is a value expressed in deciBel, and
	   must be in the interval [-32,32].

       lfe_mix_level
	   Set	LFE  mix  into	non  LFE level. It is used when there is a LFE
	   input but no LFE output. It is a value expressed  in	 deciBel,  and
	   must be in the interval [-32,32].

       rmvol, rematrix_volume
	   Set rematrix volume. Default value is 1.0.

       rematrix_maxval
	   Set	maximum	 output	 value	for  rematrixing.  This can be used to
	   prevent clipping vs. preventing volume reduction.  A value  of  1.0
	   prevents clipping.

       flags, swr_flags
	   Set flags used by the converter. Default value is 0.

	   It supports the following individual flags:

	   res force  resampling,  this flag forces resampling to be used even
	       when the input and output sample rates match.

       dither_scale
	   Set the dither scale. Default value is 1.

       dither_method
	   Set dither method. Default value is 0.

	   Supported values:

	   rectangular
	       select rectangular dither

	   triangular
	       select triangular dither

	   triangular_hp
	       select triangular dither with high pass

	   lipshitz
	       select Lipshitz noise shaping dither.

	   shibata
	       select Shibata noise shaping dither.

	   low_shibata
	       select low Shibata noise shaping dither.

	   high_shibata
	       select high Shibata noise shaping dither.

	   f_weighted
	       select f-weighted noise shaping dither

	   modified_e_weighted
	       select modified-e-weighted noise shaping dither

	   improved_e_weighted
	       select improved-e-weighted noise shaping dither

       resampler
	   Set resampling engine. Default value is swr.

	   Supported values:

	   swr select the native SW Resampler; filter  options	precision  and
	       cheby are not applicable in this case.

	   soxr
	       select  the  SoX Resampler (where available); compensation, and
	       filter  options	 filter_size,	phase_shift,   exact_rational,
	       filter_type & kaiser_beta, are not applicable in this case.

       filter_size
	   For swr only, set resampling filter size, default value is 32.

       phase_shift
	   For	swr only, set resampling phase shift, default value is 10, and
	   must be in the interval [0,30].

       linear_interp
	   Use linear interpolation when enabled (the default). Disable it  if
	   you	want  to preserve speed instead of quality when exact_rational
	   fails.

       exact_rational
	   For swr only, when enabled, try to use exact phase_count  based  on
	   input  and  output sample rate. However, if it is larger than "1 <<
	   phase_shift",  the  phase_count  will  be  "1  <<  phase_shift"  as
	   fallback. Default is enabled.

       cutoff
	   Set	cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must
	   be a float value between 0 and 1.  Default value is 0.97 with  swr,
	   and	0.91  with soxr (which, with a sample-rate of 44100, preserves
	   the entire audio band to 20kHz).

       precision
	   For soxr only, the precision in bits to which the resampled	signal
	   will	 be calculated.	 The default value of 20 (which, with suitable
	   dithering, is appropriate for a destination bit-depth of 16)	 gives
	   SoX's  'High	 Quality';  a  value  of  28  gives  SoX's  'Very High
	   Quality'.

       cheby
	   For soxr only, selects passband rolloff none (Chebyshev) &  higher-
	   precision  approximation  for 'irrational' ratios. Default value is
	   0.

       async
	   For swr only, simple 1 parameter audio  sync	 to  timestamps	 using
	   stretching, squeezing, filling and trimming. Setting this to 1 will
	   enable  filling  and	 trimming, larger values represent the maximum
	   amount in samples that the data may be stretched  or	 squeezed  for
	   each	 second.   Default value is 0, thus no compensation is applied
	   to make the samples match the audio timestamps.

       first_pts
	   For swr only, assume the first pts should be this value.  The  time
	   unit	 is  1 / sample rate.  This allows for padding/trimming at the
	   start of stream. By default, no assumption is made about the	 first
	   frame's  expected  pts,  so	no  padding  or	 trimming is done. For
	   example, this could be set to 0 to pad the beginning	 with  silence
	   if  an  audio  stream  starts after the video stream or to trim any
	   samples with a negative pts due to encoder delay.

       min_comp
	   For swr only, set the minimum  difference  between  timestamps  and
	   audio  data (in seconds) to trigger stretching/squeezing/filling or
	   trimming of the data to make it match the timestamps.  The  default
	   is  that  stretching/squeezing/filling  and	trimming  is  disabled
	   (min_comp = "FLT_MAX").

       min_hard_comp
	   For swr only, set the minimum  difference  between  timestamps  and
	   audio  data (in seconds) to trigger adding/dropping samples to make
	   it match the timestamps.  This option effectively is a threshold to
	   select  between  hard  (trim/fill)	and   soft   (squeeze/stretch)
	   compensation.  Note	that  all  compensation is by default disabled
	   through min_comp.  The default is 0.1.

       comp_duration
	   For swr  only,  set	duration  (in  seconds)	 over  which  data  is
	   stretched/squeezed  to make it match the timestamps. Must be a non-
	   negative double float value, default value is 1.0.

       max_soft_comp
	   For	 swr   only,   set   maximum   factor	by   which   data   is
	   stretched/squeezed  to make it match the timestamps. Must be a non-
	   negative double float value, default value is 0.

       matrix_encoding
	   Select matrixed stereo encoding.

	   It accepts the following values:

	   none
	       select none

	   dolby
	       select Dolby

	   dplii
	       select Dolby Pro Logic II

	   Default value is "none".

       filter_type
	   For swr only, select resampling  filter  type.  This	 only  affects
	   resampling operations.

	   It accepts the following values:

	   cubic
	       select cubic

	   blackman_nuttall
	       select Blackman Nuttall windowed sinc

	   kaiser
	       select Kaiser windowed sinc

       kaiser_beta
	   For	swr only, set Kaiser window beta value. Must be a double float
	   value in the interval [2,16], default value is 9.

       output_sample_bits
	   For swr only, set number of used output sample bits for  dithering.
	   Must	 be  an	 integer  in  the interval [0,64], default value is 0,
	   which means it's not used.

SCALER OPTIONS
       The video scaler supports the following named options.

       Options may be set by specifying -option value  in  the	FFmpeg	tools,
       with a few API-only exceptions noted below.  For programmatic use, they
       can  be	set  explicitly	 in  the  "SwsContext"	options or through the
       libavutil/opt.h API.

       sws_flags
	   Set the scaler  flags.  This	 is  also  used	 to  set  the  scaling
	   algorithm.  Only  a	single	algorithm  should be selected. Default
	   value is bicubic.

	   It accepts the following values:

	   fast_bilinear
	       Select fast bilinear scaling algorithm.

	   bilinear
	       Select bilinear scaling algorithm.

	   bicubic
	       Select bicubic scaling algorithm.

	   experimental
	       Select experimental scaling algorithm.

	   neighbor
	       Select nearest neighbor rescaling algorithm.

	   area
	       Select averaging area rescaling algorithm.

	   bicublin
	       Select  bicubic	scaling	 algorithm  for	 the  luma  component,
	       bilinear for chroma components.

	   gauss
	       Select Gaussian rescaling algorithm.

	   sinc
	       Select sinc rescaling algorithm.

	   lanczos
	       Select  Lanczos	rescaling algorithm. The default width (alpha)
	       is 3 and can be changed by setting "param0".

	   spline
	       Select natural bicubic spline rescaling algorithm.

	   print_info
	       Enable printing/debug logging.

	   accurate_rnd
	       Enable accurate rounding.

	   full_chroma_int
	       Enable full chroma interpolation.

	   full_chroma_inp
	       Select full chroma input.

	   bitexact
	       Enable bitexact output.

       srcw (API only)
	   Set source width.

       srch (API only)
	   Set source height.

       dstw (API only)
	   Set destination width.

       dsth (API only)
	   Set destination height.

       src_format (API only)
	   Set source pixel format (must be expressed as an integer).

       dst_format (API only)
	   Set destination pixel format (must be expressed as an integer).

       src_range (boolean)
	   If value is set to 1, indicates source is full range. Default value
	   is 0, which indicates source is limited range.

       dst_range (boolean)
	   If value is set to 1, enable full range  for	 destination.  Default
	   value is 0, which enables limited range.

       param0, param1
	   Set scaling algorithm parameters. The specified values are specific
	   of  some  scaling  algorithms  and ignored by others. The specified
	   values are floating point number values.

       sws_dither
	   Set the dithering algorithm. Accepts one of the  following  values.
	   Default value is auto.

	   auto
	       automatic choice

	   none
	       no dithering

	   bayer
	       bayer dither

	   ed  error diffusion dither

	   a_dither
	       arithmetic dither, based using addition

	   x_dither
	       arithmetic  dither,  based using xor (more random/less apparent
	       patterning that a_dither).

       alphablend
	   Set the alpha blending to use when the  input  has  alpha  but  the
	   output does not.  Default value is none.

	   uniform_color
	       Blend onto a uniform background color

	   checkerboard
	       Blend onto a checkerboard

	   none
	       No blending

FILTERING INTRODUCTION
       Filtering in FFmpeg is enabled through the libavfilter library.

       In libavfilter, a filter can have multiple inputs and multiple outputs.
       To  illustrate  the  sorts of things that are possible, we consider the
       following filtergraph.

			       [main]
	       input --> split ---------------------> overlay --> output
			   |				 ^
			   |[tmp]		   [flip]|
			   +-----> crop --> vflip -------+

       This filtergraph splits the input stream in two streams, then sends one
       stream through the crop filter and the vflip filter, before merging  it
       back  with  the	other  stream by overlaying it on top. You can use the
       following command to achieve this:

	       ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

       The result will be that the top half of the video is mirrored onto  the
       bottom half of the output video.

       Filters	in the same linear chain are separated by commas, and distinct
       linear chains of filters are separated by semicolons. In	 our  example,
       crop,vflip are in one linear chain, split and overlay are separately in
       another.	 The points where the linear chains join are labelled by names
       enclosed in square brackets. In the example, the split filter generates
       two outputs that are associated to the labels [main] and [tmp].

       The stream sent to the second output of split, labelled	as  [tmp],  is
       processed through the crop filter, which crops away the lower half part
       of  the video, and then vertically flipped. The overlay filter takes in
       input the first	unchanged  output  of  the  split  filter  (which  was
       labelled as [main]), and overlay on its lower half the output generated
       by the crop,vflip filterchain.

       Some  filters  take  in	input a list of parameters: they are specified
       after the filter name and an equal sign, and are	 separated  from  each
       other by a colon.

       There  exist  so-called	source filters that do not have an audio/video
       input, and sink filters that will not have audio/video output.

GRAPH
       The graph2dot program included in the FFmpeg  tools  directory  can  be
       used  to	 parse	a  filtergraph	description  and issue a corresponding
       textual representation in the dot language.

       Invoke the command:

	       graph2dot -h

       to see how to use graph2dot.

       You can then pass the dot description to	 the  dot  program  (from  the
       graphviz	 suite	of  programs) and obtain a graphical representation of
       the filtergraph.

       For example the sequence of commands:

	       echo <GRAPH_DESCRIPTION> | \
	       tools/graph2dot -o graph.tmp && \
	       dot -Tpng graph.tmp -o graph.png && \
	       display graph.png

       can be used to create and  display  an  image  representing  the	 graph
       described  by  the GRAPH_DESCRIPTION string. Note that this string must
       be a  complete  self-contained  graph,  with  its  inputs  and  outputs
       explicitly defined.  For example if your command line is of the form:

	       ffmpeg -i infile -vf scale=640:360 outfile

       your GRAPH_DESCRIPTION string will need to be of the form:

	       nullsrc,scale=640:360,nullsink

       you may also need to set the nullsrc parameters and add a format filter
       in order to simulate a specific input file.

FILTERGRAPH DESCRIPTION
       A  filtergraph is a directed graph of connected filters. It can contain
       cycles, and there can be multiple links between a pair of filters. Each
       link has one input pad on one side connecting it	 to  one  filter  from
       which  it  takes	 its  input,  and  one	output	pad  on the other side
       connecting it to one filter accepting its output.

       Each filter  in	a  filtergraph	is  an	instance  of  a	 filter	 class
       registered  in  the  application,  which	 defines  the features and the
       number of input and output pads of the filter.

       A filter with no input pads is called a "source", and a filter with  no
       output pads is called a "sink".

   Filtergraph syntax
       A  filtergraph has a textual representation, which is recognized by the
       -filter/-vf/-af and -filter_complex options in ffmpeg  and  -vf/-af  in
       ffplay,	and  by	 the  avfilter_graph_parse_ptr()  function  defined in
       libavfilter/avfilter.h.

       A filterchain consists of a sequence of	connected  filters,  each  one
       connected  to  the  previous  one  in  the  sequence.  A filterchain is
       represented by a list of ","-separated filter descriptions.

       A filtergraph consists of a sequence of	filterchains.  A  sequence  of
       filterchains  is	 represented  by  a  list of ";"-separated filterchain
       descriptions.

       A   filter   is	 represented   by    a	  string    of	  the	 form:
       [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]

       filter_name  is	the  name  of  the filter class of which the described
       filter is an instance of, and has to be the name of one of  the	filter
       classes	registered  in	the program optionally followed by "@id".  The
       name  of	 the  filter  class  is	 optionally  followed  by   a	string
       "=arguments".

       arguments  is a string which contains the parameters used to initialize
       the filter instance. It may have one of two forms:

       •   A ':'-separated list of key=value pairs.

       •   A ':'-separated list of value. In this case, the keys  are  assumed
	   to  be  the	option	names in the order they are declared. E.g. the
	   "fade" filter  declares  three  options  in	this  order  --	 type,
	   start_frame	and  nb_frames.	 Then the parameter list in:0:30 means
	   that the value in is assigned to the option type, 0 to  start_frame
	   and 30 to nb_frames.

       •   A  ':'-separated  list  of  mixed  direct  value and long key=value
	   pairs. The direct value  must  precede  the	key=value  pairs,  and
	   follow  the	same  constraints  order  of  the  previous point. The
	   following key=value pairs can be set in any preferred order.

       If the option value itself is a list of items (e.g. the "format" filter
       takes a list of pixel formats), the  items  in  the  list  are  usually
       separated by |.

       The  list  of  arguments can be quoted using the character ' as initial
       and ending mark, and the character \ for escaping the characters within
       the quoted text; otherwise the argument string is considered terminated
       when the next  special  character  (belonging  to  the  set  []=;,)  is
       encountered.

       A  special  syntax  implemented	in  the ffmpeg CLI tool allows loading
       option values from files. This is done be prepending a slash '/' to the
       option name, then the supplied value is	interpreted  as	 a  path  from
       which the actual value is loaded. E.g.

	       ffmpeg -i <INPUT> -vf drawtext=/text=/tmp/some_text <OUTPUT>

       will  load  the text to be drawn from /tmp/some_text. API users wishing
       to    implement	  a	similar	    feature	should	   use	   the
       "avfilter_graph_segment_*()" functions together with custom IO code.

       The  name  and  arguments  of  the  filter  are optionally preceded and
       followed by a list of link labels.  A link label allows one to  name  a
       link  and  associate  it to a filter output or input pad. The preceding
       labels in_link_1 ... in_link_N, are  associated	to  the	 filter	 input
       pads, the following labels out_link_1 ... out_link_M, are associated to
       the output pads.

       When two link labels with the same name are found in the filtergraph, a
       link between the corresponding input and output pad is created.

       If  an output pad is not labelled, it is linked by default to the first
       unlabelled input pad of	the  next  filter  in  the  filterchain.   For
       example in the filterchain

	       nullsrc, split[L1], [L2]overlay, nullsink

       the  split  filter instance has two output pads, and the overlay filter
       instance two input pads. The first output  pad  of  split  is  labelled
       "L1",  the  first input pad of overlay is labelled "L2", and the second
       output pad of split is linked to the second input pad of overlay, which
       are both unlabelled.

       In a filter description, if the input label of the first filter is  not
       specified,  "in"	 is assumed; if the output label of the last filter is
       not specified, "out" is assumed.

       In a complete filterchain all the unlabelled filter  input  and	output
       pads  must  be  connected. A filtergraph is considered valid if all the
       filter input and output pads of all the filterchains are connected.

       Leading	and  trailing  whitespaces  (space,  tabs,  or	 line	feeds)
       separating  tokens  in  the filtergraph specification are ignored. This
       means that the filtergraph can  be  expressed  using  empty  lines  and
       spaces to improve redability.

       For example, the filtergraph:

	       testsrc,split[L1],hflip[L2];[L1][L2] hstack

       can be represented as:

	       testsrc,
	       split [L1], hflip [L2];

	       [L1][L2] hstack

       Libavfilter  will  automatically	 insert	 scale	filters	 where	format
       conversion is required. It is possible to  specify  swscale  flags  for
       those  automatically  inserted scalers by prepending "sws_flags=flags;"
       to the filtergraph description.

       Here is a BNF description of the filtergraph syntax:

	       <NAME>		  ::= sequence of alphanumeric characters and '_'
	       <FILTER_NAME>	  ::= <NAME>["@"<NAME>]
	       <LINKLABEL>	  ::= "[" <NAME> "]"
	       <LINKLABELS>	  ::= <LINKLABEL> [<LINKLABELS>]
	       <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
	       <FILTER>		  ::= [<LINKLABELS>] <FILTER_NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
	       <FILTERCHAIN>	  ::= <FILTER> [,<FILTERCHAIN>]
	       <FILTERGRAPH>	  ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

   Notes on filtergraph escaping
       Filtergraph description composition entails several levels of escaping.
       See the "Quoting and escaping" section in  the  ffmpeg-utils(1)	manual
       for more information about the employed escaping procedure.

       A first level escaping affects the content of each filter option value,
       which may contain the special character ":" used to separate values, or
       one of the escaping characters "\'".

       A second level escaping affects the whole filter description, which may
       contain	the  escaping characters "\'" or the special characters "[],;"
       used by the filtergraph description.

       Finally, when you specify a filtergraph on  a  shell  commandline,  you
       need to perform a third level escaping for the shell special characters
       contained within it.

       For  example,  consider	the  following	string	to  be embedded in the
       drawtext filter description text value:

	       this is a 'string': may contain one, or more, special characters

       This string contains the "'" special escaping character,	 and  the  ":"
       special character, so it needs to be escaped in this way:

	       text=this is a \'string\'\: may contain one, or more, special characters

       A  second  level	 of  escaping  is  required  when embedding the filter
       description in a filtergraph description, in order to  escape  all  the
       filtergraph special characters. Thus the example above becomes:

	       drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

       (note  that  in	addition to the "\'" escaping special characters, also
       "," needs to be escaped).

       Finally an additional level of escaping	is  needed  when  writing  the
       filtergraph  description	 in  a	shell  command,	 which	depends on the
       escaping rules of the adopted shell. For example, assuming that "\"  is
       special	and  needs to be escaped with another "\", the previous string
       will finally result in:

	       -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

       In order to avoid cumbersome escaping when  using  a  commandline  tool
       accepting  a  filter  specification  as input, it is advisable to avoid
       direct inclusion of the filter or options specification in the shell.

       For example, in case of the drawtext filter, you might  prefer  to  use
       the textfile option in place of text to specify the text to render.

       When   using   the   ffmpeg   tool,  you	 might	consider  to  use  the
       -filter_script option or -filter_complex_script option.

TIMELINE EDITING
       Some  filters  support  a  generic  enable  option.  For	 the   filters
       supporting  timeline  editing,  this option can be set to an expression
       which is evaluated before  sending  a  frame  to	 the  filter.  If  the
       evaluation is non-zero, the filter will be enabled, otherwise the frame
       will be sent unchanged to the next filter in the filtergraph.

       The expression accepts the following values:

       t   timestamp  expressed	 in  seconds,  NAN  if	the input timestamp is
	   unknown

       n   sequential number of the input frame, starting from 0

       pos the position in the file  of	 the  input  frame,  NAN  if  unknown;
	   deprecated, do not use

       w
       h   width and height of the input frame if video

       Additionally,  these filters support an enable command that can be used
       to re-define the expression.

       Like any other filtering option, the enable  option  follows  the  same
       rules.

       For  example,  to enable a blur filter (smartblur) from 10 seconds to 3
       minutes, and a curves filter starting at 3 seconds:

	       smartblur = enable='between(t,10,3*60)',
	       curves	 = enable='gte(t,3)' : preset=cross_process

       See "ffmpeg -filters" to view which filters have timeline support.

CHANGING OPTIONS AT RUNTIME WITH A COMMAND
       Some options can be changed during the operation of the filter using  a
       command.	 These	options	 are  marked  'T'  on  the output of ffmpeg -h
       filter=<name of filter>.	 The name of the command is the	 name  of  the
       option and the argument is the new value.

OPTIONS FOR FILTERS WITH SEVERAL INPUTS
       Some  filters  with  several  inputs  support  a common set of options.
       These options can only be set by name, not with the short notation.

       eof_action
	   The action to take when EOF is encountered on the secondary	input;
	   it accepts one of the following values:

	   repeat
	       Repeat the last frame (the default).

	   endall
	       End both streams.

	   pass
	       Pass the main input through.

       shortest
	   If  set to 1, force the output to terminate when the shortest input
	   terminates. Default value is 0.

       repeatlast
	   If set to 1, force the filter to extend the last frame of secondary
	   streams until the end of the primary stream. A value of 0  disables
	   this behavior.  Default value is 1.

       ts_sync_mode
	   How	strictly  to sync streams based on secondary input timestamps;
	   it accepts one of the following values:

	   default
	       Frame from secondary input with	the  nearest  lower  or	 equal
	       timestamp to the primary input frame.

	   nearest
	       Frame  from secondary input with the absolute nearest timestamp
	       to the primary input frame.

AUDIO FILTERS
       When you configure your FFmpeg  build,  you  can	 disable  any  of  the
       existing	 filters using "--disable-filters".  The configure output will
       show the audio filters included in your build.

       Below is a description of the currently available audio filters.

   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

       •   Cross fade from one input to another:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

       •   Cross fade from one input to another but without overlapping:

		   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

   acrossover
       Split audio stream into several bands.

       This  filter  splits  audio  stream  into two or more frequency ranges.
       Summing all streams back will give flat output.

       The filter accepts the following options:

       split
	   Set split frequencies. Those must be positive and increasing.

       order
	   Set filter order for each band split. This controls filter roll-off
	   or steepness of filter transfer function.  Available values are:

	   2nd 12 dB per octave.

	   4th 24 dB per octave.

	   6th 36 dB per octave.

	   8th 48 dB per octave.

	   10th
	       60 dB per octave.

	   12th
	       72 dB per octave.

	   14th
	       84 dB per octave.

	   16th
	       96 dB per octave.

	   18th
	       108 dB per octave.

	   20th
	       120 dB per octave.

	   Default is 4th.

       level
	   Set input gain level. Allowed range is from 0 to 1.	Default	 value
	   is 1.

       gains
	   Set output gain for each band. Default value is 1 for all bands.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

	   Default value is "auto".

       Examples

       •   Split  input	 audio stream into two bands (low and high) with split
	   frequency of 1500 Hz, each band will be in separate stream:

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

       •   Same as above, but with higher filter order:

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500:order=8th[LOW][HIGH]' -map '[LOW]' low.wav -map '[HIGH]' high.wav

       •   Same as above, but also with additional  middle  band  (frequencies
	   between 1500 and 8000):

		   ffmpeg -i in.flac -filter_complex 'acrossover=split=1500 8000:order=8th[LOW][MID][HIGH]' -map '[LOW]' low.wav -map '[MID]' mid.wav -map '[HIGH]' high.wav

   acrusher
       Reduce audio bit resolution.

       This  filter  is bit crusher with enhanced functionality. A bit crusher
       is used to audibly reduce number of bits an  audio  signal  is  sampled
       with.  This  doesn't  change the bit depth at all, it just produces the
       effect. Material reduced in bit depth sounds more harsh and  "digital".
       This  filter  is	 able  to  even	 round to continuous values instead of
       discrete bit depths.  Additionally it has a D/C offset which results in
       different crushing of the lower and the upper half of the  signal.   An
       Anti-Aliasing setting is able to produce "softer" crushing sounds.

       Another	feature	 of this filter is the logarithmic mode.  This setting
       switches from linear distances between bits to logarithmic  ones.   The
       result is a much more "natural" sounding crusher which doesn't gate low
       signals	for  example.  The  human ear has a logarithmic perception, so
       this kind of crushing is much more pleasant.  Logarithmic  crushing  is
       also able to get anti-aliased.

       The filter accepts the following options:

       level_in
	   Set level in.

       level_out
	   Set level out.

       bits
	   Set bit reduction.

       mix Set mixing amount.

       mode
	   Can be linear: "lin" or logarithmic: "log".

       dc  Set DC.

       aa  Set anti-aliasing.

       samples
	   Set sample reduction.

       lfo Enable LFO. By default disabled.

       lforange
	   Set LFO range.

       lforate
	   Set LFO rate.

       Commands

       This filter supports the all above options as commands.

   acue
       Delay  audio  filtering	until a given wallclock timestamp. See the cue
       filter.

   adeclick
       Remove impulsive noise from input audio.

       Samples detected	 as  impulsive	noise  are  replaced  by  interpolated
       samples using autoregressive modelling.

       window, w
	   Set	window size, in milliseconds. Allowed range is from 10 to 100.
	   Default value is 55 milliseconds.  This sets size of	 window	 which
	   will be processed at once.

       overlap, o
	   Set	window overlap, in percentage of window size. Allowed range is
	   from 50 to 95. Default value is 75 percent.	Setting this to a very
	   high value  increases  impulsive  noise  removal  but  makes	 whole
	   process much slower.

       arorder, a
	   Set	autoregression	order,	in  percentage of window size. Allowed
	   range is from 0 to 25. Default value is 2 percent. This option also
	   controls quality  of	 interpolated  samples	using  neighbour  good
	   samples.

       threshold, t
	   Set threshold value. Allowed range is from 1 to 100.	 Default value
	   is 2.  This controls the strength of impulsive noise which is going
	   to  be removed.  The lower value, the more samples will be detected
	   as impulsive noise.

       burst, b
	   Set burst fusion, in percentage of window size. Allowed range is  0
	   to  10.  Default  value is 2.  If any two samples detected as noise
	   are spaced less than this value then any sample between  those  two
	   samples will be also detected as noise.

       method, m
	   Set overlap method.

	   It accepts the following values:

	   add, a
	       Select  overlap-add  method.  Even not interpolated samples are
	       slightly changed with this method.

	   save, s
	       Select overlap-save method.  Not	 interpolated  samples	remain
	       unchanged.

	   Default value is "a".

   adeclip
       Remove clipped samples from input audio.

       Samples	detected as clipped are replaced by interpolated samples using
       autoregressive modelling.

       window, w
	   Set window size, in milliseconds. Allowed range is from 10 to  100.
	   Default  value  is 55 milliseconds.	This sets size of window which
	   will be processed at once.

       overlap, o
	   Set window overlap, in percentage of window size. Allowed range  is
	   from 50 to 95. Default value is 75 percent.

       arorder, a
	   Set	autoregression	order,	in  percentage of window size. Allowed
	   range is from 0 to 25. Default value is 8 percent. This option also
	   controls quality  of	 interpolated  samples	using  neighbour  good
	   samples.

       threshold, t
	   Set threshold value. Allowed range is from 1 to 100.	 Default value
	   is 10. Higher values make clip detection less aggressive.

       hsize, n
	   Set	size  of histogram used to detect clips. Allowed range is from
	   100 to 9999.	 Default  value	 is  1000.  Higher  values  make  clip
	   detection less aggressive.

       method, m
	   Set overlap method.

	   It accepts the following values:

	   add, a
	       Select  overlap-add  method.  Even not interpolated samples are
	       slightly changed with this method.

	   save, s
	       Select overlap-save method.  Not	 interpolated  samples	remain
	       unchanged.

	   Default value is "a".

   adecorrelate
       Apply decorrelation to input audio stream.

       The filter accepts the following options:

       stages
	   Set	decorrelation  stages of filtering. Allowed range is from 1 to
	   16. Default value is 6.

       seed
	   Set random seed used for setting delay in samples across channels.

   adelay
       Delay one or more audio channels.

       Samples in delayed channel are filled with silence.

       The filter accepts the following option:

       delays
	   Set list of delays in milliseconds for each	channel	 separated  by
	   '|'.	  Unused  delays  will be silently ignored. If number of given
	   delays is smaller than number of channels  all  remaining  channels
	   will not be delayed.	 If you want to delay exact number of samples,
	   append  'S'	to  number.   If you want instead to delay in seconds,
	   append 's' to number.

       all Use last set delay  for  all	 remaining  channels.  By  default  is
	   disabled.   This  option  if enabled changes how option "delays" is
	   interpreted.

       Examples

       •   Delay first channel by  1.5	seconds,  the  third  channel  by  0.5
	   seconds  and	 leave the second channel (and any other channels that
	   may be present) unchanged.

		   adelay=1500|0|500

       •   Delay second channel by 500	samples,  the  third  channel  by  700
	   samples  and	 leave	the first channel (and any other channels that
	   may be present) unchanged.

		   adelay=0|500S|700S

       •   Delay all channels by same number of samples:

		   adelay=delays=64S:all=1

   adenorm
       Remedy denormals in audio by adding extremely low-level noise.

       This filter  shall  be  placed  before  any  filter  that  can  produce
       denormals.

       A description of the accepted parameters follows.

       level
	   Set	level of added noise in dB. Default is -351.  Allowed range is
	   from -451 to -90.

       type
	   Set type of added noise.

	   dc  Add DC signal.

	   ac  Add AC signal.

	   square
	       Add square signal.

	   pulse
	       Add pulse signal.

	   Default is "dc".

       Commands

       This filter supports the all above options as commands.

   aderivative, aintegral
       Compute derivative/integral of audio stream.

       Applying both filters one after another produces original audio.

   adrc
       Apply spectral dynamic range controller filter to input audio stream.

       A description of the accepted options follows.

       transfer
	   Set the transfer expression.

	   The expression can contain the following constants:

	   ch  current channel number

	   sn  current sample number

	   nb_channels
	       number of channels

	   t   timestamp expressed in seconds

	   sr  sample rate

	   p   current frequency power value, in dB

	   f   current frequency in Hz

	   Default value is "p".

       attack
	   Set	the  attack  in	 milliseconds.	Default	 is  50	 milliseconds.
	   Allowed range is from 1 to 1000 milliseconds.

       release
	   Set	the  release  in  milliseconds.	 Default  is 100 milliseconds.
	   Allowed range is from 5 to 2000 milliseconds.

       channels
	   Set which channels to filter, by default "all"  channels  in	 audio
	   stream are filtered.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Apply spectral compression to all frequencies with threshold of -50
	   dB and 1:6 ratio:

		   adrc=transfer='if(gt(p,-50),-50+(p-(-50))/6,p)':attack=50:release=100

       •   Similar to above but with 1:2 ratio and filtering only front center
	   channel:

		   adrc=transfer='if(gt(p,-50),-50+(p-(-50))/2,p)':attack=50:release=100:channels=FC

       •   Apply  spectral noise gate to all frequencies with threshold of -85
	   dB and with short attack time and short release time:

		   adrc=transfer='if(lte(p,-85),p-800,p)':attack=1:release=5

       •   Apply spectral expansion to all frequencies with threshold  of  -10
	   dB and 1:2 ratio:

		   adrc=transfer='if(lt(p,-10),-10+(p-(-10))*2,p)':attack=50:release=100

       •   Apply limiter to max -60 dB to all frequencies, with attack of 2 ms
	   and release of 10 ms:

		   adrc=transfer='min(p,-60)':attack=2:release=10

   adynamicequalizer
       Apply dynamic equalization to input audio stream.

       A description of the accepted options follows.

       threshold
	   Set	 the   detection   threshold  used  to	trigger	 equalization.
	   Threshold detection is using detection filter.  Default value is 0.
	   Allowed range is from 0 to 100.

       dfrequency
	   Set the detection frequency in Hz used for detection filter used to
	   trigger equalization.  Default value is 1000 Hz. Allowed  range  is
	   between 2 and 1000000 Hz.

       dqfactor
	   Set	the  detection	resonance  factor for detection filter used to
	   trigger equalization.  Default value is 1. Allowed  range  is  from
	   0.001 to 1000.

       tfrequency
	   Set	the target frequency of equalization filter.  Default value is
	   1000 Hz. Allowed range is between 2 and 1000000 Hz.

       tqfactor
	   Set the target resonance factor  for	 target	 equalization  filter.
	   Default value is 1. Allowed range is from 0.001 to 1000.

       attack
	   Set	the  amount  of	 milliseconds the signal from detection has to
	   rise above the  detection  threshold	 before	 equalization  starts.
	   Default is 20. Allowed range is between 1 and 2000.

       release
	   Set	the  amount  of	 milliseconds the signal from detection has to
	   fall	 below	the  detection	threshold  before  equalization	 ends.
	   Default is 200. Allowed range is between 1 and 2000.

       ratio
	   Set the ratio by which the equalization gain is raised.  Default is
	   1. Allowed range is between 0 and 30.

       makeup
	   Set	the  makeup  offset  by which the equalization gain is raised.
	   Default is 0. Allowed range is between 0 and 100.

       range
	   Set the max allowed cut/boost amount. Default is 50.	 Allowed range
	   is from 1 to 200.

       mode
	   Set the mode of filter operation, can be one of the following:

	   listen
	       Output only isolated detection signal.

	   cut Cut frequencies above detection threshold.

	   boost
	       Boost frequencies bellow detection threshold.

	   Default mode is cut.

       dftype
	   Set the type of detection filter, can be one of the following:

	   bandpass
	   lowpass
	   highpass
	   peak

	   Default type is bandpass.

       tftype
	   Set the type of target filter, can be one of the following:

	   bell
	   lowshelf
	   highshelf

	   Default type is bell.

       direction
	   Set processing direction relative to threshold.

	   downward
	       Boost/Cut if threshold is higher/lower than detected volume.

	   upward
	       Boost/Cut if threshold is lower/higher than detected volume.

	   Default direction is downward.

       auto
	   Automatically gather threshold from detection filter. By default is
	   disabled.  This option is useful to	detect	threshold  in  certain
	   time	 frame	of  input  audio  stream, in such case option value is
	   changed at runtime.

	   Available values are:

	   disabled
	       Disable using automatically gathered threshold value.

	   off Stop picking threshold value.

	   on  Start picking threshold value.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

       Commands

       This filter supports the all above options as commands.

   adynamicsmooth
       Apply dynamic smoothing to input audio stream.

       A description of the accepted options follows.

       sensitivity
	   Set an amount of sensitivity to frequency fluctations.  Default  is
	   2.  Allowed range is from 0 to 1e+06.

       basefreq
	   Set	a  base	 frequency  for	 smoothing.  Default  value  is 22050.
	   Allowed range is from 2 to 1e+06.

       Commands

       This filter supports the all above options as commands.

   aecho
       Apply echoing to the input audio.

       Echoes are reflected sound and can occur	 naturally  amongst  mountains
       (and  sometimes large buildings) when talking or shouting; digital echo
       effects emulate this behaviour and are often used to help fill out  the
       sound  of a single instrument or vocal. The time difference between the
       original signal and the reflection is the "delay", and the loudness  of
       the  reflected  signal  is  the	"decay".   Multiple  echoes  can  have
       different delays and decays.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain of reflected signal. Default is 0.6.

       out_gain
	   Set output gain of reflected signal. Default is 0.3.

       delays
	   Set list of time intervals in milliseconds between original	signal
	   and reflections separated by '|'. Allowed range for each "delay" is
	   "(0 - 90000.0]".  Default is 1000.

       decays
	   Set	list  of  loudness  of	reflected  signals  separated  by '|'.
	   Allowed range for each "decay" is "(0 - 1.0]".  Default is 0.5.

       Examples

       •   Make it sound as if there are twice	as  many  instruments  as  are
	   actually playing:

		   aecho=0.8:0.88:60:0.4

       •   If  delay  is  very	short,	then it sounds like a (metallic) robot
	   playing music:

		   aecho=0.8:0.88:6:0.4

       •   A longer  delay  will  sound	 like  an  open	 air  concert  in  the
	   mountains:

		   aecho=0.8:0.9:1000:0.3

       •   Same as above but with one more mountain:

		   aecho=0.8:0.9:1000|1800:0.3|0.25

   aemphasis
       Audio  emphasis filter creates or restores material directly taken from
       LPs or emphased CDs with different filter curves. E.g. to  store	 music
       on vinyl the signal has to be altered by a filter first to even out the
       disadvantages  of  this	recording medium.  Once the material is played
       back the inverse filter has to be applied to restore the distortion  of
       the frequency response.

       The filter accepts the following options:

       level_in
	   Set input gain.

       level_out
	   Set output gain.

       mode
	   Set	filter	mode.  For restoring material use "reproduction" mode,
	   otherwise use "production" mode. Default is "reproduction" mode.

       type
	   Set filter type. Selects medium. Can be one of the following:

	   col select Columbia.

	   emi select EMI.

	   bsi select BSI (78RPM).

	   riaa
	       select RIAA.

	   cd  select Compact Disc (CD).

	   50fm
	       select 50µs (FM).

	   75fm
	       select 75µs (FM).

	   50kf
	       select 50µs (FM-KF).

	   75kf
	       select 75µs (FM-KF).

       Commands

       This filter supports the all above options as commands.

   aeval
       Modify an audio signal according to the specified expressions.

       This filter accepts one or more expressions  (one  for  each  channel),
       which are evaluated and used to modify a corresponding audio signal.

       It accepts the following parameters:

       exprs
	   Set	the  '|'-separated expressions list for each separate channel.
	   If the number of input channels  is	greater	 than  the  number  of
	   expressions,	  the  last  specified	expression  is	used  for  the
	   remaining output channels.

       channel_layout, c
	   Set output channel layout. If not specified, the channel layout  is
	   specified by the number of expressions. If set to same, it will use
	   by default the same input channel layout.

       Each  expression	 in  exprs  can	 contain  the  following constants and
       functions:

       ch  channel number of the current expression

       n   number of the evaluated sample, starting from 0

       s   sample rate

       t   time of the evaluated sample expressed in seconds

       nb_in_channels
       nb_out_channels
	   input and output number of channels

       val(CH)
	   the value of input channel with number CH

       Note: this filter is slow. For  faster  processing  you	should	use  a
       dedicated filter.

       Examples

       •   Half volume:

		   aeval=val(ch)/2:c=same

       •   Invert phase of the second channel:

		   aeval=val(0)|-val(1)

   aexciter
       An  exciter  is	used  to produce high sound that is not present in the
       original signal. This is done by creating harmonic distortions  of  the
       signal  which are restricted in range and added to the original signal.
       An Exciter raises the upper end	of  an	audio  signal  without	simply
       raising	the  higher frequencies like an equalizer would do to create a
       more "crisp" or "brilliant" sound.

       The filter accepts the following options:

       level_in
	   Set input level prior processing of signal.	Allowed range is  from
	   0 to 64.  Default value is 1.

       level_out
	   Set output level after processing of signal.	 Allowed range is from
	   0 to 64.  Default value is 1.

       amount
	   Set	the  amount  of	 harmonics  added to original signal.  Allowed
	   range is from 0 to 64.  Default value is 1.

       drive
	   Set the amount of newly created harmonics.  Allowed range  is  from
	   0.1 to 10.  Default value is 8.5.

       blend
	   Set	the  octave of newly created harmonics.	 Allowed range is from
	   -10 to 10.  Default value is 0.

       freq
	   Set the  lower  frequency  limit  of	 producing  harmonics  in  Hz.
	   Allowed range is from 2000 to 12000 Hz.  Default is 7500 Hz.

       ceil
	   Set	the  upper  frequency  limit  of producing harmonics.  Allowed
	   range is from 9999 to 20000 Hz.  If value is lower than 10000 Hz no
	   limit is applied.

       listen
	   Mute the original signal  and  output  only	added  harmonics.   By
	   default is disabled.

       Commands

       This filter supports the all above options as commands.

   afade
       Apply fade-in/out effect to input audio.

       A description of the accepted parameters follows.

       type, t
	   Specify  the	 effect type, can be either "in" for fade-in, or "out"
	   for a fade-out effect. Default is "in".

       start_sample, ss
	   Specify the number of the start sample for starting	to  apply  the
	   fade effect. Default is 0.

       nb_samples, ns
	   Specify  the	 number	 of  samples  for which the fade effect has to
	   last. At the end of the fade-in effect the output audio  will  have
	   the	same  volume  as  the  input audio, at the end of the fade-out
	   transition the output audio will be silence. Default is 44100.

       start_time, st
	   Specify the start time of the fade effect. Default is 0.  The value
	   must be specified as a time duration; see the Time duration section
	   in the ffmpeg-utils(1) manual for the accepted syntax.  If set this
	   option is used instead of start_sample.

       duration, d
	   Specify the duration of the fade  effect.  See  the	Time  duration
	   section  in the ffmpeg-utils(1) manual for the accepted syntax.  At
	   the end of the fade-in effect the output audio will have  the  same
	   volume  as  the  input audio, at the end of the fade-out transition
	   the output audio will be  silence.	By  default  the  duration  is
	   determined  by  nb_samples.	 If set this option is used instead of
	   nb_samples.

       curve
	   Set curve for fade transition.

	   It accepts the following values:

	   tri select triangular, linear slope (default)

	   qsin
	       select quarter of sine wave

	   hsin
	       select half of sine wave

	   esin
	       select exponential sine wave

	   log select logarithmic

	   ipar
	       select inverted parabola

	   qua select quadratic

	   cub select cubic

	   squ select square root

	   cbr select cubic root

	   par select parabola

	   exp select exponential

	   iqsin
	       select inverted quarter of sine wave

	   ihsin
	       select inverted half of sine wave

	   dese
	       select double-exponential seat

	   desi
	       select double-exponential sigmoid

	   losi
	       select logistic sigmoid

	   sinc
	       select sine cardinal function

	   isinc
	       select inverted sine cardinal function

	   quat
	       select quartic

	   quatr
	       select quartic root

	   qsin2
	       select squared quarter of sine wave

	   hsin2
	       select squared half of sine wave

	   nofade
	       no fade applied

       silence
	   Set the initial gain	 for  fade-in  or  final  gain	for  fade-out.
	   Default value is 0.0.

       unity
	   Set	the  initial  gain  for	 fade-out  or  final gain for fade-in.
	   Default value is 1.0.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Fade in first 15 seconds of audio:

		   afade=t=in:ss=0:d=15

       •   Fade out last 25 seconds of a 900 seconds audio:

		   afade=t=out:st=875:d=25

   afftdn
       Denoise audio samples with FFT.

       A description of the accepted parameters follows.

       noise_reduction, nr
	   Set the noise reduction  in	dB,  allowed  range  is	 0.01  to  97.
	   Default value is 12 dB.

       noise_floor, nf
	   Set	the  noise  floor in dB, allowed range is -80 to -20.  Default
	   value is -50 dB.

       noise_type, nt
	   Set the noise type.

	   It accepts the following values:

	   white, w
	       Select white noise.

	   vinyl, v
	       Select vinyl noise.

	   shellac, s
	       Select shellac noise.

	   custom, c
	       Select custom noise, defined in "bn" option.

	       Default value is white noise.

       band_noise, bn
	   Set custom band noise profile for every one of 15 bands.  Bands are
	   separated by ' ' or '|'.

       residual_floor, rf
	   Set the residual floor in dB, allowed range is -80 to -20.  Default
	   value is -38 dB.

       track_noise, tn
	   Enable noise floor tracking. By default  is	disabled.   With  this
	   enabled, noise floor is automatically adjusted.

       track_residual, tr
	   Enable residual tracking. By default is disabled.

       output_mode, om
	   Set the output mode.

	   It accepts the following values:

	   input, i
	       Pass input unchanged.

	   output, o
	       Pass noise filtered out.

	   noise, n
	       Pass only noise.

	       Default value is output.

       adaptivity, ad
	   Set the adaptivity factor, used how fast to adapt gains adjustments
	   per	each  frequency bin. Value 0 enables instant adaptation, while
	   higher values react much slower.  Allowed range is  from  0	to  1.
	   Default value is 0.5.

       floor_offset, fo
	   Set	the  noise  floor offset factor. This option is used to adjust
	   offset applied to measured noise floor. It is only  effective  when
	   noise  floor	 tracking  is  enabled.	 Allowed range is from -2.0 to
	   2.0. Default value is 1.0.

       noise_link, nl
	   Set the noise link used for multichannel audio.

	   It accepts the following values:

	   none
	       Use unchanged channel's noise floor.

	   min Use measured min noise floor of all channels.

	   max Use measured max noise floor of all channels.

	   average
	       Use measured average noise floor of all channels.

	       Default value is min.

       band_multiplier, bm
	   Set the band multiplier factor,  used  how  much  to	 spread	 bands
	   across  frequency  bins.   Allowed  range is from 0.2 to 5. Default
	   value is 1.25.

       sample_noise, sn
	   Toggle capturing and measurement of noise profile from input audio.

	   It accepts the following values:

	   start, begin
	       Start sample noise capture.

	   stop, end
	       Stop sample noise capture and measure new noise band profile.

	       Default value is "none".

       gain_smooth, gs
	   Set gain smooth spatial radius, used to  smooth  gains  applied  to
	   each frequency bin.	Useful to reduce random music noise artefacts.
	   Higher  values increases smoothing of gains.	 Allowed range is from
	   0 to 50.  Default value is 0.

       Commands

       This filter supports the some above mentioned options as commands.

       Examples

       •   Reduce white noise by 10dB, and use previously measured noise floor
	   of -40dB:

		   afftdn=nr=10:nf=-40

       •   Reduce white noise by 10dB, also set initial noise floor  to	 -80dB
	   and	enable	automatic  tracking of noise floor so noise floor will
	   gradually change during processing:

		   afftdn=nr=10:nf=-80:tn=1

       •   Reduce noise by 20dB, using noise floor of -40dB and using commands
	   to take noise profile of first 0.4 seconds of input audio:

		   asendcmd=0.0 afftdn sn start,asendcmd=0.4 afftdn sn stop,afftdn=nr=20:nf=-40

   afftfilt
       Apply arbitrary expressions to samples in frequency domain.

       real
	   Set frequency domain real  expression  for  each  separate  channel
	   separated by '|'. Default is "re".  If the number of input channels
	   is  greater	than  the  number  of  expressions, the last specified
	   expression is used for the remaining output channels.

       imag
	   Set frequency domain imaginary expression for each separate channel
	   separated by '|'. Default is "im".

	   Each	 expression  in	 real  and  imag  can  contain	the  following
	   constants and functions:

	   sr  sample rate

	   b   current frequency bin number

	   nb  number of available bins

	   ch  channel number of the current expression

	   chs number of channels

	   pts current frame pts

	   re  current real part of frequency bin of current channel

	   im  current imaginary part of frequency bin of current channel

	   real(b, ch)
	       Return  the  value  of  real  part of frequency bin at location
	       (bin,channel)

	   imag(b, ch)
	       Return the value of imaginary part of frequency bin at location
	       (bin,channel)

       win_size
	   Set window size. Allowed range is from 16 to	 131072.   Default  is
	   4096

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann, hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set	window	overlap.  If  set  to  1,  the recommended overlap for
	   selected window function will be picked. Default is 0.75.

       Examples

       •   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))'"

       •   Apply robotize effect:

		   afftfilt="real='hypot(re,im)*sin(0)':imag='hypot(re,im)*cos(0)':win_size=512:overlap=0.75"

       •   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"

       •   Apply phase shift:

		   afftfilt="real=re*cos(1)-im*sin(1):imag=re*sin(1)+im*cos(1)"

   afir
       Apply an arbitrary Finite Impulse Response filter.

       This filter is designed for applying long FIR filters, up to 60 seconds
       long.

       It can be  used	as  component  for  digital  crossover	filters,  room
       equalization,	cross	 talk	cancellation,	wavefield   synthesis,
       auralization, ambiophonics, ambisonics and spatialization.

       This filter uses the streams higher than first one as FIR coefficients.
       If the non-first stream holds a single channel, it will be used for all
       input channels in the first stream, otherwise the number of channels in
       the non-first stream must be same as the	 number	 of  channels  in  the
       first stream.

       It accepts the following parameters:

       dry Set dry gain. This sets input gain.

       wet Set wet gain. This sets final output gain.

       length
	   Set Impulse Response filter length. Default is 1, which means whole
	   IR is processed.

       gtype
	   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.

	   ac  select AC gain.

	   rms select RMS gain.

       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 65536.  Lower values  decreases  latency
	   at cost of higher CPU usage.

       maxp
	   Set	maximal	 partition size used for convolution. Default is 8192.
	   Allowed range is from 8 to 65536.  Lower values  may	 increase  CPU
	   usage.

       nbirs
	   Set	number	of  input  impulse  responses  streams	which  will be
	   switchable at runtime.  Allowed range is from 1 to 32.  Default  is
	   1.

       ir  Set	IR stream which will be used for convolution, starting from 0,
	   should always be lower  than	 supplied  value  by  "nbirs"  option.
	   Default is 0.  This option can be changed at runtime via commands.

       precision
	   Set which precision to use when processing samples.

	   auto
	       Auto pick internal sample format depending on other filters.

	   float
	       Always use single-floating point precision sample format.

	   double
	       Always use double-floating point precision sample format.

	   Default value is auto.

       irload
	   Set	when  to load IR stream. Can be "init" or "access".  First one
	   load and prepares all IRs on initialization,	 second	 one  once  on
	   first access of specific IR.	 Default is "init".

       Examples

       •   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

       •   Apply  true	stereo	processing  given input stereo stream, and two
	   stereo impulse responses for left and right	channel,  the  impulse
	   response files are files with names l_ir.wav and r_ir.wav:

		   "pan=4C|c0=FL|c1=FL|c2=FR|c3=FR[a];amovie=l_ir.wav[LIR];amovie=r_ir.wav[RIR];[LIR][RIR]amerge[ir];[a][ir]afir=irfmt=input:gtype=gn:irgain=-5dB,pan=stereo|FL<c0+c2|FR<c1+c3"

   aformat
       Set  output  format constraints for the input audio. The framework will
       negotiate the most appropriate format to minimize conversions.

       It accepts the following parameters:

       sample_fmts, f
	   A '|'-separated list of requested sample formats.

       sample_rates, r
	   A '|'-separated list of requested sample rates.

       channel_layouts, cl
	   A '|'-separated list of requested channel layouts.

	   See the Channel Layout section in the  ffmpeg-utils(1)  manual  for
	   the required syntax.

       If a parameter is omitted, all values are allowed.

       Force the output to either unsigned 8-bit or signed 16-bit stereo

	       aformat=sample_fmts=u8|s16:channel_layouts=stereo

   afreqshift
       Apply frequency shift to input audio samples.

       The filter accepts the following options:

       shift
	   Specify  frequency  shift.  Allowed	range  is -INT_MAX to INT_MAX.
	   Default value is 0.0.

       level
	   Set output gain applied to final output. Allowed range is from  0.0
	   to 1.0.  Default value is 1.0.

       order
	   Set filter order used for filtering. Allowed range is from 1 to 16.
	   Default value is 8.

       Commands

       This filter supports the all above options as commands.

   afwtdn
       Reduce broadband noise from input samples using Wavelets.

       A description of the accepted options follows.

       sigma
	   Set	the  noise sigma, allowed range is from 0 to 1.	 Default value
	   is 0.  This option controls strength of denoising applied to	 input
	   samples.   Most  useful way to set this option is via decibels, eg.
	   -45dB.

       levels
	   Set the number of wavelet levels of decomposition.	Allowed	 range
	   is  from  1 to 12.  Default value is 10.  Setting this too low make
	   denoising performance very poor.

       wavet
	   Set wavelet type for decomposition of input frame.  They are sorted
	   by  number  of  coefficients,  from	lowest	 to   highest.	  More
	   coefficients	 means	worse  filtering  speed,  but  overall	better
	   quality.  Available wavelets are:

	   sym2
	   sym4
	   rbior68
	   deb10
	   sym10
	   coif5
	   bl3
       percent
	   Set percent of full denoising. Allowed  range  is  from  0  to  100
	   percent.  Default value is 85 percent or partial denoising.

       profile
	   If  enabled,	 first	input frame will be used as noise profile.  If
	   first frame samples contain	non-noise  performance	will  be  very
	   poor.

       adaptive
	   If  enabled,	 input	frames are analyzed for presence of noise.  If
	   noise is detected with high possibility then	 input	frame  profile
	   will be used for processing following frames, until new noise frame
	   is detected.

       samples
	   Set	size  of  single  frame in number of samples. Allowed range is
	   from 512 to 65536. Default frame size is 8192 samples.

       softness
	   Set softness applied inside thresholding function. Allowed range is
	   from 0 to 10. Default softness is 1.

       Commands

       This filter supports the all above options as commands.

   agate
       A gate is mainly used to reduce lower parts of a signal. This  kind  of
       signal processing reduces disturbing noise between useful signals.

       Gating  is  done by detecting the volume below a chosen level threshold
       and dividing it by the factor set with ratio. The bottom of  the	 noise
       floor  is  set  via  range. Because an exact manipulation of the signal
       would cause distortion of the waveform the reduction  can  be  levelled
       over time. This is done by setting attack and release.

       attack  determines  how long the signal has to fall below the threshold
       before any reduction will occur and release sets the  time  the	signal
       has to rise above the threshold to reduce the reduction again.  Shorter
       signals than the chosen attack time will be left untouched.

       level_in
	   Set	input  level before filtering.	Default is 1. Allowed range is
	   from 0.015625 to 64.

       mode
	   Set the mode of operation. Can be "upward" or "downward".   Default
	   is "downward". If set to "upward" mode, higher parts of signal will
	   be	amplified,   expanding	dynamic	 range	in  upward  direction.
	   Otherwise, in case of "downward" lower  parts  of  signal  will  be
	   reduced.

       range
	   Set	the  level  of	gain  reduction	 when  the signal is below the
	   threshold.  Default is 0.06125. Allowed  range  is  from  0	to  1.
	   Setting  this  to 0 disables reduction and then filter behaves like
	   expander.

       threshold
	   If a signal rises above this level the gain reduction is  released.
	   Default is 0.125. Allowed range is from 0 to 1.

       ratio
	   Set	a ratio by which the signal is reduced.	 Default is 2. Allowed
	   range is from 1 to 9000.

       attack
	   Amount of milliseconds the signal has to rise above	the  threshold
	   before  gain	 reduction stops.  Default is 20 milliseconds. Allowed
	   range is from 0.01 to 9000.

       release
	   Amount of milliseconds the signal has to fall below	the  threshold
	   before   the	  reduction   is   increased  again.  Default  is  250
	   milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
	   Set amount of amplification of signal after processing.  Default is
	   1. Allowed range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter  gain  reduction
	   more softly.	 Default is 2.828427125. Allowed range is from 1 to 8.

       detection
	   Choose if exact signal should be taken for detection or an RMS like
	   one.	 Default is "rms". Can be "peak" or "rms".

       link
	   Choose  if  the  average  level  between all channels or the louder
	   channel affects  the	 reduction.   Default  is  "average".  Can  be
	   "average" or "maximum".

       Commands

       This filter supports the all above options as commands.

   aiir
       Apply an arbitrary Infinite Impulse Response filter.

       It accepts the following parameters:

       zeros, z
	   Set B/numerator/zeros/reflection coefficients.

       poles, p
	   Set A/denominator/poles/ladder coefficients.

       gains, k
	   Set channels gains.

       dry_gain
	   Set input gain.

       wet_gain
	   Set output gain.

       format, f
	   Set coefficients format.

	   ll  lattice-ladder function

	   sf  analog transfer function

	   tf  digital transfer function

	   zp  Z-plane zeros/poles, cartesian (default)

	   pr  Z-plane zeros/poles, polar radians

	   pd  Z-plane zeros/poles, polar degrees

	   sp  S-plane zeros/poles

       process, r
	   Set type of processing.

	   d   direct processing

	   s   serial processing

	   p   parallel processing

       precision, e
	   Set filtering precision.

	   dbl double-precision floating-point (default)

	   flt single-precision floating-point

	   i32 32-bit integers

	   i16 16-bit integers

       normalize, n
	   Normalize  filter coefficients, by default is enabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       mix How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       response
	   Show	 IR  frequency	response, magnitude(magenta), phase(green) and
	   group delay(yellow) in additional video stream.  By default	it  is
	   disabled.

       channel
	   Set	for which IR channel to display frequency response. By default
	   is first channel displayed. This option is used only when  response
	   is enabled.

       size
	   Set	video  stream  size. This option is used only when response is
	   enabled.

       Coefficients in "tf" and "sf" format are separated by spaces and are in
       ascending order.

       Coefficients in "zp" format  are	 separated  by	spaces	and  order  of
       coefficients  doesn't  matter.  Coefficients in "zp" format are complex
       numbers with i imaginary unit.

       Different coefficients and gains can be provided for every channel,  in
       such  case  use	'|'  to	 separate coefficients or gains. Last provided
       coefficients will be used for all remaining channels.

       Examples

       •   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

       •   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

       •   Apply  3-rd	order  analog  normalized Butterworth low-pass filter,
	   using analog transfer function format:

		   aiir=z=1.3057 0 0 0:p=1.3057 2.3892 2.1860 1:f=sf:r=d

   alimiter
       The limiter prevents  an	 input	signal	from  rising  over  a  desired
       threshold.   This  limiter  uses	 lookahead  technology to prevent your
       signal from distorting.	It means that there is a small delay after the
       signal is processed. Keep in mind that the delay	 it  produces  is  the
       attack time you set.

       The filter accepts the following options:

       level_in
	   Set input gain. Default is 1.

       level_out
	   Set output gain. Default is 1.

       limit
	   Don't let signals above this level pass the limiter. Default is 1.

       attack
	   The limiter will reach its attenuation level in this amount of time
	   in milliseconds. Default is 5 milliseconds.

       release
	   Come	 back  from  limiting  to  attenuation	1.0  in this amount of
	   milliseconds.  Default is 50 milliseconds.

       asc When gain reduction is always needed ASC takes care of releasing to
	   an average reduction level rather than reaching a reduction of 0 in
	   the release time.

       asc_level
	   Select how much the release time is affected by ASC, 0 means nearly
	   no changes in release time while 1 produces higher release times.

       level
	   Auto level output signal.  Default  is  enabled.   This  normalizes
	   audio back to 0dB if enabled.

       latency
	   Compensate  the  delay introduced by using the lookahead buffer set
	   with attack parameter. Also flush  the  valid  audio	 data  in  the
	   lookahead buffer when the stream hits EOF.

       Depending  on  picked setting it is recommended to upsample input 2x or
       4x times with aresample before applying this filter.

   allpass
       Apply a	two-pole  all-pass  filter  with  central  frequency  (in  Hz)
       frequency,  and	filter-width  width.   An  all-pass filter changes the
       audio's frequency to phase relationship without changing its  frequency
       to amplitude relationship.

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How	much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter,  by  default  all	available  are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       order, o
	   Set the filter order, can be 1 or 2. Default is 2.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change allpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change   allpass   width_type.    Syntax   for  the	command	 is  :
	   "width_type"

       width, w
	   Change allpass width.  Syntax for the command is : "width"

       mix, m
	   Change allpass mix.	Syntax for the command is : "mix"

   aloop
       Loop audio samples.

       The filter accepts the following options:

       loop
	   Set the number of loops. Setting this value to -1  will  result  in
	   infinite loops.  Default is 0.

       size
	   Set maximal number of samples. Default is 0.

       start
	   Set first sample of loop. Default is 0.

       time
	   Set	the  time of loop start in seconds.  Only used if option named
	   start is set to -1.

   amerge
       Merge two or more audio streams into a single multi-channel stream.

       The filter accepts the following options:

       inputs
	   Set the number of inputs. Default is 2.

       If the channel layouts  of  the	inputs	are  disjoint,	and  therefore
       compatible,  the	 channel  layout of the output will be set accordingly
       and the channels will be reordered as necessary. If the channel layouts
       of the inputs are not disjoint, the output will have all	 the  channels
       of  the	first input then all the channels of the second input, in that
       order, and the channel layout of the output will be the	default	 value
       corresponding to the total number of channels.

       For  example,  if  the  first input is in 2.1 (FL+FR+LF) and the second
       input is FC+BL+BR, then the output will be in 5.1, with the channels in
       the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of
       the first input, b1 is the first channel of the second input).

       On the other hand, if both input are in	stereo,	 the  output  channels
       will  be	 in  the default order: a1, a2, b1, b2, and the channel layout
       will be arbitrarily set to 4.0, which may or may not  be	 the  expected
       value.

       All inputs must have the same sample rate, and format.

       If  inputs do not have the same duration, the output will stop with the
       shortest.

       Examples

       •   Merge two mono files into a stereo stream:

		   amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

       •   Multiple merges assuming 1 video stream  and	 6  audio  streams  in
	   input.mkv:

		   ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv

   amix
       Mixes multiple audio inputs into a single output.

       Note  that  this filter only supports float samples (the amerge and pan
       audio filters support many formats). If	the  amix  input  has  integer
       samples	then  aresample	 will be automatically inserted to perform the
       conversion to float samples.

       It accepts the following parameters:

       inputs
	   The number of inputs. If unspecified, it defaults to 2.

       duration
	   How to determine the end-of-stream.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

       dropout_transition
	   The transition time, in seconds, for volume renormalization when an
	   input stream ends. The default value is 2 seconds.

       weights
	   Specify weight of each input audio stream as a sequence of  numbers
	   separated  by  a  space. If fewer weights are specified compared to
	   number of inputs, the last weight  is  assigned  to	the  remaining
	   inputs.  Default weight for each input is 1.

       normalize
	   Always  scale  inputs  instead  of only doing summation of samples.
	   Beware of heavy clipping if inputs  are  not	 normalized  prior  or
	   after  filtering  by	 this  filter  if  this option is disabled. By
	   default is enabled.

       Examples

       •   This will mix 3 input audio streams to a  single  output  with  the
	   same duration as the first input and a dropout transition time of 3
	   seconds:

		   ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

       •   This	 will  mix  one	 vocal	and  one music input audio stream to a
	   single output with the same duration	 as  the  longest  input.  The
	   music  will	have  quarter the weight as the vocals, and the inputs
	   are not normalized:

		   ffmpeg -i VOCALS -i MUSIC -filter_complex amix=inputs=2:duration=longest:dropout_transition=0:weights="1 0.25":normalize=0 OUTPUT

       Commands

       This filter supports the following commands:

       weights
       normalize
	   Syntax is same as option with same name.

   amultiply
       Multiply first audio stream with second audio stream and	 store	result
       in  output  audio  stream.  Multiplication  is done by multiplying each
       sample from first stream with  sample  at  same	position  from	second
       stream.

       With  this  element-wise	 multiplication one can create amplitude fades
       and amplitude modulations.

   anequalizer
       High-order parametric multiband equalizer for each channel.

       It accepts the following parameters:

       params
	   This option string is in format: "cchn f=cf	w=w  g=g  t=f  |  ..."
	   Each equalizer band is separated by '|'.

	   chn Set  channel  number to which equalization will be applied.  If
	       input doesn't have that channel the entry is ignored.

	   f   Set central frequency for band.	If  input  doesn't  have  that
	       frequency the entry is ignored.

	   w   Set band width in Hertz.

	   g   Set band gain in dB.

	   t   Set filter type for band, optional, can be:

	       0   Butterworth, this is default.

	       1   Chebyshev type 1.

	       2   Chebyshev type 2.

       curves
	   With	 this  option  activated  frequency response of anequalizer is
	   displayed in video stream.

       size
	   Set video stream size. Only useful if curves option is activated.

       mgain
	   Set max gain that will be displayed. Only useful if	curves	option
	   is activated.  Setting this to a reasonable value makes it possible
	   to display gain which is derived from neighbour bands which are too
	   close  to  each  other  and	thus produce higher gain when both are
	   activated.

       fscale
	   Set frequency scale	used  to  draw	frequency  response  in	 video
	   output.  Can be linear or logarithmic. Default is logarithmic.

       colors
	   Set	color for each channel curve which is going to be displayed in
	   video stream.  This is list of color names separated by space or by
	   '|'.	 Unrecognised or missing colors	 will  be  replaced  by	 white
	   color.

       Examples

       •   Lower  gain	by  10 of central frequency 200Hz and width 100 Hz for
	   first 2 channels using Chebyshev type 1 filter:

		   anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

       Commands

       This filter supports the following commands:

       change
	   Alter existing filter parameters.  Syntax for  the  commands	 is  :
	   "fN|f=freq|w=width|g=gain"

	   fN is existing filter number, starting from 0, if no such filter is
	   available  error  is	 returned.   freq set new frequency parameter.
	   width set  new  width  parameter  in	 Hertz.	  gain	set  new  gain
	   parameter in dB.

	   Full	  filter   invocation	with  asendcmd	may  look  like	 this:
	   asendcmd=c='4.0		   anequalizer			change
	   0|f=200|w=50|g=1',anequalizer=...

   anlmdn
       Reduce	broadband   noise  in  audio  samples  using  Non-Local	 Means
       algorithm.

       Each sample is adjusted by  looking  for	 other	samples	 with  similar
       contexts.  This	context	 similarity  is	 defined  by  comparing	 their
       surrounding patches of size p. Patches are searched in  an  area	 of  r
       around the sample.

       The filter accepts the following options:

       strength, s
	   Set	denoising  strength.  Allowed  range is from 0.00001 to 10000.
	   Default value is 0.00001.

       patch, p
	   Set	patch  radius  duration.  Allowed  range  is  from  1  to  100
	   milliseconds.  Default value is 2 milliseconds.

       research, r
	   Set	research  radius  duration.  Allowed  range  is	 from 2 to 300
	   milliseconds.  Default value is 6 milliseconds.

       output, o
	   Set the output mode.

	   It accepts the following values:

	   i   Pass input unchanged.

	   o   Pass noise filtered out.

	   n   Pass only noise.

	       Default value is o.

       smooth, m
	   Set smooth factor. Default value is 11. Allowed range is from 1  to
	   1000.

       Commands

       This filter supports the all above options as commands.

   anlmf, anlms
       Apply  Normalized  Least-Mean-(Squares|Fourth)  algorithm  to the first
       audio stream using the second audio stream.

       This adaptive filter is used to mimic a desired filter by  finding  the
       filter  coefficients  that relate to producing the least mean square of
       the error signal (difference  between  the  desired,  2nd  input	 audio
       stream and the actual signal, the 1st input audio stream).

       A description of the accepted options follows.

       order
	   Set filter order.

       mu  Set filter mu.

       eps Set the filter eps.

       leakage
	   Set the filter leakage.

       out_mode
	   It accepts the following values:

	   i   Pass the 1st input.

	   d   Pass the 2nd input.

	   o   Pass  difference	 between  desired,  2nd input and error signal
	       estimate.

	   n   Pass difference between	input,	1st  input  and	 error	signal
	       estimate.

	   e   Pass error signal estimated samples.

	       Default value is o.

       Examples

       •   One	of  many usages of this filter is noise reduction, input audio
	   is filtered with same samples that are delayed by fixed amount, one
	   such example for stereo audio is:

		   asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o

       Commands

       This filter supports the same commands  as  options,  excluding	option
       "order".

   anull
       Pass the audio source unchanged to the output.

   apad
       Pad the end of an audio stream with silence.

       This can be used together with ffmpeg -shortest to extend audio streams
       to the same length as the video stream.

       A description of the accepted options follows.

       packet_size
	   Set silence packet size. Default value is 4096.

       pad_len
	   Set	the  number of samples of silence to add to the end. After the
	   value is reached, the stream is terminated. This option is mutually
	   exclusive with whole_len.

       whole_len
	   Set the minimum total number of samples in the output audio stream.
	   If the value is longer than the  input  audio  length,  silence  is
	   added  to  the  end,	 until	the  value  is reached. This option is
	   mutually exclusive with pad_len.

       pad_dur
	   Specify the duration of samples of silence to  add.	See  the  Time
	   duration  section  in  the  ffmpeg-utils(1) manual for the accepted
	   syntax. Used only if set to non-negative value.

       whole_dur
	   Specify the minimum total duration in the output audio stream.  See
	   the	Time  duration	section	 in the ffmpeg-utils(1) manual for the
	   accepted syntax. Used only if set to	 non-negative  value.  If  the
	   value  is  longer  than the input audio length, silence is added to
	   the end, until the value  is	 reached.   This  option  is  mutually
	   exclusive with pad_dur

       If  neither  the	 pad_len  nor  the whole_len nor pad_dur nor whole_dur
       option is set, the filter will add silence to  the  end	of  the	 input
       stream indefinitely.

       Note  that  for ffmpeg 4.4 and earlier a zero pad_dur or whole_dur also
       caused the filter to add silence indefinitely.

       Examples

       •   Add 1024 samples of silence to the end of the input:

		   apad=pad_len=1024

       •   Make sure the audio output will contain at least 10000 samples, pad
	   the input with silence if required:

		   apad=whole_len=10000

       •   Use ffmpeg to pad the audio input with silence, so that  the	 video
	   stream  will always result the shortest and will be converted until
	   the end in the output file when using the shortest option:

		   ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

   aphaser
       Add a phasing effect to the input audio.

       A phaser filter creates series of peaks and troughs  in	the  frequency
       spectrum.   The position of the peaks and troughs are modulated so that
       they vary over time, creating a sweeping effect.

       A description of the accepted parameters follows.

       in_gain
	   Set input gain. Default is 0.4.

       out_gain
	   Set output gain. Default is 0.74

       delay
	   Set delay in milliseconds. Default is 3.0.

       decay
	   Set decay. Default is 0.4.

       speed
	   Set modulation speed in Hz. Default is 0.5.

       type
	   Set modulation type. Default is triangular.

	   It accepts the following values:

	   triangular, t
	   sinusoidal, s

   aphaseshift
       Apply phase shift to input audio samples.

       The filter accepts the following options:

       shift
	   Specify phase shift. Allowed range is from -1.0  to	1.0.   Default
	   value is 0.0.

       level
	   Set	output gain applied to final output. Allowed range is from 0.0
	   to 1.0.  Default value is 1.0.

       order
	   Set filter order used for filtering. Allowed range is from 1 to 16.
	   Default value is 8.

       Commands

       This filter supports the all above options as commands.

   apsnr
       Measure Audio Peak Signal-to-Noise Ratio.

       This filter takes two audio streams for input, and outputs first	 audio
       stream.	Results are in dB per channel at end of either input.

   apsyclip
       Apply Psychoacoustic clipper to input audio stream.

       The filter accepts the following options:

       level_in
	   Set input gain. By default it is 1. Range is [0.015625 - 64].

       level_out
	   Set output gain. By default it is 1. Range is [0.015625 - 64].

       clip
	   Set the clipping start value. Default value is 0dBFS or 1.

       diff
	   Output   only   difference	samples,  useful  to  hear  introduced
	   distortions.	 By default is disabled.

       adaptive
	   Set strength of adaptive distortion applied. Default value is  0.5.
	   Allowed range is from 0 to 1.

       iterations
	   Set	number of iterations of psychoacoustic clipper.	 Allowed range
	   is from 1 to 20. Default value is 10.

       level
	   Auto level output signal. Default  is  disabled.   This  normalizes
	   audio back to 0dBFS if enabled.

       Commands

       This filter supports the all above options as commands.

   apulsator
       Audio  pulsator	is something between an autopanner and a tremolo.  But
       it can produce funny stereo  effects  as	 well.	Pulsator  changes  the
       volume  of  the	left  and  right channel based on a LFO (low frequency
       oscillator) with different waveforms and shifted phases.	  This	filter
       have the ability to define an offset between left and right channel. An
       offset  of 0 means that both LFO shapes match each other.  The left and
       right channel are altered equally - a conventional tremolo.  An	offset
       of  50% means that the shape of the right channel is exactly shifted in
       phase (or moved backwards about half of the frequency) - pulsator  acts
       as  an  autopanner.  At	1  both	 curves	 match again. Every setting in
       between moves the phase shift gapless between all stages	 and  produces
       some  "bypassing" sounds with sine and triangle waveforms. The more you
       set the offset near 1 (starting from the 0.5)  the  faster  the	signal
       passes from the left to the right speaker.

       The filter accepts the following options:

       level_in
	   Set input gain. By default it is 1. Range is [0.015625 - 64].

       level_out
	   Set output gain. By default it is 1. Range is [0.015625 - 64].

       mode
	   Set waveform shape the LFO will use. Can be one of: sine, triangle,
	   square, sawup or sawdown. Default is sine.

       amount
	   Set	modulation.  Define how much of original signal is affected by
	   the LFO.

       offset_l
	   Set left channel offset. Default is 0. Allowed range is [0 - 1].

       offset_r
	   Set right channel offset. Default is 0.5. Allowed range is [0 - 1].

       width
	   Set pulse width. Default is 1. Allowed range is [0 - 2].

       timing
	   Set possible timing mode. Can be one of: bpm, ms or hz. Default  is
	   hz.

       bpm Set	bpm. Default is 120. Allowed range is [30 - 300]. Only used if
	   timing is set to bpm.

       ms  Set ms. Default is 500. Allowed range is [10 - 2000]. Only used  if
	   timing is set to ms.

       hz  Set	frequency  in Hz. Default is 2. Allowed range is [0.01 - 100].
	   Only used if timing is set to hz.

   aresample
       Resample the  input  audio  to  the  specified  parameters,  using  the
       libswresample  library.	If  none  are  specified  then the filter will
       automatically convert between its input and output.

       This filter is also able to stretch/squeeze the audio data to  make  it
       match  the  timestamps  or to inject silence / cut out audio to make it
       match the timestamps, do a combination of both or do neither.

       The filter accepts the  syntax  [sample_rate:]resampler_options,	 where
       sample_rate  expresses a sample rate and resampler_options is a list of
       key=value pairs, separated by ":". See the "Resampler Options"  section
       in  the	ffmpeg-resampler(1)  manual for the complete list of supported
       options.

       Examples

       •   Resample the input audio to 44100Hz:

		   aresample=44100

       •   Stretch/squeeze samples to the given timestamps, with a maximum  of
	   1000 samples per second compensation:

		   aresample=async=1000

   areverse
       Reverse an audio clip.

       Warning:	 This  filter  requires	 memory	 to buffer the entire clip, so
       trimming is suggested.

       Examples

       •   Take the first 5 seconds of a clip, and reverse it.

		   atrim=end=5,areverse

   arls
       Apply Recursive Least Squares algorithm to the first audio stream using
       the second audio stream.

       This adaptive filter is used to mimic a desired filter  by  recursively
       finding	the  filter  coefficients that relate to producing the minimal
       weighted linear	least  squares	cost  function	of  the	 error	signal
       (difference  between the desired, 2nd input audio stream and the actual
       signal, the 1st input audio stream).

       A description of the accepted options follows.

       order
	   Set the filter order.

       lambda
	   Set the forgetting factor.

       delta
	   Set the coefficient to initialize internal covariance matrix.

       out_mode
	   Set the filter output samples. It accepts the following values:

	   i   Pass the 1st input.

	   d   Pass the 2nd input.

	   o   Pass difference between desired, 2nd  input  and	 error	signal
	       estimate.

	   n   Pass  difference	 between  input,  1st  input  and error signal
	       estimate.

	   e   Pass error signal estimated samples.

	       Default value is o.

   arnndn
       Reduce noise from speech using Recurrent Neural Networks.

       This filter accepts the following options:

       model, m
	   Set train model file to load. This option is always required.

       mix Set how much to mix filtered samples into  final  output.   Allowed
	   range  is  from  -1	to 1. Default value is 1.  Negative values are
	   special, they set how much to keep  filtered	 noise	in  the	 final
	   filter  output.  Set this option to -1 to hear actual noise removed
	   from input signal.

       Commands

       This filter supports the all above options as commands.

   asdr
       Measure Audio Signal-to-Distortion Ratio.

       This filter takes two audio streams for input, and outputs first	 audio
       stream.	Results are in dB per channel at end of either input.

   asetnsamples
       Set the number of samples per each output audio frame.

       The  last  output  packet may contain a different number of samples, as
       the filter will flush all the remaining samples when  the  input	 audio
       signals its end.

       The filter accepts the following options:

       nb_out_samples, n
	   Set the number of frames per each output audio frame. The number is
	   intended  as the number of samples per each channel.	 Default value
	   is 1024.

       pad, p
	   If set to 1, the filter will pad the last audio frame with  zeroes,
	   so  that  the last frame will contain the same number of samples as
	   the previous ones. Default value is 1.

       For example, to set the number of per-frame samples to 1234 and disable
       padding for the last frame, use:

	       asetnsamples=n=1234:p=0

   asetrate
       Set the sample rate without altering the PCM data.  This will result in
       a change of speed and pitch.

       The filter accepts the following options:

       sample_rate, r
	   Set the output sample rate. Default is 44100 Hz.

   ashowinfo
       Show a line containing various information for each input audio	frame.
       The input audio is not modified.

       The  shown  line	 contains  a  sequence	of key/value pairs of the form
       key:value.

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The presentation timestamp of the input frame, in time base	units;
	   the	time  base  depends  on	 the  filter input pad, and is usually
	   1/sample_rate.

       pts_time
	   The presentation timestamp of the input frame in seconds.

       fmt The sample format.

       chlayout
	   The channel layout.

       rate
	   The sample rate for the audio frame.

       nb_samples
	   The number of samples (per channel) in the frame.

       checksum
	   The Adler-32 checksum (printed in hexadecimal) of the  audio	 data.
	   For	planar	audio,	the  data is treated as if all the planes were
	   concatenated.

       plane_checksums
	   A list of Adler-32 checksums for each data plane.

   asisdr
       Measure Audio Scaled-Invariant Signal-to-Distortion Ratio.

       This filter takes two audio streams for input, and outputs first	 audio
       stream.	Results are in dB per channel at end of either input.

   asoftclip
       Apply audio soft clipping.

       Soft  clipping  is a type of distortion effect where the amplitude of a
       signal is saturated along a smooth curve, rather than the abrupt	 shape
       of hard-clipping.

       This filter accepts the following options:

       type
	   Set type of soft-clipping.

	   It accepts the following values:

	   hard
	   tanh
	   atan
	   cubic
	   exp
	   alg
	   quintic
	   sin
	   erf
       threshold
	   Set threshold from where to start clipping. Default value is 0dB or
	   1.

       output
	   Set gain applied to output. Default value is 0dB or 1.

       param
	   Set additional parameter which controls sigmoid function.

       oversample
	   Set oversampling factor.

       Commands

       This filter supports the all above options as commands.

   aspectralstats
       Display	frequency  domain  statistical	information  about  the	 audio
       channels.  Statistics are calculated and stored as  metadata  for  each
       audio channel and for each audio frame.

       It accepts the following option:

       win_size
	   Set	the  window length in samples. Default value is 2048.  Allowed
	   range is from 32 to 65536.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann, hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set window overlap. Allowed range is from 0 to 1. Default value  is
	   0.5.

       measure
	   Select  the parameters which are measured. The metadata keys can be
	   used as flags, default is  all  which  measures  everything.	  none
	   disables all measurement.

       A list of each metadata key follows:

       mean
       variance
       centroid
       spread
       skewness
       kurtosis
       entropy
       flatness
       crest
       flux
       slope
       decrease
       rolloff

   asr
       Automatic Speech Recognition

       This  filter  uses  PocketSphinx	 for  speech  recognition.  To	enable
       compilation  of	this  filter,  you  need  to  configure	 FFmpeg	  with
       "--enable-pocketsphinx".

       It accepts the following options:

       rate
	   Set	sampling rate of input audio. Defaults is 16000.  This need to
	   match speech models, otherwise one will get poor results.

       hmm Set dictionary containing acoustic model files.

       dict
	   Set pronunciation dictionary.

       lm  Set language model file.

       lmctl
	   Set language model set.

       lmname
	   Set which language model to use.

       logfn
	   Set output for log messages.

       The  filter  exports  recognized	  speech   as	the   frame   metadata
       "lavfi.asr.text".

   astats
       Display	time  domain statistical information about the audio channels.
       Statistics are calculated and displayed for  each  audio	 channel  and,
       where applicable, an overall figure is also given.

       It accepts the following option:

       length
	   Short  window  length  in  seconds,	used  for  peak and trough RMS
	   measurement.	 Default is 0.05 (50 milliseconds). Allowed  range  is
	   "[0 - 10]".

       metadata
	   Set	metadata  injection.  All  the metadata keys are prefixed with
	   "lavfi.astats.X", where "X" is channel number starting  from	 1  or
	   string "Overall". Default is disabled.

	   Available   keys  for  each	channel	 are:  Bit_depth  Crest_factor
	   DC_offset   Dynamic_range   Entropy	 Flat_factor	Max_difference
	   Max_level   Mean_difference	Min_difference	Min_level  Noise_floor
	   Noise_floor_count Number_of_Infs Number_of_NaNs Number_of_denormals
	   Peak_count  Abs_Peak_count	Peak_level   RMS_difference   RMS_peak
	   RMS_trough Zero_crossings Zero_crossings_rate

	   and	 for   "Overall":   Bit_depth  DC_offset  Entropy  Flat_factor
	   Max_difference Max_level Mean_difference  Min_difference  Min_level
	   Noise_floor	  Noise_floor_count    Number_of_Infs	Number_of_NaNs
	   Number_of_denormals	Number_of_samples  Peak_count	Abs_Peak_count
	   Peak_level RMS_difference RMS_level RMS_peak RMS_trough

	   For	example,  a  full key looks like "lavfi.astats.1.DC_offset" or
	   "lavfi.astats.Overall.Peak_count".

	   Read below for the description of the keys.

       reset
	   Set the number of frames over which cumulative stats are calculated
	   before being reset. Default is disabled.

       measure_perchannel
	   Select the parameters which are measured per channel. The  metadata
	   keys	  can  be  used	 as  flags,  default  is  all  which  measures
	   everything.	none disables all per channel measurement.

       measure_overall
	   Select the parameters which are measured overall. The metadata keys
	   can be used as flags, default is  all  which	 measures  everything.
	   none disables all overall measurement.

       A description of the measure keys follow:

       none
	   no measures

       all all measures

       Bit_depth
	   overall  bit	 depth	of  audio,  i.e.  number of bits used for each
	   sample

       Crest_factor
	   standard ratio of peak to RMS level (note: not in dB)

       DC_offset
	   mean amplitude displacement from zero

       Dynamic_range
	   measured dynamic range of audio in dB

       Entropy
	   entropy measured across whole audio, entropy of value near  1.0  is
	   typically measured for white noise

       Flat_factor
	   flatness  (i.e.  consecutive	 samples  with	the same value) of the
	   signal at its peak levels (i.e. either Min_level or Max_level)

       Max_difference
	   maximal difference between two consecutive samples

       Max_level
	   maximal sample level

       Mean_difference
	   mean difference between two consecutive samples, i.e.  the  average
	   of each difference between two consecutive samples

       Min_difference
	   minimal difference between two consecutive samples

       Min_level
	   minimal sample level

       Noise_floor
	   minimum local peak measured in dBFS over a short window

       Noise_floor_count
	   number  of  occasions  (not	the number of samples) that the signal
	   attained Noise floor

       Number_of_Infs
	   number of samples with an infinite value

       Number_of_NaNs
	   number of samples with a NaN (not a number) value

       Number_of_denormals
	   number of samples with a subnormal value

       Number_of_samples
	   number of samples

       Peak_count
	   number of occasions (not the number of  samples)  that  the	signal
	   attained either Min_level or Max_level

       Abs_Peak_count
	   number of occasions that the absolute samples taken from the signal
	   attained max absolute value of Min_level and Max_level

       Peak_level
	   standard peak level measured in dBFS

       RMS_difference
	   Root Mean Square difference between two consecutive samples

       RMS_level
	   standard RMS level measured in dBFS

       RMS_peak
       RMS_trough
	   peak	 and trough values for RMS level measured over a short window,
	   measured in dBFS.

       Zero crossings
	   number of points where the waveform crosses the zero level axis

       Zero crossings rate
	   rate of Zero crossings and number of audio samples

   asubboost
       Boost subwoofer frequencies.

       The filter accepts the following options:

       dry Set dry gain, how much of original signal is kept. Allowed range is
	   from 0 to 1.	 Default value is 1.0.

       wet Set wet gain, how much of filtered signal is kept. Allowed range is
	   from 0 to 1.	 Default value is 1.0.

       boost
	   Set max boost factor. Allowed range is from 1 to 12. Default	 value
	   is 2.

       decay
	   Set	delay  line  decay  gain  value. Allowed range is from 0 to 1.
	   Default value is 0.0.

       feedback
	   Set delay line feedback gain value. Allowed range is from 0	to  1.
	   Default value is 0.9.

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is 50 to 900.  Default
	   value is 100.

       slope
	   Set	slope  amount for cutoff frequency. Allowed range is 0.0001 to
	   1.  Default value is 0.5.

       delay
	   Set delay. Allowed range is from 1 to 100.  Default value is 20.

       channels
	   Set the channels to process. Default value is all available.

       Commands

       This filter supports the all above options as commands.

   asubcut
       Cut subwoofer frequencies.

       This filter allows to  set  custom,  steeper  roll  off	than  highpass
       filter,	and  thus is able to more attenuate frequency content in stop-
       band.

       The filter accepts the following options:

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is 2 to 200.   Default
	   value is 20.

       order
	   Set filter order. Available values are from 3 to 20.	 Default value
	   is 10.

       level
	   Set	input  gain level. Allowed range is from 0 to 1. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asupercut
       Cut super frequencies.

       The filter accepts the following options:

       cutoff
	   Set cutoff frequency in Hertz. Allowed range is  20000  to  192000.
	   Default value is 20000.

       order
	   Set filter order. Available values are from 3 to 20.	 Default value
	   is 10.

       level
	   Set	input  gain level. Allowed range is from 0 to 1. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asuperpass
       Apply high order Butterworth band-pass filter.

       The filter accepts the following options:

       centerf
	   Set center frequency in  Hertz.  Allowed  range  is	2  to  999999.
	   Default value is 1000.

       order
	   Set filter order. Available values are from 4 to 20.	 Default value
	   is 4.

       qfactor
	   Set	Q-factor.  Allowed range is from 0.01 to 100. Default value is
	   1.

       level
	   Set input gain level. Allowed range is from 0 to 2.	Default	 value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   asuperstop
       Apply high order Butterworth band-stop filter.

       The filter accepts the following options:

       centerf
	   Set	center	frequency  in  Hertz.  Allowed	range  is 2 to 999999.
	   Default value is 1000.

       order
	   Set filter order. Available values are from 4 to 20.	 Default value
	   is 4.

       qfactor
	   Set Q-factor. Allowed range is from 0.01 to 100. Default  value  is
	   1.

       level
	   Set	input  gain level. Allowed range is from 0 to 2. Default value
	   is 1.

       Commands

       This filter supports the all above options as commands.

   atempo
       Adjust audio tempo.

       The filter accepts exactly one  parameter,  the	audio  tempo.  If  not
       specified  then the filter will assume nominal 1.0 tempo. Tempo must be
       in the [0.5, 100.0] range.

       Note that tempo greater than 2 will skip some samples rather than blend
       them in.	 If for any reason this is a concern it is always possible  to
       daisy-chain  several instances of atempo to achieve the desired product
       tempo.

       Examples

       •   Slow down audio to 80% tempo:

		   atempo=0.8

       •   To speed up audio to 300% tempo:

		   atempo=3

       •   To speed up audio  to  300%	tempo  by  daisy-chaining  two	atempo
	   instances:

		   atempo=sqrt(3),atempo=sqrt(3)

       Commands

       This filter supports the following commands:

       tempo
	   Change  filter  tempo  scale	 factor.   Syntax for the command is :
	   "tempo"

   atilt
       Apply spectral tilt filter to audio stream.

       This filter apply  any  spectral	 roll-off  slope  over	any  specified
       frequency band.

       The filter accepts the following options:

       freq
	   Set central frequency of tilt in Hz. Default is 10000 Hz.

       slope
	   Set slope direction of tilt. Default is 0. Allowed range is from -1
	   to 1.

       width
	   Set	width  of  tilt. Default is 1000. Allowed range is from 100 to
	   10000.

       order
	   Set order of tilt filter.

       level
	   Set input volume level. Allowed range is from 0 to 4.  Defalt is 1.

       Commands

       This filter supports the all above options as commands.

   atrim
       Trim the input so that the output contains one  continuous  subpart  of
       the input.

       It accepts the following parameters:

       start
	   Timestamp  (in  seconds)  of the start of the section to keep. I.e.
	   the audio sample with the timestamp start will be the first	sample
	   in the output.

       end Specify  time  of the first audio sample that will be dropped, i.e.
	   the audio sample immediately preceding the one with	the  timestamp
	   end will be the last sample in the output.

       start_pts
	   Same	 as  start,  except  this  option  sets the start timestamp in
	   samples instead of seconds.

       end_pts
	   Same as end, except this option sets the end timestamp  in  samples
	   instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_sample
	   The number of the first sample that should be output.

       end_sample
	   The number of the first sample that should be dropped.

       start, end, and duration are expressed as time duration specifications;
       see the Time duration section in the ffmpeg-utils(1) manual.

       Note  that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the  _sample  options	simply
       count  the  samples  that pass through the filter. So start/end_pts and
       start/end_sample will give different results when  the  timestamps  are
       wrong, inexact or do not start at zero. Also note that this filter does
       not  modify  the	 timestamps. If you wish to have the output timestamps
       start at zero, insert the asetpts filter after the atrim filter.

       If multiple start or end options are  set,  this	 filter	 tries	to  be
       greedy  and  keep  all samples that match at least one of the specified
       constraints. To keep only the part that matches all the constraints  at
       once, chain multiple atrim filters.

       The  defaults are such that all the input is kept. So it is possible to
       set e.g.	 just the end values to keep everything before	the  specified
       time.

       Examples:

       •   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -af atrim=60:120

       •   Keep only the first 1000 samples:

		   ffmpeg -i INPUT -af atrim=end_sample=1000

   axcorrelate
       Calculate normalized windowed cross-correlation between two input audio
       streams.

       Resulted samples are always between -1 and 1 inclusive.	If result is 1
       it  means  two  input  samples  are  highly correlated in that selected
       segment.	 Result 0 means they are not correlated at all.	 If result  is
       -1 it means two input samples are out of phase, which means they cancel
       each other.

       The filter accepts the following options:

       size
	   Set	size  of  segment  over which cross-correlation is calculated.
	   Default is 256. Allowed range is from 2 to 131072.

       algo
	   Set algorithm for cross-correlation. Can be	"slow"	or  "fast"  or
	   "best".  Default is "best". Fast algorithm assumes mean values over
	   any	given  segment	are  always  zero  and	thus  need  much  less
	   calculations to make.  This is generally not true, but is valid for
	   typical audio streams.

       Examples

       •   Calculate correlation between channels in stereo audio stream:

		   ffmpeg -i stereo.wav -af channelsplit,axcorrelate=size=1024:algo=fast correlation.wav

   bandpass
       Apply a two-pole Butterworth band-pass filter  with  central  frequency
       frequency,  and (3dB-point) band-width width.  The csg option selects a
       constant skirt gain (peak gain = Q) instead of  the  default:  constant
       0dB  peak  gain.	  The  filter  roll  off  at  6dB per octave (20dB per
       decade).

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency. Default is 3000.

       csg Constant skirt gain if set to 1. Defaults to 0.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       channels, c
	   Specify  which  channels  to	 filter,  by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If  this  value  is
	   set	to  high  enough  value	 (higher  than impulse response length
	   truncated when reaches near	zero  values)  filtering  will	become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note	 that  filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change bandpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change  bandpass  width_type.   Syntax  for	the   command	is   :
	   "width_type"

       width, w
	   Change bandpass width.  Syntax for the command is : "width"

       mix, m
	   Change bandpass mix.	 Syntax for the command is : "mix"

   bandreject
       Apply  a two-pole Butterworth band-reject filter with central frequency
       frequency, and (3dB-point) band-width width.  The filter	 roll  off  at
       6dB per octave (20dB per decade).

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency. Default is 3000.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       mix, m
	   How	much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter,  by  default  all	available  are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set	block  size  used for reverse IIR processing. If this value is
	   set to high enough  value  (higher  than  impulse  response	length
	   truncated  when  reaches  near  zero	 values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples  when  set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change   bandreject	 frequency.   Syntax  for  the	command	 is  :
	   "frequency"

       width_type, t
	   Change  bandreject  width_type.   Syntax  for  the  command	is   :
	   "width_type"

       width, w
	   Change bandreject width.  Syntax for the command is : "width"

       mix, m
	   Change bandreject mix.  Syntax for the command is : "mix"

   bass, lowshelf
       Boost or cut the bass (lower) frequencies of the audio using a two-pole
       shelving	 filter	 with a response similar to that of a standard hi-fi's
       tone-controls. This is also known as shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
	   Give the gain at 0 Hz. Its useful range is about -20 (for  a	 large
	   cut)	 to  +20 (for a large boost).  Beware of clipping when using a
	   positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend  or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 100 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How	much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter,  by  default  all	available  are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set	block  size  used for reverse IIR processing. If this value is
	   set to high enough  value  (higher  than  impulse  response	length
	   truncated  when  reaches  near  zero	 values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples  when  set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change bass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change bass width_type.  Syntax for the command is : "width_type"

       width, w
	   Change bass width.  Syntax for the command is : "width"

       gain, g
	   Change bass gain.  Syntax for the command is : "gain"

       mix, m
	   Change bass mix.  Syntax for the command is : "mix"

   biquad
       Apply  a	 biquad IIR filter with the given coefficients.	 Where b0, b1,
       b2 and a0, a1,  a2  are	the  numerator	and  denominator  coefficients
       respectively.   and  channels,  c  specify which channels to filter, by
       default all available are filtered.

       Commands

       This filter supports the following commands:

       a0
       a1
       a2
       b0
       b1
       b2  Change biquad parameter.  Syntax for the command is : "value"

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       channels, c
	   Specify  which  channels  to	 filter,  by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If  this  value  is
	   set	to  high  enough  value	 (higher  than impulse response length
	   truncated when reaches near	zero  values)  filtering  will	become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note	 that  filter delay will be exactly this many samples when set
	   to non-zero value.

   bs2b
       Bauer stereo  to	 binaural  transformation,  which  improves  headphone
       listening of stereo audio records.

       To  enable compilation of this filter you need to configure FFmpeg with
       "--enable-libbs2b".

       It accepts the following parameters:

       profile
	   Pre-defined crossfeed level.

	   default
	       Default level (fcut=700, feed=50).

	   cmoy
	       Chu Moy circuit (fcut=700, feed=60).

	   jmeier
	       Jan Meier circuit (fcut=650, feed=95).

       fcut
	   Cut frequency (in Hz).

       feed
	   Feed level (in Hz).

   channelmap
       Remap input channels to new locations.

       It accepts the following parameters:

       map Map channels from input to output. The argument is a	 '|'-separated
	   list	  of   mappings,   each	 in  the  "in_channel-out_channel"  or
	   in_channel form. in_channel can be either the  name	of  the	 input
	   channel  (e.g. FL for front left) or its index in the input channel
	   layout.  out_channel is the name of the output channel or its index
	   in the output channel layout. If out_channel is not given  then  it
	   is  implicitly  an  index, starting with zero and increasing by one
	   for each mapping.

       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

       •   For example, assuming a 5.1+downmix input MOV file,

		   ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

	   will create an output WAV file tagged as stereo  from  the  downmix
	   channels of the input.

       •   To fix a 5.1 WAV improperly encoded in AAC's native channel order

		   ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

   channelsplit
       Split  each  channel  from an input audio stream into a separate output
       stream.

       It accepts the following parameters:

       channel_layout
	   The channel layout of the input stream. The default is "stereo".

       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

       •   For example, assuming a stereo input MP3 file,

		   ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

	   will create an output Matroska file with  two  audio	 streams,  one
	   containing only the left channel and the other the right channel.

       •   Split a 5.1 WAV file into per-channel files:

		   ffmpeg -i in.wav -filter_complex
		   'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
		   -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
		   front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
		   side_right.wav

       •   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

       •   A single delay:

		   chorus=0.7:0.9:55:0.4:0.25:2

       •   Two delays:

		   chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

       •   Fuller sounding chorus with three delays:

		   chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

   compand
       Compress or expand the audio's dynamic range.

       It accepts the following parameters:

       attacks
       decays
	   A list of  times  in	 seconds  for  each  channel  over  which  the
	   instantaneous  level	 of  the input signal is averaged to determine
	   its volume. attacks refers to increase of volume and decays	refers
	   to  decrease	 of  volume.  For  most	 situations,  the  attack time
	   (response to the audio getting louder) should be shorter  than  the
	   decay  time, because the human ear is more sensitive to sudden loud
	   audio than sudden soft audio. A typical value  for  attack  is  0.3
	   seconds and a typical value for decay is 0.8 seconds.  If specified
	   number  of  attacks	& decays is lower than number of channels, the
	   last set attack/decay will be used for all remaining channels.

       points
	   A list of  points  for  the	transfer  function,  specified	in  dB
	   relative  to the maximum possible signal amplitude. Each key points
	   list	  must	 be    defined	  using	   the	  following    syntax:
	   "x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

	   The	input  values  must  be	 in  strictly increasing order but the
	   transfer function does not have to  be  monotonically  rising.  The
	   point  "0/0"	 is  assumed  but  may be overridden (by "0/out-dBn").
	   Typical values for the transfer function are "-70/-70|-60/-20|1/0".

       soft-knee
	   Set the curve radius in dB for all joints. It defaults to 0.01.

       gain
	   Set the additional gain in dB to be applied at all  points  on  the
	   transfer  function.	This allows for easy adjustment of the overall
	   gain.  It defaults to 0.

       volume
	   Set an initial volume, in dB, to be assumed for each	 channel  when
	   filtering  starts.  This permits the user to supply a nominal level
	   initially, so that, for example, a very large gain is  not  applied
	   to  initial	signal	levels	before	the  companding	 has  begun to
	   operate. A typical value for audio which is initially quiet is  -90
	   dB. It defaults to 0.

       delay
	   Set	a  delay, in seconds. The input audio is analyzed immediately,
	   but audio is delayed before	being  fed  to	the  volume  adjuster.
	   Specifying  a  delay	 approximately equal to the attack/decay times
	   allows the filter to effectively operate in predictive rather  than
	   reactive mode. It defaults to 0.

       Examples

       •   Make music with both quiet and loud passages suitable for listening
	   to in a noisy environment:

		   compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

	   Another example for audio with whisper and explosion parts:

		   compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

       •   A  noise  gate  for	when  the  noise  is at a lower level than the
	   signal:

		   compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

       •   Here is another noise gate, this time for when the noise  is	 at  a
	   higher  level  than the signal (making it, in some ways, similar to
	   squelch):

		   compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

       •   2:1 compression starting at -6dB:

		   compand=points=-80/-80|-6/-6|0/-3.8|20/3.5

       •   2:1 compression starting at -9dB:

		   compand=points=-80/-80|-9/-9|0/-5.3|20/2.9

       •   2:1 compression starting at -12dB:

		   compand=points=-80/-80|-12/-12|0/-6.8|20/1.9

       •   2:1 compression starting at -18dB:

		   compand=points=-80/-80|-18/-18|0/-9.8|20/0.7

       •   3:1 compression starting at -15dB:

		   compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2

       •   Compressor/Gate:

		   compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6

       •   Expander:

		   compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3

       •   Hard limiter at -6dB:

		   compand=attacks=0:points=-80/-80|-6/-6|20/-6

       •   Hard limiter at -12dB:

		   compand=attacks=0:points=-80/-80|-12/-12|20/-12

       •   Hard noise gate at -35 dB:

		   compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20

       •   Soft limiter:

		   compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8

   compensationdelay
       Compensation Delay Line is a metric based delay to compensate differing
       positions of microphones or speakers.

       For example, you have recorded guitar with two  microphones  placed  in
       different locations. Because the front of sound wave has fixed speed in
       normal  conditions,  the phasing of microphones can vary and depends on
       their location and interposition. The best sound mix  can  be  achieved
       when  these  microphones	 are  in  phase	 (synchronized).  Note	that a
       distance of ~30 cm between microphones makes one microphone capture the
       signal in antiphase to the other microphone. That makes the  final  mix
       sound  moody.   This  filter  helps to solve phasing problems by adding
       different delays to each microphone track and make them synchronized.

       The best result can be reached when you take  one  track	 as  base  and
       synchronize   other   tracks   one  by  one  with  it.	Remember  that
       synchronization/delay tolerance depends on sample  rate,	 too.	Higher
       sample rates will give more tolerance.

       The filter accepts the following parameters:

       mm  Set	millimeters  distance.	This is compensation distance for fine
	   tuning.  Default is 0.

       cm  Set cm distance.  This  is  compensation  distance  for  tightening
	   distance setup.  Default is 0.

       m   Set	meters	distance.  This	 is  compensation  distance  for  hard
	   distance setup.  Default is 0.

       dry Set dry amount. Amount of unprocessed (dry) signal.	Default is 0.

       wet Set wet amount. Amount of processed (wet) signal.  Default is 1.

       temp
	   Set temperature in degrees Celsius. This is the temperature of  the
	   environment.	 Default is 20.

       Commands

       This filter supports the all above options as commands.

   crossfeed
       Apply headphone crossfeed filter.

       Crossfeed  is  the  process  of blending the left and right channels of
       stereo audio recording.	It is mainly used  to  reduce  extreme	stereo
       separation of low frequencies.

       The intent is to produce more speaker like sound to the listener.

       The filter accepts the following options:

       strength
	   Set	strength of crossfeed. Default is 0.2. Allowed range is from 0
	   to 1.  This sets gain of low shelf filter for side part  of	stereo
	   image.   Default is -6dB. Max allowed is -30db when strength is set
	   to 1.

       range
	   Set soundstage wideness. Default is 0.5. Allowed range is from 0 to
	   1.  This sets cut off frequency of low shelf filter. Default is cut
	   off near 1550 Hz. With range set to 1 cut off frequency is  set  to
	   2100 Hz.

       slope
	   Set curve slope of low shelf filter. Default is 0.5.	 Allowed range
	   is from 0.01 to 1.

       level_in
	   Set input gain. Default is 0.9.

       level_out
	   Set output gain. Default is 1.

       block_size
	   Set	block  size  used for reverse IIR processing. If this value is
	   set to high enough  value  (higher  than  impulse  response	length
	   truncated  when  reaches  near  zero	 values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples  when  set
	   to non-zero value.

       Commands

       This filter supports the all above options as commands.

   crystalizer
       Simple algorithm for audio noise sharpening.

       This filter linearly increases differences betweeen each audio sample.

       The filter accepts the following options:

       i   Sets	 the  intensity	 of  effect  (default:	2.0). Must be in range
	   between -10.0 to 0 (unchanged sound) to 10.0 (maximum effect).   To
	   inverse filtering use negative value.

       c   Enable clipping. By default is enabled.

       Commands

       This filter supports the all above options as commands.

   dcshift
       Apply a DC shift to the audio.

       This  can be useful to remove a DC offset (caused perhaps by a hardware
       problem in the recording chain) from the audio.	The  effect  of	 a  DC
       offset  is  reduced headroom and hence volume. The astats filter can be
       used to determine if a signal has a DC offset.

       shift
	   Set the DC shift, allowed range is [-1, 1]. It indicates the amount
	   to shift the audio.

       limitergain
	   Optional. It should have a value much less than  1  (e.g.  0.05  or
	   0.02) and is used to prevent clipping.

   deesser
       Apply de-essing to the audio samples.

       i   Set	intensity for triggering de-essing. Allowed range is from 0 to
	   1.  Default is 0.

       m   Set amount of ducking on treble part of  sound.  Allowed  range  is
	   from 0 to 1.	 Default is 0.5.

       f   How	much  of  original  frequency  content to keep when de-essing.
	   Allowed range is from 0 to 1.  Default is 0.5.

       s   Set the output mode.

	   It accepts the following values:

	   i   Pass input unchanged.

	   o   Pass ess filtered out.

	   e   Pass only ess.

	       Default value is o.

   dialoguenhance
       Enhance dialogue in stereo audio.

       This filter accepts stereo input and produce  surround  (3.0)  channels
       output.	 The  newly produced front center channel have enhanced speech
       dialogue originally available in both  stereo  channels.	  This	filter
       outputs front left and front right channels same as available in stereo
       input.

       The filter accepts the following options:

       original
	   Set	the  original  center  factor  to keep in front center channel
	   output.  Allowed range is from 0 to 1. Default value is 1.

       enhance
	   Set the dialogue enhance factor to  put  in	front  center  channel
	   output.  Allowed range is from 0 to 3. Default value is 1.

       voice
	   Set	the  voice  detection  factor.	Allowed range is from 2 to 32.
	   Default value is 2.

       Commands

       This filter supports the all above options as commands.

   drmeter
       Measure audio dynamic range.

       DR values of 14 and higher is found in very dynamic material. DR	 of  8
       to  13  is  found in transition material. And anything less that 8 have
       very poor dynamics and is very compressed.

       The filter accepts the following options:

       length
	   Set window length in seconds used to split audio into  segments  of
	   equal length.  Default is 3 seconds.

   dynaudnorm
       Dynamic Audio Normalizer.

       This  filter  applies  a	 certain  amount of gain to the input audio in
       order to bring its peak magnitude to a  target  level  (e.g.  0	dBFS).
       However,	 in  contrast  to  more "simple" normalization algorithms, the
       Dynamic Audio Normalizer *dynamically* re-adjusts the  gain  factor  to
       the  input  audio.   This allows for applying extra gain to the "quiet"
       sections of the audio while avoiding distortions or clipping the "loud"
       sections. In other words: The Dynamic Audio Normalizer will "even  out"
       the  volume of quiet and loud sections, in the sense that the volume of
       each section is brought to the same target level. Note,	however,  that
       the  Dynamic  Audio  Normalizer	achieves  this goal *without* applying
       "dynamic range compressing". It will retain 100% of the	dynamic	 range
       *within* each section of the audio file.

       framelen, f
	   Set	the  frame  length  in	milliseconds. In range from 10 to 8000
	   milliseconds.  Default is  500  milliseconds.   The	Dynamic	 Audio
	   Normalizer  processes  the input audio in small chunks, referred to
	   as frames. This is  required,  because  a  peak  magnitude  has  no
	   meaning  for	 just  a  single  sample  value.  Instead,  we need to
	   determine the peak magnitude for a contiguous  sequence  of	sample
	   values.  While  a  "standard"  normalizer would simply use the peak
	   magnitude of	 the  complete	file,  the  Dynamic  Audio  Normalizer
	   determines  the  peak  magnitude  individually  for each frame. The
	   length of a frame is specified in  milliseconds.  By	 default,  the
	   Dynamic  Audio  Normalizer uses a frame length of 500 milliseconds,
	   which has been found to give good results with  most	 files.	  Note
	   that	 the  exact  frame  length,  in	 number	 of  samples,  will be
	   determined  automatically,  based  on  the  sampling	 rate  of  the
	   individual input audio file.

       gausssize, g
	   Set	the  Gaussian filter window size. In range from 3 to 301, must
	   be  odd  number.  Default  is  31.	Probably  the  most  important
	   parameter  of  the Dynamic Audio Normalizer is the "window size" of
	   the	Gaussian  smoothing  filter.  The  filter's  window  size   is
	   specified  in  frames,  centered  around the current frame. For the
	   sake of simplicity, this must be an odd number.  Consequently,  the
	   default  value  of 31 takes into account the current frame, as well
	   as the 15 preceding frames and the 15 subsequent  frames.  Using  a
	   larger  window  results  in a stronger smoothing effect and thus in
	   less gain variation, i.e. slower gain adaptation. Conversely, using
	   a smaller window results in a weaker smoothing effect and  thus  in
	   more	 gain variation, i.e. faster gain adaptation.  In other words,
	   the more you increase  this	value,	the  more  the	Dynamic	 Audio
	   Normalizer  will  behave like a "traditional" normalization filter.
	   On the contrary, the more you decrease this	value,	the  more  the
	   Dynamic   Audio   Normalizer	 will  behave  like  a	dynamic	 range
	   compressor.

       peak, p
	   Set the target peak value. This specifies the  highest  permissible
	   magnitude  level  for  the normalized audio input. This filter will
	   try to approach the target peak magnitude as closely	 as  possible,
	   but	at the same time it also makes sure that the normalized signal
	   will never exceed the peak magnitude.  A frame's maximum local gain
	   factor is imposed  directly	by  the	 target	 peak  magnitude.  The
	   default value is 0.95 and thus leaves a headroom of 5%*.  It is not
	   recommended to go above this value.

       maxgain, m
	   Set the maximum gain factor. In range from 1.0 to 100.0. Default is
	   10.0.  The Dynamic Audio Normalizer determines the maximum possible
	   (local)  gain  factor  for  each input frame, i.e. the maximum gain
	   factor that does not result in clipping or distortion. The  maximum
	   gain	 factor is determined by the frame's highest magnitude sample.
	   However, the	 Dynamic  Audio	 Normalizer  additionally  bounds  the
	   frame's  maximum  gain  factor  by a predetermined (global) maximum
	   gain factor. This is done in order to avoid excessive gain  factors
	   in  "silent"	 or almost silent frames. By default, the maximum gain
	   factor is 10.0,  For	 most  inputs  the  default  value  should  be
	   sufficient  and  it	usually	 is  not  recommended to increase this
	   value. Though, for input  with  an  extremely  low  overall	volume
	   level, it may be necessary to allow even higher gain factors. Note,
	   however,  that the Dynamic Audio Normalizer does not simply apply a
	   "hard"  threshold  (i.e.  cut  off  values  above  the  threshold).
	   Instead,  a "sigmoid" threshold function will be applied. This way,
	   the gain factors will smoothly approach the	threshold  value,  but
	   never exceed that value.

       targetrms, r
	   Set	the  target  RMS.  In  range from 0.0 to 1.0. Default is 0.0 -
	   disabled.  By default, the Dynamic Audio Normalizer performs "peak"
	   normalization.  This means that the maximum local gain  factor  for
	   each	 frame	is  defined  (only)  by	 the frame's highest magnitude
	   sample. This way, the samples can be amplified as much as  possible
	   without  exceeding the maximum signal level, i.e. without clipping.
	   Optionally, however, the Dynamic Audio  Normalizer  can  also  take
	   into	 account  the  frame's	root  mean square, abbreviated RMS. In
	   electrical engineering, the RMS is commonly used to	determine  the
	   power of a time-varying signal. It is therefore considered that the
	   RMS is a better approximation of the "perceived loudness" than just
	   looking  at the signal's peak magnitude. Consequently, by adjusting
	   all frames to a constant RMS value, a uniform "perceived  loudness"
	   can	be  established.  If  a target RMS value has been specified, a
	   frame's local gain factor is	 defined  as  the  factor  that	 would
	   result  in exactly that RMS value.  Note, however, that the maximum
	   local gain factor  is  still	 restricted  by	 the  frame's  highest
	   magnitude sample, in order to prevent clipping.

       coupling, n
	   Enable  channels  coupling. By default is enabled.  By default, the
	   Dynamic Audio Normalizer will amplify  all  channels	 by  the  same
	   amount.  This  means	 the  same  gain factor will be applied to all
	   channels, i.e.  the maximum possible gain factor is	determined  by
	   the	"loudest" channel.  However, in some recordings, it may happen
	   that the volume of the  different  channels	is  uneven,  e.g.  one
	   channel may be "quieter" than the other one(s).  In this case, this
	   option  can	be used to disable the channel coupling. This way, the
	   gain factor will be	determined  independently  for	each  channel,
	   depending  only  on	the  individual	 channel's  highest  magnitude
	   sample. This allows for harmonizing the  volume  of	the  different
	   channels.

       correctdc, c
	   Enable DC bias correction. By default is disabled.  An audio signal
	   (in	the  time  domain)  is	a  sequence  of sample values.	In the
	   Dynamic Audio Normalizer these sample values are represented in the
	   -1.0 to  1.0	 range,	 regardless  of	 the  original	input  format.
	   Normally,  the  audio  signal,  or  "waveform",  should be centered
	   around the zero point.  That means if we calculate the  mean	 value
	   of  all  samples  in	 a file, or in a single frame, then the result
	   should be 0.0 or at least very close to that	 value.	 If,  however,
	   there  is  a	 significant  deviation of the mean value from 0.0, in
	   either positive or negative direction, this is referred to as a  DC
	   bias	 or  DC	 offset.  Since	 a DC bias is clearly undesirable, the
	   Dynamic Audio Normalizer  provides  optional	 DC  bias  correction.
	   With	 DC bias correction enabled, the Dynamic Audio Normalizer will
	   determine the mean value, or "DC correction" offset, of each	 input
	   frame and subtract that value from all of the frame's sample values
	   which ensures those samples are centered around 0.0 again. Also, in
	   order  to  avoid  "gaps" at the frame boundaries, the DC correction
	   offset values will be interpolated  smoothly	 between  neighbouring
	   frames.

       altboundary, b
	   Enable  alternative	boundary  mode.	 By  default is disabled.  The
	   Dynamic Audio Normalizer takes into account a certain neighbourhood
	   around each frame. This includes the preceding frames  as  well  as
	   the	subsequent frames. However, for the "boundary" frames, located
	   at the very beginning and at the very end of the  audio  file,  not
	   all neighbouring frames are available. In particular, for the first
	   few	frames	in the audio file, the preceding frames are not known.
	   And, similarly, for the last few frames  in	the  audio  file,  the
	   subsequent  frames  are  not known. Thus, the question arises which
	   gain factors should be  assumed  for	 the  missing  frames  in  the
	   "boundary"  region.	The  Dynamic  Audio  Normalizer implements two
	   modes to deal  with	this  situation.  The  default	boundary  mode
	   assumes  a  gain  factor  of	 exactly  1.0  for the missing frames,
	   resulting in a smooth "fade in" and "fade out" at the beginning and
	   at the end of the input, respectively.

       compress, s
	   Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.
	   By  default,	 the  Dynamic  Audio   Normalizer   does   not	 apply
	   "traditional" compression. This means that signal peaks will not be
	   pruned and thus the full dynamic range will be retained within each
	   local  neighbourhood. However, in some cases it may be desirable to
	   combine the Dynamic Audio Normalizer's normalization algorithm with
	   a more "traditional" compression.  For this	purpose,  the  Dynamic
	   Audio  Normalizer  provides	an optional compression (thresholding)
	   function. If (and only if) the compression feature is enabled,  all
	   input frames will be processed by a soft knee thresholding function
	   prior   to  the  actual  normalization  process.  Put  simply,  the
	   thresholding function is going to prune all samples whose magnitude
	   exceeds a certain threshold	value.	 However,  the	Dynamic	 Audio
	   Normalizer  does not simply apply a fixed threshold value. Instead,
	   the threshold value will be adjusted for each individual frame.  In
	   general, smaller parameters result  in  stronger  compression,  and
	   vice	 versa.	 Values below 3.0 are not recommended, because audible
	   distortion may appear.

       threshold, t
	   Set	the  target  threshold	value.	This  specifies	  the	lowest
	   permissible	magnitude  level  for  the  audio  input which will be
	   normalized.	If input frame volume is above this value  frame  will
	   be  normalized.   Otherwise frame may not be normalized at all. The
	   default value is set to 0, which means all  input  frames  will  be
	   normalized.	 This  option is mostly useful if digital noise is not
	   wanted to be amplified.

       channels, h
	   Specify which channels to filter, by default all available channels
	   are filtered.

       overlap, o
	   Specify overlap  for	 frames.  If  set  to  0  (default)  no	 frame
	   overlapping	is  done.   Using  >0  and  <1	values	will make less
	   conservative gain adjustments, like when framelen option is set  to
	   smaller value, if framelen option value is compensated for non-zero
	   overlap then gain adjustments will be smoother across time compared
	   to zero overlap case.

       curve, v
	   Specify the peak mapping curve expression which is going to be used
	   when	 calculating gain applied to frames. The max output frame gain
	   will still be limited by other  options  mentioned  previously  for
	   this filter.

	   The expression can contain the following constants:

	   ch  current channel number

	   sn  current sample number

	   nb_channels
	       number of channels

	   t   timestamp expressed in seconds

	   sr  sample rate

	   p   current frame peak value

       Commands

       This filter supports the all above options as commands.

   earwax
       Make audio easier to listen to on headphones.

       This  filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio
       so that when listened to on headphones the stereo image is  moved  from
       inside  your  head (standard for headphones) to outside and in front of
       the listener (standard for speakers).

       Ported from SoX.

   equalizer
       Apply a two-pole peaking equalisation (EQ) filter.  With	 this  filter,
       the signal-level at and around a selected frequency can be increased or
       decreased,  whilst (unlike bandpass and bandreject filters) that at all
       other frequencies is unchanged.

       In order to produce complex equalisation curves,	 this  filter  can  be
       given several times, each with a different central frequency.

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency in Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify the band-width of a filter in width_type units.

       gain, g
	   Set	the  required  gain  or attenuation in dB.  Beware of clipping
	   when using a positive gain.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       channels, c
	   Specify  which  channels  to	 filter,  by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If  this  value  is
	   set	to  high  enough  value	 (higher  than impulse response length
	   truncated when reaches near	zero  values)  filtering  will	become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note	 that  filter delay will be exactly this many samples when set
	   to non-zero value.

       Examples

       •   Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

		   equalizer=f=1000:t=h:width=200:g=-10

       •   Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB  at  100  Hz
	   with Q 2:

		   equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

       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

       •   lowpass at 1000 Hz:

		   firequalizer=gain='if(lt(f,1000), 0, -INF)'

       •   lowpass at 1000 Hz with gain_entry:

		   firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'

       •   custom equalization:

		   firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'

       •   higher delay with zero phase to compensate delay:

		   firequalizer=delay=0.1:fixed=on:zero_phase=on

       •   lowpass on left channel, highpass on right channel:

		   firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
		   :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on

   flanger
       Apply a flanging effect to the audio.

       The filter accepts the following options:

       delay
	   Set	base  delay in milliseconds. Range from 0 to 30. Default value
	   is 0.

       depth
	   Set added sweep delay in milliseconds. Range from 0 to 10.  Default
	   value is 2.

       regen
	   Set	percentage  regeneration (delayed signal feedback). Range from
	   -95 to 95.  Default value is 0.

       width
	   Set percentage of delayed signal mixed with original. Range from  0
	   to 100.  Default value is 71.

       speed
	   Set	sweeps per second (Hz). Range from 0.1 to 10. Default value is
	   0.5.

       shape
	   Set swept wave shape, can be	 triangular  or	 sinusoidal.   Default
	   value is sinusoidal.

       phase
	   Set	swept wave percentage-shift for multi channel. Range from 0 to
	   100.	 Default value is 25.

       interp
	   Set delay-line interpolation,  linear  or  quadratic.   Default  is
	   linear.

   haas
       Apply Haas effect to audio.

       Note  that  this	 makes most sense to apply on mono signals.  With this
       filter  applied	to  mono  signals  it  give  some  directionality  and
       stretches its stereo image.

       The filter accepts the following options:

       level_in
	   Set input level. By default is 1, or 0dB

       level_out
	   Set output level. By default is 1, or 0dB.

       side_gain
	   Set gain applied to side part of signal. By default is 1.

       middle_source
	   Set kind of middle source. Can be one of the following:

	   left
	       Pick left channel.

	   right
	       Pick right channel.

	   mid Pick middle part signal of stereo image.

	   side
	       Pick side part signal of stereo image.

       middle_phase
	   Change middle phase. By default is disabled.

       left_delay
	   Set left channel delay. By default is 2.05 milliseconds.

       left_balance
	   Set left channel balance. By default is -1.

       left_gain
	   Set left channel gain. By default is 1.

       left_phase
	   Change left phase. By default is disabled.

       right_delay
	   Set right channel delay. By defaults is 2.12 milliseconds.

       right_balance
	   Set right channel balance. By default is 1.

       right_gain
	   Set right channel gain. By default is 1.

       right_phase
	   Change right phase. By default is enabled.

   hdcd
       Decodes	High  Definition  Compatible Digital (HDCD) data. A 16-bit PCM
       stream with embedded HDCD codes is expanded into a 20-bit PCM stream.

       The filter supports the	Peak  Extend  and  Low-level  Gain  Adjustment
       features of HDCD, and detects the Transient Filter flag.

	       ffmpeg -i HDCD16.flac -af hdcd OUT24.flac

       When  using  the	 filter with wav, note the default encoding for wav is
       16-bit, so the resulting	 20-bit	 stream	 will  be  truncated  back  to
       16-bit.	Use  something	like -acodec pcm_s24le after the filter to get
       24-bit PCM output.

	       ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
	       ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav

       The filter accepts the following options:

       disable_autoconvert
	   Disable any automatic format conversion or resampling in the filter
	   graph.

       process_stereo
	   Process the stereo channels together. If target_gain does not match
	   between channels, consider  it  invalid  and	 use  the  last	 valid
	   target_gain.

       cdt_ms
	   Set the code detect timer period in ms.

       force_pe
	   Always extend peaks above -3dBFS even if PE isn't signaled.

       analyze_mode
	   Replace  audio with a solid tone and adjust the amplitude to signal
	   some specific aspect of the decoding process. The output  file  can
	   be  loaded  in  an  audio  editor  alongside	 the  original	to aid
	   analysis.

	   "analyze_mode=pe:force_pe=true" can be  used	 to  see  all  samples
	   above the PE level.

	   Modes are:

	   0, off
	       Disabled

	   1, lle
	       Gain adjustment level at each sample

	   2, pe
	       Samples where peak extend occurs

	   3, cdt
	       Samples where the code detect timer is active

	   4, tgm
	       Samples where the target gain does not match between channels

   headphone
       Apply   head-related  transfer  functions  (HRTFs)  to  create  virtual
       loudspeakers around the user for	 binaural  listening  via  headphones.
       The  HRIRs  are	provided  via additional streams, for each channel one
       stereo input stream is needed.

       The filter accepts the following options:

       map Set mapping of input streams for convolution.  The  argument	 is  a
	   '|'-separated  list	of channel names in order as they are given as
	   additional stream inputs for filter.	 This also specify  number  of
	   input streams. Number of input streams must be not less than number
	   of channels in first stream plus one.

       gain
	   Set gain applied to audio. Value is in dB. Default is 0.

       type
	   Set	processing type. Can be time or freq. time is processing audio
	   in time  domain  which  is  slow.   freq  is	 processing  audio  in
	   frequency domain which is fast.  Default is freq.

       lfe Set custom gain for LFE channels. Value is in dB. Default is 0.

       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

       •   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

       •   Full example using wav files as coefficients	 with  amovie  filters
	   for 7.1 downmix, but now in multich hrir format.

		   ffmpeg -i input.wav -filter_complex "amovie=minp.wav[hrirs];[0:a][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
		   output.wav

   highpass
       Apply  a	 high-pass filter with 3dB point frequency.  The filter can be
       either single-pole, or double-pole (the default).  The filter roll  off
       at 6dB per pole per octave (20dB per pole per decade).

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz. Default is 3000.

       poles, p
	   Set number of poles. Default is 2.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify  the	 band-width  of a filter in width_type units.  Applies
	   only to double-pole filter.	The default  is	 0.707q	 and  gives  a
	   Butterworth response.

       mix, m
	   How	much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter,  by  default  all	available  are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set	block  size  used for reverse IIR processing. If this value is
	   set to high enough  value  (higher  than  impulse  response	length
	   truncated  when  reaches  near  zero	 values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples  when  set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change highpass frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change   highpass   width_type.    Syntax  for  the	command	 is  :
	   "width_type"

       width, w
	   Change highpass width.  Syntax for the command is : "width"

       mix, m
	   Change highpass mix.	 Syntax for the command is : "mix"

   join
       Join multiple input streams into one multi-channel stream.

       It accepts the following parameters:

       inputs
	   The number of input streams. It defaults to 2.

       channel_layout
	   The desired output channel layout. It defaults to stereo.

       map Map channels from inputs to output. The argument is a '|'-separated
	   list of mappings, each  in  the  "input_idx.in_channel-out_channel"
	   form.   input_idx  is  the  0-based	index  of  the	input  stream.
	   in_channel can be either the name of the input channel (e.g. FL for
	   front left) or its index in the specified input stream. out_channel
	   is the name of the output channel.

       The filter will attempt	to  guess  the	mappings  when	they  are  not
       specified  explicitly.  It  does	 so  by first trying to find an unused
       matching input channel and if that fails	 it  picks  the	 first	unused
       input channel.

       Join 3 inputs (with properly set channel layouts):

	       ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

       Build a 5.1 output from 6 single-channel streams:

	       ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
	       'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
	       out

   ladspa
       Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

       To  enable compilation of this filter you need to configure FFmpeg with
       "--enable-ladspa".

       file, f
	   Specifies the name  of  LADSPA  plugin  library  to	load.  If  the
	   environment	variable  LADSPA_PATH is defined, the LADSPA plugin is
	   searched in each one of the	directories  specified	by  the	 colon
	   separated  list  in	LADSPA_PATH,  otherwise in the standard LADSPA
	   paths,   which    are    in	  this	  order:    HOME/.ladspa/lib/,
	   /usr/local/lib/ladspa/, /usr/lib/ladspa/.

       plugin, p
	   Specifies  the  plugin  within  the library. Some libraries contain
	   only one plugin, but others contain many of them. If	 this  is  not
	   set	filter	will  list  all available plugins within the specified
	   library.

       controls, c
	   Set the '|' separated list of  controls  which  are	zero  or  more
	   floating  point  values  that  determine the behavior of the loaded
	   plugin (for example delay, threshold or gain).  Controls need to be
	   defined	   using	 the	     following	       syntax:
	   c0=value0|c1=value1|c2=value2|..., where valuei is the value set on
	   the i-th control.  Alternatively they can be also defined using the
	   following  syntax:  value0|value1|value2|...,  where	 valuei is the
	   value set on the i-th control.  If controls is set to  "help",  all
	   available controls and their valid ranges are printed.

       sample_rate, s
	   Specify the sample rate, default to 44100. Only used if plugin have
	   zero inputs.

       nb_samples, n
	   Set	the  number  of	 samples  per  channel	per each output frame,
	   default is 1024. Only used if plugin have zero inputs.

       duration, d
	   Set the minimum  duration  of  the  sourced	audio.	See  the  Time
	   duration  section  in  the  ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater  than  the
	   specified duration, as the generated audio is always cut at the end
	   of  a  complete frame.  If not specified, or the expressed duration
	   is negative, the audio is supposed to be generated  forever.	  Only
	   used if plugin have zero inputs.

       latency, l
	   Enable  latency compensation, by default is disabled.  Only used if
	   plugin have inputs.

       Examples

       •   List all available  plugins	within	amp  (LADSPA  example  plugin)
	   library:

		   ladspa=file=amp

       •   List	 all available controls and their valid ranges for "vcf_notch"
	   plugin from "VCF" library:

		   ladspa=f=vcf:p=vcf_notch:c=help

       •   Simulate low quality audio equipment using "Computer Music Toolkit"
	   (CMT) plugin library:

		   ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

       •   Add reverberation to	 the  audio  using  TAP-plugins	 (Tom's	 Audio
	   Processing plugins):

		   ladspa=file=tap_reverb:tap_reverb

       •   Generate white noise, with 0.2 amplitude:

		   ladspa=file=cmt:noise_source_white:c=c0=.2

       •   Generate 20 bpm clicks using plugin "C* Click - Metronome" from the
	   "C* Audio Plugin Suite" (CAPS) library:

		   ladspa=file=caps:Click:c=c1=20'

       •   Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

		   ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

       •   Increase  volume  by	 20dB  using fast lookahead limiter from Steve
	   Harris "SWH Plugins" collection:

		   ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

       •   Attenuate low frequencies using Multiband EQ from Steve Harris "SWH
	   Plugins" collection:

		   ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0

       •   Reduce stereo image using "Narrower"	 from  the  "C*	 Audio	Plugin
	   Suite" (CAPS) library:

		   ladspa=caps:Narrower

       •   Another  white  noise,  now	using  "C*  Audio Plugin Suite" (CAPS)
	   library:

		   ladspa=caps:White:.2

       •   Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:

		   ladspa=caps:Fractal:c=c1=1

       •   Dynamic volume normalization using "VLevel" plugin:

		   ladspa=vlevel-ladspa:vlevel_mono

       Commands

       This filter supports the following commands:

       cN  Modify the N-th control value.

	   If the specified value is not valid, it is ignored and prior one is
	   kept.

   loudnorm
       EBU R128 loudness  normalization.  Includes  both  dynamic  and	linear
       normalization modes.  Support for both single pass (livestreams, files)
       and  double pass (files) modes.	This algorithm can target IL, LRA, and
       maximum true peak. In dynamic mode, to accurately  detect  true	peaks,
       the audio stream will be upsampled to 192 kHz.  Use the "-ar" option or
       "aresample" filter to explicitly set an output sample rate.

       The filter accepts the following options:

       I, i
	   Set	integrated  loudness  target.	Range is -70.0 - -5.0. Default
	   value is -24.0.

       LRA, lra
	   Set loudness range target.  Range is 1.0 - 50.0. Default  value  is
	   7.0.

       TP, tp
	   Set	maximum	 true  peak.   Range  is -9.0 - +0.0. Default value is
	   -2.0.

       measured_I, measured_i
	   Measured IL of input file.  Range is -99.0 - +0.0.

       measured_LRA, measured_lra
	   Measured LRA of input file.	Range is  0.0 - 99.0.

       measured_TP, measured_tp
	   Measured true peak of input file.  Range is	-99.0 - +99.0.

       measured_thresh
	   Measured threshold of input file.  Range is -99.0 - +0.0.

       offset
	   Set offset gain. Gain is  applied  before  the  true-peak  limiter.
	   Range is  -99.0 - +99.0. Default is +0.0.

       linear
	   Normalize  by  linearly  scaling  the  source audio.	 "measured_I",
	   "measured_LRA", "measured_TP", and "measured_thresh"	 must  all  be
	   specified.  Target  LRA  shouldn't be lower than source LRA and the
	   change in integrated loudness shouldn't result in a true peak which
	   exceeds the target TP. If  any  of  these  conditions  aren't  met,
	   normalization  mode	will revert to dynamic.	 Options are "true" or
	   "false". Default is "true".

       dual_mono
	   Treat mono input files as "dual-mono". If a mono file  is  intended
	   for	playback  on a stereo system, its EBU R128 measurement will be
	   perceptually	 incorrect.   If  set  to  "true",  this  option  will
	   compensate  for  this  effect.   Multi-channel  input files are not
	   affected by this option.  Options are true  or  false.  Default  is
	   false.

       print_format
	   Set	print  format  for  stats. Options are summary, json, or none.
	   Default value is none.

   lowpass
       Apply a low-pass filter with 3dB point frequency.  The  filter  can  be
       either  single-pole  or double-pole (the default).  The filter roll off
       at 6dB per pole per octave (20dB per pole per decade).

       The filter accepts the following options:

       frequency, f
	   Set frequency in Hz. Default is 500.

       poles, p
	   Set number of poles. Default is 2.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Specify the band-width of a filter in  width_type  units.   Applies
	   only	 to  double-pole  filter.   The	 default is 0.707q and gives a
	   Butterworth response.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       channels, c
	   Specify  which  channels  to	 filter,  by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If  this  value  is
	   set	to  high  enough  value	 (higher  than impulse response length
	   truncated when reaches near	zero  values)  filtering  will	become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note	 that  filter delay will be exactly this many samples when set
	   to non-zero value.

       Examples

       •   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

       •   Apply bass enhancer plugin from Calf:

		   lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2

       •   Apply vinyl plugin from Calf:

		   lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5

       •   Apply bit crusher plugin from ArtyFX:

		   lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3

       Commands

       This filter supports  all  options  that	 are  exported	by  plugin  as
       commands.

   mcompand
       Multiband Compress or expand the audio's dynamic range.

       The  input  audio  is divided into bands using 4th order Linkwitz-Riley
       IIRs.  This is akin to the crossover of a loudspeaker, and  results  in
       flat frequency response when absent compander action.

       It accepts the following parameters:

       args
	   This	 option	 syntax	 is:  attack,decay,[attack,decay..]  soft-knee
	   points  crossover_frequency	[delay	[initial_volume	  [gain]]]   |
	   attack,decay	 ...   For  explanation	 of each item refer to compand
	   filter documentation.

   pan
       Mix channels with specific gain levels. The filter accepts  the	output
       channel layout followed by a set of channels definitions.

       This  filter  is	 also designed to efficiently remap the channels of an
       audio stream.

       The filter accepts parameters of the form: "l|outdef|outdef|..."

       l   output channel layout or number of channels

       outdef
	   output      channel	    specification,	of	the	 form:
	   "out_name=[gain*]in_name[(+-)[gain*]in_name...]"

       out_name
	   output channel to define, either a channel name (FL, FR, etc.) or a
	   channel number (c0, c1, etc.)

       gain
	   multiplicative  coefficient	for  the channel, 1 leaving the volume
	   unchanged

       in_name
	   input channel to use, see out_name for details; it is not  possible
	   to mix named and numbered input channels

       If  the	`='  in	 a  channel specification is replaced by `<', then the
       gains for that specification will be renormalized so that the total  is
       1, thus avoiding clipping noise.

       Mixing examples

       For  example,  if  you want to down-mix from stereo to mono, but with a
       bigger factor for the left channel:

	       pan=1c|c0=0.9*c0+0.1*c1

       A customized down-mix to stereo that works automatically for 3-, 4-, 5-
       and 7-channels surround:

	       pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

       Note that ffmpeg integrates a default down-mix (and up-mix) system that
       should be preferred (see "-ac" option) unless you  have	very  specific
       needs.

       Remapping examples

       The channel remapping will be effective if, and only if:

       *<gain coefficients are zeroes or ones,>
       *<only one input per channel output,>

       If  all these conditions are satisfied, the filter will notify the user
       ("Pure channel mapping detected"), and use an  optimized	 and  lossless
       method to do the remapping.

       For example, if you have a 5.1 source and want a stereo audio stream by
       dropping the extra channels:

	       pan="stereo| c0=FL | c1=FR"

       Given  the  same source, you can also switch front left and front right
       channels and keep the input channel layout:

	       pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

       If the input is a stereo audio stream, you  can	mute  the  front  left
       channel (and still keep the stereo channel layout) with:

	       pan="stereo|c1=c1"

       Still  with a stereo audio stream input, you can copy the right channel
       in both front left and right:

	       pan="stereo| c0=FR | c1=FR"

   replaygain
       ReplayGain scanner filter. This filter takes  an	 audio	stream	as  an
       input  and  outputs  it	unchanged.   At	 end  of filtering it displays
       "track_gain" and "track_peak".

       The filter accepts the following exported read-only options:

       track_gain
	   Exported track gain in dB at end of stream.

       track_peak
	   Exported track peak at end of stream.

   resample
       Convert the audio sample format, sample rate and channel layout. It  is
       not meant to be used directly.

   rubberband
       Apply time-stretching and pitch-shifting with librubberband.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-librubberband".

       The filter accepts the following options:

       tempo
	   Set tempo scale factor.

       pitch
	   Set pitch scale factor.

       transients
	   Set transients detector.  Possible values are:

	   crisp
	   mixed
	   smooth
       detector
	   Set detector.  Possible values are:

	   compound
	   percussive
	   soft
       phase
	   Set phase.  Possible values are:

	   laminar
	   independent
       window
	   Set processing window size.	Possible values are:

	   standard
	   short
	   long
       smoothing
	   Set smoothing.  Possible values are:

	   off
	   on
       formant
	   Enable  formant  preservation when shift pitching.  Possible values
	   are:

	   shifted
	   preserved
       pitchq
	   Set pitch quality.  Possible values are:

	   quality
	   speed
	   consistency
       channels
	   Set channels.  Possible values are:

	   apart
	   together

       Commands

       This filter supports the following commands:

       tempo
	   Change filter tempo scale factor.  Syntax  for  the	command	 is  :
	   "tempo"

       pitch
	   Change  filter  pitch  scale	 factor.   Syntax for the command is :
	   "pitch"

   sidechaincompress
       This filter acts like normal compressor but has the ability to compress
       detected signal using second input signal.  It needs two input  streams
       and  returns  one  output stream.  First input stream will be processed
       depending on second stream signal.  The filtered	 signal	 then  can  be
       filtered	 with other filters in later stages of processing. See pan and
       amerge filter.

       The filter accepts the following options:

       level_in
	   Set input gain. Default is 1. Range is between 0.015625 and 64.

       mode
	   Set mode of compressor operation. Can be  "upward"  or  "downward".
	   Default is "downward".

       threshold
	   If a signal of second stream raises above this level it will affect
	   the	gain reduction of first stream.	 By default is 0.125. Range is
	   between 0.00097563 and 1.

       ratio
	   Set a ratio about which the signal is reduced. 1:2  means  that  if
	   the level raised 4dB above the threshold, it will be only 2dB above
	   after the reduction.	 Default is 2. Range is between 1 and 20.

       attack
	   Amount  of  milliseconds the signal has to rise above the threshold
	   before gain reduction starts. Default is 20. Range is between  0.01
	   and 2000.

       release
	   Amount  of  milliseconds the signal has to fall below the threshold
	   before reduction is decreased  again.  Default  is  250.  Range  is
	   between 0.01 and 9000.

       makeup
	   Set	the  amount  by	 how  much  signal  will  be  amplified	 after
	   processing.	Default is 1. Range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter  gain  reduction
	   more softly.	 Default is 2.82843. Range is between 1 and 8.

       link
	   Choose  if  the  "average" level between all channels of side-chain
	   stream  or  the  louder("maximum")  channel	of  side-chain	stream
	   affects the reduction. Default is "average".

       detection
	   Should the exact signal be taken in case of "peak" or an RMS one in
	   case of "rms". Default is "rms" which is mainly smoother.

       level_sc
	   Set sidechain gain. Default is 1. Range is between 0.015625 and 64.

       mix How	much  to use compressed signal in output. Default is 1.	 Range
	   is between 0 and 1.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Full ffmpeg	example	 taking	 2  audio  inputs,  1st	 input	to  be
	   compressed	depending  on  the  signal  of	2nd  input  and	 later
	   compressed signal to be merged with 2nd input:

		   ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

   sidechaingate
       A sidechain gate acts like a normal (wideband) gate but has the ability
       to filter the detected signal before sending it to the  gain  reduction
       stage.	Normally  a  gate uses the full range signal to detect a level
       above the threshold.  For example: If you  cut  all  lower  frequencies
       from  your  sidechain  signal the gate will decrease the volume of your
       track only if not enough highs appear. With this technique you are able
       to reduce the resonation of a natural  drum  or	remove	"rumbling"  of
       muted  strokes  from  a	heavily	 distorted guitar.  It needs two input
       streams and returns one output stream.	First  input  stream  will  be
       processed depending on second stream signal.

       The filter accepts the following options:

       level_in
	   Set	input  level before filtering.	Default is 1. Allowed range is
	   from 0.015625 to 64.

       mode
	   Set the mode of operation. Can be "upward" or "downward".   Default
	   is "downward". If set to "upward" mode, higher parts of signal will
	   be	amplified,   expanding	dynamic	 range	in  upward  direction.
	   Otherwise, in case of "downward" lower  parts  of  signal  will  be
	   reduced.

       range
	   Set	the  level  of	gain  reduction	 when  the signal is below the
	   threshold.  Default is 0.06125. Allowed  range  is  from  0	to  1.
	   Setting  this  to 0 disables reduction and then filter behaves like
	   expander.

       threshold
	   If a signal rises above this level the gain reduction is  released.
	   Default is 0.125. Allowed range is from 0 to 1.

       ratio
	   Set	a  ratio  about	 which	the  signal is reduced.	 Default is 2.
	   Allowed range is from 1 to 9000.

       attack
	   Amount of milliseconds the signal has to rise above	the  threshold
	   before  gain	 reduction stops.  Default is 20 milliseconds. Allowed
	   range is from 0.01 to 9000.

       release
	   Amount of milliseconds the signal has to fall below	the  threshold
	   before   the	  reduction   is   increased  again.  Default  is  250
	   milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
	   Set amount of amplification of signal after processing.  Default is
	   1. Allowed range is from 1 to 64.

       knee
	   Curve the sharp knee around the threshold to enter  gain  reduction
	   more softly.	 Default is 2.828427125. Allowed range is from 1 to 8.

       detection
	   Choose if exact signal should be taken for detection or an RMS like
	   one.	 Default is rms. Can be peak or rms.

       link
	   Choose  if  the  average  level  between all channels or the louder
	   channel affects the reduction.  Default is average. Can be  average
	   or maximum.

       level_sc
	   Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

       Commands

       This filter supports the all above options as commands.

   silencedetect
       Detect silence in an audio stream.

       This  filter logs a message when it detects that the input audio volume
       is less or equal to a noise tolerance value for a duration  greater  or
       equal to the minimum detected noise duration.

       The   printed   times  and  duration  are  expressed  in	 seconds.  The
       "lavfi.silence_start" or "lavfi.silence_start.X" metadata key is set on
       the first  frame	 whose	timestamp  equals  or  exceeds	the  detection
       duration	 and  it  contains  the	 timestamp  of	the first frame of the
       silence.

       The   "lavfi.silence_duration"	or   "lavfi.silence_duration.X"	   and
       "lavfi.silence_end"  or	"lavfi.silence_end.X" metadata keys are set on
       the first frame after the silence. If mono is enabled, and each channel
       is evaluated separately, the ".X"  suffixed  keys  are  used,  and  "X"
       corresponds to the channel number.

       The filter accepts the following options:

       noise, n
	   Set	noise  tolerance.  Can	be  specified  in  dB (in case "dB" is
	   appended to the specified value) or	amplitude  ratio.  Default  is
	   -60dB, or 0.001.

       duration, d
	   Set silence duration until notification (default is 2 seconds). See
	   the	Time  duration	section	 in the ffmpeg-utils(1) manual for the
	   accepted syntax.

       mono, m
	   Process each channel separately, instead of combined. By default is
	   disabled.

       Examples

       •   Detect 5 seconds of silence with -50dB noise tolerance:

		   silencedetect=n=-50dB:d=5

       •   Complete example with ffmpeg to detect silence  with	 0.0001	 noise
	   tolerance in silence.mp3:

		   ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

   silenceremove
       Remove silence from the beginning, middle or end of the audio.

       The filter accepts the following options:

       start_periods
	   This	 value	is  used  to  indicate	if  audio should be trimmed at
	   beginning of the audio. A value of zero indicates no silence should
	   be trimmed from the beginning. When specifying a non-zero value, it
	   trims audio up until it finds non-silence. Normally, when  trimming
	   silence  from beginning of audio the start_periods will be 1 but it
	   can be increased to higher values to trim all audio up to  specific
	   count of non-silence periods.  Default value is 0.

       start_duration
	   Specify the amount of time that non-silence must be detected before
	   it  stops  trimming	audio.	By  increasing the duration, bursts of
	   noises can be treated as silence and trimmed off. Default value  is
	   0.

       start_threshold
	   This	 indicates what sample value should be treated as silence. For
	   digital audio, a value of 0 may be fine but for audio recorded from
	   analog,  you	 may  wish  to	increase  the  value  to  account  for
	   background noise.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude ratio. Default value is 0.

       start_silence
	   Specify  max	 duration  of  silence	at beginning that will be kept
	   after trimming. Default is  0,  which  is  equal  to	 trimming  all
	   samples detected as silence.

       start_mode
	   Specify  mode of detection of silence end at start of multi-channel
	   audio.  Can be any or all. Default is any.  With  any,  any	sample
	   from	 any  channel that is detected as non-silence will trigger end
	   of silence trimming at start of audio stream.  With	all,  only  if
	   every  sample  from	every  channel is detected as non-silence will
	   trigger end of silence trimming at start of audio  stream,  limited
	   usage.

       stop_periods
	   Set	the  count  for	 trimming  silence from the end of audio. When
	   specifying  a  positive  value,  it	trims  audio  after  it	 finds
	   specified  silence  period.	To remove silence from the middle of a
	   file, specify a stop_periods that is negative. This value  is  then
	   treated  as	a  positive  value  and is used to indicate the effect
	   should restart processing as specified by stop_periods,  making  it
	   suitable  for  removing  periods  of	 silence  in the middle of the
	   audio.  Default value is 0.

       stop_duration
	   Specify a duration of silence that must exist before audio  is  not
	   copied  any	more. By specifying a higher duration, silence that is
	   wanted can be left in the audio.  Default value is 0.

       stop_threshold
	   This is the same as start_threshold but for trimming	 silence  from
	   the end of audio.  Can be specified in dB (in case "dB" is appended
	   to the specified value) or amplitude ratio. Default value is 0.

       stop_silence
	   Specify  max	 duration  of  silence	at end that will be kept after
	   trimming. Default is 0, which is  equal  to	trimming  all  samples
	   detected as silence.

       stop_mode
	   Specify  mode  of  detection of silence start after start of multi-
	   channel audio.  Can be any or all. Default is all.  With  any,  any
	   sample  from	 any  channel that is detected as silence will trigger
	   start of silence trimming after  start  of  audio  stream,  limited
	   usage.   With  all,	only  if  every	 sample	 from every channel is
	   detected as silence will trigger start of  silence  trimming	 after
	   start of audio stream.

       detection
	   Set how is silence detected.

	   avg Mean of absolute values of samples in moving window.

	   rms Root  squared  mean  of	absolute  values  of samples in moving
	       window.

	   peak
	       Maximum of absolute values of samples in moving window.

	   median
	       Median of absolute values of samples in moving window.

	   ptp Absolute of max peak to	min  peak  difference  of  samples  in
	       moving window.

	   dev Standard deviation of values of samples in moving window.

	   Default value is "rms".

       window
	   Set	duration in number of seconds used to calculate size of window
	   in  number  of  samples  for	 detecting  silence.  Using   0	  will
	   effectively	disable	 any  windowing and use only single sample per
	   channel for silence detection.  In that case it may	be  needed  to
	   also	 set  start_silence and/or stop_silence to nonzero values with
	   also	 start_duration	 and/or	 stop_duration	to   nonzero   values.
	   Default value is 0.02. Allowed range is from 0 to 10.

       timestamp
	   Set processing mode of every audio frame output timestamp.

	   write
	       Full timestamps rewrite, keep only the start time for the first
	       output frame.

	   copy
	       Non-dropped  frames are left with same timestamp as input audio
	       frame.

	   Defaults value is "write".

       Examples

       •   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

       •   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

       •   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

       •   Trim every 2nd encountered silence period  from  beginning  to  end
	   where  there is more than 1 second of silence per silence period in
	   audio:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB

       •   Similar as above, but keep maximum of 0.5 seconds of	 silence  from
	   each trimmed period:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5

       •   Similar  as	above, but keep maximum of 1.5 seconds of silence from
	   start of audio:

		   silenceremove=stop_periods=-2:stop_duration=1:stop_threshold=-90dB:stop_silence=0.5:start_periods=1:start_duration=1:start_silence=1.5:stop_threshold=-90dB

       Commands

       This filter supports some above options as commands.

   sofalizer
       SOFAlizer  uses	head-related  transfer	functions  (HRTFs)  to	create
       virtual	loudspeakers  around  the  user	 for  binaural	listening  via
       headphones (audio formats up to 9 channels supported).  The  HRTFs  are
       stored	in   SOFA  files  (see	<http://www.sofacoustics.org/>	for  a
       database).  SOFAlizer is developed at the Acoustics Research  Institute
       (ARI) of the Austrian Academy of Sciences.

       To  enable compilation of this filter you need to configure FFmpeg with
       "--enable-libmysofa".

       The filter accepts the following options:

       sofa
	   Set the SOFA file used for rendering.

       gain
	   Set gain applied to audio. Value is in dB. Default is 0.

       rotation
	   Set rotation of virtual loudspeakers in deg. Default is 0.

       elevation
	   Set elevation of virtual speakers in deg. Default is 0.

       radius
	   Set distance in meters between loudspeakers and the	listener  with
	   near-field HRTFs. Default is 1.

       type
	   Set	processing type. Can be time or freq. time is processing audio
	   in time  domain  which  is  slow.   freq  is	 processing  audio  in
	   frequency domain which is fast.  Default is freq.

       speakers
	   Set	custom	positions  of  virtual	loudspeakers.  Syntax for this
	   option is:  <CH>  <AZIM>  <ELEV>[|<CH>  <AZIM>  <ELEV>|...].	  Each
	   virtual  loudspeaker is described with short channel name following
	   with azimuth and elevation in degrees.   Each  virtual  loudspeaker
	   description	is  separated  by  '|'.	 For example to override front
	   left and front right channel positions use: 'speakers=FL  45	 15|FR
	   345 15'.  Descriptions with unrecognised channel names are ignored.

       lfegain
	   Set custom gain for LFE channels. Value is in dB. Default is 0.

       framesize
	   Set	custom	frame  size  in	 number	 of  samples. Default is 1024.
	   Allowed range is from 1024 to 96000. Only used if  option  type  is
	   set to freq.

       normalize
	   Should  all IRs be normalized upon importing SOFA file.  By default
	   is enabled.

       interpolate
	   Should nearest IRs be  interpolated	with  neighbor	IRs  if	 exact
	   position does not match. By default is disabled.

       minphase
	   Minphase all IRs upon loading of SOFA file. By default is disabled.

       anglestep
	   Set	neighbor search angle step. Only used if option interpolate is
	   enabled.

       radstep
	   Set neighbor search radius step. Only used if option interpolate is
	   enabled.

       Examples

       •   Using ClubFritz6 sofa file:

		   sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1

       •   Using ClubFritz12 sofa file and bigger radius with small rotation:

		   sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5

       •   Similar as above but with custom speaker positions for front	 left,
	   front right, back left and back right and also with custom gain:

		   "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"

   speechnorm
       Speech Normalizer.

       This  filter  expands  or  compresses  each half-cycle of audio samples
       (local set of samples all above or  all	below  zero  and  between  two
       nearest	zero crossings) depending on threshold value, so audio reaches
       target peak value under conditions controlled by below options.

       The filter accepts the following options:

       peak, p
	   Set the expansion target peak value.	 This  specifies  the  highest
	   allowed  absolute  amplitude	 level for the normalized audio input.
	   Default value is 0.95. Allowed range is from 0.0 to 1.0.

       expansion, e
	   Set the maximum expansion factor. Allowed  range  is	 from  1.0  to
	   50.0.  Default  value  is  2.0.  This option controls maximum local
	   half-cycle of samples expansion. The	 maximum  expansion  would  be
	   such	 that  local peak value reaches target peak value but never to
	   surpass it and that ratio between new and previous peak value  does
	   not surpass this option value.

       compression, c
	   Set	the  maximum  compression factor. Allowed range is from 1.0 to
	   50.0. Default value is 2.0.	This  option  controls	maximum	 local
	   half-cycle  of  samples  compression.  This	option is used only if
	   threshold option is set to value greater than  0.0,	then  in  such
	   cases  when	local  peak is lower or same as value set by threshold
	   all samples belonging to that peak's half-cycle will be  compressed
	   by current compression factor.

       threshold, t
	   Set	the  threshold	value.	Default value is 0.0. Allowed range is
	   from 0.0 to	1.0.   This  option  specifies	which  half-cycles  of
	   samples  will  be compressed and which will be expanded.  Any half-
	   cycle samples with their local peak value below  or	same  as  this
	   option  value  will	be  compressed	by current compression factor,
	   otherwise, if greater than threshold value they  will  be  expanded
	   with	 expansion factor so that it could reach peak target value but
	   never surpass it.

       raise, r
	   Set the expansion raising amount per each  half-cycle  of  samples.
	   Default  value  is  0.001.	Allowed range is from 0.0 to 1.0. This
	   controls how fast expansion factor is raised	 per  each  new	 half-
	   cycle  until	 it reaches expansion value.  Setting this options too
	   high may lead to distortions.

       fall, f
	   Set the compression raising amount per each half-cycle of  samples.
	   Default  value  is  0.001.	Allowed range is from 0.0 to 1.0. This
	   controls how fast compression factor is raised per each  new	 half-
	   cycle until it reaches compression value.

       channels, h
	   Specify which channels to filter, by default all available channels
	   are filtered.

       invert, i
	   Enable  inverted  filtering,	 by  default is disabled. This inverts
	   interpretation of threshold option. When enabled any half-cycle  of
	   samples  with  their	 local	peak  value below or same as threshold
	   option will be expanded otherwise it will be compressed.

       link, l
	   Link channels  when	calculating  gain  applied  to	each  filtered
	   channel  sample,  by	 default  is  disabled.	  When	disabled  each
	   filtered channel gain calculation is	 independent,  otherwise  when
	   this	 option	 is enabled the minimum of all possible gains for each
	   filtered channel is used.

       rms, m
	   Set the expansion target RMS	 value.	 This  specifies  the  highest
	   allowed  RMS level for the normalized audio input. Default value is
	   0.0, thus disabled. Allowed range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Weak and slow amplification:

		   speechnorm=e=3:r=0.00001:l=1

       •   Moderate and slow amplification:

		   speechnorm=e=6.25:r=0.00001:l=1

       •   Strong and fast amplification:

		   speechnorm=e=12.5:r=0.0001:l=1

       •   Very strong and fast amplification:

		   speechnorm=e=25:r=0.0001:l=1

       •   Extreme and fast amplification:

		   speechnorm=e=50:r=0.0001:l=1

   stereotools
       This filter has some handy utilities  to	 manage	 stereo	 signals,  for
       converting  M/S	stereo	recordings  to L/R signal while having control
       over the parameters or spreading the stereo image of master track.

       The filter accepts the following options:

       level_in
	   Set input level before filtering for both channels. Defaults is  1.
	   Allowed range is from 0.015625 to 64.

       level_out
	   Set	output level after filtering for both channels. Defaults is 1.
	   Allowed range is from 0.015625 to 64.

       balance_in
	   Set input balance between both channels.  Default  is  0.   Allowed
	   range is from -1 to 1.

       balance_out
	   Set	output	balance	 between both channels. Default is 0.  Allowed
	   range is from -1 to 1.

       softclip
	   Enable softclipping. Results in analog distortion instead of	 harsh
	   digital 0dB clipping. Disabled by default.

       mutel
	   Mute the left channel. Disabled by default.

       muter
	   Mute the right channel. Disabled by default.

       phasel
	   Change the phase of the left channel. Disabled by default.

       phaser
	   Change the phase of the right channel. Disabled by default.

       mode
	   Set stereo mode. Available values are:

	   lr>lr
	       Left/Right to Left/Right, this is default.

	   lr>ms
	       Left/Right to Mid/Side.

	   ms>lr
	       Mid/Side to Left/Right.

	   lr>ll
	       Left/Right to Left/Left.

	   lr>rr
	       Left/Right to Right/Right.

	   lr>l+r
	       Left/Right to Left + Right.

	   lr>rl
	       Left/Right to Right/Left.

	   ms>ll
	       Mid/Side to Left/Left.

	   ms>rr
	       Mid/Side to Right/Right.

	   ms>rl
	       Mid/Side to Right/Left.

	   lr>l-r
	       Left/Right to Left - Right.

       slev
	   Set	level  of  side	 signal.  Default is 1.	 Allowed range is from
	   0.015625 to 64.

       sbal
	   Set balance of side signal. Default is 0.  Allowed range is from -1
	   to 1.

       mlev
	   Set level of the middle signal. Default is  1.   Allowed  range  is
	   from 0.015625 to 64.

       mpan
	   Set middle signal pan. Default is 0. Allowed range is from -1 to 1.

       base
	   Set	stereo	base between mono and inversed channels. Default is 0.
	   Allowed range is from -1 to 1.

       delay
	   Set delay in milliseconds how much to delay left from right channel
	   and vice versa. Default is 0. Allowed range is from -20 to 20.

       sclevel
	   Set S/C level. Default is 1. Allowed range is from 1 to 100.

       phase
	   Set the stereo phase in degrees. Default is	0.  Allowed  range  is
	   from 0 to 360.

       bmode_in, bmode_out
	   Set balance mode for balance_in/balance_out option.

	   Can be one of the following:

	   balance
	       Classic	balance	 mode. Attenuate one channel at time.  Gain is
	       raised up to 1.

	   amplitude
	       Similar as classic mode above but gain is raised up to 2.

	   power
	       Equal power distribution, from -6dB to +6dB range.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Apply karaoke like effect:

		   stereotools=mlev=0.015625

       •   Convert M/S signal to L/R:

		   "stereotools=mode=ms>lr"

   stereowiden
       This filter enhance the stereo effect by suppressing signal  common  to
       both  channels  and  by delaying the signal of left into right and vice
       versa, thereby widening the stereo effect.

       The filter accepts the following options:

       delay
	   Time in milliseconds of the delay of left  signal  into  right  and
	   vice versa.	Default is 20 milliseconds.

       feedback
	   Amount of gain in delayed signal into right and vice versa. Gives a
	   delay  effect  of  left signal in right output and vice versa which
	   gives widening effect. Default is 0.3.

       crossfeed
	   Cross feed of left into right with inverted phase.  This  helps  in
	   suppressing	the  mono.  If	the  value is 1 it will cancel all the
	   signal common to both channels. Default is 0.3.

       drymix
	   Set level of input signal of original channel. Default is 0.8.

       Commands

       This filter supports the all above options except "delay" as commands.

   superequalizer
       Apply 18 band equalizer.

       The filter accepts the following options:

       1b  Set 65Hz band gain.

       2b  Set 92Hz band gain.

       3b  Set 131Hz band gain.

       4b  Set 185Hz band gain.

       5b  Set 262Hz band gain.

       6b  Set 370Hz band gain.

       7b  Set 523Hz band gain.

       8b  Set 740Hz band gain.

       9b  Set 1047Hz band gain.

       10b Set 1480Hz band gain.

       11b Set 2093Hz band gain.

       12b Set 2960Hz band gain.

       13b Set 4186Hz band gain.

       14b Set 5920Hz band gain.

       15b Set 8372Hz band gain.

       16b Set 11840Hz band gain.

       17b Set 16744Hz band gain.

       18b Set 20000Hz band gain.

   surround
       Apply audio surround upmix filter.

       This filter allows to produce multichannel output from audio stream.

       The filter accepts the following options:

       chl_out
	   Set output channel layout. By default, this is 5.1.

	   See the Channel Layout section in the  ffmpeg-utils(1)  manual  for
	   the required syntax.

       chl_in
	   Set input channel layout. By default, this is stereo.

	   See	the  Channel  Layout section in the ffmpeg-utils(1) manual for
	   the required syntax.

       level_in
	   Set input volume level. By default, this is 1.

       level_out
	   Set output volume level. By default, this is 1.

       lfe Enable LFE channel output if	 output	 channel  layout  has  it.  By
	   default, this is enabled.

       lfe_low
	   Set LFE low cut off frequency. By default, this is 128 Hz.

       lfe_high
	   Set LFE high cut off frequency. By default, this is 256 Hz.

       lfe_mode
	   Set	LFE mode, can be add or sub. Default is add.  In add mode, LFE
	   channel is created from input audio and added to  output.   In  sub
	   mode,  LFE  channel is created from input audio and added to output
	   but also all non-LFE output channels are subtracted with output LFE
	   channel.

       smooth
	   Set temporal smoothness strength, used to gradually change  factors
	   when	 transforming  stereo sound in time. Allowed range is from 0.0
	   to 1.0.  Useful to improve output quality with focus option	values
	   greater  than  0.0.	 Default is 0.0. Only values inside this range
	   and without edges are effective.

       angle
	   Set angle of stereo surround transform, Allowed range is from 0  to
	   360.	 Default is 90.

       focus
	   Set focus of stereo surround transform, Allowed range is from -1 to
	   1.  Default is 0.

       fc_in
	   Set front center input volume. By default, this is 1.

       fc_out
	   Set front center output volume. By default, this is 1.

       fl_in
	   Set front left input volume. By default, this is 1.

       fl_out
	   Set front left output volume. By default, this is 1.

       fr_in
	   Set front right input volume. By default, this is 1.

       fr_out
	   Set front right output volume. By default, this is 1.

       sl_in
	   Set side left input volume. By default, this is 1.

       sl_out
	   Set side left output volume. By default, this is 1.

       sr_in
	   Set side right input volume. By default, this is 1.

       sr_out
	   Set side right output volume. By default, this is 1.

       bl_in
	   Set back left input volume. By default, this is 1.

       bl_out
	   Set back left output volume. By default, this is 1.

       br_in
	   Set back right input volume. By default, this is 1.

       br_out
	   Set back right output volume. By default, this is 1.

       bc_in
	   Set back center input volume. By default, this is 1.

       bc_out
	   Set back center output volume. By default, this is 1.

       lfe_in
	   Set LFE input volume. By default, this is 1.

       lfe_out
	   Set LFE output volume. By default, this is 1.

       allx
	   Set	spread	usage  of stereo image across X axis for all channels.
	   Allowed range is from -1 to 15.  By default this value is  negative
	   -1, and thus unused.

       ally
	   Set	spread	usage  of stereo image across Y axis for all channels.
	   Allowed range is from -1 to 15.  By default this value is  negative
	   -1, and thus unused.

       fcx, flx, frx, blx, brx, slx, srx, bcx
	   Set	spread	usage  of stereo image across X axis for each channel.
	   Allowed range is from 0.06 to 15.  By default this value is 0.5.

       fcy, fly, fry, bly, bry, sly, sry, bcy
	   Set spread usage of stereo image across Y axis  for	each  channel.
	   Allowed range is from 0.06 to 15.  By default this value is 0.5.

       win_size
	   Set	window size. Allowed range is from 1024 to 65536. Default size
	   is 4096.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann, hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default is "hann".

       overlap
	   Set window overlap. If  set	to  1,	the  recommended  overlap  for
	   selected window function will be picked. Default is 0.5.

   tiltshelf
       Boost  or cut the lower frequencies and cut or boost higher frequencies
       of the audio using a two-pole shelving filter with a  response  similar
       to  that	 of  a	standard hi-fi's tone-controls.	 This is also known as
       shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
	   Give the gain at 0 Hz. Its useful range is about -20 (for  a	 large
	   cut)	 to  +20 (for a large boost).  Beware of clipping when using a
	   positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend  or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 3000 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How	much to use filtered signal in output. Default is 1.  Range is
	   between 0 and 1.

       channels, c
	   Specify which channels to filter,  by  default  all	available  are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set	block  size  used for reverse IIR processing. If this value is
	   set to high enough  value  (higher  than  impulse  response	length
	   truncated  when  reaches  near  zero	 values) filtering will become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note that filter delay will be exactly this many samples  when  set
	   to non-zero value.

       Commands

       This filter supports some options as commands.

   treble, highshelf
       Boost  or  cut treble (upper) frequencies of the audio using a two-pole
       shelving filter with a response similar to that of a  standard  hi-fi's
       tone-controls. This is also known as shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
	   Give	 the gain at whichever is the lower of ~22 kHz and the Nyquist
	   frequency. Its useful range is about -20 (for a large cut)  to  +20
	   (for a large boost). Beware of clipping when using a positive gain.

       frequency, f
	   Set	the filter's central frequency and so can be used to extend or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 3000 Hz.

       width_type, t
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

	   k   kHz

       width, w
	   Determine how steep is the filter's shelf transition.

       poles, p
	   Set number of poles. Default is 2.

       mix, m
	   How much to use filtered signal in output. Default is 1.  Range  is
	   between 0 and 1.

       channels, c
	   Specify  which  channels  to	 filter,  by default all available are
	   filtered.

       normalize, n
	   Normalize biquad coefficients, by default is disabled.  Enabling it
	   will normalize magnitude response at DC to 0dB.

       transform, a
	   Set transform type of IIR filter.

	   di
	   dii
	   tdi
	   tdii
	   latt
	   svf
	   zdf
       precision, r
	   Set precison of filtering.

	   auto
	       Pick automatic sample format depending on surround filters.

	   s16 Always use signed 16-bit.

	   s32 Always use signed 32-bit.

	   f32 Always use float 32-bit.

	   f64 Always use float 64-bit.

       block_size, b
	   Set block size used for reverse IIR processing. If  this  value  is
	   set	to  high  enough  value	 (higher  than impulse response length
	   truncated when reaches near	zero  values)  filtering  will	become
	   linear phase otherwise if not big enough it will just produce nasty
	   artifacts.

	   Note	 that  filter delay will be exactly this many samples when set
	   to non-zero value.

       Commands

       This filter supports the following commands:

       frequency, f
	   Change treble frequency.  Syntax for the command is : "frequency"

       width_type, t
	   Change treble width_type.  Syntax for the command is : "width_type"

       width, w
	   Change treble width.	 Syntax for the command is : "width"

       gain, g
	   Change treble gain.	Syntax for the command is : "gain"

       mix, m
	   Change treble mix.  Syntax for the command is : "mix"

   tremolo
       Sinusoidal amplitude modulation.

       The filter accepts the following options:

       f   Modulation  frequency  in  Hertz.  Modulation  frequencies  in  the
	   subharmonic range (20 Hz or lower) will result in a tremolo effect.
	   This	 filter	 may  also be used as a ring modulator by specifying a
	   modulation frequency higher than 20 Hz.  Range is  0.1  -  20000.0.
	   Default value is 5.0 Hz.

       d   Depth  of  modulation as a percentage. Range is 0.0 - 1.0.  Default
	   value is 0.5.

   vibrato
       Sinusoidal phase modulation.

       The filter accepts the following options:

       f   Modulation frequency in Hertz.  Range is  0.1  -  20000.0.  Default
	   value is 5.0 Hz.

       d   Depth  of  modulation as a percentage. Range is 0.0 - 1.0.  Default
	   value is 0.5.

   virtualbass
       Apply audio Virtual Bass filter.

       This filter accepts stereo input and  produce  stereo  with  LFE	 (2.1)
       channels	 output.  The newly produced LFE channel have enhanced virtual
       bass originally	obtained  from	both  stereo  channels.	  This	filter
       outputs	front  left and front right channels unchanged as available in
       stereo input.

       The filter accepts the following options:

       cutoff
	   Set the virtual bass cutoff frequency. Default  value  is  250  Hz.
	   Allowed range is from 100 to 500 Hz.

       strength
	   Set	the  virtual  bass  strength.  Allowed range is from 0.5 to 3.
	   Default value is 3.

   volume
       Adjust the input audio volume.

       It accepts the following parameters:

       volume
	   Set audio volume expression.

	   Output values are clipped to the maximum value.

	   The output audio volume is given by the relation:

		   <output_volume> = <volume> * <input_volume>

	   The default value for volume is "1.0".

       precision
	   This parameter represents the mathematical precision.

	   It determines which input sample formats  will  be  allowed,	 which
	   affects the precision of the volume scaling.

	   fixed
	       8-bit  fixed-point; this limits input sample format to U8, S16,
	       and S32.

	   float
	       32-bit floating-point; this limits input sample format to  FLT.
	       (default)

	   double
	       64-bit floating-point; this limits input sample format to DBL.

       replaygain
	   Choose  the behaviour on encountering ReplayGain side data in input
	   frames.

	   drop
	       Remove  ReplayGain  side	 data,	ignoring  its  contents	  (the
	       default).

	   ignore
	       Ignore ReplayGain side data, but leave it in the frame.

	   track
	       Prefer the track gain, if present.

	   album
	       Prefer the album gain, if present.

       replaygain_preamp
	   Pre-amplification  gain  in	dB to apply to the selected replaygain
	   gain.

	   Default value for replaygain_preamp is 0.0.

       replaygain_noclip
	   Prevent clipping by limiting the gain applied.

	   Default value for replaygain_noclip is 1.

       eval
	   Set when the volume expression is evaluated.

	   It accepts the following values:

	   once
	       only evaluate expression once during the filter initialization,
	       or when the volume command is sent

	   frame
	       evaluate expression for each incoming frame

	   Default value is once.

       The volume expression can contain the following parameters.

       n   frame number (starting at zero)

       nb_channels
	   number of channels

       nb_consumed_samples
	   number of samples consumed by the filter

       nb_samples
	   number of samples in the current frame

       pos original frame position in the file; deprecated, do not use

       pts frame PTS

       sample_rate
	   sample rate

       startpts
	   PTS at start of stream

       startt
	   time at start of stream

       t   frame time

       tb  timestamp timebase

       volume
	   last set volume value

       Note that when eval  is	set  to	 once  only  the  sample_rate  and  tb
       variables are available, all other variables will evaluate to NAN.

       Commands

       This filter supports the following commands:

       volume
	   Modify  the volume expression.  The command accepts the same syntax
	   of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

       Examples

       •   Halve the input audio volume:

		   volume=volume=0.5
		   volume=volume=1/2
		   volume=volume=-6.0206dB

	   In all the above example the named key for volume can  be  omitted,
	   for example like in:

		   volume=0.5

       •   Increase   input  audio  power  by  6  decibels  using  fixed-point
	   precision:

		   volume=volume=6dB:precision=fixed

       •   Fade volume after time 10 with an annihilation period of 5 seconds:

		   volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

   volumedetect
       Detect the volume of the input video.

       The filter has no parameters. It supports only  16-bit  signed  integer
       samples,	 so  the input will be converted when needed. Statistics about
       the volume will be printed in the log when  the	input  stream  end  is
       reached.

       In  particular it will show the mean volume (root mean square), maximum
       volume (on a per-sample basis), and the beginning of a histogram of the
       registered volume values (from the maximum value to a cumulated	1/1000
       of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

	       [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
	       [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
	       [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
	       [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
	       [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
	       [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
	       [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
	       [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
	       [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means that:

       •   The mean square energy is approximately -27 dB, or 10^-2.7.

       •   The largest sample is at -4 dB, or more precisely between -4 dB and
	   -5 dB.

       •   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

       In  other  words,  raising  the	volume	by  +4	dB  does not cause any
       clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
       Below is a description of the currently available audio sources.

   abuffer
       Buffer audio frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic  use,	in  particular
       through the interface defined in libavfilter/buffersrc.h.

       It accepts the following parameters:

       time_base
	   The timebase which will be used for timestamps of submitted frames.
	   It	 must	 be    either	 a   floating-point   number   or   in
	   numerator/denominator form.

       sample_rate
	   The sample rate of the incoming audio buffers.

       sample_fmt
	   The sample format of the incoming audio buffers.  Either  a	sample
	   format  name	 or  its corresponding integer representation from the
	   enum AVSampleFormat in libavutil/samplefmt.h

       channel_layout
	   The channel layout of the incoming audio buffers.  Either a channel
	   layout name from channel_layout_map	in  libavutil/channel_layout.c
	   or its corresponding integer representation from the AV_CH_LAYOUT_*
	   macros in libavutil/channel_layout.h

       channels
	   The	number	of  channels  of  the incoming audio buffers.  If both
	   channels and	 channel_layout	 are  specified,  then	they  must  be
	   consistent.

       Examples

	       abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

       will  instruct  the  source  to	accept	planar	16bit signed stereo at
       44100Hz.	 Since the sample format with name "s16p" corresponds  to  the
       number  6 and the "stereo" channel layout corresponds to the value 0x3,
       this is equivalent to:

	       abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

   aevalsrc
       Generate an audio signal specified by an expression.

       This source accepts in input one or  more  expressions  (one  for  each
       channel),  which	 are  evaluated	 and  used to generate a corresponding
       audio signal.

       This source accepts the following options:

       exprs
	   Set the '|'-separated expressions list for each  separate  channel.
	   In  case  the  channel_layout option is not specified, the selected
	   channel layout depends  on  the  number  of	provided  expressions.
	   Otherwise the last specified expression is applied to the remaining
	   output channels.

       channel_layout, c
	   Set	the  channel  layout.  The number of channels in the specified
	   layout must be equal to the number of specified expressions.

       duration, d
	   Set the minimum  duration  of  the  sourced	audio.	See  the  Time
	   duration  section  in  the  ffmpeg-utils(1) manual for the accepted
	   syntax.  Note that the resulting duration may be greater  than  the
	   specified duration, as the generated audio is always cut at the end
	   of a complete frame.

	   If  not specified, or the expressed duration is negative, the audio
	   is supposed to be generated forever.

       nb_samples, n
	   Set the number of  samples  per  channel  per  each	output	frame,
	   default to 1024.

       sample_rate, s
	   Specify the sample rate, default to 44100.

       Each expression in exprs can contain the following constants:

       n   number of the evaluated sample, starting from 0

       t   time of the evaluated sample expressed in seconds, starting from 0

       s   sample rate

       Examples

       •   Generate silence:

		   aevalsrc=0

       •   Generate  a sin signal with frequency of 440 Hz, set sample rate to
	   8000 Hz:

		   aevalsrc="sin(440*2*PI*t):s=8000"

       •   Generate a two channels signal, specify the channel	layout	(Front
	   Center + Back Center) explicitly:

		   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       •   Generate white noise:

		   aevalsrc="-2+random(0)"

       •   Generate an amplitude modulated signal:

		   aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

       •   Generate 2.5 Hz binaural beats on a 360 Hz carrier:

		   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

   afdelaysrc
       Generate a fractional delay FIR coefficients.

       The  resulting  stream  can  be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       delay, d
	   Set the fractional delay. Default is 0.

       sample_rate, r
	   Set the sample rate, default is 44100.

       nb_samples, n
	   Set the number of samples per each frame. Default is 1024.

       taps, t
	   Set the number  of  filter  coefficents  in	output	audio  stream.
	   Default value is 0.

       channel_layout, c
	   Specifies  the  channel  layout, and can be a string representing a
	   channel layout.  The default value of channel_layout is "stereo".

   afireqsrc
       Generate a FIR equalizer coefficients.

       The resulting stream can be used with afir  filter  for	filtering  the
       audio signal.

       The filter accepts the following options:

       preset, p
	   Set equalizer preset.  Default preset is "flat".

	   Available presets are:

	   custom
	   flat
	   acoustic
	   bass
	   beats
	   classic
	   clear
	   deep bass
	   dubstep
	   electronic
	   hard-style
	   hip-hop
	   jazz
	   metal
	   movie
	   pop
	   r&b
	   rock
	   vocal booster
       gains, g
	   Set	custom	gains for each band. Only used if the preset option is
	   set to "custom".  Gains are separated by white spaces and each gain
	   is set in dBFS.  Default is "0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0".

       bands, b
	   Set the custom bands from where custon  equalizer  gains  are  set.
	   This	 must be in strictly increasing order. Only used if the preset
	   option is set to "custom".  Bands are separated by white spaces and
	   each band represent frequency in Hz.	 Default is "25 40 63 100  160
	   250 400 630 1000 1600 2500 4000 6300 10000 16000 24000".

       taps, t
	   Set	number	of filter coefficents in output audio stream.  Default
	   value is 4096.

       sample_rate, r
	   Set sample rate of output audio stream, default is 44100.

       nb_samples, n
	   Set number of samples  per  each  frame  in	output	audio  stream.
	   Default is 1024.

       interp, i
	   Set	interpolation  method  for  FIR equalizer coefficients. Can be
	   "linear" or "cubic".

       phase, h
	   Set phase type of FIR filter. Can be "linear"  or  "min":  minimum-
	   phase.  Default is minimum-phase filter.

   afirsrc
       Generate a FIR coefficients using frequency sampling method.

       The  resulting  stream  can  be used with afir filter for filtering the
       audio signal.

       The filter accepts the following options:

       taps, t
	   Set number of filter 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.

       duration, d
	   Set the duration of	the  sourced  audio.  See  the	Time  duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If  not specified, or the expressed duration is negative, the audio
	   is supposed to be generated forever.

       Examples

       •   Set the  sample  rate  to  48000  Hz	 and  the  channel  layout  to
	   AV_CH_LAYOUT_MONO.

		   anullsrc=r=48000:cl=4

       •   Do the same operation with a more obvious syntax:

		   anullsrc=r=48000:cl=mono

       All the parameters need to be explicitly defined.

   flite
       Synthesize a voice utterance using the libflite library.

       To  enable compilation of this filter you need to configure FFmpeg with
       "--enable-libflite".

       Note that 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

       •   Read	 from  file  speech.txt,  and  synthesize  the	text using the
	   standard flite voice:

		   flite=textfile=speech.txt

       •   Read the specified text selecting the "slt" voice:

		   flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       •   Input text to ffmpeg:

		   ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       •   Make ffplay speak the specified text, using "flite" and the "lavfi"
	   device:

		   ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

       For	more	  information	    about	libflite,	check:
       <http://www.festvox.org/flite/>

   anoisesrc
       Generate a noise audio signal.

       The filter accepts the following options:

       sample_rate, r
	   Specify the sample rate. Default value is 48000 Hz.

       amplitude, a
	   Specify  the	 amplitude  (0.0 - 1.0) of the generated audio stream.
	   Default value is 1.0.

       duration, d
	   Specify the duration of the generated audio stream. Not  specifying
	   this option results in noise with an infinite length.

       color, colour, c
	   Specify the color of noise. Available noise colors are white, pink,
	   brown, blue, violet and velvet. Default color is white.

       seed, s
	   Specify a value used to seed the PRNG.

       nb_samples, n
	   Set the number of samples per each output frame, default is 1024.

       density
	   Set the density (0.0 - 1.0) for the velvet noise generator, default
	   is 0.05.

       Examples

       •   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

       •   Generate a simple 440 Hz sine wave:

		   sine

       •   Generate  a	220 Hz sine wave with a 880 Hz beep each second, for 5
	   seconds:

		   sine=220:4:d=5
		   sine=f=220:b=4:d=5
		   sine=frequency=220:beep_factor=4:duration=5

       •   Generate a 1 kHz  sine  wave	 following  "1602,1601,1602,1601,1602"
	   NTSC pattern:

		   sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'

AUDIO SINKS
       Below is a description of the currently available audio sinks.

   abuffersink
       Buffer  audio  frames,  and  make  them	available to the end of filter
       chain.

       This sink is  mainly  intended  for  programmatic  use,	in  particular
       through	the  interface	defined	 in  libavfilter/buffersink.h  or  the
       options system.

       It accepts  a  pointer  to  an  AVABufferSinkContext  structure,	 which
       defines	the  incoming  buffers'	 formats,  to  be passed as the opaque
       parameter to "avfilter_init_filter" for initialization.

   anullsink
       Null audio sink; do absolutely nothing with  the	 input	audio.	It  is
       mainly useful as a template and for use in analysis / debugging tools.

VIDEO FILTERS
       When  you  configure  your  FFmpeg  build,  you	can disable any of the
       existing filters using "--disable-filters".  The configure output  will
       show the video filters included in your build.

       Below is a description of the currently available video filters.

   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

       •   Mark the centre quarter of the frame as interesting.

		   addroi=iw/4:ih/4:iw/2:ih/2:-1/10

       •   Mark	 the  100-pixel-wide  region  on the left edge of the frame as
	   very uninteresting (to be encoded at much lower  quality  than  the
	   rest of the frame).

		   addroi=0:0:100:ih:+1/5

   alphaextract
       Extract	the  alpha component from the input as a grayscale video. This
       is especially useful with the alphamerge filter.

   alphamerge
       Add or replace the alpha	 component  of	the  primary  input  with  the
       grayscale  value	 of  a	second	input.	This  is intended for use with
       alphaextract to allow the transmission or storage  of  frame  sequences
       that have alpha in a format that doesn't support an alpha channel.

       For example, to reconstruct full frames from a normal YUV-encoded video
       and a separate video created with alphaextract, you might use:

	       movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

   amplify
       Amplify differences between current pixel and pixels of adjacent frames
       in same pixel location.

       This filter accepts the following options:

       radius
	   Set frame radius. Default is 2. Allowed range is from 1 to 63.  For
	   example  radius of 3 will instruct filter to calculate average of 7
	   frames.

       factor
	   Set factor to amplify difference. Default is 2.  Allowed  range  is
	   from 0 to 65535.

       threshold
	   Set	threshold for difference amplification. Any difference greater
	   or equal to this value will not alter source pixel. Default is  10.
	   Allowed range is from 0 to 65535.

       tolerance
	   Set tolerance for difference amplification. Any difference lower to
	   this	 value	will  not  alter  source pixel. Default is 0.  Allowed
	   range is from 0 to 65535.

       low Set lower limit  for	 changing  source  pixel.  Default  is	65535.
	   Allowed  range  is  from  0 to 65535.  This option controls maximum
	   possible value that will decrease source pixel value.

       high
	   Set high limit for changing source pixel. Default is 65535. Allowed
	   range is from 0 to 65535.  This option  controls  maximum  possible
	   value that will increase source pixel value.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Commands

       This  filter supports the following commands that corresponds to option
       of same name:

       factor
       threshold
       tolerance
       low
       high
       planes

   ass
       Same as the subtitles filter, except that it doesn't require libavcodec
       and libavformat to work. On the	other  hand,  it  is  limited  to  ASS
       (Advanced Substation Alpha) subtitles files.

       This  filter  accepts  the  following  option in addition to the common
       options from the subtitles filter:

       shaping
	   Set the shaping engine

	   Available values are:

	   auto
	       The default libass shaping engine, which is the best available.

	   simple
	       Fast, font-agnostic shaper that can do only substitutions

	   complex
	       Slower shaper using OpenType for substitutions and positioning

	   The default is "auto".

   atadenoise
       Apply an Adaptive Temporal Averaging Denoiser to the video input.

       The filter accepts the following options:

       0a  Set threshold A for 1st plane. Default is 0.02.  Valid range	 is  0
	   to 0.3.

       0b  Set	threshold  B for 1st plane. Default is 0.04.  Valid range is 0
	   to 5.

       1a  Set threshold A for 2nd plane. Default is 0.02.  Valid range	 is  0
	   to 0.3.

       1b  Set	threshold  B for 2nd plane. Default is 0.04.  Valid range is 0
	   to 5.

       2a  Set threshold A for 3rd plane. Default is 0.02.  Valid range	 is  0
	   to 0.3.

       2b  Set	threshold  B for 3rd plane. Default is 0.04.  Valid range is 0
	   to 5.

	   Threshold A is designed to react on abrupt  changes	in  the	 input
	   signal  and	threshold B is designed to react on continuous changes
	   in the input signal.

       s   Set number of frames filter will use for averaging. Default	is  9.
	   Must be odd number in range [5, 129].

       p   Set	what planes of frame filter will use for averaging. Default is
	   all.

       a   Set what variant  of	 algorithm  filter  will  use  for  averaging.
	   Default is "p" parallel.  Alternatively can be set to "s" serial.

	   Parallel can be faster then serial, while other way around is never
	   true.  Parallel will abort early on first change being greater then
	   thresholds,	while  serial  will  continue processing other side of
	   frames if they are equal or below thresholds.

       0s
       1s
       2s  Set sigma for 1st plane, 2nd plane or 3rd plane. Default is	32767.
	   Valid  range	 is from 0 to 32767.  This options controls weight for
	   each pixel in radius defined by size.  Default  value  means	 every
	   pixel  have	same  weight.	Setting	 this  option to 0 effectively
	   disables filtering.

       Commands

       This filter supports same commands as options except option  "s".   The
       command accepts the same syntax of the corresponding option.

   avgblur
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.

       planes
	   Set which planes to filter. By default all planes are filtered.

       sizeY
	   Set	vertical  radius  size,	 if  zero  it will be same as "sizeX".
	   Default is 0.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If the specified expression is not valid, it is	kept  at  its  current
       value.

   backgroundkey
       Turns a static background into transparency.

       The filter accepts the following option:

       threshold
	   Threshold for scene change detection.

       similarity
	   Similarity percentage with the background.

       blend
	   Set the blend amount for pixels that are not similar.

       Commands

       This filter supports the all above options as commands.

   bbox
       Compute	the  bounding  box for the non-black pixels in the input frame
       luma plane.

       This filter computes the bounding box containing all the pixels with  a
       luma  value  greater  than  the	minimum allowed value.	The parameters
       describing the bounding box are printed on the filter log.

       The filter accepts the following option:

       min_val
	   Set the minimal luma value. Default is 16.

       Commands

       This filter supports the all above options as commands.

   bilateral
       Apply bilateral filter, spatial smoothing while preserving edges.

       The filter accepts the following options:

       sigmaS
	   Set	sigma  of  gaussian  function  to  calculate  spatial  weight.
	   Allowed range is 0 to 512. Default is 0.1.

       sigmaR
	   Set	sigma of gaussian function to calculate range weight.  Allowed
	   range is 0 to 1. Default is 0.1.

       planes
	   Set planes to filter. Default is first only.

       Commands

       This filter supports the all above options as commands.

   bilateral_cuda
       CUDA accelerated bilateral filter, an  edge  preserving	filter.	  This
       filter	is   mathematically   accurate	 thanks	 to  the  use  of  GPU
       acceleration.   For  best  output  quality,  use	 one  to  one	chroma
       subsampling, i.e. yuv444p format.

       The filter accepts the following options:

       sigmaS
	   Set	sigma  of  gaussian function to calculate spatial weight, also
	   called sigma space.	Allowed range is 0.1 to 512. Default is 0.1.

       sigmaR
	   Set sigma of gaussian function to  calculate	 color	range  weight,
	   also	 called	 sigma color.  Allowed range is 0.1 to 512. Default is
	   0.1.

       window_size
	   Set window size of the bilateral function to determine  the	number
	   of  neighbours to loop on.  If the number entered is even, one will
	   be added automatically.  Allowed range is 1 to 255. Default is 1.

       Examples

       •   Apply the bilateral filter on a video.

		   ./ffmpeg -v verbose \
		   -hwaccel cuda -hwaccel_output_format cuda -i input.mp4  \
		   -init_hw_device cuda \
		   -filter_complex \
		   " \
		   [0:v]scale_cuda=format=yuv444p[scaled_video];
		   [scaled_video]bilateral_cuda=window_size=9:sigmaS=3.0:sigmaR=50.0" \
		   -an -sn -c:v h264_nvenc -cq 20 out.mp4

   bitplanenoise
       Show and measure bit plane noise.

       The filter accepts the following options:

       bitplane
	   Set which plane to analyze. Default is 1.

       filter
	   Filter out noisy pixels from	 "bitplane"  set  above.   Default  is
	   disabled.

   blackdetect
       Detect  video  intervals	 that  are  (almost)  completely black. Can be
       useful  to  detect  chapter  transitions,   commercials,	  or   invalid
       recordings.

       The  filter  outputs  its detection analysis to both the log as well as
       frame metadata. If a black segment of at least  the  specified  minimum
       duration	 is found, a line with the start and end timestamps as well as
       duration is printed to the log with level "info". In  addition,	a  log
       line  with  level "debug" is printed per frame showing the black amount
       detected for that frame.

       The filter also attaches metadata to the first frame of a black segment
       with key "lavfi.black_start" and to the first  frame  after  the	 black
       segment	ends  with  key	 "lavfi.black_end".  The  value is the frame's
       timestamp. This metadata is added regardless of	the  minimum  duration
       specified.

       The filter accepts the following options:

       black_min_duration, d
	   Set	the  minimum  detected black duration expressed in seconds. It
	   must be a non-negative floating point number.

	   Default value is 2.0.

       picture_black_ratio_th, pic_th
	   Set the threshold for considering a picture "black".	  Express  the
	   minimum value for the ratio:

		   <nb_black_pixels> / <nb_pixels>

	   for which a picture is considered black.  Default value is 0.98.

       pixel_black_th, pix_th
	   Set the threshold for considering a pixel "black".

	   The	threshold  expresses  the maximum pixel luma value for which a
	   pixel is considered "black". The provided value is scaled according
	   to the following equation:

		   <absolute_threshold> = <luma_minimum_value> + <pixel_black_th> * <luma_range_size>

	   luma_range_size and luma_minimum_value depend on  the  input	 video
	   format,  the	 range	is  [0-255]  for  YUV  full-range  formats and
	   [16-235] for YUV non full-range formats.

	   Default value is 0.10.

       The following example sets the maximum pixel threshold to  the  minimum
       value, and detects only black intervals of 2 or more seconds:

	       blackdetect=d=2:pix_th=0.00

   blackframe
       Detect  frames  that  are  (almost)  completely black. Can be useful to
       detect chapter transitions or commercials. Output lines consist of  the
       frame  number  of  the detected frame, the percentage of blackness, the
       position in the file if known or -1 and the timestamp in seconds.

       In order to display the output lines, you need to set the  loglevel  at
       least to the AV_LOG_INFO value.

       This  filter  exports  frame  metadata  "lavfi.blackframe.pblack".  The
       value represents the percentage of pixels in the picture that are below
       the threshold value.

       It accepts the following parameters:

       amount
	   The percentage of the pixels that have to be below  the  threshold;
	   it defaults to 98.

       threshold, thresh
	   The	threshold  below  which	 a pixel value is considered black; it
	   defaults to 32.

   blend
       Blend two video frames into each other.

       The "blend" filter takes two input streams and outputs one stream,  the
       first  input is the "top" layer and second input is "bottom" layer.  By
       default, the output terminates when the longest input terminates.

       The "tblend" (time blend) filter takes two consecutive frames from  one
       single  stream,	and  outputs  the  result obtained by blending the new
       frame on top of the old frame.

       A description of the accepted options follows.

       c0_mode
       c1_mode
       c2_mode
       c3_mode
       all_mode
	   Set blend mode for specific pixel component or all pixel components
	   in case of all_mode. Default value is "normal".

	   Available values for component modes are:

	   addition
	   and
	   average
	   bleach
	   burn
	   darken
	   difference
	   divide
	   dodge
	   exclusion
	   extremity
	   freeze
	   geometric
	   glow
	   grainextract
	   grainmerge
	   hardlight
	   hardmix
	   hardoverlay
	   harmonic
	   heat
	   interpolate
	   lighten
	   linearlight
	   multiply
	   multiply128
	   negation
	   normal
	   or
	   overlay
	   phoenix
	   pinlight
	   reflect
	   screen
	   softdifference
	   softlight
	   stain
	   subtract
	   vividlight
	   xor
       c0_opacity
       c1_opacity
       c2_opacity
       c3_opacity
       all_opacity
	   Set blend  opacity  for  specific  pixel  component	or  all	 pixel
	   components  in  case	 of all_opacity. Only used in combination with
	   pixel component blend modes.

       c0_expr
       c1_expr
       c2_expr
       c3_expr
       all_expr
	   Set blend expression for specific  pixel  component	or  all	 pixel
	   components in case of all_expr. Note that related mode options will
	   be ignored if those are set.

	   The expressions can use the following variables:

	   N   The sequential number of the filtered frame, starting from 0.

	   X
	   Y   the coordinates of the current sample

	   W
	   H   the width and height of currently filtered plane

	   SW
	   SH  Width  and height scale for the plane being filtered. It is the
	       ratio between the dimensions of the current plane to  the  luma
	       plane, e.g. for a "yuv420p" frame, the values are "1,1" for the
	       luma plane and "0.5,0.5" for the chroma planes.

	   T   Time of the current frame, expressed in seconds.

	   TOP, A
	       Value  of  pixel	 component at current location for first video
	       frame (top layer).

	   BOTTOM, B
	       Value of pixel component at current location for	 second	 video
	       frame (bottom layer).

       The "blend" filter also supports the framesync options.

       Examples

       •   Apply  transition  from  bottom  layer  to  top  layer  in first 10
	   seconds:

		   blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

       •   Apply linear horizontal transition from top layer to bottom layer:

		   blend=all_expr='A*(X/W)+B*(1-X/W)'

       •   Apply 1x1 checkerboard effect:

		   blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

       •   Apply uncover left effect:

		   blend=all_expr='if(gte(N*SW+X,W),A,B)'

       •   Apply uncover down effect:

		   blend=all_expr='if(gte(Y-N*SH,0),A,B)'

       •   Apply uncover up-left effect:

		   blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

       •   Split diagonally video and shows top and bottom layer on each side:

		   blend=all_expr='if(gt(X,Y*(W/H)),A,B)'

       •   Display differences between the current and the previous frame:

		   tblend=all_mode=grainextract

       Commands

       This filter supports same commands as options.

   blockdetect
       Determines blockiness of frames without altering the input frames.

       Based on	 Remco	Muijs  and  Ihor  Kirenko:  "A	no-reference  blocking
       artifact	 measure  for  adaptive	 video processing." 2005 13th European
       signal processing conference.

       The filter accepts the following options:

       period_min
       period_max
	   Set	minimum	 and  maximum  values  for  determining	 pixel	 grids
	   (periods).  Default values are [3,24].

       planes
	   Set planes to filter. Default is first only.

       Examples

       •   Determine  blockiness  for  the  first plane and search for periods
	   within [8,32]:

		   blockdetect=period_min=8:period_max=32:planes=1

   blurdetect
       Determines blurriness of frames without altering the input frames.

       Based on Marziliano, Pina,  et  al.  "A	no-reference  perceptual  blur
       metric."	 Allows for a block-based abbreviation.

       The filter accepts the following options:

       low
       high
	   Set	low  and  high threshold values used by the Canny thresholding
	   algorithm.

	   The high threshold selects the "strong" edge pixels, which are then
	   connected  through  8-connectivity  with  the  "weak"  edge	pixels
	   selected by the low threshold.

	   low	and  high  threshold values must be chosen in the range [0,1],
	   and low should be lesser or equal to high.

	   Default value for low is "20/255", and default value	 for  high  is
	   "50/255".

       radius
	   Define the radius to search around an edge pixel for local maxima.

       block_pct
	   Determine blurriness only for the most significant blocks, given in
	   percentage.

       block_width
	   Determine blurriness for blocks of width block_width. If set to any
	   value  smaller  1,  no  blocks  are	used  and  the	whole image is
	   processed as one no matter of block_height.

       block_height
	   Determine blurriness for blocks of height block_height. If  set  to
	   any	value  smaller	1,  no	blocks are used and the whole image is
	   processed as one no matter of block_width.

       planes
	   Set planes to filter. Default is first only.

       Examples

       •   Determine blur for 80% of most significant 32x32 blocks:

		   blurdetect=block_width=32:block_height=32:block_pct=80

   bm3d
       Denoise frames using Block-Matching 3D algorithm.

       The filter accepts the following options.

       sigma
	   Set denoising strength. Default value is 1.	Allowed range is  from
	   0 to 999.9.	The denoising algorithm is very sensitive to sigma, so
	   adjust it according to the source.

       block
	   Set local patch size. This sets dimensions in 2D.

       bstep
	   Set	sliding	 step  for  processing	blocks.	 Default  value	 is 4.
	   Allowed range is from 1 to 64.  Smaller  values  allows  processing
	   more reference blocks and is slower.

       group
	   Set	maximal	 number	 of  similar blocks for 3rd dimension. Default
	   value is 1.	When set to 1,	no  block  matching  is	 done.	Larger
	   values allows more blocks in single group.  Allowed range is from 1
	   to 256.

       range
	   Set	radius for search block matching. Default is 9.	 Allowed range
	   is from 1 to INT32_MAX.

       mstep
	   Set step between two search locations for block  matching.  Default
	   is 1.  Allowed range is from 1 to 64. Smaller is slower.

       thmse
	   Set	threshold of mean square error for block matching. Valid range
	   is 0 to INT32_MAX.

       hdthr
	   Set thresholding parameter for hard thresholding in 3D  transformed
	   domain.    Larger  values  results  in  stronger  hard-thresholding
	   filtering in frequency domain.

       estim
	   Set filtering estimation mode. Can be "basic" or "final".   Default
	   is "basic".

       ref If enabled, filter will use 2nd stream for block matching.  Default
	   is  disabled	 for "basic" value of estim option, and always enabled
	   if value of estim is "final".

       planes
	   Set planes to filter. Default is all available except alpha.

       Examples

       •   Basic filtering with bm3d:

		   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic

       •   Same as above, but filtering only luma:

		   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1

       •   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

       •   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

       •   Apply a boxblur filter with the luma, chroma, and alpha  radii  set
	   to 2:

		   boxblur=luma_radius=2:luma_power=1
		   boxblur=2:1

       •   Set the luma radius to 2, and alpha and chroma radius to 0:

		   boxblur=2:1:cr=0:ar=0

       •   Set the luma and chroma radii to a fraction of the video dimension:

		   boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

   bwdif
       Deinterlace   the   input   video   ("bwdif"  stands  for  "Bob	Weaver
       Deinterlacing Filter").

       Motion adaptive deinterlacing based on yadif with the use of w3fdif and
       cubic interpolation algorithms.	It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It  accepts  one  of	the  following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   The default value is "send_field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The	default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first  will  be
	   assumed.

       deint
	   Specify  which  frames to deinterlace. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   bwdif_cuda
       Deinterlace the input video using the bwdif algorithm, but  implemented
       in  CUDA so that it can work as part of a GPU accelerated pipeline with
       nvdec and/or nvenc.

       It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It  accepts  one  of	the  following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   The default value is "send_field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The	default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first  will  be
	   assumed.

       deint
	   Specify  which  frames to deinterlace. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   ccrepack
       Repack CEA-708 closed captioning side data

       This filter fixes various issues seen with commerical encoders  related
       to  upstream  malformed CEA-708 payloads, specifically incorrect number
       of tuples (wrong cc_count for the target FPS), and  incorrect  ordering
       of  tuples (i.e. the CEA-608 tuples are not at the first entries in the
       payload).

   cas
       Apply Contrast Adaptive Sharpen filter to video stream.

       The filter accepts the following options:

       strength
	   Set the sharpening strength. Default value is 0.

       planes
	   Set planes to filter. Default value is to filter all planes	except
	   alpha plane.

       Commands

       This filter supports same commands as options.

   chromahold
       Remove all color information for all colors except for certain one.

       The filter accepts the following options:

       color
	   The color which will not be replaced with neutral chroma.

       similarity
	   Similarity  percentage with the above color.	 0.01 matches only the
	   exact key color, while 1.0 matches everything.

       blend
	   Blend percentage.  0.0 makes pixels either fully gray, or not  gray
	   at all.  Higher values result in more preserved color.

       yuv Signals that the color passed is already in YUV instead of RGB.

	   Literal  colors  like  "green"  or "red" don't make sense with this
	   enabled anymore.  This can be used to  pass	exact  YUV  values  as
	   hexadecimal numbers.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

   chromakey
       YUV colorspace color/chroma keying.

       The filter accepts the following options:

       color
	   The color which will be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01	 matches  only	the  exact  key	 color,	 while	 1.0   matches
	   everything.

       blend
	   Blend percentage.

	   0.0	makes  pixels  either fully transparent, or not transparent at
	   all.

	   Higher values result in  semi-transparent  pixels,  with  a	higher
	   transparency the more similar the pixels color is to the key color.

       yuv Signals that the color passed is already in YUV instead of RGB.

	   Literal  colors  like  "green"  or "red" don't make sense with this
	   enabled anymore.  This can be used to  pass	exact  YUV  values  as
	   hexadecimal numbers.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

       Examples

       •   Make every green pixel in the input image transparent:

		   ffmpeg -i input.png -vf chromakey=green out.png

       •   Overlay a greenscreen-video on top of a static black background.

		   ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv

   chromakey_cuda
       CUDA accelerated YUV colorspace color/chroma keying.

       This filter works like normal chromakey filter  but  operates  on  CUDA
       frames.	for more details and parameters see chromakey.

       Examples

       •   Make all the green pixels in the input video transparent and use it
	   as an overlay for another video:

		   ./ffmpeg \
		       -hwaccel cuda -hwaccel_output_format cuda -i input_green.mp4  \
		       -hwaccel cuda -hwaccel_output_format cuda -i base_video.mp4 \
		       -init_hw_device cuda \
		       -filter_complex \
		       " \
			   [0:v]chromakey_cuda=0x25302D:0.1:0.12:1[overlay_video]; \
			   [1:v]scale_cuda=format=yuv420p[base]; \
			   [base][overlay_video]overlay_cuda" \
		       -an -sn -c:v h264_nvenc -cq 20 output.mp4

       •   Process two software sources, explicitly uploading the frames:

		   ./ffmpeg -init_hw_device cuda=cuda -filter_hw_device cuda \
		       -f lavfi -i color=size=800x600:color=white,format=yuv420p \
		       -f lavfi -i yuvtestsrc=size=200x200,format=yuv420p \
		       -filter_complex \
		       " \
			   [0]hwupload[under]; \
			   [1]hwupload,chromakey_cuda=green:0.1:0.12[over]; \
			   [under][over]overlay_cuda" \
		       -c:v hevc_nvenc -cq 18 -preset slow output.mp4

   chromanr
       Reduce chrominance noise.

       The filter accepts the following options:

       thres
	   Set	threshold  for	averaging chrominance values.  Sum of absolute
	   difference of Y, U and V pixel  components  of  current  pixel  and
	   neighbour  pixels  lower  than  this	 threshold  will  be  used  in
	   averaging. Luma component  is  left	unchanged  and	is  copied  to
	   output.  Default value is 30. Allowed range is from 1 to 200.

       sizew
	   Set	horizontal  radius  of	rectangle used for averaging.  Allowed
	   range is from 1 to 100. Default value is 5.

       sizeh
	   Set vertical radius of rectangle used for averaging.	 Allowed range
	   is from 1 to 100. Default value is 5.

       stepw
	   Set horizontal step when averaging. Default value  is  1.   Allowed
	   range is from 1 to 50.  Mostly useful to speed-up filtering.

       steph
	   Set	vertical  step	when  averaging.  Default value is 1.  Allowed
	   range is from 1 to 50.  Mostly useful to speed-up filtering.

       threy
	   Set Y  threshold  for  averaging  chrominance  values.   Set	 finer
	   control  for max allowed difference between Y components of current
	   pixel and neigbour pixels.  Default value is 200. Allowed range  is
	   from 1 to 200.

       threu
	   Set	U  threshold  for  averaging  chrominance  values.   Set finer
	   control for max allowed difference between U components of  current
	   pixel  and neigbour pixels.	Default value is 200. Allowed range is
	   from 1 to 200.

       threv
	   Set V  threshold  for  averaging  chrominance  values.   Set	 finer
	   control  for max allowed difference between V components of current
	   pixel and neigbour pixels.  Default value is 200. Allowed range  is
	   from 1 to 200.

       distance
	   Set distance type used in calculations.

	   manhattan
	       Absolute difference.

	   euclidean
	       Difference squared.

	   Default distance type is manhattan.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

   chromashift
       Shift chroma pixels horizontally and/or vertically.

       The filter accepts the following options:

       cbh Set amount to shift chroma-blue horizontally.

       cbv Set amount to shift chroma-blue vertically.

       crh Set amount to shift chroma-red horizontally.

       crv Set amount to shift chroma-red vertically.

       edge
	   Set edge mode, can be smear, default, or warp.

       Commands

       This filter supports the all above options as commands.

   ciescope
       Display CIE color diagram with pixels overlaid onto it.

       The filter accepts the following options:

       system
	   Set color system.

	   ntsc, 470m
	   ebu, 470bg
	   smpte
	   240m
	   apple
	   widergb
	   cie1931
	   rec709, hdtv
	   uhdtv, rec2020
	   dcip3
       cie Set CIE system.

	   xyy
	   ucs
	   luv
       gamuts
	   Set what gamuts to draw.

	   See "system" option for available values.

       size, s
	   Set ciescope size, by default set to 512.

       intensity, i
	   Set intensity used to map input pixel values to CIE diagram.

       contrast
	   Set	contrast  used	to  draw  tongue colors that are out of active
	   color system gamut.

       corrgamma
	   Correct gamma displayed on scope, by default enabled.

       showwhite
	   Show white point on CIE diagram, by default disabled.

       gamma
	   Set input gamma. Used only with XYZ input color space.

       fill
	   Fill with CIE colors. By default is enabled.

   codecview
       Visualize information exported by some codecs.

       Some codecs can export information through frames  using	 side-data  or
       other  means. For example, some MPEG based codecs export motion vectors
       through the export_mvs flag in the codec flags2 option.

       The filter accepts the following option:

       block
	   Display block partition structure using the luma plane.

       mv  Set motion vectors to visualize.

	   Available flags for mv are:

	   pf  forward predicted MVs of P-frames

	   bf  forward predicted MVs of B-frames

	   bb  backward predicted MVs of B-frames

       qp  Display quantization parameters using the chroma planes.

       mv_type, mvt
	   Set motion vectors type to visualize. Includes MVs from all	frames
	   unless specified by frame_type option.

	   Available flags for mv_type are:

	   fp  forward predicted MVs

	   bp  backward predicted MVs

       frame_type, ft
	   Set frame type to visualize motion vectors of.

	   Available flags for frame_type are:

	   if  intra-coded frames (I-frames)

	   pf  predicted frames (P-frames)

	   bf  bi-directionally predicted frames (B-frames)

       Examples

       •   Visualize forward predicted MVs of all frames using ffplay:

		   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp

       •   Visualize multi-directionals MVs of P and B-Frames using ffplay:

		   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb

   colorbalance
       Modify  intensity  of  primary  colors  (red,  green and blue) of input
       frames.

       The filter allows an  input  frame  to  be  adjusted  in	 the  shadows,
       midtones or highlights regions for the red-cyan, green-magenta or blue-
       yellow balance.

       A  positive  adjustment	value  shifts  the balance towards the primary
       color, a negative value towards the complementary color.

       The filter accepts the following options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

	   Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

       pl  Preserve  lightness	when  changing	color  balance.	  Default   is
	   disabled.

       Examples

       •   Add red color cast to shadows:

		   colorbalance=rs=.3

       Commands

       This filter supports the all above options as commands.

   colorcontrast
       Adjust color contrast between RGB components.

       The filter accepts the following options:

       rc  Set	the  red-cyan contrast. Defaults is 0.0. Allowed range is from
	   -1.0 to 1.0.

       gm  Set the green-magenta contrast. Defaults is 0.0. Allowed  range  is
	   from -1.0 to 1.0.

       by  Set	the  blue-yellow  contrast.  Defaults is 0.0. Allowed range is
	   from -1.0 to 1.0.

       rcw
       gmw
       byw Set the weight of each "rc", "gm", "by" option value. Default value
	   is 0.0.  Allowed range is from 0.0 to 1.0. If all weights  are  0.0
	   filtering is disabled.

       pl  Set	the  amount  of	 preserving  lightness.	 Default value is 0.0.
	   Allowed range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

   colorcorrect
       Adjust color white balance selectively for  blacks  and	whites.	  This
       filter operates in YUV colorspace.

       The filter accepts the following options:

       rl  Set	the  red  shadow  spot.	 Allowed  range	 is  from -1.0 to 1.0.
	   Default value is 0.

       bl  Set the blue shadow spot.  Allowed  range  is  from	-1.0  to  1.0.
	   Default value is 0.

       rh  Set	the  red  highlight  spot.  Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       bh  Set the blue highlight spot. Allowed range is  from	-1.0  to  1.0.
	   Default value is 0.

       saturation
	   Set	the  amount  of saturation. Allowed range is from -3.0 to 3.0.
	   Default value is 1.

       analyze
	   If set to anything other than "manual" it will analyze every	 frame
	   and use derived parameters for filtering output frame.

	   Possible values are:

	   manual
	   average
	   minmax
	   median

	   Default value is "manual".

       Commands

       This filter supports the all above options as commands.

   colorchannelmixer
       Adjust video input frames by re-mixing color channels.

       This filter modifies a color channel by adding the values associated to
       the  other  channels  of	 the  same pixels. For example if the value to
       modify is red, the output value will be:

	       <red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>

       The filter accepts the following options:

       rr
       rg
       rb
       ra  Adjust contribution of input red, green, blue  and  alpha  channels
	   for	output red channel.  Default is 1 for rr, and 0 for rg, rb and
	   ra.

       gr
       gg
       gb
       ga  Adjust contribution of input red, green, blue  and  alpha  channels
	   for	output	green  channel.	 Default is 1 for gg, and 0 for gr, gb
	   and ga.

       br
       bg
       bb
       ba  Adjust contribution of input red, green, blue  and  alpha  channels
	   for output blue channel.  Default is 1 for bb, and 0 for br, bg and
	   ba.

       ar
       ag
       ab
       aa  Adjust  contribution	 of  input red, green, blue and alpha channels
	   for output alpha channel.  Default is 1 for aa, and 0  for  ar,  ag
	   and ab.

	   Allowed ranges for options are "[-2.0, 2.0]".

       pc  Set preserve color mode. The accepted values are:

	   none
	       Disable color preserving, this is default.

	   lum Preserve luminance.

	   max Preserve max value of RGB triplet.

	   avg Preserve average value of RGB triplet.

	   sum Preserve sum value of RGB triplet.

	   nrm Preserve normalized value of RGB triplet.

	   pwr Preserve power value of RGB triplet.

       pa  Set	the  preserve color amount when changing colors. Allowed range
	   is from "[0.0, 1.0]".  Default is 0.0, thus disabled.

       Examples

       •   Convert source to grayscale:

		   colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

       •   Simulate sepia tones:

		   colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

       Commands

       This filter supports the all above options as commands.

   colorize
       Overlay a solid color on the video stream.

       The filter accepts the following options:

       hue Set the color hue. Allowed range is from 0 to 360.	Default	 value
	   is 0.

       saturation
	   Set	the  color  saturation. Allowed range is from 0 to 1.  Default
	   value is 0.5.

       lightness
	   Set the color lightness. Allowed range is from  0  to  1.   Default
	   value is 0.5.

       mix Set the mix of source lightness. By default is set to 1.0.  Allowed
	   range is from 0.0 to 1.0.

       Commands

       This filter supports the all above options as commands.

   colorkey
       RGB  colorspace color keying.  This filter operates on 8-bit RGB format
       frames by setting the alpha component of each pixel which falls	within
       the similarity radius of the key color to 0. The alpha value for pixels
       outside the similarity radius depends on the value of the blend option.

       The filter accepts the following options:

       color
	   Set the color for which alpha will be set to 0 (full transparency).
	   See	"Color"	 section  in  the  ffmpeg-utils	 manual.   Default  is
	   "black".

       similarity
	   Set the radius from the key color within which  other  colors  also
	   have	 full  transparency.   The computed distance is related to the
	   unit fractional distance in 3D space between the RGB values of  the
	   key color and the pixel's color. Range is 0.01 to 1.0. 0.01 matches
	   within  a  very  small radius around the exact key color, while 1.0
	   matches everything.	Default is 0.01.

       blend
	   Set how the alpha value for pixels that fall outside the similarity
	   radius is computed.	0.0 makes pixels either fully  transparent  or
	   fully  opaque.   Higher  values  result in semi-transparent pixels,
	   with greater transparency the more similar the pixel	 color	is  to
	   the key color.  Range is 0.0 to 1.0. Default is 0.0.

       Examples

       •   Make every green pixel in the input image transparent:

		   ffmpeg -i input.png -vf colorkey=green out.png

       •   Overlay a greenscreen-video on top of a static background image.

		   ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

   colorhold
       Remove all color information for all RGB colors except for certain one.

       The filter accepts the following options:

       color
	   The color which will not be replaced with neutral gray.

       similarity
	   Similarity percentage with the above color.	0.01 matches only  the
	   exact key color, while 1.0 matches everything.

       blend
	   Blend  percentage.  0.0  makes  pixels  fully  gray.	 Higher values
	   result in more preserved color.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If the specified expression is not valid, it is	kept  at  its  current
       value.

   colorlevels
       Adjust video input frames using levels.

       The filter accepts the following options:

       rimin
       gimin
       bimin
       aimin
	   Adjust  red,	 green,	 blue  and  alpha  input black point.  Allowed
	   ranges for options are "[-1.0, 1.0]". Defaults are 0.

       rimax
       gimax
       bimax
       aimax
	   Adjust red, green, blue  and	 alpha	input  white  point.   Allowed
	   ranges for options are "[-1.0, 1.0]". Defaults are 1.

	   Input  levels are used to lighten highlights (bright tones), darken
	   shadows (dark tones), change the balance of bright and dark tones.

       romin
       gomin
       bomin
       aomin
	   Adjust red, green, blue and	alpha  output  black  point.   Allowed
	   ranges for options are "[0, 1.0]". Defaults are 0.

       romax
       gomax
       bomax
       aomax
	   Adjust  red,	 green,	 blue  and  alpha output white point.  Allowed
	   ranges for options are "[0, 1.0]". Defaults are 1.

	   Output levels allows manual selection of a constrained output level
	   range.

       preserve
	   Set preserve color mode. The accepted values are:

	   none
	       Disable color preserving, this is default.

	   lum Preserve luminance.

	   max Preserve max value of RGB triplet.

	   avg Preserve average value of RGB triplet.

	   sum Preserve sum value of RGB triplet.

	   nrm Preserve normalized value of RGB triplet.

	   pwr Preserve power value of RGB triplet.

       Examples

       •   Make video output darker:

		   colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

       •   Increase contrast:

		   colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

       •   Make video output lighter:

		   colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

       •   Increase brightness:

		   colorlevels=romin=0.5:gomin=0.5:bomin=0.5

       Commands

       This filter supports the all above options as commands.

   colormap
       Apply custom color maps to video stream.

       This filter needs three input video streams.   First  stream  is	 video
       stream that is going to be filtered out.	 Second and third video stream
       specify color patches for source color to target color mapping.

       The filter accepts the following options:

       patch_size
	   Set the source and target video stream patch size in pixels.

       nb_patches
	   Set	the  max  number  of used patches from source and target video
	   stream.  Default value is number of patches available in additional
	   video streams.  Max allowed number of patches is 64.

       type
	   Set the adjustments used for target colors. Can  be	"relative"  or
	   "absolute".	Defaults is "absolute".

       kernel
	   Set	the  kernel  used  to measure color differences between mapped
	   colors.

	   The accepted values are:

	   euclidean
	   weuclidean

	   Default is "euclidean".

   colormatrix
       Convert color matrix.

       The filter accepts the following options:

       src
       dst Specify the source and destination color matrix. Both  values  must
	   be specified.

	   The accepted values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt601
	       BT.601

	   bt470
	       BT.470

	   bt470bg
	       BT.470BG

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       For example to convert from BT.601 to SMPTE-240M, use the command:

	       colormatrix=bt601:smpte240m

   colorspace
       Convert colorspace, transfer characteristics or color primaries.	 Input
       video needs to have an even size.

       The filter accepts the following options:

       all Specify all color properties at once.

	   The accepted values are:

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   bt601-6-525
	       BT.601-6 525

	   bt601-6-625
	       BT.601-6 625

	   bt709
	       BT.709

	   smpte170m
	       SMPTE-170M

	   smpte240m
	       SMPTE-240M

	   bt2020
	       BT.2020

       space
	   Specify output colorspace.

	   The accepted values are:

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   ycgco
	       YCgCo

	   bt2020ncl
	       BT.2020 with non-constant luminance

       trc Specify output transfer characteristics.

	   The accepted values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG

	   gamma22
	       Constant gamma of 2.2

	   gamma28
	       Constant gamma of 2.8

	   smpte170m
	       SMPTE-170M, BT.601-6 625 or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   srgb
	       SRGB

	   iec61966-2-1
	       iec61966-2-1

	   iec61966-2-4
	       iec61966-2-4

	   xvycc
	       xvycc

	   bt2020-10
	       BT.2020 for 10-bits content

	   bt2020-12
	       BT.2020 for 12-bits content

       primaries
	   Specify output color primaries.

	   The accepted values are:

	   bt709
	       BT.709

	   bt470m
	       BT.470M

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   film
	       film

	   smpte431
	       SMPTE-431

	   smpte432
	       SMPTE-432

	   bt2020
	       BT.2020

	   jedec-p22
	       JEDEC P22 phosphors

       range
	   Specify output color range.

	   The accepted values are:

	   tv  TV (restricted) range

	   mpeg
	       MPEG (restricted) range

	   pc  PC (full) range

	   jpeg
	       JPEG (full) range

       format
	   Specify output color format.

	   The accepted values are:

	   yuv420p
	       YUV 4:2:0 planar 8-bits

	   yuv420p10
	       YUV 4:2:0 planar 10-bits

	   yuv420p12
	       YUV 4:2:0 planar 12-bits

	   yuv422p
	       YUV 4:2:2 planar 8-bits

	   yuv422p10
	       YUV 4:2:2 planar 10-bits

	   yuv422p12
	       YUV 4:2:2 planar 12-bits

	   yuv444p
	       YUV 4:4:4 planar 8-bits

	   yuv444p10
	       YUV 4:4:4 planar 10-bits

	   yuv444p12
	       YUV 4:4:4 planar 12-bits

       fast
	   Do  a  fast	conversion, which skips gamma/primary correction. This
	   will take  significantly  less  CPU,	 but  will  be	mathematically
	   incorrect.  To  get	output	compatible  with  that produced by the
	   colormatrix filter, use fast=1.

       dither
	   Specify dithering mode.

	   The accepted values are:

	   none
	       No dithering

	   fsb Floyd-Steinberg dithering

       wpadapt
	   Whitepoint adaptation mode.

	   The accepted values are:

	   bradford
	       Bradford whitepoint adaptation

	   vonkries
	       von Kries whitepoint adaptation

	   identity
	       identity whitepoint adaptation (i.e. no whitepoint adaptation)

       iall
	   Override all input properties at once. Same accepted values as all.

       ispace
	   Override input colorspace. Same accepted values as space.

       iprimaries
	   Override input color primaries. Same accepted values as primaries.

       itrc
	   Override input transfer characteristics. Same  accepted  values  as
	   trc.

       irange
	   Override input color range. Same accepted values as range.

       The filter converts the transfer characteristics, color space and color
       primaries  to  the  specified  user  values.  The  output value, if not
       specified, is set to a default value based on the  "all"	 property.  If
       that  property is also not specified, the filter will log an error. The
       output color range and format default to the same value	as  the	 input
       color  range  and  format.  The	input  transfer characteristics, color
       space, color primaries and color range should be set on the input data.
       If any of these are missing, the	 filter	 will  log  an	error  and  no
       conversion will take place.

       For example to convert the input to SMPTE-240M, use the command:

	       colorspace=smpte240m

   colorspace_cuda
       CUDA accelerated implementation of the colorspace filter.

       It  is by no means feature complete compared to the software colorspace
       filter, and at the current time only supports  color  range  conversion
       between jpeg/full and mpeg/limited range.

       The filter accepts the following options:

       range
	   Specify output color range.

	   The accepted values are:

	   tv  TV (restricted) range

	   mpeg
	       MPEG (restricted) range

	   pc  PC (full) range

	   jpeg
	       JPEG (full) range

   colortemperature
       Adjust  color  temperature  in  video to simulate variations in ambient
       color temperature.

       The filter accepts the following options:

       temperature
	   Set the temperature in Kelvin. Allowed range is from 1000 to 40000.
	   Default value is 6500 K.

       mix Set mixing with filtered output. Allowed range  is  from  0	to  1.
	   Default value is 1.

       pl  Set	the amount of preserving lightness. Allowed range is from 0 to
	   1.  Default value is 0.

       Commands

       This filter supports same commands as options.

   convolution
       Apply convolution of 3x3, 5x5, 7x7  or  horizontal/vertical  up	to  49
       elements.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set	matrix	for  each  plane.   Matrix  is sequence of 9, 25 or 49
	   signed integers in square mode, and from 1  to  49  odd  number  of
	   signed integers in row mode.

       0rdiv
       1rdiv
       2rdiv
       3rdiv
	   Set multiplier for calculated value for each plane.	If unset or 0,
	   it will be sum of all matrix elements.

       0bias
       1bias
       2bias
       3bias
	   Set	bias  for each plane. This value is added to the result of the
	   multiplication.  Useful for making the overall  image  brighter  or
	   darker. Default is 0.0.

       0mode
       1mode
       2mode
       3mode
	   Set	matrix	mode  for  each	 plane.	 Can be square, row or column.
	   Default is square.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Apply sharpen:

		   convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"

       •   Apply blur:

		   convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"

       •   Apply edge enhance:

		   convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"

       •   Apply edge detect:

		   convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"

       •   Apply laplacian edge detector which includes diagonals:

		   convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"

       •   Apply emboss:

		   convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"

   convolve
       Apply 2D convolution of video stream in frequency domain	 using	second
       stream as impulse.

       The filter accepts the following options:

       planes
	   Set which planes to process.

       impulse
	   Set	which  impulse video frames will be processed, can be first or
	   all. Default is all.

       The "convolve" filter also supports the framesync options.

   copy
       Copy the input video source unchanged to the  output.  This  is	mainly
       useful for testing purposes.

   coreimage
       Video filtering on GPU using Apple's CoreImage API on OSX.

       Hardware	 acceleration  is  based  on  an OpenGL context. Usually, this
       means it is processed by video hardware. However, software-based OpenGL
       implementations exist which means there is no  guarantee	 for  hardware
       processing. It depends on the respective OSX.

       There are many filters and image generators provided by Apple that come
       with a large variety of options. The filter has to be referenced by its
       name along with its options.

       The coreimage filter accepts the following options:

       list_filters
	   List	 all  available	 filters  and  generators along with all their
	   respective options as well as possible minimum and  maximum	values
	   along with the default values.

		   list_filters=true

       filter
	   Specify  all	 filters  by  their  respective name and options.  Use
	   list_filters to determine  all  valid  filter  names	 and  options.
	   Numerical   options	 are  specified	 by  a	float  value  and  are
	   automatically clamped to their respective value range.  Vector  and
	   color  options  have	 to  be specified by a list of space separated
	   float values. Character escaping has to be done.  A special	option
	   name "default" is available to use default options for a filter.

	   It  is  required to specify either "default" or at least one of the
	   filter options.  All omitted options are used  with	their  default
	   values.  The syntax of the filter string is as follows:

		   filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]

       output_rect
	   Specify  a rectangle where the output of the filter chain is copied
	   into the input image. It is given by	 a  list  of  space  separated
	   float values:

		   output_rect=x\ y\ width\ height

	   If  not  given,  the	 output rectangle equals the dimensions of the
	   input image.	 The output rectangle is automatically cropped at  the
	   borders  of	the  input  image.  Negative values are valid for each
	   component.

		   output_rect=25\ 25\ 100\ 100

       Several filters can be chained for successive processing	 without  GPU-
       HOST  transfers	allowing for fast processing of complex filter chains.
       Currently, only filters with zero (generators) or exactly one (filters)
       input image and	one  output  image  are	 supported.  Also,  transition
       filters are not yet usable as intended.

       Some  filters  generate output images with additional padding depending
       on the respective filter kernel. The padding is	automatically  removed
       to ensure the filter output has the same size as the input image.

       For image generators, the size of the output image is determined by the
       previous	 output	 image	of  the filter chain or the input image of the
       whole filterchain, respectively. The generators do not  use  the	 pixel
       information  of	this  image  to	 generate  their  output. However, the
       generated output is blended onto this image, resulting  in  partial  or
       complete coverage of the output image.

       The  coreimagesrc  video source can be used for generating input images
       which are directly fed into the filter chain. By	 using	it,  providing
       input images by another video source or an input video is not required.

       Examples

       •   List all filters available:

		   coreimage=list_filters=true

       •   Use the CIBoxBlur filter with default options to blur an image:

		   coreimage=filter=CIBoxBlur@default

       •   Use	 a  filter  chain  with	 CISepiaTone  at  default  values  and
	   CIVignetteEffect with its center at 100x100	and  a	radius	of  50
	   pixels:

		   coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50

       •   Use	nullsrc	 and  CIQRCodeGenerator	 to  create  a QR code for the
	   FFmpeg homepage, given as complete  and  escaped  command-line  for
	   Apple's standard bash shell:

		   ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

   corr
       Obtain the correlation between two input videos.

       This filter takes two input videos.

       Both  input  videos  must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that  both  inputs  have
       the same number of frames, which are compared one by one.

       The obtained per component, average, min and max correlation is printed
       through the logging system.

       The  filter  stores  the	 calculated correlation of each frame in frame
       metadata.

       This filter also supports the framesync options.

       In the below  example  the  input  file	main.mpg  being	 processed  is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi corr -f null -

   cover_rect
       Cover a rectangular object

       It accepts the following options:

       cover
	   Filepath of the optional cover image, needs to be in yuv420.

       mode
	   Set covering mode.

	   It accepts the following values:

	   cover
	       cover it by the supplied image

	   blur
	       cover it by interpolating the surrounding pixels

	   Default value is blur.

       Examples

       •   Cover  a  rectangular object by the supplied image of a given video
	   using ffmpeg:

		   ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

   crop
       Crop the input video to given dimensions.

       It accepts the following parameters:

       w, out_w
	   The	width  of  the	output	video.	It  defaults  to  "iw".	  This
	   expression  is evaluated only once during the filter configuration,
	   or when the w or out_w command is sent.

       h, out_h
	   The height  of  the	output	video.	It  defaults  to  "ih".	  This
	   expression  is evaluated only once during the filter configuration,
	   or when the h or out_h command is sent.

       x   The horizontal position, in the input video, of the	left  edge  of
	   the output video. It defaults to "(in_w-out_w)/2".  This expression
	   is evaluated per-frame.

       y   The	vertical  position, in the input video, of the top edge of the
	   output video.  It defaults to "(in_h-out_h)/2".  This expression is
	   evaluated per-frame.

       keep_aspect
	   If set to 1 will force the output display aspect ratio  to  be  the
	   same	 of  the input, by changing the output sample aspect ratio. It
	   defaults to 0.

       exact
	   Enable exact	 cropping.  If	enabled,  subsampled  videos  will  be
	   cropped  at	exact  width/height/x/y	 as  specified and will not be
	   rounded to nearest smaller value.  It defaults to 0.

       The out_w, out_h,  x,  y	 parameters  are  expressions  containing  the
       following constants:

       x
       y   The	computed  values  for x and y. They are evaluated for each new
	   frame.

       in_w
       in_h
	   The input width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (cropped) width and height.

       ow
       oh  These are the same as out_w and out_h.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
	   horizontal and vertical chroma subsample values.  For  example  for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       n   The number of the input frame, starting from 0.

       pos the	position  in  the  file	 of  the  input frame, NAN if unknown;
	   deprecated, do not use

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
	   is unknown.

       The expression for out_w may depend on the  value  of  out_h,  and  the
       expression  for	out_h may depend on out_w, but they cannot depend on x
       and y, as x and y are evaluated after out_w and out_h.

       The x and y parameters specify the expressions for the position of  the
       top-left	 corner	 of  the output (non-cropped) area. They are evaluated
       for each frame. If the evaluated value is not valid, it is approximated
       to the nearest valid value.

       The expression for x may depend on y, and  the  expression  for	y  may
       depend on x.

       Examples

       •   Crop area with size 100x100 at position (12,34).

		   crop=100:100:12:34

	   Using named options, the example above becomes:

		   crop=w=100:h=100:x=12:y=34

       •   Crop the central input area with size 100x100:

		   crop=100:100

       •   Crop the central input area with size 2/3 of the input video:

		   crop=2/3*in_w:2/3*in_h

       •   Crop the input video central square:

		   crop=out_w=in_h
		   crop=in_h

       •   Delimit  the	 rectangle with the top-left corner placed at position
	   100:100 and the right-bottom corner	corresponding  to  the	right-
	   bottom corner of the input image.

		   crop=in_w-100:in_h-100:100:100

       •   Crop	 10 pixels from the left and right borders, and 20 pixels from
	   the top and bottom borders

		   crop=in_w-2*10:in_h-2*20

       •   Keep only the bottom right quarter of the input image:

		   crop=in_w/2:in_h/2:in_w/2:in_h/2

       •   Crop height for getting Greek harmony:

		   crop=in_w:1/PHI*in_w

       •   Apply trembling effect:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

       •   Apply erratic camera effect depending on timestamp:

		   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)

       •   Set x depending on the value of y:

		   crop=in_w/2:in_h/2:y:10+10*sin(n/10)

       Commands

       This filter supports the following commands:

       w, out_w
       h, out_h
       x
       y   Set width/height of the output video	 and  the  horizontal/vertical
	   position  in	 the input video.  The command accepts the same syntax
	   of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   cropdetect
       Auto-detect the crop size.

       It  calculates  the  necessary  cropping	 parameters  and  prints   the
       recommended  parameters via the logging system. The detected dimensions
       correspond to the non-black or video area of the input video  according
       to mode.

       It accepts the following parameters:

       mode
	   Depending  on mode crop detection is based on either the mere black
	   value of surrounding pixels or a combination of motion vectors  and
	   edge pixels.

	   black
	       Detect  black  pixels  surrounding  the playing video. For fine
	       control use option limit.

	   mvedges
	       Detect the playing video by the motion vectors inside the video
	       and scanning for edge pixels typically forming the border of  a
	       playing video.

       limit
	   Set higher black value threshold, which can be optionally specified
	   from	 nothing  (0)  to everything (255 for 8-bit based formats). An
	   intensity value greater to the set value is	considered  non-black.
	   It  defaults	 to  24.  You can also specify a value between 0.0 and
	   1.0 which will be scaled depending on the  bitdepth	of  the	 pixel
	   format.

       round
	   The	value  which  the  width/height	 should	 be  divisible	by. It
	   defaults to 16. The offset is automatically adjusted to center  the
	   video.  Use 2 to get only even dimensions (needed for 4:2:2 video).
	   16 is best when encoding to most video codecs.

       skip
	   Set the number of initial frames for which evaluation  is  skipped.
	   Default is 2. Range is 0 to INT_MAX.

       reset_count, reset
	   Set	the  counter  that determines after how many frames cropdetect
	   will reset the previously detected largest  video  area  and	 start
	   over to detect the current optimal crop area. Default value is 0.

	   This	 can  be  useful  when channel logos distort the video area. 0
	   indicates 'never reset', and returns the largest  area  encountered
	   during playback.

       mv_threshold
	   Set	motion	in  pixel  units as threshold for motion detection. It
	   defaults to 8.

       low
       high
	   Set low and high threshold values used by  the  Canny  thresholding
	   algorithm.

	   The high threshold selects the "strong" edge pixels, which are then
	   connected  through  8-connectivity  with  the  "weak"  edge	pixels
	   selected by the low threshold.

	   low and high threshold values must be chosen in  the	 range	[0,1],
	   and low should be lesser or equal to high.

	   Default  value  for	low  is "5/255", and default value for high is
	   "15/255".

       Examples

       •   Find video area surrounded by black borders:

		   ffmpeg -i file.mp4 -vf cropdetect,metadata=mode=print -f null -

       •   Find an embedded video area, generate motion vectors beforehand:

		   ffmpeg -i file.mp4 -vf mestimate,cropdetect=mode=mvedges,metadata=mode=print -f null -

       •   Find an embedded video area, use motion vectors from decoder:

		   ffmpeg -flags2 +export_mvs -i file.mp4 -vf cropdetect=mode=mvedges,metadata=mode=print -f null -

       Commands

       This filter supports the following commands:

       limit
	   The command accepts the same syntax of  the	corresponding  option.
	   If the specified expression is not valid, it is kept at its current
	   value.

   cue
       Delay  video  filtering	until  a given wallclock timestamp. The filter
       first passes on preroll amount of  frames,  then	 it  buffers  at  most
       buffer  amount  of frames and waits for the cue. After reaching the cue
       it forwards the buffered frames and also any subsequent	frames	coming
       in its input.

       The  filter  can	 be  used  synchronize	the  output of multiple ffmpeg
       processes for realtime output devices like  decklink.  By  putting  the
       delay  in  the filtering chain and pre-buffering frames the process can
       pass on data to output almost immediately after	the  target  wallclock
       timestamp is reached.

       Perfect	frame  accuracy	 cannot	 be guaranteed, but the result is good
       enough for some use cases.

       cue The cue timestamp expressed in a UNIX  timestamp  in	 microseconds.
	   Default is 0.

       preroll
	   The duration of content to pass on as preroll expressed in seconds.
	   Default is 0.

       buffer
	   The	maximum	 duration  of content to buffer before waiting for the
	   cue expressed in seconds. Default is 0.

   curves
       Apply color adjustments using curves.

       This filter is similar to the Adobe Photoshop and  GIMP	curves	tools.
       Each  component	(red,  green and blue) has its values defined by N key
       points tied from each other using a smooth curve. The x-axis represents
       the pixel values from the input frame, and the  y-axis  the  new	 pixel
       values to be set for the output frame.

       By  default,  a	component curve is defined by the two points (0;0) and
       (1;1). This creates a straight line where each original pixel value  is
       "adjusted" to its own value, which means no change to the image.

       The filter allows you to redefine these two points and add some more. A
       new  curve  will	 be  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. The curve is formed by using a	 natural  or  monotonic	 cubic
       spline	interpolation,	 depending  on	the  interp  option  (default:
       "natural"). The "natural" spline produces a smoother curve  in  general
       while the monotonic ("pchip") spline guarantees the transitions between
       the  specified  points to be monotonic. If the computed curves happened
       to  go  outside	the  vector  spaces,  the  values  will	  be   clipped
       accordingly.

       The filter accepts the following options:

       preset
	   Select  one of the available color presets. This option can be used
	   in addition to the r, g, b parameters;  in  this  case,  the	 later
	   options  takes  priority  on	 the preset values.  Available presets
	   are:

	   none
	   color_negative
	   cross_process
	   darker
	   increase_contrast
	   lighter
	   linear_contrast
	   medium_contrast
	   negative
	   strong_contrast
	   vintage

	   Default is "none".

       master, m
	   Set the master key points. These points will define a  second  pass
	   mapping.  It	 is sometimes called a "luminance" or "value" mapping.
	   It can be used with r, g, b or all  since  it  acts	like  a	 post-
	   processing LUT.

       red, r
	   Set the key points for the red component.

       green, g
	   Set the key points for the green component.

       blue, b
	   Set the key points for the blue component.

       all Set	the key points for all components (not including master).  Can
	   be used in addition to the other key points component  options.  In
	   this	 case,	the  unset  component(s)  will	fallback  on  this all
	   setting.

       psfile
	   Specify a Photoshop curves file (".acv")  to	 import	 the  settings
	   from.

       plot
	   Save Gnuplot script of the curves in specified file.

       interp
	   Specify the kind of interpolation. Available algorithms are:

	   natural
	       Natural	cubic  spline using a piece-wise cubic polynomial that
	       is twice continuously differentiable.

	   pchip
	       Monotonic  cubic	 spline	 using	a  piecewise   cubic   Hermite
	       interpolating polynomial (PCHIP).

       To  avoid  some filtergraph syntax conflicts, each key points list need
       to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

       Commands

       This filter supports same commands as options.

       Examples

       •   Increase slightly the middle level of blue:

		   curves=blue='0/0 0.5/0.58 1/1'

       •   Vintage effect:

		   curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'

	   Here we obtain the following coordinates for each components:

	   red "(0;0.11) (0.42;0.51) (1;0.95)"

	   green
	       "(0;0) (0.50;0.48) (1;1)"

	   blue
	       "(0;0.22) (0.49;0.44) (1;0.80)"

       •   The previous example can  also  be  achieved	 with  the  associated
	   built-in preset:

		   curves=preset=vintage

       •   Or simply:

		   curves=vintage

       •   Use	a  Photoshop  preset  and  redefine  the  points  of the green
	   component:

		   curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

       •   Check out the curves of the "cross_process"	profile	 using	ffmpeg
	   and gnuplot:

		   ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
		   gnuplot -p /tmp/curves.plt

   datascope
       Video data analysis filter.

       This filter shows hexadecimal pixel values of part of video.

       The filter accepts the following options:

       size, s
	   Set output video size.

       x   Set x offset from where to pick pixels.

       y   Set y offset from where to pick pixels.

       mode
	   Set scope mode, can be one of the following:

	   mono
	       Draw  hexadecimal  pixel	 values	 with  white  color  on	 black
	       background.

	   color
	       Draw hexadecimal pixel values with input video pixel  color  on
	       black background.

	   color2
	       Draw  hexadecimal  pixel values on color background picked from
	       input video, the text color is picked in such way so its always
	       visible.

       axis
	   Draw rows and columns numbers on left and top of video.

       opacity
	   Set background opacity.

       format
	   Set display number format. Can  be  "hex",  or  "dec".  Default  is
	   "hex".

       components
	   Set	pixel  components  to display. By default all pixel components
	   are displayed.

       Commands

       This filter supports same commands as options excluding "size" option.

   dblur
       Apply Directional blur filter.

       The filter accepts the following options:

       angle
	   Set angle of directional blur. Default is 45.

       radius
	   Set radius of directional blur. Default is 5.

       planes
	   Set which planes to filter. By default all planes are filtered.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If the specified expression is not valid, it is	kept  at  its  current
       value.

   dctdnoiz
       Denoise frames using 2D DCT (frequency domain filtering).

       This filter is not designed for real time.

       The filter accepts the following options:

       sigma, s
	   Set the noise sigma constant.

	   This	 sigma	defines	 a  hard  threshold  of "3 * sigma"; every DCT
	   coefficient (absolute value) below this threshold with be dropped.

	   If you need a more advanced filtering, see expr.

	   Default is 0.

       overlap
	   Set number overlapping pixels for each block. Since the filter  can
	   be  slow,  you may want to reduce this value, at the cost of a less
	   effective filter and the risk of various artefacts.

	   If the overlapping value doesn't permit processing the whole	 input
	   width  or height, a warning will be displayed and according borders
	   won't be denoised.

	   Default value is blocksize-1, which is the best possible setting.

       expr, e
	   Set the coefficient factor expression.

	   For each coefficient of  a  DCT  block,  this  expression  will  be
	   evaluated as a multiplier value for the coefficient.

	   If this is option is set, the sigma option will be ignored.

	   The absolute value of the coefficient can be accessed through the c
	   variable.

       n   Set	the  blocksize	using  the  number of bits. "1<<n" defines the
	   blocksize, which is the width and height of the processed blocks.

	   The default value is 3 (8x8) and can be raised to 4 for a blocksize
	   of 16x16. Note that changing this setting has huge consequences  on
	   the	 speed	 processing.  Also,  a	larger	block  size  does  not
	   necessarily means a better de-noising.

       Examples

       Apply a denoise with a sigma of 4.5:

	       dctdnoiz=4.5

       The same operation can be achieved using the expression system:

	       dctdnoiz=e='gte(c, 4.5*3)'

       Violent denoise using a block size of "16x16":

	       dctdnoiz=15:n=4

   deband
       Remove banding artifacts from  input  video.   It  works	 by  replacing
       banded pixels with average value of referenced pixels.

       The filter accepts the following options:

       1thr
       2thr
       3thr
       4thr
	   Set	banding	 detection  threshold for each plane. Default is 0.02.
	   Valid range is 0.00003 to 0.5.  If difference between current pixel
	   and reference pixel is less than threshold, it will	be  considered
	   as banded.

       range, r
	   Banding  detection  range  in  pixels.  Default is 16. If positive,
	   random number in range 0 to set value will be  used.	 If  negative,
	   exact  absolute  value  will	 be used.  The range defines square of
	   four pixels around current pixel.

       direction, d
	   Set direction in radians from which four pixel will be compared. If
	   positive, random direction from 0 to set direction will be  picked.
	   If  negative,  exact	 of absolute value will be picked. For example
	   direction 0, -PI or -2*PI radians will pick only pixels on same row
	   and -PI/2 will pick only pixels on same column.

       blur, b
	   If enabled, current pixel is compared with  average	value  of  all
	   four	 surrounding  pixels.  The  default  is	 enabled.  If disabled
	   current pixel is compared with all  four  surrounding  pixels.  The
	   pixel  is  considered  banded  if  only  all	 four differences with
	   surrounding pixels are less than threshold.

       coupling, c
	   If enabled, current pixel is changed	 if  and  only	if  all	 pixel
	   components	are   banded,  e.g.  banding  detection	 threshold  is
	   triggered for all color components.	The default is disabled.

       Commands

       This filter supports the all above options as commands.

   deblock
       Remove blocking artifacts from input video.

       The filter accepts the following options:

       filter
	   Set filter type, can be weak or strong. Default  is	strong.	  This
	   controls what kind of deblocking is applied.

       block
	   Set size of block, allowed range is from 4 to 512. Default is 8.

       alpha
       beta
       gamma
       delta
	   Set	blocking  detection  thresholds.  Allowed  range  is  0	 to 1.
	   Defaults are: 0.098 for alpha and 0.05 for the rest.	 Using	higher
	   threshold  gives  more deblocking strength.	Setting alpha controls
	   threshold detection at exact	 edge  of  block.   Remaining  options
	   controls   threshold	  detection   near  the	 edge.	Each  one  for
	   below/above or left/right. Setting  any  of	those  to  0  disables
	   deblocking.

       planes
	   Set planes to filter. Default is to filter all available planes.

       Examples

       •   Deblock using weak filter and block size of 4 pixels.

		   deblock=filter=weak:block=4

       •   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

       •   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

       •   Similar as above, but filter only second and third plane.

		   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6

       Commands

       This filter supports the all above options as commands.

   decimate
       Drop duplicated frames at regular intervals.

       The filter accepts the following options:

       cycle
	   Set the number of frames from which one will	 be  dropped.  Setting
	   this	 to  N	means  one  frame  in  every batch of N frames will be
	   dropped.  Default is 5.

       dupthresh
	   Set the threshold for duplicate detection. If the difference metric
	   for a frame is less than  or	 equal	to  this  value,  then	it  is
	   declared as duplicate. Default is 1.1

       scthresh
	   Set scene change threshold. Default is 15.

       blockx
       blocky
	   Set	the  size  of  the  x  and  y-axis  blocks  used during metric
	   calculations.  Larger blocks give  better  noise  suppression,  but
	   also	 give  worse  detection of small movements. Must be a power of
	   two. Default is 32.

       ppsrc
	   Mark main input as a pre-processed input and activate clean	source
	   input  stream.  This	 allows	 the  input  to	 be pre-processed with
	   various filters to help the metrics calculation while  keeping  the
	   frame  selection  lossless.	When set to 1, the first stream is for
	   the pre-processed input, and the second stream is the clean	source
	   from where the kept frames are chosen. Default is 0.

       chroma
	   Set whether or not chroma is considered in the metric calculations.
	   Default is 1.

       mixed
	   Set	whether or not the input only partially contains content to be
	   decimated.  Default is "false".  If	enabled	 video	output	stream
	   will be in variable frame rate.

   deconvolve
       Apply 2D deconvolution of video stream in frequency domain using second
       stream as impulse.

       The filter accepts the following options:

       planes
	   Set which planes to process.

       impulse
	   Set	which  impulse video frames will be processed, can be first or
	   all. Default is all.

       noise
	   Set noise when doing divisions. Default is 0.0000001.  Useful  when
	   width and height are not same and not power of 2 or if stream prior
	   to convolving had noise.

       The "deconvolve" filter also supports the framesync options.

   dedot
       Reduce  cross-luminance	(dot-crawl)  and  cross-color  (rainbows) from
       video.

       It accepts the following options:

       m   Set mode of operation. Can be combination of	 dotcrawl  for	cross-
	   luminance reduction and/or rainbows for cross-color reduction.

       lt  Set	spatial	 luma  threshold.  Lower values increases reduction of
	   cross-luminance.

       tl  Set tolerance for temporal luma. Higher values increases  reduction
	   of cross-luminance.

       tc  Set	 tolerance   for  chroma  temporal  variation.	Higher	values
	   increases reduction of cross-color.

       ct  Set temporal chroma threshold. Lower values increases reduction  of
	   cross-color.

   deflate
       Apply deflate effect to the video.

       This filter replaces the pixel by the local(3x3) average by taking into
       account only values lower than the pixel.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit  the  maximum change for each plane, default is 65535.	 If 0,
	   plane will remain unchanged.

       Commands

       This filter supports the all above options as commands.

   deflicker
       Remove temporal frame luminance variations.

       It accepts the following options:

       size, s
	   Set moving-average filter size in frames.  Default  is  5.  Allowed
	   range is 2 - 129.

       mode, m
	   Set averaging mode to smooth temporal luminance variations.

	   Available values are:

	   am  Arithmetic mean

	   gm  Geometric mean

	   hm  Harmonic mean

	   qm  Quadratic mean

	   cm  Cubic mean

	   pm  Power mean

	   median
	       Median

       bypass
	   Do not actually modify frame. Useful when one only wants metadata.

   dejudder
       Remove judder produced by partially interlaced telecined content.

       Judder  can  be	introduced,  for  instance,  by	 pullup filter. If the
       original source was partially telecined	content	 then  the  output  of
       "pullup,dejudder"  will	have  a	 variable  frame  rate. May change the
       recorded frame rate of the container.  Aside  from  that	 change,  this
       filter will not affect constant frame rate video.

       The option available in this filter is:

       cycle
	   Specify the length of the window over which the judder repeats.

	   Accepts any integer greater than 1. Useful values are:

	   4   If the original was telecined from 24 to 30 fps (Film to NTSC).

	   5   If the original was telecined from 25 to 30 fps (PAL to NTSC).

	   20  If a mixture of the two.

	   The default is 4.

   delogo
       Suppress a TV station logo by a simple interpolation of the surrounding
       pixels.	Just  set a rectangle covering the logo and watch it disappear
       (and sometimes something even uglier appear - your mileage may vary).

       It accepts the following parameters:

       x
       y   Specify the top left corner coordinates of the logo. They  must  be
	   specified.

       w
       h   Specify  the	 width	and  height of the logo to clear. They must be
	   specified.

       show
	   When set to 1, a green rectangle is drawn on the screen to simplify
	   finding the right x, y, w, and h parameters.	 The default value  is
	   0.

	   The	rectangle  is  drawn  on  the  outermost  pixels which will be
	   (partly) replaced with interpolated values. The values of the  next
	   pixels immediately outside this rectangle in each direction will be
	   used to compute the interpolated pixel values inside the rectangle.

       Examples

       •   Set	a rectangle covering the area with top left corner coordinates
	   0,0 and size 100x77:

		   delogo=x=0:y=0:w=100:h=77

   derain
       Remove the rain in the input image/video by applying the derain methods
       based on convolutional neural networks. Supported models:

       •   Recurrent Squeeze-and-Excitation Context Aggregation Net  (RESCAN).
	   See
	   <http://openaccess.thecvf.com/content_ECCV_2018/papers/Xia_Li_Recurrent_Squeeze-and-Excitation_Context_ECCV_2018_paper.pdf>.

       Training	 as  well  as  model  generation  scripts  are provided in the
       repository at <https://github.com/XueweiMeng/derain_filter.git>.

       The filter accepts the following options:

       filter_type
	   Specify which filter to use.	 This  option  accepts	the  following
	   values:

	   derain
	       Derain  filter.	To  conduct  derain  filter, you need to use a
	       derain model.

	   dehaze
	       Dehaze filter. To conduct dehaze filter,	 you  need  to	use  a
	       dehaze model.

	   Default value is derain.

       dnn_backend
	   Specify  which  DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to  install
	       the	 TensorFlow	  for	    C	    library	  (see
	       <https://www.tensorflow.org/install/lang_c>)   and    configure
	       FFmpeg with "--enable-libtensorflow"

       model
	   Set	path  to  model	 file  specifying network architecture and its
	   parameters.	 Note  that  different	backends  use  different  file
	   formats. TensorFlow can load files for only its format.

       To  get	full  functionality  (such as async execution), please use the
       dnn_processing filter.

   deshake
       Attempt to fix small changes in horizontal and/or vertical shift.  This
       filter  helps remove camera shake from hand-holding a camera, bumping a
       tripod, moving on a vehicle, etc.

       The filter accepts the following options:

       x
       y
       w
       h   Specify a rectangular area where to limit  the  search  for	motion
	   vectors.   If  desired the search for motion vectors can be limited
	   to a rectangular area of the frame defined by its top left  corner,
	   width  and  height.	These  parameters have the same meaning as the
	   drawbox filter which can be used to visualise the position  of  the
	   bounding box.

	   This	 is  useful  when simultaneous movement of subjects within the
	   frame might be confused for camera  motion  by  the	motion	vector
	   search.

	   If any or all of x, y, w and h are set to -1 then the full frame is
	   used.  This	allows	later options to be set without specifying the
	   bounding box for the motion vector search.

	   Default - search the whole frame.

       rx
       ry  Specify the maximum extent of movement in x and y directions in the
	   range 0-64 pixels. Default 16.

       edge
	   Specify how to generate pixels to fill blanks at the	 edge  of  the
	   frame. Available values are:

	   blank, 0
	       Fill zeroes at blank locations

	   original, 1
	       Original image at blank locations

	   clamp, 2
	       Extruded edge value at blank locations

	   mirror, 3
	       Mirrored edge at blank locations

	   Default value is mirror.

       blocksize
	   Specify the blocksize to use for motion search. Range 4-128 pixels,
	   default 8.

       contrast
	   Specify  the	 contrast  threshold for blocks. Only blocks with more
	   than	 the  specified	 contrast  (difference	between	 darkest   and
	   lightest pixels) will be considered. Range 1-255, default 125.

       search
	   Specify the search strategy. Available values are:

	   exhaustive, 0
	       Set exhaustive search

	   less, 1
	       Set less exhaustive search.

	   Default value is exhaustive.

       filename
	   If  set  then a detailed log of the motion search is written to the
	   specified file.

   despill
       Remove unwanted contamination of foreground colors, caused by reflected
       color of greenscreen or bluescreen.

       This filter accepts the following options:

       type
	   Set what type of despill to use.

       mix Set how spillmap will be generated.

       expand
	   Set how much to get rid of still remaining spill.

       red Controls amount of red in spill area.

       green
	   Controls  amount  of	 green	in  spill  area.   Should  be  -1  for
	   greenscreen.

       blue
	   Controls   amount  of  blue	in  spill  area.   Should  be  -1  for
	   bluescreen.

       brightness
	   Controls brightness of spill area, preserving colors.

       alpha
	   Modify alpha from generated spillmap.

       Commands

       This filter supports the all above options as commands.

   detelecine
       Apply an exact  inverse	of  the	 telecine  operation.  It  requires  a
       predefined pattern specified using the pattern option which must be the
       same as that passed to the telecine filter.

       This filter accepts the following options:

       first_field
	   top, t
	       top field first

	   bottom, b
	       bottom field first The default value is "top".

       pattern
	   A  string  of numbers representing the pulldown pattern you wish to
	   apply.  The default value is 23.

       start_frame
	   A number representing position of the first frame with  respect  to
	   the	telecine pattern. This is to be used if the stream is cut. The
	   default value is 0.

   dilation
       Apply dilation effect to the video.

       This filter replaces the pixel by the local(3x3) maximum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit the maximum change for each plane, default is 65535.	If  0,
	   plane will remain unchanged.

       coordinates
	   Flag which specifies the pixel to refer to. Default is 255 i.e. all
	   eight pixels are used.

	   Flags to local 3x3 coordinates maps like this:

	       1 2 3
	       4   5
	       6 7 8

       Commands

       This filter supports the all above options as commands.

   displace
       Displace pixels as indicated by second and third input stream.

       It takes three input streams and outputs one stream, the first input is
       the source, and second and third input are displacement maps.

       The  second  input  specifies  how  much	 to  displace pixels along the
       x-axis, while the third input specifies how  much  to  displace	pixels
       along  the y-axis.  If one of displacement map streams terminates, last
       frame from that displacement map will be used.

       Note that once generated, displacements maps can	 be  reused  over  and
       over again.

       A description of the accepted options follows.

       edge
	   Set displace behavior for pixels that are out of range.

	   Available values are:

	   blank
	       Missing pixels are replaced by black pixels.

	   smear
	       Adjacent pixels will spread out to replace missing pixels.

	   wrap
	       Out  of	range  pixels  are  wrapped so they point to pixels of
	       other side.

	   mirror
	       Out of range pixels will be replaced with mirrored pixels.

	   Default is smear.

       Examples

       •   Add ripple effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT

       •   Add wave effect to rgb input of video size hd720:

		   ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT

   dnn_classify
       Do classification with deep neural networks based on bounding boxes.

       The filter accepts the following options:

       dnn_backend
	   Specify which DNN backend to use for model loading  and  execution.
	   This	 option accepts only openvino now, tensorflow backends will be
	   added.

       model
	   Set path to model file  specifying  network	architecture  and  its
	   parameters.	 Note  that  different	backends  use  different  file
	   formats.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       confidence
	   Set the confidence threshold (default: 0.5).

       labels
	   Set path to label file specifying the mapping between label id  and
	   name.   Each	 label name is written in one line, tailing spaces and
	   empty lines are skipped.  The first line is the name of label id 0,
	   and the second line is the name of label id 1, etc.	The  label  id
	   is considered as name if the label file is not provided.

       backend_configs
	   Set the configs to be passed into backend

	   For	tensorflow  backend,  you can set its configs with sess_config
	   options,  please  use  tools/python/tf_sess_config.py  to  get  the
	   configs for your system.

   dnn_detect
       Do object detection with deep neural networks.

       The filter accepts the following options:

       dnn_backend
	   Specify  which  DNN backend to use for model loading and execution.
	   This option accepts only openvino now, tensorflow backends will  be
	   added.

       model
	   Set	path  to  model	 file  specifying network architecture and its
	   parameters.	 Note  that  different	backends  use  different  file
	   formats.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       confidence
	   Set the confidence threshold (default: 0.5).

       labels
	   Set	path to label file specifying the mapping between label id and
	   name.  Each label name is written in one line, tailing  spaces  and
	   empty  lines are skipped.  The first line is the name of label id 0
	   (usually it is 'background'), and the second line is	 the  name  of
	   label  id  1, etc.  The label id is considered as name if the label
	   file is not provided.

       backend_configs
	   Set the configs to be passed into backend. To use async  execution,
	   set	async  (default:  set).	  Roll	back  to sync execution if the
	   backend does not support async.

   dnn_processing
       Do image processing with deep neural networks. It works	together  with
       another filter which converts the pixel format of the Frame to what the
       dnn network requires.

       The filter accepts the following options:

       dnn_backend
	   Specify  which  DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to  install
	       the	 TensorFlow	  for	    C	    library	  (see
	       <https://www.tensorflow.org/install/lang_c>)   and    configure
	       FFmpeg with "--enable-libtensorflow"

	   openvino
	       OpenVINO	 backend. To enable this backend you need to build and
	       install	   the	   OpenVINO	for	C     library	  (see
	       <https://github.com/openvinotoolkit/openvino/blob/master/build-instruction.md>)
	       and     configure     FFmpeg	with	"--enable-libopenvino"
	       (--extra-cflags=-I... --extra-ldflags=-L... might be needed  if
	       the  header  files  and libraries are not installed into system
	       path)

       model
	   Set path to model file  specifying  network	architecture  and  its
	   parameters.	 Note  that  different	backends  use  different  file
	   formats. TensorFlow, OpenVINO backend can load files for  only  its
	   format.

       input
	   Set the input name of the dnn network.

       output
	   Set the output name of the dnn network.

       backend_configs
	   Set	the configs to be passed into backend. To use async execution,
	   set async (default: set).  Roll  back  to  sync  execution  if  the
	   backend does not support async.

	   For	tensorflow  backend,  you can set its configs with sess_config
	   options,  please  use  tools/python/tf_sess_config.py  to  get  the
	   configs of TensorFlow backend for your system.

       Examples

       •   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

       •   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

       •   Handle the Y channel with espcn.pb (see sr filter),	which  changes
	   frame  size,	 for  format  yuv420p  (planar YUV formats supported),
	   please use tools/python/tf_sess_config.py to	 get  the  configs  of
	   TensorFlow backend for your system.

		   ./ffmpeg -i 480p.jpg -vf format=yuv420p,dnn_processing=dnn_backend=tensorflow:model=espcn.pb:input=x:output=y:backend_configs=sess_config=0x10022805320e09cdccccccccccec3f20012a01303801 -y tmp.espcn.jpg

   drawbox
       Draw a colored box on the input image.

       It accepts the following parameters:

       x
       y   The	expressions  which  specify the top left corner coordinates of
	   the box. It defaults to 0.

       width, w
       height, h
	   The expressions which specify the width and height of the box; if 0
	   they are interpreted as the input width and height. It defaults  to
	   0.

       color, c
	   Specify  the	 color	of the box to write. For the general syntax of
	   this option, check the "Color" section in the ffmpeg-utils  manual.
	   If  the  special  value "invert" is used, the box edge color is the
	   same as the video with inverted luma.

       thickness, t
	   The expression which sets the thickness of the box edge.   A	 value
	   of "fill" will create a filled box. Default value is 3.

	   See below for the list of accepted constants.

       replace
	   Applicable  if the input has alpha. With value 1, the pixels of the
	   painted box will overwrite the  video's  color  and	alpha  pixels.
	   Default  is 0, which composites the box onto the input, leaving the
	   video's alpha intact.

       The parameters for x, y, w and h and t are expressions  containing  the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal  and  vertical  chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y offset coordinates where the box is drawn.

       w
       h   The width and height of the drawn box.

       box_source
	   Box source can be set as side_data_detection_bboxes if you want  to
	   use box data in detection bboxes of side data.

	   If  box_source  is  set, the x, y, width and height will be ignored
	   and still use box data in detection bboxes of side data. So	please
	   do  not  use	 this  parameter  if  you  were not sure about the box
	   source.

       t   The thickness of the drawn box.

	   These constants allow the x, y, w, h and t expressions to refer  to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

       •   Draw a black box around the edge of the input image:

		   drawbox

       •   Draw a box with color red and an opacity of 50%:

		   drawbox=10:20:200:60:red@0.5

	   The previous example can be specified as:

		   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       •   Fill the box with pink color:

		   drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=fill

       •   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

       •   Draw	 a  grid  with	cell  100x100 pixels, thickness 2 pixels, with
	   color red and an opacity of 50%:

		   drawgrid=width=100:height=100:thickness=2:color=red@0.5

       •   Draw a white 3x3 grid with an opacity of 50%:

		   drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If the specified expression is not valid, it is	kept  at  its  current
       value.

   drawtext
       Draw  a	text  string  or text from a specified file on top of a video,
       using the libfreetype library.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-libfreetype" and "--enable-libharfbuzz".  To  enable  default
       font  fallback  and  the	 font option you need to configure FFmpeg with
       "--enable-libfontconfig".  To enable the text_shaping option, you  need
       to configure FFmpeg with "--enable-libfribidi".

       Syntax

       It accepts the following parameters:

       box Used	 to  draw  a  box around text using the background color.  The
	   value must be either 1 (enable) or 0 (disable).  The default	 value
	   of box is 0.

       boxborderw
	   Set	the  width  of	the  border  to	 be drawn around the box using
	   boxcolor.  The value must be specified using one of	the  following
	   formats:

	   *<"boxborderw=10" set the width of all the borders to 10>
	   *<"boxborderw=10|20" set the width of the top and bottom borders to
	   10>
		   and the width of the left and right borders to 20

	   *<"boxborderw=10|20|30" set the width of the top border to 10, the
	   width>
		   of the bottom border to 30 and the width of the left and right borders to 20

	   *<"boxborderw=10|20|30|40" set the borders width to 10 (top), 20
	   (right),>
		   30 (bottom), 40 (left)

	   The default value of boxborderw is "0".

       boxcolor
	   The color to be used for drawing box around text. For the syntax of
	   this option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of boxcolor is "white".

       line_spacing
	   Set	the  line spacing in pixels. The default value of line_spacing
	   is 0.

       text_align
	   Set the vertical and horizontal alignment of the text with  respect
	   to  the box boundaries.  The value is combination of flags, one for
	   the vertical alignment (T=top, M=middle, B=bottom) and one for  the
	   horizontal alignment (L=left, C=center, R=right).  Please note that
	   tab	 characters  are  only	supported  with	 the  left  horizontal
	   alignment.

       y_align
	   Specify what the y value is referred to. Possible values are:

	   *<"text" the top of the highest glyph of the first text line is
	   placed at y>
	   *<"baseline" the baseline of the first text line is placed at y>
	   *<"font" the baseline of the first text line is placed at y plus
	   the>
		   ascent (in pixels) defined in the font metrics

	   The default value of y_align is "text" for backward compatibility.

       borderw
	   Set the width of the border to  be  drawn  around  the  text	 using
	   bordercolor.	 The default value of borderw is 0.

       bordercolor
	   Set	the  color  to be used for drawing border around text. For the
	   syntax of this option, check the "Color"  section  in  the  ffmpeg-
	   utils manual.

	   The default value of bordercolor is "black".

       expansion
	   Select  how	the text is expanded. Can be either "none", "strftime"
	   (deprecated) or "normal"  (default).	 See  the  drawtext_expansion,
	   Text expansion section below for details.

       basetime
	   Set	a  start  time	for  the count. Value is in microseconds. Only
	   applied in the deprecated "strftime" expansion mode. To emulate  in
	   normal  expansion  mode use the "pts" function, supplying the start
	   time (in seconds) as the second argument.

       fix_bounds
	   If true, check and fix text coords to avoid clipping.

       fontcolor
	   The color to be used for drawing fonts.  For	 the  syntax  of  this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of fontcolor is "black".

       fontcolor_expr
	   String  which  is  expanded	the same way as text to obtain dynamic
	   fontcolor value. By default this option has empty value and is  not
	   processed. When this option is set, it overrides fontcolor option.

       font
	   The font family to be used for drawing text. By default Sans.

       fontfile
	   The	font  file  to	be  used  for  drawing	text. The path must be
	   included.  This parameter is mandatory if the fontconfig support is
	   disabled.

       alpha
	   Draw the text applying alpha blending. The value can	 be  a	number
	   between  0.0 and 1.0.  The expression accepts the same variables x,
	   y as well.  The default value is 1.	Please see fontcolor_expr.

       fontsize
	   The font size to be used for drawing text.  The  default  value  of
	   fontsize is 16.

       text_shaping
	   If  set  to	1, attempt to shape the text (for example, reverse the
	   order of right-to-left text	and  join  Arabic  characters)	before
	   drawing  it.	  Otherwise,  just draw the text exactly as given.  By
	   default 1 (if supported).

       ft_load_flags
	   The flags to be used for loading the fonts.

	   The flags map the corresponding flags supported by libfreetype, and
	   are a combination of the following values:

	   default
	   no_scale
	   no_hinting
	   render
	   no_bitmap
	   vertical_layout
	   force_autohint
	   crop_bitmap
	   pedantic
	   ignore_global_advance_width
	   no_recurse
	   ignore_transform
	   monochrome
	   linear_design
	   no_autohint

	   Default value is "default".

	   For more information consult the documentation  for	the  FT_LOAD_*
	   libfreetype flags.

       shadowcolor
	   The	color  to  be used for drawing a shadow behind the drawn text.
	   For the syntax of this option, check the  "Color"  section  in  the
	   ffmpeg-utils manual.

	   The default value of shadowcolor is "black".

       boxw
	   Set	the  width  of	the  box to be drawn around text.  The default
	   value of boxw is computed automatically to match the text width

       boxh
	   Set the height of the box to be drawn  around  text.	  The  default
	   value of boxh is computed automatically to match the text height

       shadowx
       shadowy
	   The	x  and	y offsets for the text shadow position with respect to
	   the position of the text. They can be either positive  or  negative
	   values. The default value for both is "0".

       start_number
	   The starting frame number for the n/frame_num variable. The default
	   value is "0".

       tabsize
	   The size in number of spaces to use for rendering the tab.  Default
	   value is 4.

       timecode
	   Set	 the  initial  timecode	 representation	 in  "hh:mm:ss[:;.]ff"
	   format.  It	can  be	 used  with   or   without   text   parameter.
	   timecode_rate option must be specified.

       timecode_rate, rate, r
	   Set	the timecode frame rate (timecode only). Value will be rounded
	   to nearest integer. Minimum value is "1".  Drop-frame  timecode  is
	   supported for frame rates 30 & 60.

       tc24hmax
	   If  set to 1, the output of the timecode option will wrap around at
	   24 hours.  Default is 0 (disabled).

       text
	   The text string to be drawn. The text must be a sequence  of	 UTF-8
	   encoded  characters.	  This	parameter  is  mandatory if no file is
	   specified with the parameter textfile.

       textfile
	   A text file containing text	to  be	drawn.	The  text  must	 be  a
	   sequence of UTF-8 encoded characters.

	   This parameter is mandatory if no text string is specified with the
	   parameter text.

	   If both text and textfile are specified, an error is thrown.

       text_source
	   Text source should be set as side_data_detection_bboxes if you want
	   to use text data in detection bboxes of side data.

	   If  text source is set, text and textfile will be ignored and still
	   use text data in detection bboxes of side data. So  please  do  not
	   use this parameter if you are not sure about the text source.

       reload
	   The textfile will be reloaded at specified frame interval.  Be sure
	   to update textfile atomically, or it may be read partially, or even
	   fail.  Range is 0 to INT_MAX. Default is 0.

       x
       y   The	expressions which specify the offsets where text will be drawn
	   within the video frame. They are relative to the top/left border of
	   the output image.

	   The default value of x and y is "0".

	   See below for the list of accepted constants and functions.

       The parameters for x and y are  expressions  containing	the  following
       constants and functions:

       dar input display aspect ratio, it is the same as (w / h) * sar

       hsub
       vsub
	   horizontal  and  vertical  chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       line_h, lh
	   the height of each text line

       main_h, h, H
	   the input height

       main_w, w, W
	   the input width

       max_glyph_a, ascent
	   the maximum distance from the baseline to  the  highest/upper  grid
	   coordinate  used  to	 place	a  glyph  outline  point,  for all the
	   rendered glyphs.  It	 is  a	positive  value,  due  to  the	grid's
	   orientation with the Y axis upwards.

       max_glyph_d, descent
	   the	 maximum  distance  from  the  baseline	 to  the  lowest  grid
	   coordinate used to  place  a	 glyph	outline	 point,	 for  all  the
	   rendered  glyphs.   This  is	 a  negative  value, due to the grid's
	   orientation, with the Y axis upwards.

       max_glyph_h
	   maximum glyph height, that is the maximum height for all the glyphs
	   contained in the rendered  text,  it	 is  equivalent	 to  ascent  -
	   descent.

       max_glyph_w
	   maximum  glyph  width, that is the maximum width for all the glyphs
	   contained in the rendered text

       font_a
	   the ascent size defined in the font metrics

       font_d
	   the descent size defined in the font metrics

       top_a
	   the maximum ascender of the glyphs of the first text line

       bottom_d
	   the maximum descender of the glyphs of the last text line

       n   the number of input frame, starting from 0

       rand(min, max)
	   return a random number included between min and max

       sar The input sample aspect ratio.

       t   timestamp expressed in seconds,  NAN	 if  the  input	 timestamp  is
	   unknown

       text_h, th
	   the height of the rendered text

       text_w, tw
	   the width of the rendered text

       x
       y   the x and y offset coordinates where the text is drawn.

	   These  parameters  allow  the  x and y expressions to refer to each
	   other, so you can for example specify "y=x/dar".

       pict_type
	   A one character description of the current frame's picture type.

       pkt_pos
	   The current packet's position in  the  input	 file  or  stream  (in
	   bytes,  from	 the start of the input). A value of -1 indicates this
	   info is not available.

       duration
	   The current packet's duration, in seconds.

       pkt_size
	   The current packet's size (in bytes).

       Text expansion

       If expansion is set to  "strftime",  the	 filter	 recognizes  sequences
       accepted	 by the "strftime" C function in the provided text and expands
       them accordingly. Check the documentation of "strftime".	 This  feature
       is  deprecated  in  favor  of  "normal"	expansion with the "gmtime" or
       "localtime" expansion functions.

       If expansion is set to "none", the text is printed verbatim.

       If expansion is set to "normal" (which is the default),	the  following
       expansion mechanism is used.

       The backslash character \, followed by any character, always expands to
       the second character.

       Sequences  of  the  form	 "%{...}"  are	expanded. The text between the
       braces is a function name, possibly followed by arguments separated  by
       ':'.  If the arguments contain special characters or delimiters (':' or
       '}'), they should be escaped.

       Note  that they probably must also be escaped as the value for the text
       option in the filter argument string and as the filter argument in  the
       filtergraph description, and possibly also for the shell, that makes up
       to  four levels of escaping; using a text file with the textfile option
       avoids these problems.

       The following functions are available:

       expr, e
	   The expression evaluation result.

	   It  must  take  one	argument  specifying  the  expression  to   be
	   evaluated,  which accepts the same constants and functions as the x
	   and y values. Note that not	all  constants	should	be  used,  for
	   example  the text size is not known when evaluating the expression,
	   so the constants text_w and text_h will have an undefined value.

       expr_int_format, eif
	   Evaluate the expression's value and output as formatted integer.

	   The first argument is the expression to be evaluated, just  as  for
	   the	expr  function.	  The  second  argument	 specifies  the output
	   format. Allowed values are x, X, d and u. They are treated  exactly
	   as  in  the "printf" function.  The third parameter is optional and
	   sets the number of positions taken by the output.  It can  be  used
	   to add padding with zeros from the left.

       gmtime
	   The	time at which the filter is running, expressed in UTC.	It can
	   accept an argument: a "strftime" C  function	 format	 string.   The
	   format  string  is  extended	 to support the variable %[1-6]N which
	   prints fractions of the second with optionally specified number  of
	   digits.

       localtime
	   The	time  at  which	 the filter is running, expressed in the local
	   time zone.  It can accept an	 argument:  a  "strftime"  C  function
	   format  string.   The  format  string  is  extended	to support the
	   variable  %[1-6]N  which  prints  fractions	of  the	 second	  with
	   optionally specified number of digits.

       metadata
	   Frame metadata. Takes one or two arguments.

	   The first argument is mandatory and specifies the metadata key.

	   The second argument is optional and specifies a default value, used
	   when the metadata key is not found or empty.

	   Available metadata can be identified by inspecting entries starting
	   with	 TAG  included	within	each  frame section printed by running
	   "ffprobe -show_frames".

	   String metadata generated in filters leading to the drawtext filter
	   are also available.

       n, frame_num
	   The frame number, starting from 0.

       pict_type
	   A one character description of the current picture type.

       pts The timestamp of the current	 frame.	  It  can  take	 up  to	 three
	   arguments.

	   The	first  argument is the format of the timestamp; it defaults to
	   "flt" for seconds as a decimal number  with	microsecond  accuracy;
	   "hms"   stands  for	a  formatted  [-]HH:MM:SS.mmm  timestamp  with
	   millisecond accuracy.  "gmtime" stands for  the  timestamp  of  the
	   frame  formatted  as UTC time; "localtime" stands for the timestamp
	   of the frame formatted as local time zone time.

	   The second argument is an offset added to the timestamp.

	   If the format is set to "hms",  a  third  argument  "24HH"  may  be
	   supplied to present the hour part of the formatted timestamp in 24h
	   format (00-23).

	   If  the  format is set to "localtime" or "gmtime", a third argument
	   may be  supplied:  a	 "strftime"  C	function  format  string.   By
	   default, YYYY-MM-DD HH:MM:SS format will be used.

       Commands

       This filter supports altering parameters via commands:

       reinit
	   Alter existing filter parameters.

	   Syntax for the argument is the same as for filter invocation, e.g.

		   fontsize=56:fontcolor=green:text='Hello World'

	   Full filter invocation with sendcmd would look like this:

		   sendcmd=c='56.0 drawtext reinit fontsize=56\:fontcolor=green\:text=Hello\\ World'

	   If  the  entire argument can't be parsed or applied as valid values
	   then the filter will continue with its existing parameters.

       The following options are also supported as commands:

       *<x>
       *<y>
       *<alpha>
       *<fontsize>
       *<fontcolor>
       *<boxcolor>
       *<bordercolor>
       *<shadowcolor>
       *<box>
       *<boxw>
       *<boxh>
       *<boxborderw>
       *<line_spacing>
       *<text_align>
       *<shadowx>
       *<shadowy>
       *<borderw>

       Examples

       •   Draw "Test Text" with font FreeSerif, using the default values  for
	   the optional parameters.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

       •   Draw	 'Test	Text' with font FreeSerif of size 24 at position x=100
	   and y=50 (counting from the top-left corner of the screen), text is
	   yellow with a red box around it. Both the text and the box have  an
	   opacity of 20%.

		   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
			     x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

	   Note	 that  the  double  quotes are not necessary if spaces are not
	   used within the parameter list.

       •   Show the text at the center of the video frame:

		   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"

       •   Show the text at a random position, switching  to  a	 new  position
	   every 30 seconds:

		   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"

       •   Show	 a text line sliding from right to left in the last row of the
	   video frame. The file LONG_LINE is assumed to contain a single line
	   with no newlines.

		   drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"

       •   Show the content of file CREDITS off the bottom of  the  frame  and
	   scroll up.

		   drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"

       •   Draw	 a  single green letter "g", at the center of the input video.
	   The glyph baseline is placed at half screen height.

		   drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"

       •   Show text for 1 second every 3 seconds:

		   drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"

       •   Use fontconfig to set the font. Note that the  colons  need	to  be
	   escaped.

		   drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'

       •   Draw "Test Text" with font size dependent on height of the video.

		   drawtext="text='Test Text': fontsize=h/30: x=(w-text_w)/2: y=(h-text_h*2)"

       •   Print  the  date of a real-time encoding (see documentation for the
	   "strftime" C function):

		   drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

       •   Show text fading in and out (appearing/disappearing):

		   #!/bin/sh
		   DS=1.0 # display start
		   DE=10.0 # display end
		   FID=1.5 # fade in duration
		   FOD=5 # fade out duration
		   ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"

       •   Horizontally align multiple separate texts. Note  that  max_glyph_a
	   and the fontsize value are included in the y offset.

		   drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
		   drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a

       •   Plot	 special  lavf.image2dec.source_basename  metadata  onto  each
	   frame if such metadata exists. Otherwise,  plot  the	 string	 "NA".
	   Note	 that  image2 demuxer must have option -export_path_metadata 1
	   for the special metadata fields to be available for filters.

		   drawtext="fontsize=20:fontcolor=white:fontfile=FreeSans.ttf:text='%{metadata\:lavf.image2dec.source_basename\:NA}':x=10:y=10"

       For	more	  information	   about      libfreetype,	check:
       <http://www.freetype.org/>.

       For	 more	   information	    about      fontconfig,	check:
       <http://freedesktop.org/software/fontconfig/fontconfig-user.html>.

       For more information about libfribidi, check: <http://fribidi.org/>.

       For	more	  information	   about      libharfbuzz,	check:
       <https://github.com/harfbuzz/harfbuzz>.

   edgedetect
       Detect  and  draw  edges.  The  filter  uses  the  Canny Edge Detection
       algorithm.

       The filter accepts the following options:

       low
       high
	   Set low and high threshold values used by  the  Canny  thresholding
	   algorithm.

	   The high threshold selects the "strong" edge pixels, which are then
	   connected  through  8-connectivity  with  the  "weak"  edge	pixels
	   selected by the low threshold.

	   low and high threshold values must be chosen in  the	 range	[0,1],
	   and low should be lesser or equal to high.

	   Default  value  for	low is "20/255", and default value for high is
	   "50/255".

       mode
	   Define the drawing mode.

	   wires
	       Draw white/gray wires on black background.

	   colormix
	       Mix the colors to create a paint/cartoon effect.

	   canny
	       Apply Canny edge detector on all selected planes.

	   Default value is wires.

       planes
	   Select planes for filtering. By default all	available  planes  are
	   filtered.

       Examples

       •   Standard  edge  detection  with  custom  values  for the hysteresis
	   thresholding:

		   edgedetect=low=0.1:high=0.4

       •   Painting effect without thresholding:

		   edgedetect=mode=colormix:high=0

   elbg
       Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.

       For each input image, the filter will compute the optimal mapping  from
       the  input  to the output given the codebook length, that is the number
       of distinct output colors.

       This filter accepts the following options.

       codebook_length, l
	   Set codebook length. The value must	be  a  positive	 integer,  and
	   represents  the  number of distinct output colors. Default value is
	   256.

       nb_steps, n
	   Set the maximum number of iterations to  apply  for	computing  the
	   optimal mapping. The higher the value the better the result and the
	   higher the computation time. Default value is 1.

       seed, s
	   Set	a  random  seed,  must	be  an	integer included between 0 and
	   UINT32_MAX. If not specified, or  if	 explicitly  set  to  -1,  the
	   filter will try to use a good random seed on a best effort basis.

       pal8
	   Set	pal8  output  pixel  format.  This  option  does not work with
	   codebook length greater than 256. Default is disabled.

       use_alpha
	   Include  alpha  values  in  the  quantization  calculation.	Allows
	   creating  palettized	 output images (e.g. PNG8) with multiple alpha
	   smooth blending.

   entropy
       Measure graylevel entropy in  histogram	of  color  channels  of	 video
       frames.

       It accepts the following parameters:

       mode
	   Can be either normal or diff. Default is normal.

	   diff	 mode  measures	 entropy  of  histogram delta values, absolute
	   differences between neighbour histogram values.

   epx
       Apply the EPX magnification filter which is designed for pixel art.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "2xEPX", 3 for  "3xEPX".   Default
	   is 3.

   eq
       Set brightness, contrast, saturation and approximate gamma adjustment.

       The filter accepts the following options:

       contrast
	   Set	the  contrast  expression.  The value must be a float value in
	   range -1000.0 to 1000.0. The default value is "1".

       brightness
	   Set the brightness expression. The value must be a float  value  in
	   range -1.0 to 1.0. The default value is "0".

       saturation
	   Set	the  saturation expression. The value must be a float in range
	   0.0 to 3.0. The default value is "1".

       gamma
	   Set the gamma expression. The value must be a float in range 0.1 to
	   10.0.  The default value is "1".

       gamma_r
	   Set the gamma expression for red. The value	must  be  a  float  in
	   range 0.1 to 10.0. The default value is "1".

       gamma_g
	   Set	the  gamma  expression for green. The value must be a float in
	   range 0.1 to 10.0. The default value is "1".

       gamma_b
	   Set the gamma expression for blue. The value must  be  a  float  in
	   range 0.1 to 10.0. The default value is "1".

       gamma_weight
	   Set	the  gamma  weight  expression.	 It  can be used to reduce the
	   effect of a high gamma value on bright image areas, e.g. keep  them
	   from	 getting overamplified and just plain white. The value must be
	   a float in range 0.0 to  1.0.  A  value  of	0.0  turns  the	 gamma
	   correction  all  the	 way  down  while  1.0	leaves	it at its full
	   strength. Default is "1".

       eval
	   Set when the expressions for brightness, contrast,  saturation  and
	   gamma expressions are evaluated.

	   It accepts the following values:

	   init
	       only evaluate expressions once during the filter initialization
	       or when a command is processed

	   frame
	       evaluate expressions for each incoming frame

	   Default value is init.

       The expressions accept the following parameters:

       n   frame count of the input frame starting from 0

       pos byte position of the corresponding packet in the input file, NAN if
	   unspecified; deprecated, do not use

       r   frame  rate	of  the	 input	video,	NAN if the input frame rate is
	   unknown

       t   timestamp expressed in seconds,  NAN	 if  the  input	 timestamp  is
	   unknown

       Commands

       The filter supports the following commands:

       contrast
	   Set the contrast expression.

       brightness
	   Set the brightness expression.

       saturation
	   Set the saturation expression.

       gamma
	   Set the gamma expression.

       gamma_r
	   Set the gamma_r expression.

       gamma_g
	   Set gamma_g expression.

       gamma_b
	   Set gamma_b expression.

       gamma_weight
	   Set gamma_weight expression.

	   The command accepts the same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   erosion
       Apply erosion effect to the video.

       This filter replaces the pixel by the local(3x3) minimum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit  the  maximum change for each plane, default is 65535.	 If 0,
	   plane will remain unchanged.

       coordinates
	   Flag which specifies the pixel to refer to. Default is 255 i.e. all
	   eight pixels are used.

	   Flags to local 3x3 coordinates maps like this:

	       1 2 3
	       4   5
	       6 7 8

       Commands

       This filter supports the all above options as commands.

   estdif
       Deinterlace the input video ("estdif" stands for	 "Edge	Slope  Tracing
       Deinterlacing Filter").

       Spatial	 only  filter  that  uses  edge	 slope	tracing	 algorithm  to
       interpolate missing lines.  It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It  accepts  one  of	the  following
	   values:

	   frame
	       Output one frame for each frame.

	   field
	       Output one frame for each field.

	   The default value is "field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   tff Assume the top field is first.

	   bff Assume the bottom field is first.

	   auto
	       Enable automatic detection of field parity.

	   The	default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first  will  be
	   assumed.

       deint
	   Specify  which  frames to deinterlace. Accepts one of the following
	   values:

	   all Deinterlace all frames.

	   interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

       rslope
	   Specify the search radius for edge slope tracing. Default value  is
	   1.  Allowed range is from 1 to 15.

       redge
	   Specify  the search radius for best edge matching. Default value is
	   2.  Allowed range is from 0 to 15.

       ecost
	   Specify the edge cost  for  edge  matching.	Default	 value	is  2.
	   Allowed range is from 0 to 50.

       mcost
	   Specify  the	 middle	 cost  for  edge matching. Default value is 1.
	   Allowed range is from 0 to 50.

       dcost
	   Specify the distance cost for edge matching. Default	 value	is  1.
	   Allowed range is from 0 to 50.

       interp
	   Specify  the	 interpolation used. Default is 4-point interpolation.
	   It accepts one of the following values:

	   2p  Two-point interpolation.

	   4p  Four-point interpolation.

	   6p  Six-point interpolation.

       Commands

       This filter supports same commands as options.

   exposure
       Adjust exposure of the video stream.

       The filter accepts the following options:

       exposure
	   Set the exposure correction in EV. Allowed range is	from  -3.0  to
	   3.0 EV Default value is 0 EV.

       black
	   Set	the black level correction. Allowed range is from -1.0 to 1.0.
	   Default value is 0.

       Commands

       This filter supports same commands as options.

   extractplanes
       Extract color channel components from input video stream into  separate
       grayscale video streams.

       The filter accepts the following option:

       planes
	   Set plane(s) to extract.

	   Available values for planes are:

	   y
	   u
	   v
	   a
	   r
	   g
	   b

	   Choosing planes not available in the input will result in an error.
	   That	 means	you  cannot select "r", "g", "b" planes with "y", "u",
	   "v" planes at same time.

       Examples

       •   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

       •   Fade in the first 30 frames of video:

		   fade=in:0:30

	   The command above is equivalent to:

		   fade=t=in:s=0:n=30

       •   Fade out the last 45 frames of a 200-frame video:

		   fade=out:155:45
		   fade=type=out:start_frame=155:nb_frames=45

       •   Fade	 in  the  first 25 frames and fade out the last 25 frames of a
	   1000-frame video:

		   fade=in:0:25, fade=out:975:25

       •   Make the first 5 frames yellow, then fade in from frame 5-24:

		   fade=in:5:20:color=yellow

       •   Fade in alpha over first 25 frames of video:

		   fade=in:0:25:alpha=1

       •   Make the first 5.5 seconds black, then fade in for 0.5 seconds:

		   fade=t=in:st=5.5:d=0.5

   feedback
       Apply feedback video filter.

       This filter pass cropped input frames to 2nd output.  From there it can
       be filtered with other video filters.  After filter receives frame from
       2nd input, that frame is combined on top of  original  frame  from  1st
       input and passed to 1st output.

       The typical usage is filter only part of frame.

       The filter accepts the following options:

       x
       y   Set the top left crop position.

       w
       h   Set the crop size.

       Examples

       •   Blur	 only  top  left  rectangular part of video frame size 100x100
	   with gblur filter.

		   [in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]gblur=8[blurin]

       •   Draw black box on top left part of video frame of size 100x100 with
	   drawbox filter.

		   [in][blurin]feedback=x=0:y=0:w=100:h=100[out][blurout];[blurout]drawbox=x=0:y=0:w=100:h=100:t=100[blurin]

   fftdnoiz
       Denoise frames using 3D FFT (frequency domain filtering).

       The filter accepts the following options:

       sigma
	   Set	the  noise  sigma  constant.  This  sets  denoising  strength.
	   Default value is 1. Allowed range is from 0 to 30.  Using very high
	   sigma with low overlap may give blocking artifacts.

       amount
	   Set	amount of denoising. By default all detected noise is reduced.
	   Default value is 1. Allowed range is from 0 to 1.

       block
	   Set size of block in pixels, Default is 32, can be 8 to 256.

       overlap
	   Set block overlap. Default is 0.5. Allowed range  is	 from  0.2  to
	   0.8.

       method
	   Set denoising method. Default is "wiener", can also be "hard".

       prev
	   Set	number	of previous frames to use for denoising. By default is
	   set to 0.

       next
	   Set number of next frames to to use for denoising.  By  default  is
	   set to 0.

       planes
	   Set	planes	which  will  be filtered, by default are all available
	   filtered except alpha.

   fftfilt
       Apply arbitrary expressions to samples in frequency domain

       dc_Y
	   Adjust the dc value (gain) of the luma  plane  of  the  image.  The
	   filter  accepts  an	integer	 value in range 0 to 1000. The default
	   value is set to 0.

       dc_U
	   Adjust the dc value (gain) of the 1st chroma plane  of  the	image.
	   The filter accepts an integer value in range 0 to 1000. The default
	   value is set to 0.

       dc_V
	   Adjust  the	dc  value (gain) of the 2nd chroma plane of the image.
	   The filter accepts an integer value in range 0 to 1000. The default
	   value is set to 0.

       weight_Y
	   Set the frequency domain weight expression for the luma plane.

       weight_U
	   Set the frequency domain  weight  expression	 for  the  1st	chroma
	   plane.

       weight_V
	   Set	the  frequency	domain	weight	expression  for the 2nd chroma
	   plane.

       eval
	   Set when the expressions are evaluated.

	   It accepts the following values:

	   init
	       Only   evaluate	 expressions   once    during	 the	filter
	       initialization.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

	   The filter accepts the following variables:

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       N   The number of input frame, starting from 0.

       WS
       HS  The size of FFT array for horizontal and vertical processing.

       Examples

       •   High-pass:

		   fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

       •   Low-pass:

		   fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

       •   Sharpen:

		   fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

       •   Blur:

		   fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'

   field
       Extract a single field from an interlaced image using stride arithmetic
       to  avoid  wasting  CPU	time.  The  output  frames  are marked as non-
       interlaced.

       The filter accepts the following options:

       type
	   Specify whether to extract the top (if the value is 0 or "top")  or
	   the bottom field (if the value is 1 or "bottom").

   fieldhint
       Create new frames by copying the top and bottom fields from surrounding
       frames supplied as numbers by the hint file.

       hint
	   Set file containing hints: absolute/relative frame numbers.

	   There  must	be  one	 line for each frame in a clip. Each line must
	   contain two numbers separated by the comma, optionally followed  by
	   "-"	or  "+".  Numbers supplied on each line of file can not be out
	   of [N-1,N+1] where N is current frame number for "absolute" mode or
	   out of [-1, 1] range for "relative" mode. First number  tells  from
	   which frame to pick up top field and second number tells from which
	   frame to pick up bottom field.

	   If  optionally  followed  by	 "+"  output  frame  will be marked as
	   interlaced, else if followed by "-" output frame will be marked  as
	   progressive,	 else  it  will	 be  marked  same  as input frame.  If
	   optionally followed by "t" output frame will use only top field, or
	   in case of "b" it will use only bottom field.  If line starts  with
	   "#" or ";" that line is skipped.

       mode
	   Can	be  item  "absolute"  or  "relative"  or "pattern". Default is
	   "absolute".	The "pattern" mode is same as "relative" mode,	except
	   at  last  entry  of	file  if there are more frames to process than
	   "hint" file is seek back to start.

       Example of first several lines of "hint" file for "relative" mode:

	       0,0 - # first frame
	       1,0 - # second frame, use third's frame top field and second's frame bottom field
	       1,0 - # third frame, use fourth's frame top field and third's frame bottom field
	       1,0 -
	       0,0 -
	       0,0 -
	       1,0 -
	       1,0 -
	       1,0 -
	       0,0 -
	       0,0 -
	       1,0 -
	       1,0 -
	       1,0 -
	       0,0 -

   fieldmatch
       Field matching filter for inverse telecine. It is meant to  reconstruct
       the  progressive	 frames	 from  a telecined stream. The filter does not
       drop duplicated frames, so  to  achieve	a  complete  inverse  telecine
       "fieldmatch"  needs  to	be  followed  by  a  decimation filter such as
       decimate in the filtergraph.

       The separation of the field matching  and  the  decimation  is  notably
       motivated  by  the  possibility	of  inserting  a de-interlacing filter
       fallback between the two.  If the source has mixed telecined  and  real
       interlaced  content,  "fieldmatch" will not be able to match fields for
       the interlaced parts.  But these remaining combed frames will be marked
       as interlaced, and thus can be de-interlaced by a later filter such  as
       yadif before decimation.

       In addition to the various configuration options, "fieldmatch" can take
       an  optional  second  stream,  activated	 through  the ppsrc option. If
       enabled, the frames reconstruction will be  based  on  the  fields  and
       frames  from this second stream. This allows the first input to be pre-
       processed in order to help the various algorithms of the filter,	 while
       keeping the output lossless (assuming the fields are matched properly).
       Typically,  a  field-aware denoiser, or brightness/contrast adjustments
       can help.

       Note that this filter uses the same algorithms as  TIVTC/TFM  (AviSynth
       project)	 and  VIVTC/VFM	 (VapourSynth  project).  The later is a light
       clone of TFM from which "fieldmatch" is based on.  While	 the  semantic
       and usage are very close, some behaviour and options names can differ.

       The decimate filter currently only works for constant frame rate input.
       If  your input has mixed telecined (30fps) and progressive content with
       a lower framerate like 24fps use the following filterchain  to  produce
       the		   necessary		    cfr		       stream:
       "dejudder,fps=30000/1001,fieldmatch,decimate".

       The filter accepts the following options:

       order
	   Specify the assumed field order  of	the  input  stream.  Available
	   values are:

	   auto
	       Auto detect parity (use FFmpeg's internal parity value).

	   bff Assume bottom field first.

	   tff Assume top field first.

	   Note	 that  it  is  sometimes  recommended  not to trust the parity
	   announced by the stream.

	   Default value is auto.

       mode
	   Set the matching mode or strategy to use. pc mode is the safest  in
	   the	sense  that  it won't risk creating jerkiness due to duplicate
	   frames when possible, but if there are bad edits or blended	fields
	   it  will  end  up  outputting combed frames when a good match might
	   actually exist. On the other hand, pcn_ub mode is the most risky in
	   terms of creating jerkiness, but will almost	 always	 find  a  good
	   frame  if  there  is	 one.  The  other  values are all somewhere in
	   between pc and pcn_ub in terms of risking  jerkiness	 and  creating
	   duplicate  frames  versus finding good matches in sections with bad
	   edits, orphaned fields, blended fields, etc.

	   More details about p/c/n/u/b are  available	in  p/c/n/u/b  meaning
	   section.

	   Available values are:

	   pc  2-way matching (p/c)

	   pc_n
	       2-way matching, and trying 3rd match if still combed (p/c + n)

	   pc_u
	       2-way  matching,	 and  trying  3rd  match (same order) if still
	       combed (p/c + u)

	   pc_n_ub
	       2-way matching, trying 3rd match if still  combed,  and	trying
	       4th/5th matches if still combed (p/c + n + u/b)

	   pcn 3-way matching (p/c/n)

	   pcn_ub
	       3-way  matching,	 and  trying  4th/5th  matches if all 3 of the
	       original matches are detected as combed (p/c/n + u/b)

	   The parenthesis at the end indicate the matches that would be  used
	   for that mode assuming order=tff (and field on auto or top).

	   In  terms  of speed pc mode is by far the fastest and pcn_ub is the
	   slowest.

	   Default value is pc_n.

       ppsrc
	   Mark the main input stream as a pre-processed input, and enable the
	   secondary input stream as the clean source to pick the fields from.
	   See the filter introduction for more details. It is similar to  the
	   clip2 feature from VFM/TFM.

	   Default value is 0 (disabled).

       field
	   Set	the  field to match from. It is recommended to set this to the
	   same value as order unless you experience  matching	failures  with
	   that	 setting.  In certain circumstances changing the field that is
	   used to match from can have a large impact on matching performance.
	   Available values are:

	   auto
	       Automatic (same value as order).

	   bottom
	       Match from the bottom field.

	   top Match from the top field.

	   Default value is auto.

       mchroma
	   Set whether or not chroma is included during the match comparisons.
	   In most cases it is recommended to leave this enabled.  You	should
	   set	this  to  0  only if your clip has bad chroma problems such as
	   heavy rainbowing or other artifacts. Setting this to 0  could  also
	   be used to speed things up at the cost of some accuracy.

	   Default value is 1.

       y0
       y1  These  define an exclusion band which excludes the lines between y0
	   and y1 from being included  in  the	field  matching	 decision.  An
	   exclusion  band  can	 be used to ignore subtitles, a logo, or other
	   things that may interfere with the matching. y0 sets	 the  starting
	   scan	 line and y1 sets the ending line; all lines in between y0 and
	   y1 (including y0 and y1) will be ignored. Setting y0 and y1 to  the
	   same value will disable the feature.	 y0 and y1 defaults to 0.

       scthresh
	   Set the scene change detection threshold as a percentage of maximum
	   change  on  the  luma  plane.  Good values are in the "[8.0, 14.0]"
	   range.  Scene  change  detection   is   only	  relevant   in	  case
	   combmatch=sc.  The range for scthresh is "[0.0, 100.0]".

	   Default value is 12.0.

       combmatch
	   When	 combatch is not none, "fieldmatch" will take into account the
	   combed scores of matches when deciding what match  to  use  as  the
	   final match. Available values are:

	   none
	       No final matching based on combed scores.

	   sc  Combed scores are only used when a scene change is detected.

	   full
	       Use combed scores all the time.

	   Default is sc.

       combdbg
	   Force  "fieldmatch"	to  calculate  the  combed metrics for certain
	   matches and print them. This setting is known as micout in  TFM/VFM
	   vocabulary.	Available values are:

	   none
	       No forced calculation.

	   pcn Force p/c/n calculations.

	   pcnub
	       Force p/c/n/u/b calculations.

	   Default value is none.

       cthresh
	   This is the area combing threshold used for combed frame detection.
	   This essentially controls how "strong" or "visible" combing must be
	   to  be  detected.   Larger values mean combing must be more visible
	   and smaller values mean combing can be less visible or  strong  and
	   still  be detected. Valid settings are from -1 (every pixel will be
	   detected as combed) to 255 (no pixel will be detected  as  combed).
	   This	 is  basically	a pixel difference value. A good range is "[8,
	   12]".

	   Default value is 9.

       chroma
	   Sets whether or not	chroma	is  considered	in  the	 combed	 frame
	   decision.   Only  disable  this  if your source has chroma problems
	   (rainbowing, etc.) that are causing problems for the	 combed	 frame
	   detection  with chroma enabled. Actually, using chroma=0 is usually
	   more reliable, except for the  case	where  there  is  chroma  only
	   combing in the source.

	   Default value is 0.

       blockx
       blocky
	   Respectively	 set  the  x-axis  and	y-axis size of the window used
	   during combed frame detection. This has to do with the size of  the
	   area	 in which combpel pixels are required to be detected as combed
	   for a frame to  be  declared	 combed.  See  the  combpel  parameter
	   description	for more info.	Possible values are any number that is
	   a power of 2 starting at 4 and going up to 512.

	   Default value is 16.

       combpel
	   The number of combed pixels inside any of the blocky by blockx size
	   blocks on the frame for the frame to be detected as	combed.	 While
	   cthresh  controls  how  "visible" the combing must be, this setting
	   controls "how much" combing there must be in any localized area  (a
	   window  defined  by	the  blockx and blocky settings) on the frame.
	   Minimum value is 0 and maximum is "blocky x blockx" (at which point
	   no frames will ever be detected as combed). This setting  is	 known
	   as MI in TFM/VFM vocabulary.

	   Default value is 80.

       p/c/n/u/b meaning

       p/c/n

       We assume the following telecined stream:

	       Top fields:     1 2 2 3 4
	       Bottom fields:  1 2 3 4 4

       The  numbers  correspond to the progressive frame the fields relate to.
       Here, the first two frames are progressive, the 3rd and 4th are combed,
       and so on.

       When  "fieldmatch"  is  configured  to  run  a  matching	 from	bottom
       (field=bottom) this is how this input stream get transformed:

	       Input stream:
			       T     1 2 2 3 4
			       B     1 2 3 4 4	 <-- matching reference

	       Matches:		     c c n n c

	       Output stream:
			       T     1 2 3 4 4
			       B     1 2 3 4 4

       As  a  result  of  the  field matching, we can see that some frames get
       duplicated.  To perform a complete inverse telecine, you need  to  rely
       on  a  decimation  filter  after	 this  operation. See for instance the
       decimate filter.

       The same operation now matching from top fields (field=top) looks  like
       this:

	       Input stream:
			       T     1 2 2 3 4	 <-- matching reference
			       B     1 2 3 4 4

	       Matches:		     c c p p c

	       Output stream:
			       T     1 2 2 3 4
			       B     1 2 2 3 4

       In  these  examples,  we	 can see what p, c and n mean; basically, they
       refer to the frame and field of the opposite parity:

       *<p matches the field of the opposite parity in the previous frame>
       *<c matches the field of the opposite parity in the current frame>
       *<n matches the field of the opposite parity in the next frame>

       u/b

       The u and b matching are a bit special in the  sense  that  they	 match
       from  the  opposite  parity  flag. In the following examples, we assume
       that we	are  currently	matching  the  2nd  frame  (Top:2,  bottom:2).
       According  to  the  match, a 'x' is placed above and below each matched
       fields.

       With bottom matching (field=bottom):

	       Match:		c	  p	      n		 b	    u

				x	x		x	 x	    x
		 Top	      1 2 2	1 2 2	    1 2 2      1 2 2	  1 2 2
		 Bottom	      1 2 3	1 2 3	    1 2 3      1 2 3	  1 2 3
				x	  x	      x	       x	      x

	       Output frames:
				2	   1	      2		 2	    2
				2	   2	      2		 1	    3

       With top matching (field=top):

	       Match:		c	  p	      n		 b	    u

				x	  x	      x	       x	      x
		 Top	      1 2 2	1 2 2	    1 2 2      1 2 2	  1 2 2
		 Bottom	      1 2 3	1 2 3	    1 2 3      1 2 3	  1 2 3
				x	x		x	 x	    x

	       Output frames:
				2	   2	      2		 1	    2
				2	   1	      3		 2	    2

       Examples

       Simple IVTC of a top field first telecined stream:

	       fieldmatch=order=tff:combmatch=none, decimate

       Advanced IVTC, with fallback on yadif for still combed frames:

	       fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

   fieldorder
       Transform the field order of the input video.

       It accepts the following parameters:

       order
	   The output field order. Valid values are tff for top field first or
	   bff for bottom field first.

       The default value is tff.

       The transformation is done by shifting the picture content up  or  down
       by  one	line,  and filling the remaining line with appropriate picture
       content.	 This method is consistent with	 most  broadcast  field	 order
       converters.

       If the input video is not flagged as being interlaced, or it is already
       flagged	as  being of the required output field order, then this filter
       does not alter the incoming video.

       It is very useful when converting to or from PAL DV material, which  is
       bottom field first.

       For example:

	       ffmpeg -i in.vob -vf "fieldorder=bff" out.dv

   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 (half sample symmetric)

	   fixed
	       fill pixels with constant value

	   reflect
	       fill pixels using reflecting (whole sample symmetric)

	   wrap
	       fill pixels using wrapping

	   fade
	       fade pixels to constant value

	   margins
	       fill pixels at top and bottom  with  weighted  averages	pixels
	       near borders

	   Default is smear.

       color
	   Set color for pixels in fixed or fade mode. Default is black.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

   find_rect
       Find a rectangular object in the input video.

       The object to search for must be specified as a gray8  image  specified
       with the object option.

       For  each possible match, a score is computed. If the score reaches the
       specified threshold, the object is considered found.

       If the input video contains  multiple  instances	 of  the  object,  the
       filter will find only one of them.

       When  an object is found, the following metadata entries are set in the
       matching frame:

       lavfi.rect.w
	   width of object

       lavfi.rect.h
	   height of object

       lavfi.rect.x
	   x position of object

       lavfi.rect.y
	   y position of object

       lavfi.rect.score
	   match score of the found object

       It accepts the following options:

       object
	   Filepath of the object image, needs to be in gray8.

       threshold
	   Detection threshold, expressed as a decimal	number	in  the	 range
	   0-1.

	   A  threshold value of 0.01 means only exact matches, a threshold of
	   0.99 means almost everything matches.

	   Default value is 0.5.

       mipmaps
	   Number of mipmaps, default is 3.

       xmin, ymin, xmax, ymax
	   Specifies the rectangle in which to search.

       discard
	   Discard frames where object is not detected. Default is disabled.

       Examples

       •   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

       •   Find	 the  position	of  an	object in each frame using ffprobe and
	   write it to a log file:

		   ffprobe -f lavfi movie=test.mp4,find_rect=object=object.pgm:threshold=0.3 \
		     -show_entries frame=pkt_pts_time:frame_tags=lavfi.rect.x,lavfi.rect.y \
		     -of csv -o find_rect.csv

   floodfill
       Flood area with values of same pixel components with another values.

       It accepts the following options:

       x   Set pixel x coordinate.

       y   Set pixel y coordinate.

       s0  Set source #0 component value.

       s1  Set source #1 component value.

       s2  Set source #2 component value.

       s3  Set source #3 component value.

       d0  Set destination #0 component value.

       d1  Set destination #1 component value.

       d2  Set destination #2 component value.

       d3  Set destination #3 component value.

   format
       Convert the  input  video  to  one  of  the  specified  pixel  formats.
       Libavfilter  will try to pick one that is suitable as input to the next
       filter.

       It accepts the following parameters:

       pix_fmts
	   A   '|'-separated   list   of   pixel   format   names,   such   as
	   "pix_fmts=yuv420p|monow|rgb24".

       Examples

       •   Convert the input video to the yuv420p format

		   format=pix_fmts=yuv420p

	   Convert the input video to any of the formats in the list

		   format=pix_fmts=yuv420p|yuv444p|yuv410p

   fps
       Convert	the  video  to specified constant frame rate by duplicating or
       dropping frames as necessary.

       It accepts the following parameters:

       fps The desired output frame rate. It  accepts  expressions  containing
	   the following constants:

	   source_fps
	       The input's frame rate

	   ntsc
	       NTSC frame rate of "30000/1001"

	   pal PAL frame rate of 25.0

	   film
	       Film frame rate of 24.0

	   ntsc_film
	       NTSC-film frame rate of "24000/1001"

	   The default is 25.

       start_time
	   Assume  the	first  PTS should be the given value, in seconds. This
	   allows for padding/trimming at the start of stream. By default,  no
	   assumption  is  made	 about	the  first frame's expected PTS, so no
	   padding or trimming is done.	 For example, this could be set	 to  0
	   to  pad the beginning with duplicates of the first frame if a video
	   stream starts after the audio stream or to trim any frames  with  a
	   negative PTS.

       round
	   Timestamp (PTS) rounding method.

	   Possible values are:

	   zero
	       round towards 0

	   inf round away from 0

	   down
	       round towards -infinity

	   up  round towards +infinity

	   near
	       round to nearest

	   The default is "near".

       eof_action
	   Action performed when reading the last frame.

	   Possible values are:

	   round
	       Use same timestamp rounding method as used for other frames.

	   pass
	       Pass  through last frame if input duration has not been reached
	       yet.

	   The default is "round".

       Alternatively,  the  options  can  be  specified	 as  a	flat   string:
       fps[:start_time[:round]].

       See also the setpts filter.

       Examples

       •   A typical usage in order to set the fps to 25:

		   fps=fps=25

       •   Sets the fps to 24, using abbreviation and rounding method to round
	   to nearest:

		   fps=fps=film:round=near

   framepack
       Pack  two  different  video  streams into a stereoscopic video, setting
       proper metadata on supported codecs. The two views should have the same
       size and framerate and processing will  stop  when  the	shorter	 video
       ends. Please note that you may conveniently adjust view properties with
       the scale and fps filters.

       It accepts the following parameters:

       format
	   The desired packing format. Supported values are:

	   sbs The views are next to each other (default).

	   tab The views are on top of each other.

	   lines
	       The views are packed by line.

	   columns
	       The views are packed by column.

	   frameseq
	       The views are temporally interleaved.

       Some examples:

	       # Convert left and right views into a frame-sequential video
	       ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

	       # Convert views into a side-by-side video with the same output resolution as the input
	       ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

   framerate
       Change the frame rate by interpolating new video output frames from the
       source frames.

       This  filter  is	 not  designed	to  function correctly with interlaced
       media. If you wish to change the frame rate of  interlaced  media  then
       you  are	 required  to  deinterlace before this filter and re-interlace
       after this filter.

       A description of the accepted options follows.

       fps Specify the output frames per  second.  This	 option	 can  also  be
	   specified as a value alone. The default is 50.

       interp_start
	   Specify the start of a range where the output frame will be created
	   as  a linear interpolation of two frames. The range is [0-255], the
	   default is 15.

       interp_end
	   Specify the end of a range where the output frame will  be  created
	   as  a linear interpolation of two frames. The range is [0-255], the
	   default is 240.

       scene
	   Specify the level at which a scene change is detected  as  a	 value
	   between  0  and 100 to indicate a new scene; a low value reflects a
	   low probability for the current frame to  introduce	a  new	scene,
	   while  a  higher value means the current frame is more likely to be
	   one.	 The default is 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

       •   Apply   the	 distort0r   effect,  setting  the  first  two	double
	   parameters:

		   frei0r=filter_name=distort0r:filter_params=0.5|0.01

       •   Apply the  colordistance  effect,  taking  a	 color	as  the	 first
	   parameter:

		   frei0r=colordistance:0.2/0.3/0.4
		   frei0r=colordistance:violet
		   frei0r=colordistance:0x112233

       •   Apply the perspective effect, specifying the top left and top right
	   image positions:

		   frei0r=perspective:0.2/0.2|0.8/0.2

       For more information, see <http://frei0r.dyne.org>

       Commands

       This filter supports the filter_params option as commands.

   fspp
       Apply fast and simple postprocessing. It is a faster version of spp.

       It  splits  (I)DCT  into	 horizontal/vertical passes. Unlike the simple
       post- processing filter, one of them is performed once per  block,  not
       per pixel.  This allows for much higher speed.

       The filter accepts the following options:

       quality
	   Set	 quality.  This	 option	 defines  the  number  of  levels  for
	   averaging. It accepts an integer in the range 4-5. Default value is
	   4.

       qp  Force a constant quantization parameter. It accepts an  integer  in
	   range  0-63.	 If not set, the filter will use the QP from the video
	   stream (if available).

       strength
	   Set filter strength. It accepts an integer  in  range  -15  to  32.
	   Lower  values  mean	more  details  but  also more artifacts, while
	   higher values make the image smoother but  also  blurrier.  Default
	   value is 0 − 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 luma expression.

       cb_expr, cb
	   Set the chrominance blue expression.

       cr_expr, cr
	   Set the chrominance red expression.

       alpha_expr, a
	   Set the alpha expression.

       red_expr, r
	   Set the red expression.

       green_expr, g
	   Set the green expression.

       blue_expr, b
	   Set the blue expression.

       The colorspace is selected according to the specified options.  If  one
       of  the	lum_expr, cb_expr, or cr_expr options is specified, the filter
       will automatically select a YCbCr colorspace. If one of	the  red_expr,
       green_expr,  or	blue_expr  options is specified, it will select an RGB
       colorspace.

       If one of the chrominance expression is not defined, it falls  back  on
       the  other one. If no alpha expression is specified it will evaluate to
       opaque value.  If none of chrominance expressions are  specified,  they
       will evaluate to the luma expression.

       The expressions can use the following variables and functions:

       N   The sequential number of the filtered frame, starting from 0.

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       SW
       SH  Width  and  height scale depending on the currently filtered plane.
	   It is the ratio between the	corresponding  luma  plane  number  of
	   pixels and the current plane ones. E.g. for YUV4:2:0 the values are
	   "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

       T   Time of the current frame, expressed in seconds.

       p(x, y)
	   Return  the	value  of  the	pixel at location (x,y) of the current
	   plane.

       lum(x, y)
	   Return the value of the pixel at location (x,y) of the luma plane.

       cb(x, y)
	   Return the value of the  pixel  at  location	 (x,y)	of  the	 blue-
	   difference chroma plane. Return 0 if there is no such plane.

       cr(x, y)
	   Return  the	value  of  the	pixel  at  location  (x,y) of the red-
	   difference chroma plane. Return 0 if there is no such plane.

       r(x, y)
       g(x, y)
       b(x, y)
	   Return  the	value  of  the	pixel  at  location   (x,y)   of   the
	   red/green/blue component. Return 0 if there is no such component.

       alpha(x, y)
	   Return the value of the pixel at location (x,y) of the alpha plane.
	   Return 0 if there is no such plane.

       psum(x,y), lumsum(x, y), cbsum(x,y), crsum(x,y), rsum(x,y), gsum(x,y),
       bsum(x,y), alphasum(x,y)
	   Sum	of  sample  values  in the rectangle from (0,0) to (x,y), this
	   allows obtaining sums  of  samples  within  a  rectangle.  See  the
	   functions without the sum postfix.

       interpolation
	   Set one of interpolation methods:

	   nearest, n
	   bilinear, b

	   Default is bilinear.

       For  functions,	if  x  and  y  are outside the area, the value will be
       automatically clipped to the closer edge.

       Please note that this filter can use multiple  threads  in  which  case
       each  slice will have its own expression state. If you want to use only
       a single expression state because your expressions depend  on  previous
       state then you should limit the number of filter threads to 1.

       Examples

       •   Flip the image horizontally:

		   geq=p(W-X\,Y)

       •   Generate  a	bidimensional  sine  wave,  with  angle	 "PI/3"	 and a
	   wavelength of 100 pixels:

		   geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

       •   Generate a fancy enigmatic moving light:

		   nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128

       •   Generate a quick emboss effect:

		   format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

       •   Modify RGB components depending on pixel position:

		   geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

       •   Create a radial gradient that is the same size as the  input	 (also
	   see the vignette filter):

		   geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

   gradfun
       Fix  the	 banding  artifacts  that are sometimes introduced into nearly
       flat regions by truncation  to  8-bit  color  depth.   Interpolate  the
       gradients that should go where the bands are, and dither them.

       It  is  designed	 for  playback	only.	Do  not	 use it prior to lossy
       compression, because compression tends to lose  the  dither  and	 bring
       back the bands.

       It accepts the following parameters:

       strength
	   The	maximum	 amount by which the filter will change any one pixel.
	   This is also the  threshold	for  detecting	nearly	flat  regions.
	   Acceptable  values  range from .51 to 64; the default value is 1.2.
	   Out-of-range values will be clipped to the valid range.

       radius
	   The neighborhood to fit the gradient to. A larger radius makes  for
	   smoother gradients, but also prevents the filter from modifying the
	   pixels  near	 detailed  regions.  Acceptable	 values	 are 8-32; the
	   default value is 16. Out-of-range values will  be  clipped  to  the
	   valid range.

       Alternatively,	the  options  can  be  specified  as  a	 flat  string:
       strength[:radius]

       Examples

       •   Apply the filter with a 3.5 strength and radius of 8:

		   gradfun=3.5:8

       •   Specify radius, omitting the strength (which will fall-back to  the
	   default value):

		   gradfun=radius=8

   graphmonitor
       Show various filtergraph stats.

       With this filter one can debug complete filtergraph.  Especially issues
       with links filling with queued frames.

       The filter accepts the following options:

       size, s
	   Set video output size. Default is hd720.

       opacity, o
	   Set video opacity. Default is 0.9. Allowed range is from 0 to 1.

       mode, m
	   Set output mode flags.

	   Available values for flags are:

	   full
	       No any filtering. Default.

	   compact
	       Show only filters with queued frames.

	   nozero
	       Show only filters with non-zero stats.

	   noeof
	       Show only filters with non-eof stat.

	   nodisabled
	       Show only filters that are enabled in timeline.

       flags, f
	   Set flags which enable which stats are shown in video.

	   Available values for flags are:

	   none
	       All flags turned off.

	   all All flags turned on.

	   queue
	       Display number of queued frames in each link.

	   frame_count_in
	       Display number of frames taken from filter.

	   frame_count_out
	       Display number of frames given out from filter.

	   frame_count_delta
	       Display delta number of frames between above two values.

	   pts Display current filtered frame pts.

	   pts_delta
	       Display pts delta between current and previous frame.

	   time
	       Display current filtered frame time.

	   time_delta
	       Display time delta between current and previous frame.

	   timebase
	       Display time base for filter link.

	   format
	       Display used format for filter link.

	   size
	       Display video size or number of audio channels in case of audio
	       used by filter link.

	   rate
	       Display	video  frame rate or sample rate in case of audio used
	       by filter link.

	   eof Display link output status.

	   sample_count_in
	       Display number of samples taken from filter.

	   sample_count_out
	       Display number of samples given out from filter.

	   sample_count_delta
	       Display delta number of samples between above two values.

	   disabled
	       Show the timeline filter status.

       rate, r
	   Set upper limit for video rate of output stream, Default  value  is
	   25.	This guarantee that output video frame rate will not be higher
	   than this value.

   grayworld
       A  color	 constancy  filter  that applies color correction based on the
       grayworld assumption

       See:
       <https://www.researchgate.net/publication/275213614_A_New_Color_Correction_Method_for_Underwater_Imaging>

       The algorithm  uses linear light, so input data	should	be  linearized
       beforehand (and possibly correctly tagged).

	       ffmpeg -i INPUT -vf zscale=transfer=linear,grayworld,zscale=transfer=bt709,format=yuv420p OUTPUT

   greyedge
       A  color	 constancy variation filter which estimates scene illumination
       via grey edge algorithm and corrects the scene colors accordingly.

       See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>

       The filter accepts the following options:

       difford
	   The order of differentiation to be applied on the  scene.  Must  be
	   chosen in the range [0,2] and default value is 1.

       minknorm
	   The	Minkowski  parameter  to be used for calculating the Minkowski
	   distance. Must be chosen in the range [0,20] and default  value  is
	   1.  Set to 0 for getting max value instead of calculating Minkowski
	   distance.

       sigma
	   The standard deviation of Gaussian blur to be applied on the scene.
	   Must be chosen in the range	[0,1024.0]  and	 default  value	 =  1.
	   floor(  sigma * break_off_sigma(3) ) can't be equal to 0 if difford
	   is greater than 0.

       Examples

       •   Grey Edge:

		   greyedge=difford=1:minknorm=5:sigma=2

       •   Max Edge:

		   greyedge=difford=1:minknorm=0:sigma=2

   guided
       Apply guided filter for edge-preserving smoothing, dehazing and so on.

       The filter accepts the following options:

       radius
	   Set the box radius in pixels.  Allowed range is 1 to 20. Default is
	   3.

       eps Set regularization parameter (with square).	Allowed range is 0  to
	   1. Default is 0.01.

       mode
	   Set filter mode. Can be "basic" or "fast".  Default is "basic".

       sub Set	subsampling  ratio for "fast" mode.  Range is 2 to 64. Default
	   is 4.  No subsampling occurs in "basic" mode.

       guidance
	   Set guidance mode. Can be "off" or  "on".  Default  is  "off".   If
	   "off",  single  input is required.  If "on", two inputs of the same
	   resolution and pixel format are required.  The second input	serves
	   as the guidance.

       planes
	   Set planes to filter. Default is first only.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Edge-preserving smoothing with guided filter:

		   ffmpeg -i in.png -vf guided out.png

       •   Dehazing, structure-transferring filtering, detail enhancement with
	   guided  filter.   For  the  generation  of guidance image, refer to
	   paper	"Guided	       Image	     Filtering".	  See:
	   <http://kaiminghe.com/publications/pami12guidedfilter.pdf>.

		   ffmpeg -i in.png -i guidance.png -filter_complex guided=guidance=on out.png

   haldclut
       Apply a Hald CLUT to a video stream.

       First  input is the video stream to process, and second one is the Hald
       CLUT.  The Hald CLUT input can be a simple picture or a complete	 video
       stream.

       The filter accepts the following options:

       clut
	   Set	which  CLUT  video  frames will be processed from second input
	   stream, can be first or all. Default is all.

       shortest
	   Force termination when the shortest input terminates. Default is 0.

       repeatlast
	   Continue applying the last CLUT after the  end  of  the  stream.  A
	   value  of  0 disable the filter after the last frame of the CLUT is
	   reached.  Default is 1.

       "haldclut" also has the	same  interpolation  options  as  lut3d	 (both
       filters share the same internals).

       This filter also supports the framesync options.

       More  information about the Hald CLUT can be found on Eskil Steenberg's
       website		 (Hald		 CLUT		 author)	    at
       <http://www.quelsolaar.com/technology/clut.html>.

       Commands

       This filter supports the "interp" option as commands.

       Workflow examples

       Hald CLUT video stream

       Generate an identity Hald CLUT stream altered with various effects:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

       Note: make sure you use a lossless codec.

       Then use it with "haldclut" to apply it on some random stream:

	       ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

       The  Hald  CLUT	will  be  applied to the 10 first seconds (duration of
       clut.nut), then the latest picture of that CLUT stream will be  applied
       to the remaining frames of the "mandelbrot" stream.

       Hald CLUT with preview

       A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by
       "Level*Level*Level"  pixels.  For a given Hald CLUT, FFmpeg will select
       the biggest possible square starting at the top left  of	 the  picture.
       The  remaining  padding	pixels (bottom or right) will be ignored. This
       area can be used to add a preview of the Hald CLUT.

       Typically, the following generated Hald CLUT will be supported  by  the
       "haldclut" filter:

	       ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
		  pad=iw+320 [padded_clut];
		  smptebars=s=320x256, split [a][b];
		  [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
		  [main][b] overlay=W-320" -frames:v 1 clut.png

       It contains the original and a preview of the effect of the CLUT: SMPTE
       color  bars  are	 displayed  on the right-top, and below the same color
       bars processed by the color changes.

       Then, the effect of this Hald CLUT can be visualized with:

	       ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

   hflip
       Flip the input video horizontally.

       For example, to horizontally flip the input video with ffmpeg:

	       ffmpeg -i in.avi -vf "hflip" out.avi

   histeq
       This filter applies a global color histogram  equalization  on  a  per-
       frame basis.

       It  can	be  used to correct video that has a compressed range of pixel
       intensities.   The  filter  redistributes  the  pixel  intensities   to
       equalize	 their	distribution  across  the  intensity  range. It may be
       viewed as an "automatically adjusting contrast filter". This filter  is
       useful only for correcting degraded or poorly captured source video.

       The filter accepts the following options:

       strength
	   Determine  the  amount  of  equalization  to	 be  applied.	As the
	   strength is reduced, the distribution of  pixel  intensities	 more-
	   and-more  approaches	 that  of the input frame. The value must be a
	   float number in the range [0,1] and defaults to 0.200.

       intensity
	   Set the maximum intensity that can generated and scale  the	output
	   values  appropriately.   The	 strength should be set as desired and
	   then the intensity can be limited if needed to  avoid  washing-out.
	   The value must be a float number in the range [0,1] and defaults to
	   0.210.

       antibanding
	   Set the antibanding level. If enabled the filter will randomly vary
	   the	luminance  of output pixels by a small amount to avoid banding
	   of the histogram. Possible values are "none", "weak"	 or  "strong".
	   It defaults to "none".

   histogram
       Compute and draw a color distribution histogram for the input video.

       The  computed  histogram	 is  a	representation	of the color component
       distribution in an image.

       Standard histogram displays the color  components  distribution	in  an
       image.	 Displays   color   graph  for	each  color  component.	 Shows
       distribution of the Y, U, V, A or R,  G,	 B  components,	 depending  on
       input  format, in the current frame. Below each graph a color component
       scale meter is shown.

       The filter accepts the following options:

       level_height
	   Set height of level. Default value is 200.  Allowed range  is  [50,
	   2048].

       scale_height
	   Set	height	of color scale. Default value is 12.  Allowed range is
	   [0, 40].

       display_mode
	   Set display mode.  It accepts the following values:

	   stack
	       Per color component graphs are placed below each other.

	   parade
	       Per color component graphs are placed side by side.

	   overlay
	       Presents information identical to that in the "parade",	except
	       that  the graphs representing color components are superimposed
	       directly over one another.

	   Default is "stack".

       levels_mode
	   Set mode. Can be either "linear",  or  "logarithmic".   Default  is
	   "linear".

       components
	   Set what color components to display.  Default is 7.

       fgopacity
	   Set foreground opacity. Default is 0.7.

       bgopacity
	   Set background opacity. Default is 0.5.

       colors_mode
	   Set colors mode.  It accepts the following values:

	   whiteonblack
	   blackonwhite
	   whiteongray
	   blackongray
	   coloronblack
	   coloronwhite
	   colorongray
	   blackoncolor
	   whiteoncolor
	   grayoncolor

	   Default is "whiteonblack".

       Examples

       •   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:

       •   Hardware frame input, normal frame output

	   Map the input frames to system memory and pass them to the  output.
	   If  the  original  hardware	frame  is later required (for example,
	   after overlaying something else on part of it),  the	 hwmap	filter
	   can be used again in the next mode to retrieve it.

       •   Normal frame input, hardware frame output

	   If  the  input  is  actually a software-mapped hardware frame, then
	   unmap it - that is, return the original hardware frame.

	   Otherwise, a device must be provided.  Create new hardware surfaces
	   on that device for the output, then map them back to	 the  software
	   format  at the input and give those frames to the preceding filter.
	   This will then act like the hwupload filter, but  may  be  able  to
	   avoid  an additional copy when the input is already in a compatible
	   format.

       •   Hardware frame input and output

	   A device must be supplied for the output, either directly  or  with
	   the	derive_device option.  The input and output devices must be of
	   different types and compatible -  the  exact	 meaning  of  this  is
	   system-dependent,  but  typically  it means that they must refer to
	   the same underlying hardware context (for  example,	refer  to  the
	   same graphics card).

	   If  the  input frames were originally created on the output device,
	   then unmap to retrieve the original frames.

	   Otherwise, map the  frames  to  the	output	device	-  create  new
	   hardware  frames  on	 the output corresponding to the frames on the
	   input.

       The following additional parameters are accepted:

       mode
	   Set the frame mapping mode.	Some combination of:

	   read
	       The mapped frame should be readable.

	   write
	       The mapped frame should be writeable.

	   overwrite
	       The mapping will always overwrite the entire frame.

	       This may improve performance in some  cases,  as	 the  original
	       contents of the frame need not be loaded.

	   direct
	       The mapping must not involve any copying.

	       Indirect mappings to copies of frames are created in some cases
	       where  either  direct  mapping is not possible or it would have
	       unexpected properties.  Setting	this  flag  ensures  that  the
	       mapping is direct and will fail if that is not possible.

	   Defaults to read+write if not specified.

       derive_device type
	   Rather  than	 using	the device supplied at initialisation, instead
	   derive a new device of type type from the device the	 input	frames
	   exist on.

       reverse
	   In  a  hardware to hardware mapping, map in reverse - create frames
	   in the sink and map them back to the source.	 This may be necessary
	   in some cases where a mapping in one direction is required but only
	   the opposite direction is supported by the devices being used.

	   This option is dangerous - it may break  the	 preceding  filter  in
	   undefined  ways  if	there  are  any additional constraints on that
	   filter's output.  Do not use it  without  fully  understanding  the
	   implications of its use.

   hwupload
       Upload system memory frames to hardware surfaces.

       The   device  to	 upload	 to  must  be  supplied	 when  the  filter  is
       initialised.  If using ffmpeg, select the appropriate device  with  the
       -filter_hw_device  option  or with the derive_device option.  The input
       and output devices must be of different	types  and  compatible	-  the
       exact  meaning of this is system-dependent, but typically it means that
       they must refer to the same underlying hardware context	(for  example,
       refer to the same graphics card).

       The following additional parameters are accepted:

       derive_device type
	   Rather  than	 using	the device supplied at initialisation, instead
	   derive a new device of type type from the device the	 input	frames
	   exist on.

   hwupload_cuda
       Upload system memory frames to a CUDA device.

       It accepts the following optional parameters:

       device
	   The number of the CUDA device to use

   hqx
       Apply  a high-quality magnification filter designed for pixel art. This
       filter was originally created by Maxim Stepin.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "hq2x", 3 for  "hq3x"  and	4  for
	   "hq4x".  Default is 3.

   hstack
       Stack input videos horizontally.

       All streams must be of same pixel format and of same height.

       Note  that  this	 filter is faster than using overlay and pad filter to
       create same output.

       The filter accepts the following option:

       inputs
	   Set number of input streams. Default is 2.

       shortest
	   If set to 1, force the output to terminate when the shortest	 input
	   terminates. Default value is 0.

   hsvhold
       Turns a certain HSV range into gray values.

       This  filter measures color difference between set HSV color in options
       and ones measured in video stream. Depending on options, output	colors
       can be changed to be gray or not.

       The filter accepts the following options:

       hue Set	 the  hue  value  which	 will  be  used	 in  color  difference
	   calculation.	 Allowed range is from -360 to 360. Default  value  is
	   0.

       sat Set	the  saturation	 value	which will be used in color difference
	   calculation.	 Allowed range is from -1 to 1. Default value is 0.

       val Set the value which will be used in color  difference  calculation.
	   Allowed range is from -1 to 1. Default value is 0.

       similarity
	   Set	similarity  percentage	with  the key color.  Allowed range is
	   from 0 to 1. Default value is 0.01.

	   0.00001 matches  only  the  exact  key  color,  while  1.0  matches
	   everything.

       blend
	   Blend  percentage.	Allowed range is from 0 to 1. Default value is
	   0.

	   0.0 makes pixels either fully gray, or not gray at all.

	   Higher values result in more gray pixels, with a higher gray	 pixel
	   the more similar the pixels color is to the key color.

   hsvkey
       Turns a certain HSV range into transparency.

       This  filter measures color difference between set HSV color in options
       and ones measured in video stream. Depending on options, output	colors
       can be changed to transparent by adding alpha channel.

       The filter accepts the following options:

       hue Set	 the  hue  value  which	 will  be  used	 in  color  difference
	   calculation.	 Allowed range is from -360 to 360. Default  value  is
	   0.

       sat Set	the  saturation	 value	which will be used in color difference
	   calculation.	 Allowed range is from -1 to 1. Default value is 0.

       val Set the value which will be used in color  difference  calculation.
	   Allowed range is from -1 to 1. Default value is 0.

       similarity
	   Set	similarity  percentage	with  the key color.  Allowed range is
	   from 0 to 1. Default value is 0.01.

	   0.00001 matches  only  the  exact  key  color,  while  1.0  matches
	   everything.

       blend
	   Blend  percentage.	Allowed range is from 0 to 1. Default value is
	   0.

	   0.0 makes pixels either fully transparent, or  not  transparent  at
	   all.

	   Higher  values  result  in  semi-transparent	 pixels, with a higher
	   transparency the more similar the pixels color is to the key color.

   hue
       Modify the hue and/or the saturation of the input.

       It accepts the following parameters:

       h   Specify the hue angle  as  a	 number	 of  degrees.  It  accepts  an
	   expression, and defaults to "0".

       s   Specify  the	 saturation  in	 the  [-10,10]	range.	It  accepts an
	   expression and defaults to "1".

       H   Specify the hue angle  as  a	 number	 of  radians.  It  accepts  an
	   expression, and defaults to "0".

       b   Specify  the	 brightness  in	 the  [-10,10]	range.	It  accepts an
	   expression and defaults to "0".

       h and H are mutually exclusive, and can't  be  specified	 at  the  same
       time.

       The  b,	h,  H  and  s  option  values  are  expressions containing the
       following constants:

       n   frame count of the input frame starting from 0

       pts presentation timestamp of the input frame expressed	in  time  base
	   units

       r   frame  rate	of  the	 input	video,	NAN if the input frame rate is
	   unknown

       t   timestamp expressed in seconds,  NAN	 if  the  input	 timestamp  is
	   unknown

       tb  time base of the input video

       Examples

       •   Set the hue to 90 degrees and the saturation to 1.0:

		   hue=h=90:s=1

       •   Same command but expressing the hue in radians:

		   hue=H=PI/2:s=1

       •   Rotate  hue	and  make  the saturation swing between 0 and 2 over a
	   period of 1 second:

		   hue="H=2*PI*t: s=sin(2*PI*t)+1"

       •   Apply a 3 seconds saturation fade-in effect starting at 0:

		   hue="s=min(t/3\,1)"

	   The general fade-in expression can be written as:

		   hue="s=min(0\, max((t-START)/DURATION\, 1))"

       •   Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

		   hue="s=max(0\, min(1\, (8-t)/3))"

	   The general fade-out expression can be written as:

		   hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

       Commands

       This filter supports the following commands:

       b
       s
       h
       H   Modify the hue and/or the saturation and/or brightness of the input
	   video.  The command accepts the same syntax	of  the	 corresponding
	   option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   huesaturation
       Apply hue-saturation-intensity adjustments to input video stream.

       This filter operates in RGB colorspace.

       This filter accepts the following options:

       hue Set the hue shift in degrees to apply. Default is 0.	 Allowed range
	   is from -180 to 180.

       saturation
	   Set	the  saturation shift. Default is 0.  Allowed range is from -1
	   to 1.

       intensity
	   Set the intensity shift. Default is 0.  Allowed range is from -1 to
	   1.

       colors
	   Set	which  primary	and  complementary  colors  are	 going	to  be
	   adjusted.  This options is set by providing one or multiple values.
	   This	 can select multiple colors at once. By default all colors are
	   selected.

	   r   Adjust reds.

	   y   Adjust yellows.

	   g   Adjust greens.

	   c   Adjust cyans.

	   b   Adjust blues.

	   m   Adjust magentas.

	   a   Adjust all colors.

       strength
	   Set strength of filtering. Allowed range is from 0 to 100.  Default
	   value is 1.

       rw, gw, bw
	   Set weight for each RGB component. Allowed range is from  0	to  1.
	   By  default	is set to 0.333, 0.334, 0.333.	Those options are used
	   in saturation and lightess processing.

       lightness
	   Set preserving lightness, by default is disabled.   Adjusting  hues
	   can	change	lightness  from original RGB triplet, with this option
	   enabled lightness is kept at same value.

   hysteresis
       Grow first stream into second stream by	connecting  components.	  This
       makes it possible to build more robust edge masks.

       This filter accepts the following options:

       planes
	   Set	which  planes  will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       threshold
	   Set threshold which is used in filtering. If pixel component	 value
	   is	higher	 than  this  value  filter  algorithm  for  connecting
	   components is activated.  By default value is 0.

       The "hysteresis" filter also supports the framesync options.

   iccdetect
       Detect the colorspace  from an embedded ICC profile (if	present),  and
       update the frame's tags accordingly.

       This filter accepts the following options:

       force
	   If  true,  the  frame's  existing  colorspace  tags	will always be
	   overridden by values detected from an ICC profile. Otherwise,  they
	   will	 only  be  assigned  if	 they  contain	"unknown".  Enabled by
	   default.

   iccgen
       Generate ICC profiles and attach them to frames.

       This filter accepts the following options:

       color_primaries
       color_trc
	   Configure the colorspace that the ICC  profile  will	 be  generated
	   for.	 The  default  value of "auto" infers the value from the input
	   frame's metadata, defaulting to BT.709/sRGB as appropriate.

	   See the setparams filter for a list of possible  values,  but  note
	   that "unknown" are not valid values for this filter.

       force
	   If  true,  an  ICC  profile	will  be  generated  even  if it would
	   overwrite an already existing ICC profile. Disabled by default.

   identity
       Obtain the identity score between two input videos.

       This filter takes two input videos.

       Both input videos must have the same resolution and  pixel  format  for
       this  filter  to	 work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained per component, average, min	 and  max  identity  score  is
       printed through the logging system.

       The filter stores the calculated identity scores of each frame in frame
       metadata.

       This filter also supports the framesync options.

       In  the	below  example	the  input  file  main.mpg  being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi identity -f null -

   idet
       Detect video interlacing type.

       This filter tries  to  detect  if  the  input  frames  are  interlaced,
       progressive,  top  or  bottom  field  first. It will also try to detect
       fields that are repeated between adjacent frames (a sign of telecine).

       Single frame detection considers only immediately adjacent frames  when
       classifying  each  frame.   Multiple  frame  detection incorporates the
       classification history of previous frames.

       The filter will log these metadata values:

       single.current_frame
	   Detected type of current frame using	 single-frame  detection.  One
	   of:	``tff''	 (top  field  first),  ``bff''	(bottom	 field first),
	   ``progressive'', or ``undetermined''

       single.tff
	   Cumulative number of frames	detected  as  top  field  first	 using
	   single-frame detection.

       multiple.tff
	   Cumulative  number  of  frames  detected  as	 top field first using
	   multiple-frame detection.

       single.bff
	   Cumulative number of frames detected as bottom  field  first	 using
	   single-frame detection.

       multiple.current_frame
	   Detected  type of current frame using multiple-frame detection. One
	   of: ``tff''	(top  field  first),  ``bff''  (bottom	field  first),
	   ``progressive'', or ``undetermined''

       multiple.bff
	   Cumulative  number  of  frames detected as bottom field first using
	   multiple-frame detection.

       single.progressive
	   Cumulative number of frames detected as progressive	using  single-
	   frame detection.

       multiple.progressive
	   Cumulative number of frames detected as progressive using multiple-
	   frame detection.

       single.undetermined
	   Cumulative  number  of  frames  that	 could not be classified using
	   single-frame detection.

       multiple.undetermined
	   Cumulative number of frames that  could  not	 be  classified	 using
	   multiple-frame detection.

       repeated.current_frame
	   Which  field in the current frame is repeated from the last. One of
	   ``neither'', ``top'', or ``bottom''.

       repeated.neither
	   Cumulative number of frames with no repeated field.

       repeated.top
	   Cumulative number of frames with the top field  repeated  from  the
	   previous frame's top field.

       repeated.bottom
	   Cumulative number of frames with the bottom field repeated from the
	   previous frame's bottom field.

       The filter accepts the following options:

       intl_thres
	   Set interlacing threshold.

       prog_thres
	   Set progressive threshold.

       rep_thres
	   Threshold for repeated field detection.

       half_life
	   Number  of  frames  after which a given frame's contribution to the
	   statistics  is  halved  (i.e.,  it  contributes  only  0.5  to  its
	   classification).  The  default  of 0 means that all frames seen are
	   given full weight of 1.0 forever.

       analyze_interlaced_flag
	   When this is not 0 then idet	 will  use  the	 specified  number  of
	   frames to determine if the interlaced flag is accurate, it will not
	   count  undetermined frames.	If the flag is found to be accurate it
	   will be used without any further computations, if it is found to be
	   inaccurate it will be cleared  without  any	further	 computations.
	   This allows inserting the idet filter as a low computational method
	   to clean up the interlaced flag

   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

       •   Apply default values:

		   kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

       •   Enable additional sharpening:

		   kerndeint=sharp=1

       •   Paint processed pixels in white:

		   kerndeint=map=1

   kirsch
       Apply kirsch operator to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes will be  processed,	 unprocessed  planes  will  be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

       Commands

       This filter supports the all above options as commands.

   lagfun
       Slowly update darker pixels.

       This  filter  makes  short flashes of light appear longer.  This filter
       accepts the following options:

       decay
	   Set factor for decaying. Default is .95. Allowed range is from 0 to
	   1.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Commands

       This filter supports the all above options as commands.

   lenscorrection
       Correct radial lens distortion

       This filter can be used to correct for radial distortion as can	result
       from the use of wide angle lenses, and thereby re-rectify the image. To
       find  the  right	 parameters one can use tools available for example as
       part of opencv or  simply  trial-and-error.   To	 use  opencv  use  the
       calibration  sample  (under  samples/cpp)  from	the opencv sources and
       extract the k1 and k2 coefficients from the resulting matrix.

       Note that effectively the same filter is available in  the  open-source
       tools Krita and Digikam from the KDE project.

       In  contrast  to	 the  vignette	filter,	 which	can  also  be  used to
       compensate lens errors, this filter  corrects  the  distortion  of  the
       image,  whereas	vignette  corrects the brightness distribution, so you
       may want to use both filters together in certain cases, though you will
       have to take care  of  ordering,	 i.e.  whether	vignetting  should  be
       applied before or after lens correction.

       Options

       The filter accepts the following options:

       cx  Relative  x-coordinate of the focal point of the image, and thereby
	   the center of the distortion. This value has a range [0,1]  and  is
	   expressed as fractions of the image width. Default is 0.5.

       cy  Relative  y-coordinate of the focal point of the image, and thereby
	   the center of the distortion. This value has a range [0,1]  and  is
	   expressed as fractions of the image height. Default is 0.5.

       k1  Coefficient	of  the	 quadratic  correction	term. This value has a
	   range [-1,1]. 0 means no correction. Default is 0.

       k2  Coefficient of the double quadratic correction term. This value has
	   a range [-1,1].  0 means no correction. Default is 0.

       i   Set interpolation type. Can be "nearest" or "bilinear".  Default is
	   "nearest".

       fc  Specify the color of the unmapped pixels. For the  syntax  of  this
	   option,  check  the	"Color"	 section  in  the ffmpeg-utils manual.
	   Default color is "black@0".

       The formula that generates the correction is:

       r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

       where r_0 is halve of the image diagonal and r_src and  r_tgt  are  the
       distances  from	the  focal  point  in  the  source  and target images,
       respectively.

       Commands

       This filter supports the all above options as commands.

   lensfun
       Apply	 lens	  correction	 via	 the	  lensfun      library
       (<http://lensfun.sourceforge.net/>).

       The  "lensfun"  filter requires the camera make, camera model, and lens
       model to apply the lens correction. The filter will  load  the  lensfun
       database and query it to find the corresponding camera and lens entries
       in  the	database. As long as these entries can be found with the given
       options, the filter  can	 perform  corrections  on  frames.  Note  that
       incomplete  strings  will  result in the filter choosing the best match
       with the given options, and the filter will output  the	chosen	camera
       and  lens models (logged with level "info"). You must provide the make,
       camera model, and lens model as they are required.

       To obtain a list of available makes and models, leave out one  or  both
       of  "make"  and	"model" options. The filter will send the full list to
       the log with level "INFO".  The first column is the make and the second
       column is the model.  To obtain a list of  available  lenses,  set  any
       values  for  make  and model and leave out the "lens_model" option. The
       filter will send the full list of lenses in the log with level  "INFO".
       The ffmpeg tool will exit after the list is printed.

       The filter accepts the following options:

       make
	   The	make  of  the  camera  (for  example, "Canon"). This option is
	   required.

       model
	   The model of the camera  (for  example,  "Canon  EOS	 100D").  This
	   option is required.

       lens_model
	   The	model  of the lens (for example, "Canon EF-S 18-55mm f/3.5-5.6
	   IS STM"). This option is required.

       db_path
	   The full path to the lens database folder. If not set,  the	filter
	   will	 attempt  to  load the database from the install path when the
	   library was built. Default is unset.

       mode
	   The type of correction to apply. The	 following  values  are	 valid
	   options:

	   vignetting
	       Enables fixing lens vignetting.

	   geometry
	       Enables fixing lens geometry. This is the default.

	   subpixel
	       Enables fixing chromatic aberrations.

	   vig_geo
	       Enables fixing lens vignetting and lens geometry.

	   vig_subpixel
	       Enables fixing lens vignetting and chromatic aberrations.

	   distortion
	       Enables fixing both lens geometry and chromatic aberrations.

	   all Enables all possible corrections.

       focal_length
	   The	focal  length  of the image/video (zoom; expected constant for
	   video). For example, a 18--55mm lens	 has  focal  length  range  of
	   [18--55], so a value in that range should be chosen when using that
	   lens. Default 18.

       aperture
	   The aperture of the image/video (expected constant for video). Note
	   that aperture is only used for vignetting correction. Default 3.5.

       focus_distance
	   The	focus  distance	 of  the  image/video  (expected  constant for
	   video). Note that focus distance is only used  for  vignetting  and
	   only	  slightly  affects  the  vignetting  correction  process.  If
	   unknown, leave it at the default value (which is 1000).

       scale
	   The scale factor  which  is	applied	 after	transformation.	 After
	   correction  the  video  is  no longer necessarily rectangular. This
	   parameter controls how much of the resulting image is visible.  The
	   value  0  means that a value will be chosen automatically such that
	   there is little or no unmapped area in the output image. 1.0	 means
	   that no additional scaling is done. Lower values may result in more
	   of the corrected image being visible, while higher values may avoid
	   unmapped areas in the output.

       target_geometry
	   The target geometry of the output image/video. The following values
	   are valid options:

	   rectilinear (default)
	   fisheye
	   panoramic
	   equirectangular
	   fisheye_orthographic
	   fisheye_stereographic
	   fisheye_equisolid
	   fisheye_thoby
       reverse
	   Apply  the  reverse	of  image  correction  (instead	 of correcting
	   distortion, apply it).

       interpolation
	   The type of interpolation  used  when  correcting  distortion.  The
	   following values are valid options:

	   nearest
	   linear (default)
	   lanczos

       Examples

       •   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

       •   Apply  the  same  as	 before,  but  only for the first 5 seconds of
	   video.

		   ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8:enable='lte(t\,5)' -c:v h264 -b:v 8000k output.mov

   libplacebo
       Flexible	 GPU-accelerated  processing  filter   based   on   libplacebo
       (<https://code.videolan.org/videolan/libplacebo>).

       Options

       The options for this filter are divided into the following sections:

       Output mode

       These  options  control the overall output mode. By default, libplacebo
       will try to preserve the source colorimetry and size as best as it can,
       but it will apply any embedded film grain,  dolby  vision  metadata  or
       anamorphic SAR present in source frames.

       inputs
	   Set	the  number  of	 inputs. This can be used, alongside the "idx"
	   variable, to allow  placing/blending	 multiple  inputs  inside  the
	   output  frame.  This	 effectively  enables functionality similar to
	   hstack, overlay, etc.

       w
       h   Set the output video dimension expression. Default values are  "iw"
	   and "ih".

	   Allows for the same expressions as the scale filter.

       crop_x
       crop_y
	   Set	the input crop x/y expressions, default values are "(iw-cw)/2"
	   and "(ih-ch)/2".

       crop_w
       crop_h
	   Set the input crop width/height  expressions,  default  values  are
	   "iw" and "ih".

       pos_x
       pos_y
	   Set	the  output  placement	x/y  expressions,  default  values are
	   "(ow-pw)/2" and "(oh-ph)/2".

       pos_w
       pos_h
	   Set the output placement width/height expressions,  default	values
	   are "ow" and "oh".

       fps Set the output frame rate. This can be rational, e.g. "60000/1001".
	   If set to the special string "none" (the default), input timestamps
	   will instead be passed through to the output unmodified. Otherwise,
	   the input video frames will be interpolated as necessary to rescale
	   the	video  to  the	specified  target  framerate,  in  a manner as
	   determined by the frame_mixer option.

       format
	   Set the output format override. If unset (the default), frames will
	   be output in the  same  format  as  the  respective	input  frames.
	   Otherwise, format conversion will be performed.

       force_original_aspect_ratio
       force_divisible_by
	   Work the same as the identical scale filter options.

       normalize_sar
	   If  enabled, output frames will always have a pixel aspect ratio of
	   1:1. This will introduce additional padding/cropping as  necessary.
	   If  disabled	 (the default), any aspect ratio mismatches, including
	   those from e.g. anamorphic video  sources,  are  forwarded  to  the
	   output pixel aspect ratio.

       pad_crop_ratio
	   Specifies  a	 ratio	(between  0.0  and  1.0)  between  padding and
	   cropping when the input aspect ratio	 does  not  match  the	output
	   aspect  ratio  and  normalize_sar  is in effect. The default of 0.0
	   always pads the content with black borders, while a	value  of  1.0
	   always  crops  off  parts  of  the content. Intermediate values are
	   possible, leading to a mix of the two approaches.

       fillcolor
	   Set the color used to fill the  output  area	 not  covered  by  the
	   output  image,  for	example	 as a result of normalize_sar. For the
	   general syntax of this option, check the  "Color"  section  in  the
	   ffmpeg-utils manual. Defaults to "black".

       corner_rounding
	   Render  frames  with	 rounded  corners. The value, given as a float
	   ranging from 0.0 to 1.0, indicates the relative degree of rounding,
	   from fully square to fully circular. In other words, it  gives  the
	   radius divided by half the smaller side length. Defaults to 0.0.

       extra_opts
	   Pass	 extra libplacebo internal configuration options. These can be
	   specified as a list	of  key=value  pairs  separated	 by  ':'.  The
	   following  example  shows  how  to configure a custom filter kernel
	   ("EWA  LanczosSharp")  and  use  it	to  double  the	 input	 image
	   resolution:

		   -vf "libplacebo=w=iw*2:h=ih*2:extra_opts='upscaler=custom\:upscaler_preset=ewa_lanczos\:upscaler_blur=0.9812505644269356'"

       colorspace
       color_primaries
       color_trc
       range
	   Configure  the  colorspace that output frames will be delivered in.
	   The default value of "auto" outputs frames in the  same  format  as
	   the	input  frames,	leading	 to  no	 change.  For any other value,
	   conversion will be performed.

	   See the setparams filter for a list of possible values.

       apply_filmgrain
	   Apply film grain (e.g. AV1 or H.274) if present in  source  frames,
	   and strip it from the output. Enabled by default.

       apply_dolbyvision
	   Apply  Dolby	 Vision	 RPU metadata if present in source frames, and
	   strip it from the output.  Enabled  by  default.  Note  that	 Dolby
	   Vision  will	 always	 output BT.2020+PQ, overriding the usual input
	   frame metadata. These will also be picked as the values  of	"auto"
	   for the respective frame output options.

       In  addition  to	 the  expression  constants  documented	 for the scale
       filter, the crop_w, crop_h, crop_x, crop_y,  pos_w,  pos_h,  pos_x  and
       pos_y options can also contain the following constants:

       in_idx, idx
	   The (0-based) numeric index of the currently active input stream.

       crop_w, cw
       crop_h, ch
	   The computed values of crop_w and crop_h.

       pos_w, pw
       pos_h, ph
	   The computed values of pos_w and pos_h.

       in_t, t
	   The	input  frame  timestamp, in seconds. NAN if input timestamp is
	   unknown.

       out_t, ot
	   The input frame timestamp, in seconds. NAN if  input	 timestamp  is
	   unknown.

       n   The input frame number, starting with 0.

       Scaling

       The  options  in this section control how libplacebo performs upscaling
       and (if	necessary)  downscaling.  Note	that  libplacebo  will	always
       internally  operate on 4:4:4 content, so any sub-sampled chroma formats
       such as "yuv420p" will necessarily be upsampled and downsampled as part
       of the rendering process. That means scaling might be in effect even if
       the source and destination resolution are the same.

       upscaler
       downscaler
	   Configure the filter kernel used for upscaling and downscaling. The
	   respective defaults are "spline36" and "mitchell". For a full  list
	   of  possible	 values,  pass	"help"	to  these  options.  The  most
	   important values are:

	   none
	       Forces the use of  built-in  GPU	 texture  sampling  (typically
	       bilinear).  Extremely  fast  but	 poor quality, especially when
	       downscaling.

	   bilinear
	       Bilinear interpolation. Can generally be done for free on GPUs,
	       except when doing so would  lead	 to  aliasing.	Fast  and  low
	       quality.

	   nearest
	       Nearest-neighbour interpolation. Sharp but highly aliasing.

	   oversample
	       Algorithm  that	looks  visually	 similar  to nearest-neighbour
	       interpolation but tries to preserve pixel  aspect  ratio.  Good
	       for  pixel  art,	 since it results in minimal distortion of the
	       artistic appearance.

	   lanczos
	       Standard sinc-sinc interpolation kernel.

	   spline36
	       Cubic  spline  approximation  of	 lanczos.  No  difference   in
	       performance, but has very slightly less ringing.

	   ewa_lanczos
	       Elliptically  weighted  average	version of lanczos, based on a
	       jinc-sinc kernel.  This is also popularly referred to  as  just
	       "Jinc scaling". Slow but very high quality.

	   gaussian
	       Gaussian kernel. Has certain ideal mathematical properties, but
	       subjectively very blurry.

	   mitchell
	       Cubic  BC  spline  with	parameters recommended by Mitchell and
	       Netravali. Very little ringing.

       frame_mixer
	   Controls the kernel used for mixing frames temporally. The  default
	   value  is  "none",  which disables frame mixing. For a full list of
	   possible values, pass "help" to this	 option.  The  most  important
	   values are:

	   none
	       Disables	 frame	mixing, giving a result equivalent to "nearest
	       neighbour" semantics.

	   oversample
	       Oversamples the input video to create  a	 "Smooth  Motion"-type
	       effect: if an output frame would exactly fall on the transition
	       between	two  video  frames,  it	 is  blended  according to the
	       relative overlap.  This	is  the	 recommended  option  whenever
	       preserving the original subjective appearance is desired.

	   mitchell_clamp
	       Larger filter kernel that smoothly interpolates multiple frames
	       in  a  manner designed to eliminate ringing and other artefacts
	       as much as possible. This is the	 recommended  option  wherever
	       maximum visual smoothness is desired.

	   linear
	       Linear	blend/fade   between  frames.  Especially  useful  for
	       constructing e.g.  slideshows.

       lut_entries
	   Configures the size of scaler LUTs, ranging	from  1	 to  256.  The
	   default of 0 will pick libplacebo's internal default, typically 64.

       antiringing
	   Enables  anti-ringing (for non-EWA filters). The value (between 0.0
	   and 1.0) configures the strength of the anti-ringing algorithm. May
	   increase aliasing if set too high. Disabled by default.

       sigmoid
	   Enable sigmoidal  compression  during  upscaling.  Reduces  ringing
	   slightly.  Enabled by default.

       Debanding

       Libplacebo  comes  with	a  built-in  debanding	filter that is good at
       counteracting many common sources of banding and blocking. Turning this
       on is highly recommended whenever quality is desired.

       deband
	   Enable (fast) debanding algorithm. Disabled by default.

       deband_iterations
	   Number of  deband  iterations  of  the  debanding  algorithm.  Each
	   iteration  is  performed  with  progressively increased radius (and
	   diminished threshold).  Recommended values are in the range 1 to 4.
	   Defaults to 1.

       deband_threshold
	   Debanding filter strength. Higher numbers lead to  more  aggressive
	   debanding.  Defaults to 4.0.

       deband_radius
	   Debanding  filter  radius.  A  higher  radius  is  better  for slow
	   gradients, while a lower radius  is	better	for  steep  gradients.
	   Defaults to 16.0.

       deband_grain
	   Amount  of  extra  output  grain  to add. Helps hide imperfections.
	   Defaults to 6.0.

       Color adjustment

       A collection of subjective color controls. Not very  rigorous,  so  the
       exact  effect  will  vary somewhat depending on the input primaries and
       colorspace.

       brightness
	   Brightness boost, between -1.0 and 1.0. Defaults to 0.0.

       contrast
	   Contrast gain, between 0.0 and 16.0. Defaults to 1.0.

       saturation
	   Saturation gain, between 0.0 and 16.0. Defaults to 1.0.

       hue Hue shift in radians, between -3.14 and 3.14. Defaults to 0.0. This
	   will rotate the UV subvector, defaulting to BT.709 coefficients for
	   RGB inputs.

       gamma
	   Gamma adjustment, between 0.0 and 16.0. Defaults to 1.0.

       cones
	   Cone model to use  for  color  blindness  simulation.  Accepts  any
	   combination of "l", "m" and "s". Here are some examples:

	   m   Deuteranomaly   /   deuteranopia	  (affecting   3%-4%   of  the
	       population)

	   l   Protanomaly / protanopia (affecting 1%-2% of the population)

	   l+m Monochromacy (very rare)

	   l+m+s
	       Achromatopsy (complete loss of daytime vision, extremely rare)

       cone-strength
	   Gain factor for the cones specified by  "cones",  between  0.0  and
	   10.0.  A value of 1.0 results in no change to color vision. A value
	   of 0.0 (the default) simulates complete loss of those cones. Values
	   above 1.0 result in exaggerating  the  differences  between	cones,
	   which may help compensate for reduced color vision.

       Peak detection

       To  help	 deal with sources that only have static HDR10 metadata (or no
       tagging whatsoever), libplacebo uses its own  internal  frame  analysis
       compute	shader	to  analyze  source  frames and adapt the tone mapping
       function in realtime. If this is too slow, or if	 exactly  reproducible
       frame-perfect results are needed, it's recommended to turn this feature
       off.

       peak_detect
	   Enable  HDR peak detection. Ignores static MaxCLL/MaxFALL values in
	   favor of dynamic detection from the input. Note that	 the  detected
	   values  do  not  get written back to the output frames, they merely
	   guide the internal tone mapping process. Enabled by default.

       smoothing_period
	   Peak detection smoothing period, between  0.0  and  1000.0.	Higher
	   values result in peak detection becoming less responsive to changes
	   in the input. Defaults to 100.0.

       minimum_peak
	   Lower  bound	 on the detected peak (relative to SDR white), between
	   0.0 and 100.0. Defaults to 1.0.

       scene_threshold_low
       scene_threshold_high
	   Lower and upper thresholds for scene change detection. Expressed in
	   a logarithmic scale between 0.0 and 100.0. Default to 5.5 and 10.0,
	   respectively. Setting either to  a  negative	 value	disables  this
	   functionality.

       percentile
	   Which  percentile  of  the frame brightness histogram to use as the
	   source  peak	 for  tone-mapping.  Defaults  to  99.995,  a	fairly
	   conservative value.	Setting this to 100.0 disables frame histogram
	   measurement	and  instead  uses  the true peak brightness for tone-
	   mapping.

       Tone mapping

       The options in this  section  control  how  libplacebo  performs	 tone-
       mapping	and  gamut-mapping  when dealing with mismatches between wide-
       gamut or HDR content.  In general, libplacebo relies on accurate source
       tagging and mastering display gamut information	to  produce  the  best
       results.

       gamut_mode
	   How	to  handle  out-of-gamut  colors that can occur as a result of
	   colorimetric gamut mapping.

	   clip
	       Do nothing, simply clip out-of-range colors to the RGB  volume.
	       Low quality but extremely fast.

	   perceptual
	       Perceptually  soft-clip colors to the gamut volume. This is the
	       default.

	   relative
	       Relative colorimetric hard-clip. Similar	 to  "perceptual"  but
	       without the soft knee.

	   saturation
	       Saturation mapping, maps primaries directly to primaries in RGB
	       space.  Not recommended except for artificial computer graphics
	       for which a bright, saturated display is desired.

	   absolute
	       Absolute	 colorimetric hard-clip. Performs no adjustment of the
	       white point.

	   desaturate
	       Hard-desaturates	 out-of-gamut  colors  towards	white,	 while
	       preserving  the luminance. Has a tendency to distort the visual
	       appearance of bright objects.

	   darken
	       Linearly reduces	 content  brightness  to  preserves  saturated
	       details,	  followed  by	clipping  the  remaining  out-of-gamut
	       colors.

	   warn
	       Highlight out-of-gamut pixels (by inverting/marking them).

	   linear
	       Linearly reduces chromaticity of the entire image  to  make  it
	       fit  within the target color volume. Be careful when using this
	       on BT.2020 sources without proper mastering metadata, as	 doing
	       so will lead to excessive desaturation.

       tonemapping
	   Tone-mapping algorithm to use. Available values are:

	   auto
	       Automatic  selection  based on internal heuristics. This is the
	       default.

	   clip
	       Performs	 no  tone-mapping,  just  clips	 out-of-range  colors.
	       Retains	 perfect   color  accuracy  for	 in-range  colors  but
	       completely destroys out-of-range information.  Does not perform
	       any black point adaptation. Not configurable.

	   st2094-40
	       EETF from SMPTE ST 2094-40 Annex B, which  applies  the	Bezier
	       curves  from  HDR10+ dynamic metadata based on Bezier curves to
	       perform tone-mapping. The OOTF used is adjusted	based  on  the
	       ratio between the targeted and actual display peak luminances.

	   st2094-10
	       EETF  from SMPTE ST 2094-10 Annex B.2, which takes into account
	       the  input  signal  average  luminance  in  addition   to   the
	       maximum/minimum. The configurable contrast parameter influences
	       the  slope  of the linear output segment, defaulting to 1.0 for
	       no increase/decrease in	contrast.  Note	 that  this  does  not
	       currently  include  the	subjective  gain/offset/gamma controls
	       defined in Annex B.3.

	   bt.2390
	       EETF from the ITU-R Report BT.2390, a hermite  spline  roll-off
	       with  linear  segment.  The  knee point offset is configurable.
	       Note that this parameter defaults to 1.0, rather than the value
	       of 0.5 from the ITU-R spec.

	   bt.2446a
	       EETF from ITU-R Report BT.2446, method A.  Designed  for	 well-
	       mastered	 HDR sources. Can be used for both forward and inverse
	       tone mapping. Not configurable.

	   spline
	       Simple spline consisting of two polynomials, joined by a single
	       pivot point.  The  parameter  gives  the	 pivot	point  (in  PQ
	       space),	defaulting  to 0.30.  Can be used for both forward and
	       inverse tone mapping.

	   reinhard
	       Simple non-linear, global tone mapping algorithm. The parameter
	       specifies the local contrast coefficient at the	display	 peak.
	       Essentially,  a	parameter  of  0.5  implies that the reference
	       white will be about half as bright as when  clipping.  Defaults
	       to  0.5,	 which	results	 in  the  simplest formulation of this
	       function.

	   mobius
	       Generalization  of  the	reinhard  tone	mapping	 algorithm  to
	       support an additional linear slope near black. The tone mapping
	       parameter  indicates  the  trade-off between the linear section
	       and the non-linear section. Essentially, for a given  parameter
	       x,  every  color	 value	below x will be mapped linearly, while
	       higher values get non-linearly  tone-mapped.  Values  near  1.0
	       make  this curve behave like "clip", while values near 0.0 make
	       this curve behave like "reinhard". The default  value  is  0.3,
	       which provides a good balance between colorimetric accuracy and
	       preserving out-of-gamut details.

	   hable
	       Piece-wise,  filmic  tone-mapping  algorithm  developed by John
	       Hable for use in Uncharted  2,  inspired	 by  a	similar	 tone-
	       mapping	algorithm  used	 by  Kodak.  Popularized by its use in
	       video games with HDR rendering. Preserves both dark and	bright
	       details	very well, but comes with the drawback of changing the
	       average brightness quite significantly. This is sort of similar
	       to "reinhard" with parameter 0.24.

	   gamma
	       Fits a gamma (power) function to transfer  between  the	source
	       and  target color spaces, effectively resulting in a perceptual
	       hard-knee joining two roughly linear sections.  This  preserves
	       details	at  all scales fairly accurately, but can result in an
	       image with a muted or dull appearance. The parameter is used as
	       the cutoff point, defaulting to 0.5.

	   linear
	       Linearly stretches the input range to the output range,	in  PQ
	       space.  This  will preserve all details accurately, but results
	       in a significantly different average brightness.	 Can  be  used
	       for  inverse  tone-mapping in addition to regular tone-mapping.
	       The  parameter  can  be	used  as  an  additional  linear  gain
	       coefficient (defaulting to 1.0).

       tonemapping_param
	   For	tunable	 tone mapping functions, this parameter can be used to
	   fine-tune  the  curve  behavior.  Refer  to	the  documentation  of
	   "tonemapping".  The default value of 0.0 is replaced by the curve's
	   preferred default setting.

       inverse_tonemapping
	   If enabled, this filter will also attempt stretching SDR signals to
	   fill HDR output color volumes. Disabled by default.

       tonemapping_lut_size
	   Size of the tone-mapping LUT, between 2 and 1024. Defaults to  256.
	   Note that this figure is squared when combined with "peak_detect".

       contrast_recovery
	   Contrast recovery strength. If set to a value above 0.0, the source
	   image   will	 be  divided  into  high-frequency  and	 low-frequency
	   components, and a portion of the high-frequency image is added back
	   onto the tone-mapped output.	 May cause excessive ringing artifacts
	   for some HDR sources, but can improve the subjective sharpness  and
	   detail  left	 over  in  the	image after tone-mapping.  Defaults to
	   0.30.

       contrast_smoothness
	   Contrast recovery lowpass kernel size. Defaults to 3.5.  Increasing
	   or decreasing this will affect the visual appearance substantially.
	   Has no effect when "contrast_recovery" is disabled.

       Dithering

       By  default,  libplacebo will dither whenever necessary, which includes
       rendering  to  any  integer  format  below   16-bit   precision.	  It's
       recommended  to	always leave this on, since not doing so may result in
       visible banding in the  output,	even  if  the  "debanding"  filter  is
       enabled.	 If maximum performance is needed, use "ordered_fixed" instead
       of disabling dithering.

       dithering
	   Dithering method to use. Accepts the following values:

	   none
	       Disables dithering completely. May result in visible banding.

	   blue
	       Dither with pseudo-blue noise. This is the default.

	   ordered
	       Tunable ordered dither pattern.

	   ordered_fixed
	       Faster ordered dither with a fixed size of 6. Texture-less.

	   white
	       Dither with white noise. Texture-less.

       dither_lut_size
	   Dither LUT size, as log base2 between  1  and  8.  Defaults	to  6,
	   corresponding to a LUT size of "64x64".

       dither_temporal
	   Enables temporal dithering. Disabled by default.

       Custom shaders

       libplacebo  supports  a number of custom shaders based on the mpv .hook
       GLSL  syntax.  A	 collection  of	 such  shaders	can  be	 found	 here:
       <https://github.com/mpv-player/mpv/wiki/User-Scripts#user-shaders>

       A full description of the mpv shader format is beyond the scope of this
       section,	     but      a	    summary	can	be     found	 here:
       <https://mpv.io/manual/master/#options-glsl-shader>

       custom_shader_path
	   Specifies a path to a custom shader file to load at runtime.

       custom_shader_bin
	   Specifies a complete custom shader as a raw string.

       Debugging / performance

       All of the options  in  this  section  default  off.  They  may	be  of
       assistance  when	 attempting  to squeeze the maximum performance at the
       cost of quality.

       skip_aa
	   Disable anti-aliasing when downscaling.

       polar_cutoff
	   Truncate polar (EWA) scaler kernels below this absolute  magnitude,
	   between 0.0 and 1.0.

       disable_linear
	   Disable linear light scaling.

       disable_builtin
	   Disable built-in GPU sampling (forces LUT).

       disable_fbos
	   Forcibly   disable	FBOs,	resulting   in	 loss  of  almost  all
	   functionality, but offering the maximum possible speed.

       Commands

       This filter supports almost all of the above options as commands.

       Examples

       •   Tone-map input to standard gamut BT.709 output:

		   libplacebo=colorspace=bt709:color_primaries=bt709:color_trc=bt709:range=tv

       •   Rescale input  to  fit  into	 standard  1080p,  with	 high  quality
	   scaling:

		   libplacebo=w=1920:h=1080:force_original_aspect_ratio=decrease:normalize_sar=true:upscaler=ewa_lanczos:downscaler=ewa_lanczos

       •   Interpolate low FPS / VFR input to smoothed constant 60 fps output:

		   libplacebo=fps=60:frame_mixer=mitchell_clamp

       •   Convert input to standard sRGB JPEG:

		   libplacebo=format=yuv420p:colorspace=bt470bg:color_primaries=bt709:color_trc=iec61966-2-1:range=pc

       •   Use higher quality debanding settings:

		   libplacebo=deband=true:deband_iterations=3:deband_radius=8:deband_threshold=6

       •   Run	this  filter  on  the CPU, on systems with Mesa installed (and
	   with the most expensive options disabled):

		   ffmpeg ... -init_hw_device vulkan:llvmpipe ... -vf libplacebo=upscaler=none:downscaler=none:peak_detect=false

       •   Suppress CPU-based AV1/H.274 film grain application in the decoder,
	   in favor of doing it with this filter. Note that  this  is  only  a
	   gain	 if  the  frames  are  either already on the GPU, or if you're
	   using libplacebo for	 other	purposes,  since  otherwise  the  VRAM
	   roundtrip will more than offset any expected speedup.

		   ffmpeg -export_side_data +film_grain ... -vf libplacebo=apply_filmgrain=true

       •   Interop with VAAPI hwdec to avoid round-tripping through RAM:

		   ffmpeg -init_hw_device vulkan -hwaccel vaapi -hwaccel_output_format vaapi ... -vf libplacebo

   libvmaf
       Calulate	 the  VMAF  (Video Multi-Method Assessment Fusion) score for a
       reference/distorted pair of input videos.

       The first input is the distorted video, and the	second	input  is  the
       reference video.

       The obtained VMAF score is printed through the logging system.

       It requires Netflix's vmaf library (libvmaf) as a pre-requisite.	 After
       installing   the	  library   it	can  be	 enabled  using:  "./configure
       --enable-libvmaf".

       The filter has following options:

       model
	   A `|` delimited list of vmaf models. Each model can	be  configured
	   with a number of parameters.	 Default value: "version=vmaf_v0.6.1"

       feature
	   A  `|`  delimited  list of features. Each feature can be configured
	   with a number of parameters.

       log_path
	   Set the file path to be used to store log files.

       log_fmt
	   Set the format of the log file (xml, json, csv, or sub).

       n_threads
	   Set number  of  threads  to	be  used  when	initializing  libvmaf.
	   Default value: 0, no threads.

       n_subsample
	   Set frame subsampling interval to be used.

       This filter also supports the framesync options.

       Examples

       •   In  the examples below, a distorted video distorted.mpg is compared
	   with a reference file reference.mpg.

       •   Basic usage:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf=log_path=output.xml -f null -

       •   Example with multiple models:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='model=version=vmaf_v0.6.1\\:name=vmaf|version=vmaf_v0.6.1neg\\:name=vmaf_neg' -f null -

       •   Example with multiple addtional features:

		   ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null -

       •   Example with options and different containers:

		   ffmpeg -i distorted.mpg -i reference.mkv -lavfi "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]libvmaf=log_fmt=json:log_path=output.json" -f null -

   libvmaf_cuda
       This is the CUDA variant of the libvmaf filter. It  only	 accepts  CUDA
       frames.

       It requires Netflix's vmaf library (libvmaf) as a pre-requisite.	 After
       installing   the	  library   it	can  be	 enabled  using:  "./configure
       --enable-nonfree --enable-ffnvcodec --enable-libvmaf".

       Examples

       •   Basic usage showing CUVID hardware decoding and CUDA	 scaling  with
	   scale_cuda:

		   ffmpeg \
		       -hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid -i dis.obu \
		       -hwaccel cuda -hwaccel_output_format cuda -codec:v av1_cuvid -i ref.obu \
		       -filter_complex "
			   [0:v]scale_cuda=format=yuv420p[ref]; \
			   [1:v]scale_cuda=format=yuv420p[dis]; \
			   [dis][ref]libvmaf_cuda=log_fmt=json:log_path=output.json
		       " \
		       -f null -

   limitdiff
       Apply limited difference filter using second and optionally third video
       stream.

       The filter accepts the following options:

       threshold
	   Set	the threshold to use when allowing certain differences between
	   video streams.  Any absolute difference value lower or  exact  than
	   this threshold will pick pixel components from first video stream.

       elasticity
	   Set	the  elasticity	 of  soft  thresholding	 when processing video
	   streams.   This  value  multiplied  with  first  one	 sets	second
	   threshold.	Any  absolute  difference  value greater or exact than
	   second threshold will  pick	pixel  components  from	 second	 video
	   stream. For values between those two threshold linear interpolation
	   between first and second video stream will be used.

       reference
	   Enable the reference (third) video stream processing. By default is
	   disabled.   If  set, this video stream will be used for calculating
	   absolute difference with first video stream.

       planes
	   Specify which planes will be processed. Defaults to all available.

       Commands

       This filter supports the all above options as  commands	except	option
       reference.

   limiter
       Limits the pixel components values to the specified range [min, max].

       The filter accepts the following options:

       min Lower bound. Defaults to the lowest allowed value for the input.

       max Upper bound. Defaults to the highest allowed value for the input.

       planes
	   Specify which planes will be processed. Defaults to all available.

       Commands

       This filter supports the all above options as commands.

   loop
       Loop video frames.

       The filter accepts the following options:

       loop
	   Set	the  number  of loops. Setting this value to -1 will result in
	   infinite loops.  Default is 0.

       size
	   Set maximal size in number of frames. Default is 0.

       start
	   Set first frame of loop. Default is 0.

       time
	   Set the time of loop start in seconds.  Only used if	 option	 named
	   start is set to -1.

       Examples

       •   Loop single first frame infinitely:

		   loop=loop=-1:size=1:start=0

       •   Loop single first frame 10 times:

		   loop=loop=10:size=1:start=0

       •   Loop 10 first frames 5 times:

		   loop=loop=5:size=10:start=0

   lut1d
       Apply a 1D LUT to an input video.

       The filter accepts the following options:

       file
	   Set the 1D LUT file name.

	   Currently supported formats:

	   cube
	       Iridas

	   csp cineSpace

       interp
	   Select interpolation mode.

	   Available values are:

	   nearest
	       Use values from the nearest defined point.

	   linear
	       Interpolate values using the linear interpolation.

	   cosine
	       Interpolate values using the cosine interpolation.

	   cubic
	       Interpolate values using the cubic interpolation.

	   spline
	       Interpolate values using the spline interpolation.

       Commands

       This filter supports the all above options as commands.

   lut3d
       Apply a 3D LUT to an input video.

       The filter accepts the following options:

       file
	   Set the 3D LUT file name.

	   Currently supported formats:

	   3dl AfterEffects

	   cube
	       Iridas

	   dat DaVinci

	   m3d Pandora

	   csp cineSpace

       interp
	   Select interpolation mode.

	   Available values are:

	   nearest
	       Use values from the nearest defined point.

	   trilinear
	       Interpolate values using the 8 points defining a cube.

	   tetrahedral
	       Interpolate values using a tetrahedron.

	   pyramid
	       Interpolate values using a pyramid.

	   prism
	       Interpolate values using a prism.

       Commands

       This filter supports the "interp" option as commands.

   lumakey
       Turn certain luma values into transparency.

       The filter accepts the following options:

       threshold
	   Set	the luma which will be used as base for transparency.  Default
	   value is 0.

       tolerance
	   Set the range of luma values to be keyed  out.   Default  value  is
	   0.01.

       softness
	   Set the range of softness. Default value is 0.  Use this to control
	   gradual transition from zero to full transparency.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

   lut, lutrgb, lutyuv
       Compute a look-up table for binding each pixel component input value to
       an output value, and apply it to the input video.

       lutyuv applies a lookup table to a YUV input video, lutrgb  to  an  RGB
       input video.

       These filters accept the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set	fourth	pixel  component  expression, corresponds to the alpha
	   component

       r   set red component expression

       g   set green component expression

       b   set blue component expression

       a   alpha component expression

       y   set Y/luma component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

       Each of them specifies the expression to use for computing  the	lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c* options depends on the
       format in input.

       The  lut	 filter	 requires  either  YUV	or RGB pixel formats in input,
       lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       clipval
	   The input value, clipped to the minval-maxval range.

       maxval
	   The maximum value for the pixel component.

       minval
	   The minimum value for the pixel component.

       negval
	   The negated value for the pixel component  value,  clipped  to  the
	   minval-maxval    range;    it   corresponds	 to   the   expression
	   "maxval-clipval+minval".

       clip(val)
	   The computed value in val, clipped to the minval-maxval range.

       gammaval(gamma)
	   The computed gamma correction value of the pixel  component	value,
	   clipped   to	  the  minval-maxval  range.  It  corresponds  to  the
	   expression
	   "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "clipval".

       Commands

       This filter supports same commands as options.

       Examples

       •   Negate input video:

		   lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
		   lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

	   The above is the same as:

		   lutrgb="r=negval:g=negval:b=negval"
		   lutyuv="y=negval:u=negval:v=negval"

       •   Negate luma:

		   lutyuv=y=negval

       •   Remove chroma components, turning the video into a graytone image:

		   lutyuv="u=128:v=128"

       •   Apply a luma burning effect:

		   lutyuv="y=2*val"

       •   Remove green and blue components:

		   lutrgb="g=0:b=0"

       •   Set a constant alpha channel value on input:

		   format=rgba,lutrgb=a="maxval-minval/2"

       •   Correct luma gamma by a factor of 0.5:

		   lutyuv=y=gammaval(0.5)

       •   Discard least significant bits of luma:

		   lutyuv=y='bitand(val, 128+64+32)'

       •   Technicolor like effect:

		   lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'

   lut2, tlut2
       The "lut2" filter takes two input streams and outputs one stream.

       The "tlut2" (time lut2) filter takes two consecutive  frames  from  one
       single stream.

       This filter accepts the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set	fourth	pixel  component  expression, corresponds to the alpha
	   component

       d   set output bit depth, only available for "lut2" filter. By  default
	   is  0,  which  means	 bit  depth is automatically picked from first
	   input format.

       The "lut2" filter also supports the framesync options.

       Each of them specifies the expression to use for computing  the	lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c* options depends on the
       format in inputs.

       The expressions can contain the following constants:

       w
       h   The input width and height.

       x   The first input value for the pixel component.

       y   The second input value for the pixel component.

       bdx The first input video bit depth.

       bdy The second input video bit depth.

       All expressions default to "x".

       Commands

       This  filter  supports  the all above options as commands except option
       "d".

       Examples

       •   Highlight differences between two RGB video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

       •   Highlight differences between two YUV video streams:

		   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

       •   Show max difference between two video streams:

		   lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'

   maskedclamp
       Clamp the first input stream with the  second  input  and  third	 input
       stream.

       Returns	the  value of first stream to be between second input stream -
       "undershoot" and third input stream + "overshoot".

       This filter accepts the following options:

       undershoot
	   Default value is 0.

       overshoot
	   Default value is 0.

       planes
	   Set which planes will be processed as  bitmap,  unprocessed	planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       Commands

       This filter supports the all above options as commands.

   maskedmax
       Merge  the  second  and	third  input  stream  into output stream using
       absolute differences between second input stream and first input stream
       and absolute difference between third  input  stream  and  first	 input
       stream.	The  picked  value  will be from second input stream if second
       absolute difference is greater than  first  one	or  from  third	 input
       stream otherwise.

       This filter accepts the following options:

       planes
	   Set	which  planes  will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       Commands

       This filter supports the all above options as commands.

   maskedmerge
       Merge the first input stream with the second  input  stream  using  per
       pixel weights in the third input stream.

       A  value	 of  0	in  the	 third stream pixel component means that pixel
       component from first stream is returned unchanged, while maximum	 value
       (eg.  255  for  8-bit  videos)  means  that pixel component from second
       stream is returned unchanged. Intermediate values define the amount  of
       merging between both input stream's pixel components.

       This filter accepts the following options:

       planes
	   Set	which  planes  will be processed as bitmap, unprocessed planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       Commands

       This filter supports the all above options as commands.

   maskedmin
       Merge the second and  third  input  stream  into	 output	 stream	 using
       absolute differences between second input stream and first input stream
       and  absolute  difference  between  third  input stream and first input
       stream. The picked value will be from second  input  stream  if	second
       absolute	 difference  is less than first one or from third input stream
       otherwise.

       This filter accepts the following options:

       planes
	   Set which planes will be processed as  bitmap,  unprocessed	planes
	   will be copied from first stream.  By default value 0xf, all planes
	   will be processed.

       Commands

       This filter supports the all above options as commands.

   maskedthreshold
       Pick  pixels  comparing	absolute  difference of two video streams with
       fixed threshold.

       If absolute difference between pixel  component	of  first  and	second
       video  stream is equal or lower than user supplied threshold than pixel
       component from first video stream is picked, otherwise pixel  component
       from second video stream is picked.

       This filter accepts the following options:

       threshold
	   Set	threshold  used	 when  picking pixels from absolute difference
	   from two input video streams.

       planes
	   Set which planes will be processed as  bitmap,  unprocessed	planes
	   will	 be  copied  from  second  stream.   By default value 0xf, all
	   planes will be processed.

       mode
	   Set mode of filter operation. Can be "abs" or "diff".   Default  is
	   "abs".

       Commands

       This filter supports the all above options as commands.

   maskfun
       Create mask from input video.

       For example it is useful to create motion masks after "tblend" filter.

       This filter accepts the following options:

       low Set	low  threshold.	 Any  pixel component lower or exact than this
	   value will be set to 0.

       high
	   Set high threshold. Any pixel component higher than this value will
	   be set to max value allowed for current pixel format.

       planes
	   Set planes to filter, by default all available planes are filtered.

       fill
	   Fill all frame pixels with this value.

       sum Set max average  pixel  value  for  frame.  If  sum	of  all	 pixel
	   components  is  higher  that	 this  average,	 output	 frame will be
	   completely filled with value set by fill option.  Typically	useful
	   for scene changes when used in combination with "tblend" filter.

       Commands

       This filter supports the all above options as commands.

   mcdeint
       Apply motion-compensation deinterlacing.

       It  needs  one  field per frame as input and must thus be used together
       with yadif=1/3 or equivalent.

       This filter accepts the following options:

       mode
	   Set the deinterlacing mode.

	   It accepts one of the following values:

	   fast
	   medium
	   slow
	       use iterative motion estimation

	   extra_slow
	       like slow, but use multiple reference frames.

	   Default value is fast.

       parity
	   Set the picture field parity assumed for the input video.  It  must
	   be one of the following values:

	   0, tff
	       assume top field first

	   1, bff
	       assume bottom field first

	   Default value is bff.

       qp  Set	per-block  quantization	 parameter  (QP)  used by the internal
	   encoder.

	   Higher values should result in a smoother motion vector  field  but
	   less optimal individual vectors. Default value is 1.

   median
       Pick median pixel from certain rectangle defined by radius.

       This filter accepts the following options:

       radius
	   Set	horizontal  radius size. Default value is 1.  Allowed range is
	   integer from 1 to 127.

       planes
	   Set which planes to process. Default is 15, which is all  available
	   planes.

       radiusV
	   Set	vertical  radius  size.	 Default value is 0.  Allowed range is
	   integer from 0 to 127.  If it is  0,	 value	will  be  picked  from
	   horizontal "radius" option.

       percentile
	   Set	median percentile. Default value is 0.5.  Default value of 0.5
	   will pick always median values, while 0 will pick  minimum  values,
	   and 1 maximum values.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

   mergeplanes
       Merge color channel components from several video streams.

       The filter accepts up to 4 input	 streams,  and	merge  selected	 input
       planes to the output video.

       This filter accepts the following options:

       mapping
	   Set input to output plane mapping. Default is 0.

	   The	mappings is specified as a bitmap. It should be specified as a
	   hexadecimal number in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the
	   mapping for the first plane of the  output  stream.	'A'  sets  the
	   number  of the input stream to use (from 0 to 3), and 'a' the plane
	   number of the corresponding input to use (from 0 to 3). The rest of
	   the mappings is similar, 'Bb' describes the mapping for the	output
	   stream  second  plane,  'Cc'	 describes  the mapping for the output
	   stream third plane and 'Dd' describes the mapping  for  the	output
	   stream fourth plane.

       format
	   Set output pixel format. Default is "yuva444p".

       map0s
       map1s
       map2s
       map3s
	   Set input to output stream mapping for output Nth plane. Default is
	   0.

       map0p
       map1p
       map2p
       map3p
	   Set	input to output plane mapping for output Nth plane. Default is
	   0.

       Examples

       •   Merge three gray video streams of same width and height into single
	   video stream:

		   [a0][a1][a2]mergeplanes=0x001020:yuv444p

       •   Merge 1st yuv444p stream and 2nd gray video	stream	into  yuva444p
	   video stream:

		   [a0][a1]mergeplanes=0x00010210:yuva444p

       •   Swap Y and A plane in yuva444p stream:

		   format=yuva444p,mergeplanes=0x03010200:yuva444p

       •   Swap U and V plane in yuv420p stream:

		   format=yuv420p,mergeplanes=0x000201:yuv420p

       •   Cast a rgb24 clip to yuv444p:

		   format=rgb24,mergeplanes=0x000102:yuv444p

   mestimate
       Estimate	 and  export  motion  vectors using block matching algorithms.
       Motion vectors are stored in frame  side	 data  to  be  used  by	 other
       filters.

       This filter accepts the following options:

       method
	   Specify  the motion estimation method. Accepts one of the following
	   values:

	   esa Exhaustive search algorithm.

	   tss Three step search algorithm.

	   tdls
	       Two dimensional logarithmic search algorithm.

	   ntss
	       New three step search algorithm.

	   fss Four step search algorithm.

	   ds  Diamond search algorithm.

	   hexbs
	       Hexagon-based search algorithm.

	   epzs
	       Enhanced predictive zonal search algorithm.

	   umh Uneven multi-hexagon search algorithm.

	   Default value is esa.

       mb_size
	   Macroblock size. Default 16.

       search_param
	   Search parameter. Default 7.

   midequalizer
       Apply Midway Image Equalization effect using two video streams.

       Midway Image Equalization adjusts a pair of images  to  have  the  same
       histogram,  while  maintaining their dynamics as much as possible. It's
       useful for e.g. matching exposures from a pair of stereo cameras.

       This filter has two inputs and one output, which must be of same	 pixel
       format,	but  may  be of different sizes. The output of filter is first
       input adjusted with midway histogram of both inputs.

       This filter accepts the following option:

       planes
	   Set which planes to process. Default is 15, which is all  available
	   planes.

   minterpolate
       Convert the video to specified frame rate using motion interpolation.

       This filter accepts the following options:

       fps Specify   the   output  frame  rate.	 This  can  be	rational  e.g.
	   "60000/1001". Frames are dropped if fps is lower than  source  fps.
	   Default 60.

       mi_mode
	   Motion interpolation mode. Following values are accepted:

	   dup Duplicate previous or next frame for interpolating new ones.

	   blend
	       Blend source frames. Interpolated frame is mean of previous and
	       next frames.

	   mci Motion	compensated   interpolation.   Following  options  are
	       effective when this mode is selected:

	       mc_mode
		   Motion compensation mode. Following values are accepted:

		   obmc
		       Overlapped block motion compensation.

		   aobmc
		       Adaptive overlapped block motion	 compensation.	Window
		       weighting   coefficients	  are	controlled  adaptively
		       according  to  the  reliabilities  of  the  neighboring
		       motion vectors to reduce oversmoothing.

		   Default mode is obmc.

	       me_mode
		   Motion estimation mode. Following values are accepted:

		   bidir
		       Bidirectional  motion  estimation.  Motion  vectors are
		       estimated for each source frame	in  both  forward  and
		       backward directions.

		   bilat
		       Bilateral   motion   estimation.	  Motion  vectors  are
		       estimated directly for interpolated frame.

		   Default mode is bilat.

	       me  The algorithm to be used for motion	estimation.  Following
		   values are accepted:

		   esa Exhaustive search algorithm.

		   tss Three step search algorithm.

		   tdls
		       Two dimensional logarithmic search algorithm.

		   ntss
		       New three step search algorithm.

		   fss Four step search algorithm.

		   ds  Diamond search algorithm.

		   hexbs
		       Hexagon-based search algorithm.

		   epzs
		       Enhanced predictive zonal search algorithm.

		   umh Uneven multi-hexagon search algorithm.

		   Default algorithm is epzs.

	       mb_size
		   Macroblock size. Default 16.

	       search_param
		   Motion estimation search parameter. Default 32.

	       vsbmc
		   Enable  variable-size  block	 motion	 compensation.	Motion
		   estimation is applied with smaller block  sizes  at	object
		   boundaries  in order to make the them less blur. Default is
		   0 (disabled).

       scd Scene change detection method. Scene change leads motion vectors to
	   be in random direction. Scene change detection replace interpolated
	   frames by duplicate ones.  May  not	be  needed  for	 other	modes.
	   Following values are accepted:

	   none
	       Disable scene change detection.

	   fdiff
	       Frame  difference.  Corresponding pixel values are compared and
	       if it satisfies scd_threshold scene change is detected.

	   Default method is fdiff.

       scd_threshold
	   Scene change detection threshold. Default is 10..

   mix
       Mix several video input streams into one video stream.

       A description of the accepted options follows.

       inputs
	   The number of inputs. If unspecified, it defaults to 2.

       weights
	   Specify weight of each input video stream as sequence.  Each weight
	   is separated by space. If number of weights is smaller than	number
	   of  frames  last  specified	weight	will be used for all remaining
	   unset weights.

       scale
	   Specify scale, if it is set it will be multiplied with sum of  each
	   weight multiplied with pixel values to give final destination pixel
	   value. By default scale is auto scaled to sum of weights.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       duration
	   Specify how end of stream is determined.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

       Commands

       This filter supports the following commands:

       weights
       scale
       planes
	   Syntax is same as option with same name.

   monochrome
       Convert video to gray using custom color filter.

       A description of the accepted options follows.

       cb  Set	the  chroma blue spot. Allowed range is from -1 to 1.  Default
	   value is 0.

       cr  Set the chroma red spot. Allowed range is from -1  to  1.   Default
	   value is 0.

       size
	   Set the color filter size. Allowed range is from .1 to 10.  Default
	   value is 1.

       high
	   Set the highlights strength. Allowed range is from 0 to 1.  Default
	   value is 0.

       Commands

       This filter supports the all above options as commands.

   morpho
       This  filter  allows  to apply main morphological grayscale transforms,
       erode and dilate with arbitrary structures set in second input stream.

       Unlike naive implementation and much slower performance in erosion  and
       dilation filters, when speed is critical "morpho" filter should be used
       instead.

       A description of accepted options follows,

       mode
	   Set morphological transform to apply, can be:

	   erode
	   dilate
	   open
	   close
	   gradient
	   tophat
	   blackhat

	   Default is "erode".

       planes
	   Set	planes	to  filter,  by	 default  all  planes except alpha are
	   filtered.

       structure
	   Set which structure video frames  will  be  processed  from	second
	   input stream, can be first or all. Default is all.

       The "morpho" filter also supports the framesync options.

       Commands

       This filter supports same commands as options.

   mpdecimate
       Drop frames that do not differ greatly from the previous frame in order
       to reduce frame rate.

       The  main  use  of  this	 filter is for very-low-bitrate encoding (e.g.
       streaming over dialup modem), but it could in theory be used for fixing
       movies that were inverse-telecined incorrectly.

       A description of the accepted options follows.

       max Set the maximum number of consecutive frames which can  be  dropped
	   (if	positive),  or the minimum interval between dropped frames (if
	   negative). If the value is 0, the frame is dropped disregarding the
	   number of previous sequentially dropped frames.

	   Default value is 0.

       keep
	   Set the maximum number of  consecutive  similar  frames  to	ignore
	   before  to  start  dropping	them.  If the value is 0, the frame is
	   dropped disregarding the number of  previous	 sequentially  similar
	   frames.

	   Default value is 0.

       hi
       lo
       frac
	   Set the dropping threshold values.

	   Values  for hi and lo are for 8x8 pixel blocks and represent actual
	   pixel value differences, so a threshold of 64 corresponds to 1 unit
	   of difference for each pixel, or the same  spread  out  differently
	   over the block.

	   A frame is a candidate for dropping if no 8x8 blocks differ by more
	   than	 a threshold of hi, and if no more than frac blocks (1 meaning
	   the whole image) differ by more than a threshold of lo.

	   Default value for hi is 64*12, default value for lo	is  64*5,  and
	   default value for frac is 0.33.

   msad
       Obtain  the  MSAD  (Mean Sum of Absolute Differences) between two input
       videos.

       This filter takes two input videos.

       Both input videos must have the same resolution and  pixel  format  for
       this  filter  to	 work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained per component,  average,  min  and	max  MSAD  is  printed
       through the logging system.

       The filter stores the calculated MSAD of each frame in frame metadata.

       This filter also supports the framesync options.

       In  the	below  example	the  input  file  main.mpg  being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi msad -f null -

   multiply
       Multiply first video stream pixels  values  with	 second	 video	stream
       pixels values.

       The filter accepts the following options:

       scale
	   Set	the  scale  applied  to	 second video stream. By default is 1.
	   Allowed range is from 0 to 9.

       offset
	   Set the offset applied to second video stream. By default  is  0.5.
	   Allowed range is from -1 to 1.

       planes
	   Specify  planes from input video stream that will be processed.  By
	   default all planes are processed.

       Commands

       This filter supports same commands as options.

   negate
       Negate (invert) the input video.

       It accepts the following option:

       components
	   Set components to negate.

	   Available values for components are:

	   y
	   u
	   v
	   a
	   r
	   g
	   b
       negate_alpha
	   With value 1, it negates the alpha component, if  present.  Default
	   value is 0.

       Commands

       This filter supports same commands as options.

   nlmeans
       Denoise frames using Non-Local Means algorithm.

       Each  pixel  is	adjusted  by  looking  for  other  pixels with similar
       contexts.  This	context	 similarity  is	 defined  by  comparing	 their
       surrounding patches of size pxp. Patches are searched in an area of rxr
       around the pixel.

       Note  that  the	research area defines centers for patches, which means
       some patches will be made of pixels outside that research area.

       The filter accepts the following options.

       s   Set denoising strength. Default is 1.0.  Must  be  in  range	 [1.0,
	   30.0].

       p   Set patch size. Default is 7. Must be odd number in range [0, 99].

       pc  Same as p but for chroma planes.

	   The default value is 0 and means automatic.

       r   Set	research  size. Default is 15. Must be odd number in range [0,
	   99].

       rc  Same as r but for chroma planes.

	   The default value is 0 and means automatic.

   nnedi
       Deinterlace video using neural network edge directed interpolation.

       This filter accepts the following options:

       weights
	   Mandatory  option,  without	binary	file  filter  can  not	 work.
	   Currently	    file	can	   be	     found	 here:
	   https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

       deint
	   Set which frames to deinterlace, by default it is  "all".   Can  be
	   "all" or "interlaced".

       field
	   Set mode of operation.

	   Can be one of the following:

	   af  Use frame flags, both fields.

	   a   Use frame flags, single field.

	   t   Use top field only.

	   b   Use bottom field only.

	   tf  Use both fields, top first.

	   bf  Use both fields, bottom first.

       planes
	   Set which planes to process, by default filter process all frames.

       nsize
	   Set	size  of  local	 neighborhood  around  each pixel, used by the
	   predictor neural network.

	   Can be one of the following:

	   s8x6
	   s16x6
	   s32x6
	   s48x6
	   s8x4
	   s16x4
	   s32x4
       nns Set the number of neurons in predictor neural network.  Can be  one
	   of the following:

	   n16
	   n32
	   n64
	   n128
	   n256
       qual
	   Controls  the  number  of different neural network predictions that
	   are blended together to compute the	final  output  value.  Can  be
	   "fast", default or "slow".

       etype
	   Set	which  set  of weights to use in the predictor.	 Can be one of
	   the following:

	   a, abs
	       weights trained to minimize absolute error

	   s, mse
	       weights trained to minimize squared error

       pscrn
	   Controls whether or not the prescreener neural network is  used  to
	   decide  which  pixels  should  be processed by the predictor neural
	   network and which can be handled  by	 simple	 cubic	interpolation.
	   The prescreener is trained to know whether cubic interpolation will
	   be  sufficient for a pixel or whether it should be predicted by the
	   predictor nn.  The computational complexity of the  prescreener  nn
	   is  much  less than that of the predictor nn. Since most pixels can
	   be handled by cubic interpolation, using the prescreener  generally
	   results  in	much  faster  processing.   The	 prescreener is pretty
	   accurate, so the difference between using it and not	 using	it  is
	   almost always unnoticeable.

	   Can be one of the following:

	   none
	   original
	   new
	   new2
	   new3

	   Default is "new".

       Commands

       This  filter  supports  same  commands  as  options,  excluding weights
       option.

   noformat
       Force libavfilter not to use any of the specified pixel formats for the
       input to the next filter.

       It accepts the following parameters:

       pix_fmts
	   A   '|'-separated   list   of   pixel   format   names,   such   as
	   pix_fmts=yuv420p|monow|rgb24".

       Examples

       •   Force  libavfilter  to  use a format different from yuv420p for the
	   input to the vflip filter:

		   noformat=pix_fmts=yuv420p,vflip

       •   Convert the input video to any of the formats not contained in  the
	   list:

		   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
	   Set noise seed for specific pixel component or all pixel components
	   in case of all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
	   Set	noise  strength	 for  specific	pixel  component  or all pixel
	   components in case all_strength. Default value is 0. Allowed	 range
	   is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
	   Set	pixel  component  flags	 or  set  flags	 for all components if
	   all_flags.  Available values for component flags are:

	   a   averaged temporal noise (smoother)

	   p   mix random noise with a (semi)regular pattern

	   t   temporal noise (noise pattern changes between frames)

	   u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

	       noise=alls=20:allf=t+u

   normalize
       Normalize RGB video (aka histogram  stretching,	contrast  stretching).
       See: https://en.wikipedia.org/wiki/Normalization_(image_processing)

       For each channel of each frame, the filter computes the input range and
       maps  it	 linearly to the user-specified output range. The output range
       defaults to the full dynamic range from pure black to pure white.

       Temporal smoothing can be used on the input range to reduce  flickering
       (rapid  changes in brightness) caused when small dark or bright objects
       enter or	 leave	the  scene.  This  is  similar	to  the	 auto-exposure
       (automatic  gain	 control) on a video camera, and, like a video camera,
       it may cause a period of over- or under-exposure of the video.

       The R,G,B channels can be normalized  independently,  which  may	 cause
       some  color  shifting,  or  linked  together as a single channel, which
       prevents	 color	 shifting.   Linked   normalization   preserves	  hue.
       Independent  normalization  does	 not, so it can be used to remove some
       color casts. Independent and linked normalization can  be  combined  in
       any ratio.

       The normalize filter accepts the following options:

       blackpt
       whitept
	   Colors  which  define  the output range. The minimum input value is
	   mapped to the blackpt. The maximum input value  is  mapped  to  the
	   whitept.  The defaults are black and white respectively. Specifying
	   white  for  blackpt and black for whitept will give color-inverted,
	   normalized video. Shades of grey can be used to reduce the  dynamic
	   range  (contrast). Specifying saturated colors here can create some
	   interesting effects.

       smoothing
	   The number of previous frames to use for  temporal  smoothing.  The
	   input  range	 of  each  channel is smoothed using a rolling average
	   over the current frame  and	the  smoothing	previous  frames.  The
	   default is 0 (no temporal smoothing).

       independence
	   Controls   the   ratio  of  independent  (color  shifting)  channel
	   normalization to linked (color preserving)  normalization.  0.0  is
	   fully  linked,  1.0	is  fully  independent. Defaults to 1.0 (fully
	   independent).

       strength
	   Overall strength of the filter. 1.0 is  full	 strength.  0.0	 is  a
	   rather expensive no-op. Defaults to 1.0 (full strength).

       Commands

       This  filter  supports  same  commands  as options, excluding smoothing
       option.	The command accepts  the  same	syntax	of  the	 corresponding
       option.

       If  the	specified  expression  is not valid, it is kept at its current
       value.

       Examples

       Stretch video contrast to use the full dynamic range, with no  temporal
       smoothing; may flicker depending on the source content:

	       normalize=blackpt=black:whitept=white:smoothing=0

       As  above,  but with 50 frames of temporal smoothing; flicker should be
       reduced, depending on the source content:

	       normalize=blackpt=black:whitept=white:smoothing=50

       As above, but with hue-preserving linked channel normalization:

	       normalize=blackpt=black:whitept=white:smoothing=50:independence=0

       As above, but with half strength:

	       normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5

       Map the darkest input color to red, the brightest input color to cyan:

	       normalize=blackpt=red:whitept=cyan

   null
       Pass the video source unchanged to the output.

   ocr
       Optical Character Recognition

       This filter uses Tesseract for optical character recognition. To enable
       compilation  of	this  filter,  you  need  to  configure	 FFmpeg	  with
       "--enable-libtesseract".

       It accepts the following options:

       datapath
	   Set	datapath to tesseract data. Default is to use whatever was set
	   at installation.

       language
	   Set language, default is "eng".

       whitelist
	   Set character whitelist.

       blacklist
	   Set character blacklist.

       The  filter   exports   recognized   text   as	the   frame   metadata
       "lavfi.ocr.text".  The filter exports confidence of recognized words as
       the frame metadata "lavfi.ocr.confidence".

   ocv
       Apply a video transform using libopencv.

       To  enable  this	 filter, install the libopencv library and headers and
       configure FFmpeg with "--enable-libopencv".

       It accepts the following parameters:

       filter_name
	   The name of the libopencv filter to apply.

       filter_params
	   The parameters to pass to the libopencv filter. If  not  specified,
	   the default values are assumed.

       Refer   to  the	official  libopencv  documentation  for	 more  precise
       information:
       <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

       Several libopencv filters are supported; see the following subsections.

       dilate

       Dilate  an  image  by  using  a	specific  structuring	element.    It
       corresponds to the libopencv function "cvDilate".

       It accepts the parameters: struct_el|nb_iterations.

       struct_el  represents  a	 structuring  element,	and  has  the  syntax:
       colsxrows+anchor_xxanchor_y/shape

       cols and	 rows  represent  the  number  of  columns  and	 rows  of  the
       structuring  element, anchor_x and anchor_y the anchor point, and shape
       the shape for the structuring element. shape must be  "rect",  "cross",
       "ellipse", or "custom".

       If  the value for shape is "custom", it must be followed by a string of
       the form "=filename".  The  file	 with  name  filename  is  assumed  to
       represent  a  binary image, with each printable character corresponding
       to a bright pixel. When a custom shape  is  used,  cols	and  rows  are
       ignored,	 the  number  or columns and rows of the read file are assumed
       instead.

       The default value for struct_el is "3x3+0x0/rect".

       nb_iterations specifies the number of times the transform is applied to
       the image, and defaults to 1.

       Some examples:

	       # Use the default values
	       ocv=dilate

	       # Dilate using a structuring element with a 5x5 cross, iterating two times
	       ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

	       # Read the shape from the file diamond.shape, iterating two times.
	       # The file diamond.shape may contain a pattern of characters like this
	       #   *
	       #  ***
	       # *****
	       #  ***
	       #   *
	       # The specified columns and rows are ignored
	       # but the anchor point coordinates are not
	       ocv=dilate:0x0+2x2/custom=diamond.shape|2

       erode

       Erode an image by using a specific structuring element.	It corresponds
       to the libopencv function "cvErode".

       It accepts  the	parameters:  struct_el:nb_iterations,  with  the  same
       syntax and semantics as the dilate filter.

       smooth

       Smooth the input video.

       The	 filter	      takes	  the	    following	   parameters:
       type|param1|param2|param3|param4.

       type is the type of smooth filter to apply, and	must  be  one  of  the
       following  values:  "blur",  "blur_no_scale",  "median", "gaussian", or
       "bilateral". The default value is "gaussian".

       The meaning of param1, param2, param3, and param4 depends on the smooth
       type. param1 and param2 accept integer positive values or 0. param3 and
       param4 accept floating point values.

       The default value for param1 is 3. The  default	value  for  the	 other
       parameters is 0.

       These parameters correspond to the parameters assigned to the libopencv
       function "cvSmooth".

   oscilloscope
       2D Video Oscilloscope.

       Useful to measure spatial impulse, step responses, chroma delays, etc.

       It accepts the following parameters:

       x   Set scope center x position.

       y   Set scope center y position.

       s   Set scope size, relative to frame diagonal.

       t   Set scope tilt/rotation.

       o   Set trace opacity.

       tx  Set trace center x position.

       ty  Set trace center y position.

       tw  Set trace width, relative to width of frame.

       th  Set trace height, relative to height of frame.

       c   Set	which  components  to  trace. By default it traces first three
	   components.

       g   Draw trace grid. By default is enabled.

       st  Draw some statistics. By default is enabled.

       sc  Draw scope. By default is enabled.

       Commands

       This filter supports same commands as options.  The command accepts the
       same syntax of the corresponding option.

       If the specified expression is not valid, it is	kept  at  its  current
       value.

       Examples

       •   Inspect full first row of video frame.

		   oscilloscope=x=0.5:y=0:s=1

       •   Inspect full last row of video frame.

		   oscilloscope=x=0.5:y=1:s=1

       •   Inspect full 5th line of video frame of height 1080.

		   oscilloscope=x=0.5:y=5/1080:s=1

       •   Inspect full last column of video frame.

		   oscilloscope=x=1:y=0.5:s=1:t=1

   overlay
       Overlay one video on top of another.

       It  takes  two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.

       It accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and  y	coordinates  of	 the  overlaid
	   video on the main video. Default value is "0" for both expressions.
	   In  case  the  expression  is  invalid,  it	is set to a huge value
	   (meaning that the overlay will not be displayed within  the	output
	   visible area).

       eof_action
	   See framesync.

       eval
	   Set when the expressions for x, and y are evaluated.

	   It accepts the following values:

	   init
	       only evaluate expressions once during the filter initialization
	       or when a command is processed

	   frame
	       evaluate expressions for each incoming frame

	   Default value is frame.

       shortest
	   See framesync.

       format
	   Set the format for the output video.

	   It accepts the following values:

	   yuv420
	       force YUV 4:2:0 8-bit planar output

	   yuv420p10
	       force YUV 4:2:0 10-bit planar output

	   yuv422
	       force YUV 4:2:2 8-bit planar output

	   yuv422p10
	       force YUV 4:2:2 10-bit planar output

	   yuv444
	       force YUV 4:4:4 8-bit planar output

	   yuv444p10
	       force YUV 4:4:4 10-bit planar output

	   rgb force RGB 8-bit packed output

	   gbrp
	       force RGB 8-bit planar output

	   auto
	       automatically pick format

	   Default value is yuv420.

       repeatlast
	   See framesync.

       alpha
	   Set	format	of  alpha of the overlaid video, it can be straight or
	   premultiplied. Default is straight.

       The x, and y expressions can contain the following parameters.

       main_w, W
       main_h, H
	   The main input width and height.

       overlay_w, w
       overlay_h, h
	   The overlay input width and height.

       x
       y   The computed values for x and y. They are evaluated	for  each  new
	   frame.

       hsub
       vsub
	   horizontal  and  vertical  chroma  subsample	 values	 of the output
	   format. For example for the pixel format "yuv422p" hsub  is	2  and
	   vsub is 1.

       n   the number of input frame, starting from 0

       pos the	position  in  the  file	 of  the  input frame, NAN if unknown;
	   deprecated, do not use

       t   The	timestamp,  expressed  in  seconds.  It's  NAN	if  the	 input
	   timestamp is unknown.

       This filter also supports the framesync options.

       Note that the n, t variables are available only when evaluation is done
       per frame, and will evaluate to NAN when eval is set to init.

       Be  aware  that	frames	are  taken  from each input video in timestamp
       order, hence, if their initial timestamps differ, it is a good idea  to
       pass  the  two inputs through a setpts=PTS-STARTPTS filter to have them
       begin in the same zero timestamp, as the example for the	 movie	filter
       does.

       You can chain together more overlays but you should test the efficiency
       of such approach.

       Commands

       This filter supports the following commands:

       x
       y   Modify  the	x and y of the overlay input.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

       Examples

       •   Draw the overlay at 10 pixels from the bottom right corner  of  the
	   main video:

		   overlay=main_w-overlay_w-10:main_h-overlay_h-10

	   Using named options the example above becomes:

		   overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

       •   Insert  a  transparent  PNG	logo  in the bottom left corner of the
	   input, using the ffmpeg tool with the "-filter_complex" option:

		   ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

       •   Insert 2 different transparent PNG logos  (second  logo  on	bottom
	   right corner) using the ffmpeg tool:

		   ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

       •   Add	a transparent color layer on top of the main video; "WxH" must
	   specify the size of the main input to the overlay filter:

		   color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

       •   Play an original video  and	a  filtered  version  (here  with  the
	   deshake filter) side by side using the ffplay tool:

		   ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

	   The above command is the same as:

		   ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

       •   Make	 a  sliding  overlay  appearing from the left to the right top
	   part of the screen starting since time 2:

		   overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

       •   Compose output by putting two input videos side to side:

		   ffmpeg -i left.avi -i right.avi -filter_complex "
		   nullsrc=size=200x100 [background];
		   [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
		   [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
		   [background][left]	    overlay=shortest=1	     [background+left];
		   [background+left][right] overlay=shortest=1:x=100 [left+right]
		   "

       •   Mask 10-20 seconds of a video by applying the delogo	 filter	 to  a
	   section

		   ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
		   -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
		   masked.avi

       •   Chain several overlays in cascade:

		   nullsrc=s=200x200 [bg];
		   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
		   [in0] lutrgb=r=0, [bg]   overlay=0:0	    [mid0];
		   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
		   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
		   [in3] null,	     [mid2] overlay=100:100 [out0]

   overlay_cuda
       Overlay one video on top of another.

       This  is	 the CUDA variant of the overlay filter.  It only accepts CUDA
       frames. The underlying input pixel formats have to match.

       It takes two inputs and has one output. The first input is  the	"main"
       video on which the second input is overlaid.

       It accepts the following parameters:

       x
       y   Set	expressions  for the x and y coordinates of the overlaid video
	   on the main video.

	   They can contain the following parameters:

	   main_w, W
	   main_h, H
	       The main input width and height.

	   overlay_w, w
	   overlay_h, h
	       The overlay input width and height.

	   x
	   y   The computed values for x and y. They are  evaluated  for  each
	       new frame.

	   n   The ordinal index of the main input frame, starting from 0.

	   pos The  byte  offset position in the file of the main input frame,
	       NAN if unknown.	Deprecated, do not use.

	   t   The timestamp of the main input frame,  expressed  in  seconds,
	       NAN if unknown.

	   Default value is "0" for both expressions.

       eval
	   Set when the expressions for x and y are evaluated.

	   It accepts the following values:

	   init
	       Evaluate	 expressions once during filter initialization or when
	       a command is processed.

	   frame
	       Evaluate expressions for each incoming frame

	   Default value is frame.

       eof_action
	   See framesync.

       shortest
	   See framesync.

       repeatlast
	   See framesync.

       This filter also supports the framesync options.

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
	   Set depth.

	   Larger depth values will denoise lower frequency  components	 more,
	   but slow down filtering.

	   Must be an int in the range 8-16, default is 8.

       luma_strength, ls
	   Set luma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
	   Set chroma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add  paddings  to  the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following parameters:

       width, w
       height, h
	   Specify an expression for the size of the  output  image  with  the
	   paddings  added.  If	 the  value  for  width	 or  height  is 0, the
	   corresponding input size is used for the output.

	   The width expression can reference the  value  set  by  the	height
	   expression, and vice versa.

	   The default value of width and height is 0.

       x
       y   Specify  the	 offsets to place the input image at within the padded
	   area, with respect to the top/left border of the output image.

	   The x expression can reference the value set by the	y  expression,
	   and vice versa.

	   The default value of x and y is 0.

	   If  x or y evaluate to a negative number, they'll be changed so the
	   input image is centered on the padded area.

       color
	   Specify the color of the  padded  area.  For	 the  syntax  of  this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of color is "black".

       eval
	   Specify when to evaluate  width, height, x and y expression.

	   It accepts the following values:

	   init
	       Only evaluate expressions once during the filter initialization
	       or when a command is processed.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

       aspect
	   Pad to aspect instead to a resolution.

       The  value  for	the  width,  height,  x, and y options are expressions
       containing the following constants:

       in_w
       in_h
	   The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output width and height (the  size  of  the  padded  area),  as
	   specified by the width and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN
	   if not yet specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
	   The	horizontal  and	 vertical chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       •   Add paddings with the color "violet" to the input video. The output
	   video size is 640x480, and the top-left corner of the  input	 video
	   is placed at column 0, row 40

		   pad=640:480:0:40:violet

	   The example above is equivalent to the following command:

		   pad=width=640:height=480:x=0:y=40:color=violet

       •   Pad	the  input  to get an output with dimensions increased by 3/2,
	   and put the input video at the center of the padded area:

		   pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

       •   Pad the input to get a  squared  output  with  size	equal  to  the
	   maximum value between the input width and height, and put the input
	   video at the center of the padded area:

		   pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

       •   Pad the input to get a final w/h ratio of 16:9:

		   pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

       •   In  case  of	 anamorphic  video, in order to set the output display
	   aspect correctly, it is necessary to use  sar  in  the  expression,
	   according to the relation:

		   (ih * X / ih) * sar = output_dar
		   X = output_dar / sar

	   Thus the previous example needs to be modified to:

		   pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

       •   Double  the output size and put the input video in the bottom-right
	   corner of the output padded area:

		   pad="2*iw:2*ih:ow-iw:oh-ih"

   palettegen
       Generate one palette for a whole video stream.

       It accepts the following options:

       max_colors
	   Set the maximum number of colors to quantize in the palette.	 Note:
	   the palette will still  contain  256	 colors;  the  unused  palette
	   entries will be black.

       reserve_transparent
	   Create a palette of 255 colors maximum and reserve the last one for
	   transparency.  Reserving  the  transparency color is useful for GIF
	   optimization.  If not set, the maximum of  colors  in  the  palette
	   will	 be  256.  You	probably  want	to  disable  this option for a
	   standalone image.  Set by default.

       transparency_color
	   Set the color that will be used as background for transparency.

       stats_mode
	   Set statistics mode.

	   It accepts the following values:

	   full
	       Compute full frame histograms.

	   diff
	       Compute histograms only for the part that differs from previous
	       frame. This might be relevant to give more  importance  to  the
	       moving part of your input if the background is static.

	   single
	       Compute new histogram for each frame.

	   Default value is full.

       The  filter  also  exports the frame metadata "lavfi.color_quant_ratio"
       ("nb_color_in / nb_color_out") which you can use to evaluate the degree
       of color quantization of the palette. This information is also  visible
       at info logging level.

       Examples

       •   Generate a representative palette of a given video using ffmpeg:

		   ffmpeg -i input.mkv -vf palettegen palette.png

   paletteuse
       Use a palette to downsample an input video stream.

       The  filter  takes  two	inputs:	 one  video  stream and a palette. The
       palette must be a 256 pixels image.

       It accepts the following options:

       dither
	   Select dithering mode. Available algorithms are:

	   bayer
	       Ordered 8x8 bayer dithering (deterministic)

	   heckbert
	       Dithering as defined by Paul Heckbert  in  1982	(simple	 error
	       diffusion).   Note:  this  dithering  is	 sometimes  considered
	       "wrong" and is included as a reference.

	   floyd_steinberg
	       Floyd and Steingberg dithering (error diffusion)

	   sierra2
	       Frankie Sierra dithering v2 (error diffusion)

	   sierra2_4a
	       Frankie Sierra dithering v2 "Lite" (error diffusion)

	   sierra3
	       Frankie Sierra dithering v3 (error diffusion)

	   burkes
	       Burkes dithering (error diffusion)

	   atkinson
	       Atkinson dithering by Bill Atkinson at  Apple  Computer	(error
	       diffusion)

	   none
	       Disable dithering.

	   Default is sierra2_4a.

       bayer_scale
	   When	 bayer dithering is selected, this option defines the scale of
	   the pattern (how much the crosshatch pattern	 is  visible).	A  low
	   value means more visible pattern for less banding, and higher value
	   means less visible pattern at the cost of more banding.

	   The	option must be an integer value in the range [0,5]. Default is
	   2.

       diff_mode
	   If set, define the zone to process

	   rectangle
	       Only the	 changing  rectangle  will  be	reprocessed.  This  is
	       similar	to GIF cropping/offsetting compression mechanism. This
	       option can be useful for speed if only a part of the  image  is
	       changing,  and  has use cases such as limiting the scope of the
	       error diffusal dither to the rectangle that bounds  the	moving
	       scene  (it  leads  to  more  deterministic  output if the scene
	       doesn't change much, and as a  result  less  moving  noise  and
	       better GIF compression).

	   Default is none.

       new Take new palette for each output frame.

       alpha_threshold
	   Sets	 the alpha threshold for transparency. Alpha values above this
	   threshold will be treated as completely opaque,  and	 values	 below
	   this threshold will be treated as completely transparent.

	   The	option	must be an integer value in the range [0,255]. Default
	   is 128.

       Examples

       •   Use a palette (generated for example with palettegen) to  encode  a
	   GIF using ffmpeg:

		   ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

   perspective
       Correct perspective of video not recorded perpendicular to the screen.

       A description of the accepted parameters follows.

       x0
       y0
       x1
       y1
       x2
       y2
       x3
       y3  Set coordinates expression for top left, top right, bottom left and
	   bottom  right  corners.   Default values are "0:0:W:0:0:H:W:H" with
	   which perspective will remain unchanged.  If the "sense" option  is
	   set	to  "source",  then  the  specified points will be sent to the
	   corners of the  destination.	 If  the  "sense"  option  is  set  to
	   "destination",  then	 the corners of the source will be sent to the
	   specified coordinates.

	   The expressions can use the following variables:

	   W
	   H   the width and height of video frame.

	   in  Input frame count.

	   on  Output frame count.

       interpolation
	   Set interpolation for perspective correction.

	   It accepts the following values:

	   linear
	   cubic

	   Default value is linear.

       sense
	   Set interpretation of coordinate options.

	   It accepts the following values:

	   0, source
	       Send point in the source specified by the given coordinates  to
	       the corners of the destination.

	   1, destination
	       Send  the corners of the source to the point in the destination
	       specified by the given coordinates.

	       Default value is source.

       eval
	   Set	when  the  expressions	for  coordinates  x0,y0,...x3,y3   are
	   evaluated.

	   It accepts the following values:

	   init
	       only evaluate expressions once during the filter initialization
	       or when a command is processed

	   frame
	       evaluate expressions for each incoming frame

	   Default value is init.

   phase
       Delay  interlaced  video	 by  one  field	 time  so that the field order
       changes.

       The intended use is to fix PAL movies that have been captured with  the
       opposite field order to the film-to-video transfer.

       A description of the accepted parameters follows.

       mode
	   Set phase mode.

	   It accepts the following values:

	   t   Capture	field  order top-first, transfer bottom-first.	Filter
	       will delay the bottom field.

	   b   Capture field order bottom-first, transfer  top-first.	Filter
	       will delay the top field.

	   p   Capture	and transfer with the same field order. This mode only
	       exists for the documentation of the other options to refer  to,
	       but  if	you  actually select it, the filter will faithfully do
	       nothing.

	   a   Capture field order determined automatically  by	 field	flags,
	       transfer	 opposite.   Filter  selects  among t and b modes on a
	       frame by frame basis using field flags. If no field information
	       is available, then this works just like u.

	   u   Capture unknown or varying, transfer opposite.  Filter  selects
	       among t and b on a frame by frame basis by analyzing the images
	       and  selecting the alternative that produces best match between
	       the fields.

	   T   Capture top-first, transfer unknown or varying.	Filter selects
	       among t and p using image analysis.

	   B   Capture bottom-first,  transfer	unknown	 or  varying.	Filter
	       selects among b and p using image analysis.

	   A   Capture determined by field flags, transfer unknown or varying.
	       Filter  selects	among  t,  b and p using field flags and image
	       analysis. If no field information is available, then this works
	       just like U. This is the default mode.

	   U   Both capture and transfer unknown or varying.   Filter  selects
	       among t, b and p using image analysis only.

       Commands

       This filter supports the all above options as commands.

   photosensitivity
       Reduce various flashes in video, so to help users with epilepsy.

       It accepts the following options:

       frames, f
	   Set how many frames to use when filtering. Default is 30.

       threshold, t
	   Set detection threshold factor. Default is 1.  Lower is stricter.

       skip
	   Set	how  many  pixels  to skip when sampling frames. Default is 1.
	   Allowed range is from 1 to 1024.

       bypass
	   Leave frames unchanged. Default is disabled.

   pixdesctest
       Pixel  format  descriptor  test	filter,	 mainly	 useful	 for  internal
       testing. The output video should be equal to the input video.

       For example:

	       format=monow, pixdesctest

       can be used to test the monowhite pixel format descriptor definition.

   pixelize
       Apply pixelization to video stream.

       The filter accepts the following options:

       width, w
       height, h
	   Set	block  dimensions that will be used for pixelization.  Default
	   value is 16.

       mode, m
	   Set the mode of pixelization used.

	   Possible values are:

	   avg
	   min
	   max

	   Default value is "avg".

       planes, p
	   Set what planes to filter. Default is to filter all planes.

       Commands

       This filter supports all options as commands.

   pixscope
       Display sample values of color channels.	 Mainly	 useful	 for  checking
       color and levels. Minimum supported resolution is 640x480.

       The filters accept the following options:

       x   Set scope X position, relative offset on X axis.

       y   Set scope Y position, relative offset on Y axis.

       w   Set scope width.

       h   Set scope height.

       o   Set	window	opacity. This window also holds statistics about pixel
	   area.

       wx  Set window X position, relative offset on X axis.

       wy  Set window Y position, relative offset on Y axis.

       Commands

       This filter supports same commands as options.

   pp
       Enable  the  specified  chain  of   postprocessing   subfilters	 using
       libpostproc.  This  library should be automatically selected with a GPL
       build ("--enable-gpl").	Subfilters must be separated by '/' and can be
       disabled by prepending a '-'.  Each subfilter and some options  have  a
       short  and a long name that can be used interchangeably, i.e. dr/dering
       are the same.

       The filters accept the following options:

       subfilters
	   Set postprocessing subfilters string.

       All subfilters share common options to determine their scope:

       a/autoq
	   Honor the quality commands for this subfilter.

       c/chrom
	   Do chrominance filtering, too (default).

       y/nochrom
	   Do luma filtering only (no chrominance).

       n/noluma
	   Do chrominance filtering only (no luma).

       These options can be appended after the subfilter name, separated by  a
       '|'.

       Available subfilters are:

       hb/hdeblock[|difference[|flatness]]
	   Horizontal deblocking filter

	   difference
	       Difference  factor  where  higher  values  mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where	 lower	values	mean  more  deblocking
	       (default: 39).

       vb/vdeblock[|difference[|flatness]]
	   Vertical deblocking filter

	   difference
	       Difference  factor  where  higher  values  mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where	 lower	values	mean  more  deblocking
	       (default: 39).

       ha/hadeblock[|difference[|flatness]]
	   Accurate horizontal deblocking filter

	   difference
	       Difference  factor  where  higher  values  mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where	 lower	values	mean  more  deblocking
	       (default: 39).

       va/vadeblock[|difference[|flatness]]
	   Accurate vertical deblocking filter

	   difference
	       Difference  factor  where  higher  values  mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where	 lower	values	mean  more  deblocking
	       (default: 39).

       The horizontal and vertical deblocking filters share the difference and
       flatness	 values	 so  you  cannot set different horizontal and vertical
       thresholds.

       h1/x1hdeblock
	   Experimental horizontal deblocking filter

       v1/x1vdeblock
	   Experimental vertical deblocking filter

       dr/dering
	   Deringing filter

       tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
       reducer
	   threshold1
	       larger -> stronger filtering

	   threshold2
	       larger -> stronger filtering

	   threshold3
	       larger -> stronger filtering

       al/autolevels[:f/fullyrange], automatic brightness / contrast
       correction
	   f/fullyrange
	       Stretch luma to "0-255".

       lb/linblenddeint
	   Linear blend deinterlacing filter that deinterlaces the given block
	   by filtering all lines with a "(1 2 1)" filter.

       li/linipoldeint
	   Linear interpolating deinterlacing  filter  that  deinterlaces  the
	   given block by linearly interpolating every second line.

       ci/cubicipoldeint
	   Cubic  interpolating	 deinterlacing	filter	deinterlaces the given
	   block by cubically interpolating every second line.

       md/mediandeint
	   Median deinterlacing filter that deinterlaces the  given  block  by
	   applying a median filter to every second line.

       fd/ffmpegdeint
	   FFmpeg  deinterlacing  filter  that deinterlaces the given block by
	   filtering every second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
	   Vertically  applied	 FIR   lowpass	 deinterlacing	 filter	  that
	   deinterlaces the given block by filtering all lines with a "(-1 2 6
	   2 -1)" filter.

       fq/forceQuant[|quantizer]
	   Overrides  the  quantizer  table  from  the input with the constant
	   quantizer you specify.

	   quantizer
	       Quantizer to use

       de/default
	   Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
	   Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       •   Apply horizontal and vertical deblocking, deringing	and  automatic
	   brightness/contrast:

		   pp=hb/vb/dr/al

       •   Apply default filters without brightness/contrast correction:

		   pp=de/-al

       •   Apply default filters and temporal denoiser:

		   pp=default/tmpnoise|1|2|3

       •   Apply deblocking on luma only, and switch vertical deblocking on or
	   off automatically depending on available CPU time:

		   pp=hb|y/vb|a

   pp7
       Apply Postprocessing filter 7. It is variant of the spp filter, similar
       to spp = 6 with 7 point DCT, where only the center sample is used after
       IDCT.

       The filter accepts the following options:

       qp  Force  a  constant quantization parameter. It accepts an integer in
	   range 0 to 63. If not set, the filter will  use  the	 QP  from  the
	   video stream (if available).

       mode
	   Set thresholding mode. Available modes are:

	   hard
	       Set hard thresholding.

	   soft
	       Set  soft  thresholding	(better	 de-ringing effect, but likely
	       blurrier).

	   medium
	       Set medium thresholding (good results, default).

   premultiply
       Apply alpha premultiply effect to input video stream using first	 plane
       of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       inplace
	   Do not require 2nd input for processing, instead  use  alpha	 plane
	   from input stream.

   prewitt
       Apply prewitt operator to input video stream.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

       Commands

       This filter supports the all above options as commands.

   pseudocolor
       Alter frame colors in video with pseudocolors.

       This filter accepts the following options:

       c0  set pixel first component expression

       c1  set pixel second component expression

       c2  set pixel third component expression

       c3  set pixel fourth component expression,  corresponds	to  the	 alpha
	   component

       index, i
	   set component to use as base for altering colors

       preset, p
	   Pick one of built-in LUTs. By default is set to none.

	   Available LUTs:

	   magma
	   inferno
	   plasma
	   viridis
	   turbo
	   cividis
	   range1
	   range2
	   shadows
	   highlights
	   solar
	   nominal
	   preferred
	   total
	   spectral
	   cool
	   heat
	   fiery
	   blues
	   green
	   helix
       opacity
	   Set	opacity	 of  output  colors.  Allowed  range  is  from 0 to 1.
	   Default value is set to 1.

       Each of the expression options specifies	 the  expression  to  use  for
       computing  the  lookup  table  for  the	corresponding  pixel component
       values.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       ymin, umin, vmin, amin
	   The minimum allowed component value.

       ymax, umax, vmax, amax
	   The maximum allowed component value.

       All expressions default to "val".

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Change too high luma values to gradient:

		   pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"

   psnr
       Obtain the average, maximum and minimum	PSNR  (Peak  Signal  to	 Noise
       Ratio) between two input videos.

       This  filter  takes  in	input  two  input  videos,  the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the PSNR.

       Both video inputs must have the same resolution and  pixel  format  for
       this  filter  to	 work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The obtained average PSNR is printed through the logging system.

       The filter stores the accumulated MSE  (mean  squared  error)  of  each
       frame,  and  at	the  end  of  the processing it is averaged across all
       frames equally, and the following formula  is  applied  to  obtain  the
       PSNR:

	       PSNR = 10*log10(MAX^2/MSE)

       Where MAX is the average of the maximum values of each component of the
       image.

       The description of the accepted parameters follows.

       stats_file, f
	   If specified the filter will use the named file to save the PSNR of
	   each individual frame. When filename equals "-" the data is sent to
	   standard output.

       stats_version
	   Specifies which version of the stats file format to use. Details of
	   each format are written below.  Default value is 1.

       stats_add_max
	   Determines  whether	the  max  value	 is  output  to the stats log.
	   Default value is 0.	Requires stats_version >= 2. If	 this  is  set
	   and stats_version < 2, the filter will return an error.

       This filter also supports the framesync options.

       The  file  printed  if  stats_file  is selected, contains a sequence of
       key/value pairs of the form  key:value  for  each  compared  couple  of
       frames.

       If  a stats_version greater than 1 is specified, a header line precedes
       the list of per-frame-pair stats, with key value	 pairs	following  the
       frame format with the following parameters:

       psnr_log_version
	   The version of the log file format. Will match stats_version.

       fields
	   A comma separated list of the per-frame-pair parameters included in
	   the log.

       A description of each shown per-frame-pair parameter follows:

       n   sequential number of the input frame, starting from 1

       mse_avg
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames, averaged over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_b, mse_a
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames for the component specified by the suffix.

       psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
	   Peak Signal to Noise ratio of the compared frames for the component
	   specified by the suffix.

       max_avg, max_y, max_u, max_v
	   Maximum  allowed  value  for	 each  channel,	 and  average over all
	   channels.

       Examples

       •   For example:

		   movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
		   [main][ref] psnr="stats_file=stats.log" [out]

	   On this example the input file being processed is compared with the
	   reference file ref_movie.mpg. The PSNR of each individual frame  is
	   stored in stats.log.

       •   Another example with different containers:

		   ffmpeg -i main.mpg -i ref.mkv -lavfi	 "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]psnr" -f null -

   pullup
       Pulldown	 reversal (inverse telecine) filter, capable of handling mixed
       hard-telecine,  24000/1001  fps	 progressive,	and   30000/1001   fps
       progressive content.

       The  pullup  filter  is designed to take advantage of future context in
       making its decisions. This filter is stateless in  the  sense  that  it
       does not lock onto a pattern to follow, but it instead looks forward to
       the   following	fields	in  order  to  identify	 matches  and  rebuild
       progressive frames.

       To produce content with an even framerate, insert the fps filter	 after
       pullup,	use  "fps=24000/1001"  if  the	input  frame rate is 29.97fps,
       "fps=24" for 30fps and the (rare) telecined 25fps input.

       The filter accepts the following options:

       jl
       jr
       jt
       jb  These options set the amount of  "junk"  to	ignore	at  the	 left,
	   right,  top,	 and bottom of the image, respectively. Left and right
	   are in units of 8 pixels, while top and bottom are in  units	 of  2
	   lines.  The default is 8 pixels on each side.

       sb  Set	the  strict  breaks.  Setting this option to 1 will reduce the
	   chances of filter generating an occasional mismatched frame, but it
	   may also cause an excessive number of frames to be  dropped	during
	   high	 motion	 sequences.   Conversely,  setting  it to -1 will make
	   filter match fields more easily.  This may help processing of video
	   where there is slight blurring between the  fields,	but  may  also
	   cause  there	 to be interlaced frames in the output.	 Default value
	   is 0.

       mp  Set the metric plane to use. It accepts the following values:

	   l   Use luma plane.

	   u   Use chroma blue plane.

	   v   Use chroma red plane.

	   This option may be set to use chroma plane instead of  the  default
	   luma	 plane	for  doing  filter's  computations.  This  may improve
	   accuracy on very  clean  source  material,  but  more  likely  will
	   decrease  accuracy,	especially  if	there is chroma noise (rainbow
	   effect) or any grayscale video.  The main purpose of setting mp  to
	   a  chroma  plane  is	 to  reduce CPU load and make pullup usable in
	   realtime on slow machines.

       For best results (without duplicated frames in the output file)	it  is
       necessary  to  change  the  output  frame rate. For example, to inverse
       telecine NTSC input:

	       ffmpeg -i input -vf pullup -r 24000/1001 ...

   qp
       Change video quantization parameters (QP).

       The filter accepts the following option:

       qp  Set expression for quantization parameter.

       The expression is evaluated through the eval API and can contain, among
       others, the following constants:

       known
	   1 if index is not 129, 0 otherwise.

       qp  Sequential index starting from -129 to 128.

       Examples

       •   Some equation like:

		   qp=2+2*sin(PI*qp)

   random
       Flush video frames from internal cache of frames into a	random	order.
       No frame is discarded.  Inspired by frei0r nervous filter.

       frames
	   Set	size in number of frames of internal cache, in range from 2 to
	   512. Default is 30.

       seed
	   Set seed for random number generator, must be an  integer  included
	   between  0 and "UINT32_MAX". If not specified, or if explicitly set
	   to less than 0, the filter will try to use a good random seed on  a
	   best effort basis.

   readeia608
       Read  closed  captioning	 (EIA-608) information from the top lines of a
       video frame.

       This  filter  adds  frame  metadata  for	 "lavfi.readeia608.X.cc"   and
       "lavfi.readeia608.X.line",  where  "X"  is the number of the identified
       line with EIA-608  data	(starting  from	 0).  A	 description  of  each
       metadata value follows:

       lavfi.readeia608.X.cc
	   The two bytes stored as EIA-608 data (printed in hexadecimal).

       lavfi.readeia608.X.line
	   The number of the line on which the EIA-608 data was identified and
	   read.

       This filter accepts the following options:

       scan_min
	   Set the line to start scanning for EIA-608 data. Default is 0.

       scan_max
	   Set the line to end scanning for EIA-608 data. Default is 29.

       spw Set	the  ratio of width reserved for sync code detection.  Default
	   is 0.27. Allowed range is "[0.1 - 0.7]".

       chp Enable checking the parity bit. In the event of a parity error, the
	   filter will output 0x00 for that character. Default is false.

       lp  Lowpass lines prior to further processing. Default is enabled.

       Commands

       This filter supports the all above options as commands.

       Examples

       •   Output a csv with presentation time and  the	 first	two  lines  of
	   identified EIA-608 captioning data.

		   ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv

   readvitc
       Read  vertical  interval timecode (VITC) information from the top lines
       of a video frame.

       The filter adds frame metadata  key  "lavfi.readvitc.tc_str"  with  the
       timecode value, if a valid timecode has been detected. Further metadata
       key  "lavfi.readvitc.found" is set to 0/1 depending on whether timecode
       data has been found or not.

       This filter accepts the following options:

       scan_max
	   Set the maximum number of lines to scan for VITC data. If the value
	   is set to -1 the full video frame is scanned. Default is 45.

       thr_b
	   Set the luma threshold for black.  Accepts  float  numbers  in  the
	   range  [0.0,1.0],  default value is 0.2. The value must be equal or
	   less than "thr_w".

       thr_w
	   Set the luma threshold for white.  Accepts  float  numbers  in  the
	   range  [0.0,1.0],  default value is 0.6. The value must be equal or
	   greater than "thr_b".

       Examples

       •   Detect and draw VITC data onto the video frame; if no valid VITC is
	   detected, draw "--:--:--:--" as a placeholder:

		   ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

   remap
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x,  y)
       position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
       out of range, zero value for pixel will be used for destination pixel.

       Xmap  and  Ymap	input video streams must be of same dimensions. Output
       video stream will have Xmap/Ymap video  stream  dimensions.   Xmap  and
       Ymap input video streams are 16bit depth, single channel.

       format
	   Specify  pixel format of output from this filter. Can be "color" or
	   "gray".  Default is "color".

       fill
	   Specify the color of the unmapped pixels. For the  syntax  of  this
	   option,  check  the	"Color"	 section  in  the ffmpeg-utils manual.
	   Default color is "black".

   removegrain
       The removegrain filter is a spatial denoiser for progressive video.

       m0  Set mode for the first plane.

       m1  Set mode for the second plane.

       m2  Set mode for the third plane.

       m3  Set mode for the fourth plane.

       Range of mode is from 0 to 24. Description of each mode follows:

       0   Leave input plane unchanged. Default.

       1   Clips the pixel with the minimum and maximum	 of  the  8  neighbour
	   pixels.

       2   Clips  the  pixel  with  the	 second	 minimum  and maximum of the 8
	   neighbour pixels.

       3   Clips the pixel with	 the  third  minimum  and  maximum  of	the  8
	   neighbour pixels.

       4   Clips  the  pixel  with  the	 fourth	 minimum  and maximum of the 8
	   neighbour pixels.  This is equivalent to a median filter.

       5   Line-sensitive clipping giving the minimal change.

       6   Line-sensitive clipping, intermediate.

       7   Line-sensitive clipping, intermediate.

       8   Line-sensitive clipping, intermediate.

       9   Line-sensitive clipping on a line where the neighbours  pixels  are
	   the closest.

       10  Replaces the target pixel with the closest neighbour.

       11  [1 2 1] horizontal and vertical kernel blur.

       12  Same as mode 11.

       13  Bob mode, interpolates top field from the line where the neighbours
	   pixels are the closest.

       14  Bob	mode,  interpolates  bottom  field  from  the  line  where the
	   neighbours pixels are the closest.

       15  Bob mode, interpolates top field.  Same  as	13  but	 with  a  more
	   complicated interpolation formula.

       16  Bob	mode,  interpolates  bottom  field. Same as 14 but with a more
	   complicated interpolation formula.

       17  Clips the pixel with the minimum and maximum	 of  respectively  the
	   maximum and minimum of each pair of opposite neighbour pixels.

       18  Line-sensitive  clipping  using  opposite neighbours whose greatest
	   distance from the current pixel is minimal.

       19  Replaces the pixel with the average of its 8 neighbours.

       20  Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

       21  Clips pixels using the averages of opposite neighbour.

       22  Same as mode 21 but simpler and faster.

       23  Small edge and halo removal, but reputed useless.

       24  Similar as 23.

   removelogo
       Suppress a TV station logo, using an  image  file  to  determine	 which
       pixels  comprise	 the  logo.  It	 works	by  filling in the pixels that
       comprise the logo with neighboring pixels.

       The filter accepts the following options:

       filename, f
	   Set the filter bitmap file, which can be any image format supported
	   by libavformat. The width and height of the image file  must	 match
	   those of the video stream being processed.

       Pixels  in  the	provided  bitmap  image	 with  a value of zero are not
       considered part of the logo, non-zero pixels are considered part of the
       logo. If you use white (255) for the logo and black (0) for  the	 rest,
       you  will  be  safe. For making the filter bitmap, it is recommended to
       take a screen capture of a black frame with the logo visible, and  then
       using a threshold filter followed by the erode filter once or twice.

       If  needed,  little  splotches  can be fixed manually. Remember that if
       logo pixels are not covered, the filter quality will be	much  reduced.
       Marking	too many pixels as part of the logo does not hurt as much, but
       it will increase the amount of blurring needed to cover over the	 image
       and will destroy more information than necessary, and extra pixels will
       slow things down on a large logo.

   repeatfields
       This  filter  uses  the repeat_field flag from the Video ES headers and
       hard repeats fields based on its value.

   reverse
       Reverse a video clip.

       Warning: This filter requires memory to	buffer	the  entire  clip,  so
       trimming is suggested.

       Examples

       •   Take the first 5 seconds of a clip, and reverse it.

		   trim=end=5,reverse

   rgbashift
       Shift R/G/B/A pixels horizontally and/or vertically.

       The filter accepts the following options:

       rh  Set amount to shift red horizontally.

       rv  Set amount to shift red vertically.

       gh  Set amount to shift green horizontally.

       gv  Set amount to shift green vertically.

       bh  Set amount to shift blue horizontally.

       bv  Set amount to shift blue vertically.

       ah  Set amount to shift alpha horizontally.

       av  Set amount to shift alpha vertically.

       edge
	   Set edge mode, can be smear, default, or warp.

       Commands

       This filter supports the all above options as commands.

   roberts
       Apply roberts cross operator to input video stream.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

       Commands

       This filter supports the all above options as commands.

   rotate
       Rotate video by an arbitrary angle expressed in radians.

       The filter accepts the following options:

       A description of the optional parameters follows.

       angle, a
	   Set an expression for the angle by which to rotate the input	 video
	   clockwise,  expressed as a number of radians. A negative value will
	   result in a counter-clockwise rotation. By default  it  is  set  to
	   "0".

	   This expression is evaluated for each frame.

       out_w, ow
	   Set	the  output  width  expression,	 default  value is "iw".  This
	   expression is evaluated just once during configuration.

       out_h, oh
	   Set the output height expression,  default  value  is  "ih".	  This
	   expression is evaluated just once during configuration.

       bilinear
	   Enable  bilinear  interpolation  if set to 1, a value of 0 disables
	   it. Default value is 1.

       fillcolor, c
	   Set the color used to fill the  output  area	 not  covered  by  the
	   rotated  image.  For	 the  general syntax of this option, check the
	   "Color" section in the ffmpeg-utils manual.	If the	special	 value
	   "none"  is  selected	 then  no  background  is  printed (useful for
	   example if the background is never shown).

	   Default value is "black".

       The expressions for the angle and  the  output  size  can  contain  the
       following constants and functions:

       n   sequential number of the input frame, starting from 0. It is always
	   NAN before the first frame is filtered.

       t   time	 in seconds of the input frame, it is set to 0 when the filter
	   is configured. It is always NAN before the first frame is filtered.

       hsub
       vsub
	   horizontal and vertical chroma subsample values.  For  example  for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_w, iw
       in_h, ih
	   the input video width and height

       out_w, ow
       out_h, oh
	   the output width and height, that is the size of the padded area as
	   specified by the width and height expressions

       rotw(a)
       roth(a)
	   the	minimal	 width/height  required	 for completely containing the
	   input video rotated by a radians.

	   These are  only  available  when  computing	the  out_w  and	 out_h
	   expressions.

       Examples

       •   Rotate the input by PI/6 radians clockwise:

		   rotate=PI/6

       •   Rotate the input by PI/6 radians counter-clockwise:

		   rotate=-PI/6

       •   Rotate the input by 45 degrees clockwise:

		   rotate=45*PI/180

       •   Apply  a constant rotation with period T, starting from an angle of
	   PI/3:

		   rotate=PI/3+2*PI*t/T

       •   Make the input video	 rotation  oscillating	with  a	 period	 of  T
	   seconds and an amplitude of A radians:

		   rotate=A*sin(2*PI/T*t)

       •   Rotate  the video, output size is chosen so that the whole rotating
	   input video is always completely contained in the output:

		   rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

       •   Rotate the video, reduce the output size so that no	background  is
	   ever shown:

		   rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

       Commands

       The filter supports the following commands:

       a, angle
	   Set	the  angle expression.	The command accepts the same syntax of
	   the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   sab
       Apply Shape Adaptive Blur.

       The filter accepts the following options:

       luma_radius, lr
	   Set luma blur filter strength, must be a value  in  range  0.1-4.0,
	   default value is 1.0. A greater value will result in a more blurred
	   image, and in slower processing.

       luma_pre_filter_radius, lpfr
	   Set	luma  pre-filter radius, must be a value in the 0.1-2.0 range,
	   default value is 1.0.

       luma_strength, ls
	   Set luma maximum difference between pixels to still be  considered,
	   must be a value in the 0.1-100.0 range, default value is 1.0.

       chroma_radius, cr
	   Set chroma blur filter strength, must be a value in range -0.9-4.0.
	   A  greater value will result in a more blurred image, and in slower
	   processing.

       chroma_pre_filter_radius, cpfr
	   Set chroma pre-filter radius, must  be  a  value  in	 the  -0.9-2.0
	   range.

       chroma_strength, cs
	   Set	 chroma	  maximum   difference	between	 pixels	 to  still  be
	   considered, must be a value in the -0.9-100.0 range.

       Each chroma option value, if not explicitly specified, is  set  to  the
       corresponding luma option value.

   scale
       Scale (resize) the input video, using the libswscale library.

       The  scale filter forces the output display aspect ratio to be the same
       of the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next filter, the scale filter will convert the input to	the  requested
       format.

       Options

       The  filter  accepts  the  following  options,  or  any	of the options
       supported by the libswscale scaler.

       See the ffmpeg-scaler manual for the complete list of scaler options.

       width, w
       height, h
	   Set the output video dimension expression.  Default	value  is  the
	   input dimension.

	   If  the  width  or  w  value	 is 0, the input width is used for the
	   output. If the height or h value is 0, the input height is used for
	   the output.

	   If one and only one of the values is -n with	 n  >=	1,  the	 scale
	   filter  will	 use  a	 value	that maintains the aspect ratio of the
	   input image, calculated from the other specified  dimension.	 After
	   that	 it  will, however, make sure that the calculated dimension is
	   divisible by n and adjust the value if necessary.

	   If both values are -n with n >= 1, the behavior will	 be  identical
	   to both values being set to 0 as previously detailed.

	   See	below  for  the	 list  of  accepted  constants	for use in the
	   dimension expression.

       eval
	   Specify when to evaluate width and height  expression.  It  accepts
	   the following values:

	   init
	       Only evaluate expressions once during the filter initialization
	       or when a command is processed.

	   frame
	       Evaluate expressions for each incoming frame.

	   Default value is init.

       interl
	   Set the interlacing mode. It accepts the following values:

	   1   Force interlaced aware scaling.

	   0   Do not apply interlaced scaling.

	   -1  Select interlaced aware scaling depending on whether the source
	       frames are flagged as interlaced or not.

	   Default value is 0.

       flags
	   Set	libswscale scaling flags. See the ffmpeg-scaler manual for the
	   complete list of values. If not  explicitly	specified  the	filter
	   applies the default flags.

       param0, param1
	   Set	libswscale  input  parameters for scaling algorithms that need
	   them. See the ffmpeg-scaler manual for the complete	documentation.
	   If not explicitly specified the filter applies empty parameters.

       size, s
	   Set the video size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual.

       in_color_matrix
       out_color_matrix
	   Set in/output YCbCr color space type.

	   This	 allows	 the  autodetected  value  to be overridden as well as
	   allows forcing a specific value used for the output and encoder.

	   If not specified, the color space type depends on the pixel format.

	   Possible values:

	   auto
	       Choose automatically.

	   bt709
	       Format  conforming  to  International  Telecommunication	 Union
	       (ITU) Recommendation BT.709.

	   fcc Set  color  space  conforming  to  the  United  States  Federal
	       Communications Commission (FCC)	Code  of  Federal  Regulations
	       (CFR) Title 47 (2003) 73.682 (a).

	   bt601
	   bt470
	   smpte170m
	       Set color space conforming to:

	       •   ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

	       •   ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

	       •   Society  of Motion Picture and Television Engineers (SMPTE)
		   ST 170:2004

	   smpte240m
	       Set color space conforming to SMPTE ST 240:1999.

	   bt2020
	       Set  color  space  conforming  to  ITU-R	 BT.2020  non-constant
	       luminance system.

       in_range
       out_range
	   Set in/output YCbCr sample range.

	   This	 allows	 the  autodetected  value  to be overridden as well as
	   allows forcing a specific value used for the output and encoder. If
	   not specified, the range depends  on	 the  pixel  format.  Possible
	   values:

	   auto/unknown
	       Choose automatically.

	   jpeg/full/pc
	       Set full range (0-255 in case of 8-bit luma).

	   mpeg/limited/tv
	       Set "MPEG" range (16-235 in case of 8-bit luma).

       force_original_aspect_ratio
	   Enable  decreasing  or  increasing  output video width or height if
	   necessary to keep the original aspect ratio. Possible values:

	   disable
	       Scale the video as specified and disable this feature.

	   decrease
	       The output video dimensions will automatically be decreased  if
	       needed.

	   increase
	       The  output video dimensions will automatically be increased if
	       needed.

	   One useful instance of this option is that when you know a specific
	   device's maximum allowed resolution, you can use this to limit  the
	   output  video  to  that,  while  retaining  the  aspect  ratio. For
	   example, device A allows  1280x720  playback,  and  your  video  is
	   1920x800.  Using  this  option  (set it to decrease) and specifying
	   1280x720 to the command line makes the output 1280x533.

	   Please note that this is a different thing than specifying -1 for w
	   or h, you still need to specify  the	 output	 resolution  for  this
	   option to work.

       force_divisible_by
	   Ensures  that  both	the  output  dimensions, width and height, are
	   divisible  by  the  given   integer	 when	used   together	  with
	   force_original_aspect_ratio.	 This  works  similar to using "-n" in
	   the w and h options.

	   This option respects the value set for force_original_aspect_ratio,
	   increasing or decreasing the resolution  accordingly.  The  video's
	   aspect ratio may be slightly modified.

	   This	 option can be handy if you need to have a video fit within or
	   exceed a defined resolution using  force_original_aspect_ratio  but
	   also have encoder restrictions on width or height divisibility.

       The  values  of	the  w	and  h	options are expressions containing the
       following constants:

       in_w
       in_h
	   The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
	   horizontal and vertical input chroma subsample values. For  example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
	   horizontal and vertical output chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       n   The	(sequential) number of the input frame, starting from 0.  Only
	   available with "eval=frame".

       t   The presentation timestamp of  the  input  frame,  expressed	 as  a
	   number of seconds. Only available with "eval=frame".

       pos The position (byte offset) of the frame in the input stream, or NaN
	   if  this information is unavailable and/or meaningless (for example
	   in case of synthetic video).	  Only	available  with	 "eval=frame".
	   Deprecated, do not use.

       Examples

       •   Scale the input video to a size of 200x100

		   scale=w=200:h=100

	   This is equivalent to:

		   scale=200:100

	   or:

		   scale=200x100

       •   Specify a size abbreviation for the output size:

		   scale=qcif

	   which can also be written as:

		   scale=size=qcif

       •   Scale the input to 2x:

		   scale=w=2*iw:h=2*ih

       •   The above is the same as:

		   scale=2*in_w:2*in_h

       •   Scale the input to 2x with forced interlaced scaling:

		   scale=2*iw:2*ih:interl=1

       •   Scale the input to half size:

		   scale=w=iw/2:h=ih/2

       •   Increase the width, and set the height to the same size:

		   scale=3/2*iw:ow

       •   Seek Greek harmony:

		   scale=iw:1/PHI*iw
		   scale=ih*PHI:ih

       •   Increase the height, and set the width to 3/2 of the height:

		   scale=w=3/2*oh:h=3/5*ih

       •   Increase  the  size,	 making	 the  size  a  multiple	 of the chroma
	   subsample values:

		   scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub"

       •   Increase the width to a maximum of 500  pixels,  keeping  the  same
	   aspect ratio as the input:

		   scale=w='min(500\, iw*3/2):h=-1'

       •   Make pixels square by combining scale and setsar:

		   scale='trunc(ih*dar):ih',setsar=1/1

       •   Make	 pixels	 square by combining scale and setsar, making sure the
	   resulting resolution is even (required by some codecs):

		   scale='trunc(ih*dar/2)*2:trunc(ih/2)*2',setsar=1/1

       Commands

       This filter supports the following commands:

       width, w
       height, h
	   Set the output video dimension expression.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   scale_cuda
       Scale (resize) and  convert  (pixel  format)  the  input	 video,	 using
       accelerated CUDA kernels.  Setting the output width and height works in
       the same way as for the scale filter.

       The filter accepts the following options:

       w
       h   Set	the  output  video  dimension expression. Default value is the
	   input dimension.

	   Allows for the same expressions as the scale filter.

       interp_algo
	   Sets the algorithm used for scaling:

	   nearest
	       Nearest neighbour

	       Used by default if input parameters match the desired output.

	   bilinear
	       Bilinear

	   bicubic
	       Bicubic

	       This is the default.

	   lanczos
	       Lanczos

       format
	   Controls the output	pixel  format.	By  default,  or  if  none  is
	   specified, the input pixel format is used.

	   The	filter	does  not support converting between YUV and RGB pixel
	   formats.

       passthrough
	   If set to 0, every frame is processed, even	if  no	conversion  is
	   neccesary.	This  mode can be useful to use the filter as a buffer
	   for a downstream frame-consumer that exhausts the  limited  decoder
	   frame pool.

	   If  set  to	1,  frames  are passed through as-is if they match the
	   desired output parameters. This is the default behaviour.

       param
	   Algorithm-Specific parameter.

	   Affects the curves of the bicubic algorithm.

       force_original_aspect_ratio
       force_divisible_by
	   Work the same as the identical scale filter options.

       Examples

       •   Scale input to 720p, keeping aspect ratio and ensuring  the	output
	   is yuv420p.

		   scale_cuda=-2:720:format=yuv420p

       •   Upscale to 4K using nearest neighbour algorithm.

		   scale_cuda=4096:2160:interp_algo=nearest

       •   Don't  do any conversion or scaling, but copy all input frames into
	   newly allocated ones.  This can be useful to deal with a filter and
	   encode chain that otherwise exhausts the decoders frame pool.

		   scale_cuda=passthrough=0

   scale_npp
       Use the NVIDIA  Performance  Primitives	(libnpp)  to  perform  scaling
       and/or pixel format conversion on CUDA video frames. Setting the output
       width and height works in the same way as for the scale filter.

       The following additional options are accepted:

       format
	   The	pixel  format  of the output CUDA frames. If set to the string
	   "same" (the default), the input format  will	 be  kept.  Note  that
	   automatic  format  negotiation  and conversion is not yet supported
	   for hardware frames

       interp_algo
	   The	interpolation  algorithm  used	for  resizing.	One   of   the
	   following:

	   nn  Nearest neighbour.

	   linear
	   cubic
	   cubic2p_bspline
	       2-parameter cubic (B=1, C=0)

	   cubic2p_catmullrom
	       2-parameter cubic (B=0, C=1/2)

	   cubic2p_b05c03
	       2-parameter cubic (B=1/2, C=3/10)

	   super
	       Supersampling

	   lanczos
       force_original_aspect_ratio
	   Enable  decreasing  or  increasing  output video width or height if
	   necessary to keep the original aspect ratio. Possible values:

	   disable
	       Scale the video as specified and disable this feature.

	   decrease
	       The output video dimensions will automatically be decreased  if
	       needed.

	   increase
	       The  output video dimensions will automatically be increased if
	       needed.

	   One useful instance of this option is that when you know a specific
	   device's maximum allowed resolution, you can use this to limit  the
	   output  video  to  that,  while  retaining  the  aspect  ratio. For
	   example, device A allows  1280x720  playback,  and  your  video  is
	   1920x800.  Using  this  option  (set it to decrease) and specifying
	   1280x720 to the command line makes the output 1280x533.

	   Please note that this is a different thing than specifying -1 for w
	   or h, you still need to specify  the	 output	 resolution  for  this
	   option to work.

       force_divisible_by
	   Ensures  that  both	the  output  dimensions, width and height, are
	   divisible  by  the  given   integer	 when	used   together	  with
	   force_original_aspect_ratio.	 This  works  similar to using "-n" in
	   the w and h options.

	   This option respects the value set for force_original_aspect_ratio,
	   increasing or decreasing the resolution  accordingly.  The  video's
	   aspect ratio may be slightly modified.

	   This	 option can be handy if you need to have a video fit within or
	   exceed a defined resolution using  force_original_aspect_ratio  but
	   also have encoder restrictions on width or height divisibility.

       eval
	   Specify  when  to  evaluate width and height expression. It accepts
	   the following values:

	   init
	       Only evaluate expressions once during the filter initialization
	       or when a command is processed.

	   frame
	       Evaluate expressions for each incoming frame.

       The values of the w  and	 h  options  are  expressions  containing  the
       following constants:

       in_w
       in_h
	   The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       n   The	(sequential) number of the input frame, starting from 0.  Only
	   available with "eval=frame".

       t   The presentation timestamp of  the  input  frame,  expressed	 as  a
	   number of seconds. Only available with "eval=frame".

       pos The position (byte offset) of the frame in the input stream, or NaN
	   if  this information is unavailable and/or meaningless (for example
	   in case of synthetic video).	  Only	available  with	 "eval=frame".
	   Deprecated, do not use.

   scale2ref
       Scale (resize) the input video, based on a reference video.

       See the scale filter for available options, scale2ref supports the same
       but  uses  the  reference  video	 instead  of  the main input as basis.
       scale2ref also supports the following additional constants  for	the  w
       and h options:

       main_w
       main_h
	   The main input video's width and height

       main_a
	   The same as main_w / main_h

       main_sar
	   The main input video's sample aspect ratio

       main_dar, mdar
	   The	main  input  video's  display  aspect  ratio.  Calculated from
	   "(main_w / main_h) * main_sar".

       main_hsub
       main_vsub
	   The main input video's horizontal  and  vertical  chroma  subsample
	   values.   For  example for the pixel format "yuv422p" hsub is 2 and
	   vsub is 1.

       main_n
	   The (sequential) number of the main input frame, starting  from  0.
	   Only available with "eval=frame".

       main_t
	   The	presentation timestamp of the main input frame, expressed as a
	   number of seconds. Only available with "eval=frame".

       main_pos
	   The position (byte offset) of the frame in the main	input  stream,
	   or  NaN  if this information is unavailable and/or meaningless (for
	   example  in	case  of  synthetic  video).   Only   available	  with
	   "eval=frame".

       Examples

       •   Scale  a  subtitle  stream  (b) to match the main video (a) in size
	   before overlaying

		   'scale2ref[b][a];[a][b]overlay'

       •   Scale a logo to 1/10th the height of a video, while preserving  its
	   display aspect ratio.

		   [logo-in][video-in]scale2ref=w=oh*mdar:h=ih/10[logo-out][video-out]

       Commands

       This filter supports the following commands:

       width, w
       height, h
	   Set the output video dimension expression.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

   scale2ref_npp
       Use  the	 NVIDIA	 Performance Primitives (libnpp) to scale (resize) the
       input video, based on a reference video.

       See the scale_npp filter for available options, scale2ref_npp  supports
       the  same  but  uses  the  reference video instead of the main input as
       basis. scale2ref_npp also supports the following	 additional  constants
       for the w and h options:

       main_w
       main_h
	   The main input video's width and height

       main_a
	   The same as main_w / main_h

       main_sar
	   The main input video's sample aspect ratio

       main_dar, mdar
	   The	main  input  video's  display  aspect  ratio.  Calculated from
	   "(main_w / main_h) * main_sar".

       main_n
	   The (sequential) number of the main input frame, starting  from  0.
	   Only available with "eval=frame".

       main_t
	   The	presentation timestamp of the main input frame, expressed as a
	   number of seconds. Only available with "eval=frame".

       main_pos
	   The position (byte offset) of the frame in the main	input  stream,
	   or  NaN  if this information is unavailable and/or meaningless (for
	   example  in	case  of  synthetic  video).   Only   available	  with
	   "eval=frame".

       Examples

       •   Scale  a  subtitle  stream  (b) to match the main video (a) in size
	   before overlaying

		   'scale2ref_npp[b][a];[a][b]overlay_cuda'

       •   Scale a logo to 1/10th the height of a video, while preserving  its
	   display aspect ratio.

		   [logo-in][video-in]scale2ref_npp=w=oh*mdar:h=ih/10[logo-out][video-out]

   scale_vt
       Scale and convert the color parameters using VTPixelTransferSession.

       The filter accepts the following options:

       w
       h   Set	the  output  video  dimension expression. Default value is the
	   input dimension.

       color_matrix
	   Set the output colorspace matrix.

       color_primaries
	   Set the output color primaries.

       color_transfer
	   Set the output transfer characteristics.

   scharr
       Apply scharr operator to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes will be  processed,	 unprocessed  planes  will  be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

       Commands

       This filter supports the all above options as commands.

   scroll
       Scroll input video horizontally and/or vertically by constant speed.

       The filter accepts the following options:

       horizontal, h
	   Set	the horizontal scrolling speed. Default is 0. Allowed range is
	   from -1 to 1.  Negative values changes scrolling direction.

       vertical, v
	   Set the vertical scrolling speed. Default is 0.  Allowed  range  is
	   from -1 to 1.  Negative values changes scrolling direction.

       hpos
	   Set	the  initial  horizontal  scrolling  position.	Default	 is 0.
	   Allowed range is from 0 to 1.

       vpos
	   Set the initial vertical scrolling position. Default is 0.  Allowed
	   range is from 0 to 1.

       Commands

       This filter supports the following commands:

       horizontal, h
	   Set the horizontal scrolling speed.

       vertical, v
	   Set the vertical scrolling speed.

   scdet
       Detect video scene change.

       This  filter  sets  frame  metadata  with mafd between frame, the scene
       score, and forward the frame to the next filter, so they can use	 these
       metadata to detect scene change or others.

       In addition, this filter logs a message and sets frame metadata when it
       detects a scene change by threshold.

       "lavfi.scd.mafd" metadata keys are set with mafd for every frame.

       "lavfi.scd.score"  metadata  keys  are  set with scene change score for
       every frame to detect scene change.

       "lavfi.scd.time" metadata keys are set with current filtered frame time
       which detect scene change with threshold.

       The filter accepts the following options:

       threshold, t
	   Set the scene change detection threshold as a percentage of maximum
	   change. Good values are in the "[8.0, 14.0]" range. The  range  for
	   threshold is "[0., 100.]".

	   Default value is 10..

       sc_pass, s
	   Set	the  flag  to  pass  scene  change  frames to the next filter.
	   Default value is 0 You can enable it if you want to get snapshot of
	   scene change frames only.

   selectivecolor
       Adjust cyan, magenta, yellow and black  (CMYK)  to  certain  ranges  of
       colors  (such  as  "reds",  "yellows",  "greens",  "cyans",  ...).  The
       adjustment range is defined by the "purity" of the color (that is,  how
       saturated it already is).

       This filter is similar to the Adobe Photoshop Selective Color tool.

       The filter accepts the following options:

       correction_method
	   Select color correction method.

	   Available values are:

	   absolute
	       Specified  adjustments are applied "as-is" (added/subtracted to
	       original pixel component value).

	   relative
	       Specified adjustments are relative to  the  original  component
	       value.

	   Default is "absolute".

       reds
	   Adjustments	for  red pixels (pixels where the red component is the
	   maximum)

       yellows
	   Adjustments for yellow pixels (pixels where the blue	 component  is
	   the minimum)

       greens
	   Adjustments	for  green pixels (pixels where the green component is
	   the maximum)

       cyans
	   Adjustments for cyan pixels (pixels where the red component is  the
	   minimum)

       blues
	   Adjustments for blue pixels (pixels where the blue component is the
	   maximum)

       magentas
	   Adjustments for magenta pixels (pixels where the green component is
	   the minimum)

       whites
	   Adjustments	for  white  pixels  (pixels  where  all components are
	   greater than 128)

       neutrals
	   Adjustments for all pixels except pure black and pure white

       blacks
	   Adjustments for black  pixels  (pixels  where  all  components  are
	   lesser than 128)

       psfile
	   Specify  a  Photoshop  selective  color file (".asv") to import the
	   settings from.

       All the adjustment settings (reds, yellows, ...) accept up to  4	 space
       separated  floating  point  adjustment  values  in  the	[-1,1]	range,
       respectively to adjust the amount of cyan, magenta,  yellow  and	 black
       for the pixels of its range.

       Examples

       •   Increase cyan by 50% and reduce yellow by 33% in every green areas,
	   and increase magenta by 27% in blue areas:

		   selectivecolor=greens=.5 0 -.33 0:blues=0 .27

       •   Use a Photoshop selective color preset:

		   selectivecolor=psfile=MySelectiveColorPresets/Misty.asv

   separatefields
       The  "separatefields"  takes  a frame-based video input and splits each
       frame into its components fields, producing a new half height clip with
       twice the frame rate and twice the frame count.

       This filter use field-dominance information in frame to decide which of
       each pair of fields to place first in the output.  If it gets it	 wrong
       use setfield filter before "separatefields" filter.

   setdar, setsar
       The "setdar" filter sets the Display Aspect Ratio for the filter output
       video.

       This is done by changing the specified Sample (aka Pixel) Aspect Ratio,
       according to the following equation:

	       <DAR> = <HORIZONTAL_RESOLUTION> / <VERTICAL_RESOLUTION> * <SAR>

       Keep  in	 mind  that  the  "setdar"  filter  does  not modify the pixel
       dimensions of the video frame. Also, the display aspect	ratio  set  by
       this filter may be changed by later filters in the filterchain, e.g. in
       case of scaling or if another "setdar" or a "setsar" filter is applied.

       The  "setsar"  filter  sets the Sample (aka Pixel) Aspect Ratio for the
       filter output video.

       Note that as a consequence of  the  application	of  this  filter,  the
       output  display	aspect	ratio  will  change  according to the equation
       above.

       Keep in mind that the sample aspect ratio set by	 the  "setsar"	filter
       may  be	changed	 by  later filters in the filterchain, e.g. if another
       "setsar" or a "setdar" filter is applied.

       It accepts the following parameters:

       r, ratio, dar ("setdar" only), sar ("setsar" only)
	   Set the aspect ratio used by the filter.

	   The parameter  can  be  a  floating	point  number  string,	or  an
	   expression.	If  the	 parameter  is not specified, the value "0" is
	   assumed, meaning that the same input value is used.

       max Set the maximum integer value to use for expressing	numerator  and
	   denominator when reducing the expressed aspect ratio to a rational.
	   Default value is 100.

       The parameter sar is an expression containing the following constants:

       w, h
	   The input width and height.

       a   Same as w / h.

       sar The input sample aspect ratio.

       dar The input display aspect ratio. It is the same as (w / h) * sar.

       hsub, vsub
	   Horizontal  and  vertical chroma subsample values. For example, for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       •   To change the display aspect ratio to  16:9,	 specify  one  of  the
	   following:

		   setdar=dar=1.77777
		   setdar=dar=16/9

       •   To change the sample aspect ratio to 10:11, specify:

		   setsar=sar=10/11

       •   To  set  a  display	aspect	ratio  of  16:9, and specify a maximum
	   integer value of 1000  in  the  aspect  ratio  reduction,  use  the
	   command:

		   setdar=ratio=16/9:max=1000

   setfield
       Force field for the output video frame.

       The  "setfield"	filter	marks  the interlace type field for the output
       frames. It  does	 not  change  the  input  frame,  but  only  sets  the
       corresponding  property,	 which	affects	 how  the  frame is treated by
       following filters (e.g. "fieldorder" or "yadif").

       The filter accepts the following options:

       mode
	   Available values are:

	   auto
	       Keep the same field property.

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

   setparams
       Force frame parameter for the output video frame.

       The "setparams" filter marks interlace and color range for  the	output
       frames.	It  does  not  change  the  input  frame,  but	only  sets the
       corresponding property, which affects  how  the	frame  is  treated  by
       filters/encoders.

       field_mode
	   Available values are:

	   auto
	       Keep the same field property (default).

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

       range
	   Available values are:

	   auto
	       Keep the same color range property (default).

	   unspecified, unknown
	       Mark the frame as unspecified color range.

	   limited, tv, mpeg
	       Mark the frame as limited range.

	   full, pc, jpeg
	       Mark the frame as full range.

       color_primaries
	   Set the color primaries.  Available values are:

	   auto
	       Keep the same color primaries property (default).

	   bt709
	   unknown
	   bt470m
	   bt470bg
	   smpte170m
	   smpte240m
	   film
	   bt2020
	   smpte428
	   smpte431
	   smpte432
	   jedec-p22
       color_trc
	   Set the color transfer.  Available values are:

	   auto
	       Keep the same color trc property (default).

	   bt709
	   unknown
	   bt470m
	   bt470bg
	   smpte170m
	   smpte240m
	   linear
	   log100
	   log316
	   iec61966-2-4
	   bt1361e
	   iec61966-2-1
	   bt2020-10
	   bt2020-12
	   smpte2084
	   smpte428
	   arib-std-b67
       colorspace
	   Set the colorspace.	Available values are:

	   auto
	       Keep the same colorspace property (default).

	   gbr
	   bt709
	   unknown
	   fcc
	   bt470bg
	   smpte170m
	   smpte240m
	   ycgco
	   bt2020nc
	   bt2020c
	   smpte2085
	   chroma-derived-nc
	   chroma-derived-c
	   ictcp

   sharpen_npp
       Use  the	 NVIDIA	 Performance  Primitives  (libnpp)  to	perform	 image
       sharpening with border control.

       The following additional options are accepted:

       border_type
	   Type of sampling to be used ad frame borders. One of the following:

	   replicate
	       Replicate pixel values.

   shear
       Apply shear transform to input video.

       This filter supports the following options:

       shx Shear factor in X-direction. Default value is 0.  Allowed range  is
	   from -2 to 2.

       shy Shear  factor in Y-direction. Default value is 0.  Allowed range is
	   from -2 to 2.

       fillcolor, c
	   Set the color used to fill the  output  area	 not  covered  by  the
	   transformed video. For the general syntax of this option, check the
	   "Color"  section  in the ffmpeg-utils manual.  If the special value
	   "none" is selected  then  no	 background  is	 printed  (useful  for
	   example if the background is never shown).

	   Default value is "black".

       interp
	   Set	interpolation type. Can be "bilinear" or "nearest". Default is
	   "bilinear".

       Commands

       This filter supports the all above options as commands.

   showinfo
       Show a line containing various information for each input video	frame.
       The input video is not modified.

       This filter supports the following options:

       checksum
	   Calculate checksums of each plane. By default enabled.

       The  shown  line	 contains  a  sequence	of key/value pairs of the form
       key:value.

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The Presentation TimeStamp of  the  input  frame,  expressed	 as  a
	   number of time base units. The time base unit depends on the filter
	   input pad.

       pts_time
	   The	Presentation  TimeStamp	 of  the  input	 frame, expressed as a
	   number of seconds.

       fmt The pixel format name.

       sar The sample aspect ratio of the input frame, expressed in  the  form
	   num/den.

       s   The	size  of the input frame. For the syntax of this option, check
	   the "Video size" section in the ffmpeg-utils manual.

       i   The type of interlaced mode ("P" for	 "progressive",	 "T"  for  top
	   field first, "B" for bottom field first).

       iskey
	   This is 1 if the frame is a key frame, 0 otherwise.

       type
	   The	picture type of the input frame ("I" for an I-frame, "P" for a
	   P-frame, "B" for a B-frame, or "?"  for  an	unknown	 type).	  Also
	   refer  to  the documentation of the "AVPictureType" enum and of the
	   "av_get_picture_type_char" function defined in libavutil/avutil.h.

       checksum
	   The Adler-32 checksum (printed in hexadecimal) of all the planes of
	   the input frame.

       plane_checksum
	   The Adler-32 checksum (printed in hexadecimal) of each plane of the
	   input frame, expressed in the form "[c0 c1 c2 c3]".

       mean
	   The mean value  of  pixels  in  each	 plane	of  the	 input	frame,
	   expressed in the form "[mean0 mean1 mean2 mean3]".

       stdev
	   The	standard  deviation of pixel values in each plane of the input
	   frame, expressed in the form "[stdev0 stdev1 stdev2 stdev3]".

   showpalette
       Displays the 256 colors palette of each	frame.	This  filter  is  only
       relevant for pal8 pixel format frames.

       It accepts the following option:

       s   Set	the size of the box used to represent one palette color entry.
	   Default is 30 (for a "30x30" pixel box).

   shuffleframes
       Reorder and/or duplicate and/or drop video frames.

       It accepts the following parameters:

       mapping
	   Set the destination indexes of input frames.	 This is space or  '|'
	   separated  list of indexes that maps input frames to output frames.
	   Number of indexes also sets maximal value that each index may have.
	   '-1' index have special meaning and that is to drop frame.

       The first frame has the index 0. The  default  is  to  keep  the	 input
       unchanged.

       Examples

       •   Swap second and third frame of every three frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=0 2 1" OUTPUT

       •   Swap 10th and 1st frame of every ten frames of the input:

		   ffmpeg -i INPUT -vf "shuffleframes=9 1 2 3 4 5 6 7 8 0" OUTPUT

   shufflepixels
       Reorder pixels in video frames.

       This filter accepts the following options:

       direction, d
	   Set	shuffle	 direction.  Can  be  forward  or  inverse  direction.
	   Default direction is forward.

       mode, m
	   Set shuffle mode. Can be horizontal, vertical or block mode.

       width, w
       height, h
	   Set shuffle block_size. In case of  horizontal  shuffle  mode  only
	   width  part	of  size is used, and in case of vertical shuffle mode
	   only height part of size is used.

       seed, s
	   Set random seed used with shuffling pixels. Mainly useful to set to
	   be able to reverse filtering process to get	original  input.   For
	   example, to reverse forward shuffle you need to use same parameters
	   and exact same seed and to set direction to inverse.

   shuffleplanes
       Reorder and/or duplicate video planes.

       It accepts the following parameters:

       map0
	   The index of the input plane to be used as the first output plane.

       map1
	   The index of the input plane to be used as the second output plane.

       map2
	   The index of the input plane to be used as the third output plane.

       map3
	   The index of the input plane to be used as the fourth output plane.

       The  first  plane  has  the  index  0. The default is to keep the input
       unchanged.

       Examples

       •   Swap the second and third planes of the input:

		   ffmpeg -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

   signalstats
       Evaluate various visual	metrics	 that  assist  in  determining	issues
       associated with the digitization of analog video media.

       By default the filter will log these metadata values:

       YMIN
	   Display  the	 minimal  Y  value  contained  within the input frame.
	   Expressed in range of [0-255].

       YLOW
	   Display the Y value at the 10% percentile within the	 input	frame.
	   Expressed in range of [0-255].

       YAVG
	   Display  the	 average  Y value within the input frame. Expressed in
	   range of [0-255].

       YHIGH
	   Display the Y value at the 90% percentile within the	 input	frame.
	   Expressed in range of [0-255].

       YMAX
	   Display  the	 maximum  Y  value  contained  within the input frame.
	   Expressed in range of [0-255].

       UMIN
	   Display the minimal U  value	 contained  within  the	 input	frame.
	   Expressed in range of [0-255].

       ULOW
	   Display  the	 U value at the 10% percentile within the input frame.
	   Expressed in range of [0-255].

       UAVG
	   Display the average U value within the input	 frame.	 Expressed  in
	   range of [0-255].

       UHIGH
	   Display  the	 U value at the 90% percentile within the input frame.
	   Expressed in range of [0-255].

       UMAX
	   Display the maximum U  value	 contained  within  the	 input	frame.
	   Expressed in range of [0-255].

       VMIN
	   Display  the	 minimal  V  value  contained  within the input frame.
	   Expressed in range of [0-255].

       VLOW
	   Display the V value at the 10% percentile within the	 input	frame.
	   Expressed in range of [0-255].

       VAVG
	   Display  the	 average  V value within the input frame. Expressed in
	   range of [0-255].

       VHIGH
	   Display the V value at the 90% percentile within the	 input	frame.
	   Expressed in range of [0-255].

       VMAX
	   Display  the	 maximum  V  value  contained  within the input frame.
	   Expressed in range of [0-255].

       SATMIN
	   Display the minimal saturation value	 contained  within  the	 input
	   frame.  Expressed in range of [0-~181.02].

       SATLOW
	   Display the saturation value at the 10% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATAVG
	   Display  the	 average  saturation  value  within  the  input frame.
	   Expressed in range of [0-~181.02].

       SATHIGH
	   Display the saturation value at the 90% percentile within the input
	   frame.  Expressed in range of [0-~181.02].

       SATMAX
	   Display the maximum saturation value	 contained  within  the	 input
	   frame.  Expressed in range of [0-~181.02].

       HUEMED
	   Display  the median value for hue within the input frame. Expressed
	   in range of [0-360].

       HUEAVG
	   Display the average value for hue within the input frame. Expressed
	   in range of [0-360].

       YDIF
	   Display the average of sample value difference between  all	values
	   of the Y plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       UDIF
	   Display  the	 average of sample value difference between all values
	   of the U plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       VDIF
	   Display the average of sample value difference between  all	values
	   of the V plane in the current frame and corresponding values of the
	   previous input frame.  Expressed in range of [0-255].

       YBITDEPTH
	   Display  bit depth of Y plane in current frame.  Expressed in range
	   of [0-16].

       UBITDEPTH
	   Display bit depth of U plane in current frame.  Expressed in	 range
	   of [0-16].

       VBITDEPTH
	   Display  bit depth of V plane in current frame.  Expressed in range
	   of [0-16].

       The filter accepts the following options:

       stat
       out stat specify an additional form  of	image  analysis.   out	output
	   video with the specified type of pixel highlighted.

	   Both options accept the following values:

	   tout
	       Identify	 temporal  outliers  pixels.  A	 temporal outlier is a
	       pixel unlike the neighboring pixels of the same field. Examples
	       of temporal outliers include the	 results  of  video  dropouts,
	       head clogs, or tape tracking issues.

	   vrep
	       Identify	 vertical  line	 repetition.  Vertical line repetition
	       includes similar rows of pixels within a frame. In born-digital
	       video vertical line repetition is common, but this  pattern  is
	       uncommon	 in  video  digitized  from  an analog source. When it
	       occurs in video that results from the digitization of an analog
	       source it can indicate concealment from a dropout compensator.

	   brng
	       Identify pixels that fall outside of legal broadcast range.

       color, c
	   Set the highlight color for the out option. The  default  color  is
	   yellow.

       Examples

       •   Output data of various video metrics:

		   ffprobe -f lavfi movie=example.mov,signalstats="stat=tout+vrep+brng" -show_frames

       •   Output  specific data about the minimum and maximum values of the Y
	   plane per frame:

		   ffprobe -f lavfi movie=example.mov,signalstats -show_entries frame_tags=lavfi.signalstats.YMAX,lavfi.signalstats.YMIN

       •   Playback video  while  highlighting	pixels	that  are  outside  of
	   broadcast range in red.

		   ffplay example.mov -vf signalstats="out=brng:color=red"

       •   Playback video with signalstats metadata drawn over the frame.

		   ffplay example.mov -vf signalstats=stat=brng+vrep+tout,drawtext=fontfile=FreeSerif.ttf:textfile=signalstat_drawtext.txt

	   The contents of signalstat_drawtext.txt used in the command are:

		   time %{pts:hms}
		   Y (%{metadata:lavfi.signalstats.YMIN}-%{metadata:lavfi.signalstats.YMAX})
		   U (%{metadata:lavfi.signalstats.UMIN}-%{metadata:lavfi.signalstats.UMAX})
		   V (%{metadata:lavfi.signalstats.VMIN}-%{metadata:lavfi.signalstats.VMAX})
		   saturation maximum: %{metadata:lavfi.signalstats.SATMAX}

   signature
       Calculates  the MPEG-7 Video Signature. The filter can handle more than
       one input. In  this  case  the  matching	 between  the  inputs  can  be
       calculated  additionally.   The	filter always passes through the first
       input. The signature of each stream can be written into a file.

       It accepts the following options:

       detectmode
	   Enable or disable the matching process.

	   Available values are:

	   off Disable the calculation of a matching (default).

	   full
	       Calculate the matching for the whole video and  output  whether
	       the whole video matches or only parts.

	   fast
	       Calculate  only	until  a  matching is found or the video ends.
	       Should be faster in some cases.

       nb_inputs
	   Set the number of inputs. The option value must be a	 non  negative
	   integer.  Default value is 1.

       filename
	   Set	the path to which the output is written. If there is more than
	   one input, the path must be a prototype, i.e. must  contain	%d  or
	   %0nd	 (where	 n  is a positive integer), that will be replaced with
	   the input number. If no filename is specified, no  output  will  be
	   written. This is the default.

       format
	   Choose the output format.

	   Available values are:

	   binary
	       Use the specified binary representation (default).

	   xml Use the specified xml representation.

       th_d
	   Set	threshold to detect one word as similar. The option value must
	   be an integer greater than zero. The default value is 9000.

       th_dc
	   Set threshold to detect all words as similar. The option value must
	   be an integer greater than zero. The default value is 60000.

       th_xh
	   Set threshold to detect frames as similar. The option value must be
	   an integer greater than zero. The default value is 116.

       th_di
	   Set the minimum length of a sequence in frames to recognize	it  as
	   matching  sequence. The option value must be a non negative integer
	   value.  The default value is 0.

       th_it
	   Set the minimum relation, that matching frames to all  frames  must
	   have.  The option value must be a double value between 0 and 1. The
	   default value is 0.5.

       Examples

       •   To  calculate  the  signature  of  an  input	 video and store it in
	   signature.bin:

		   ffmpeg -i input.mkv -vf signature=filename=signature.bin -map 0:v -f null -

       •   To detect whether two videos match and store the signatures in  XML
	   format in signature0.xml and signature1.xml:

		   ffmpeg -i input1.mkv -i input2.mkv -filter_complex "[0:v][1:v] signature=nb_inputs=2:detectmode=full:format=xml:filename=signature%d.xml" -map :v -f null -

   siti
       Calculate Spatial Information (SI) and Temporal Information (TI) scores
       for  a  video, as defined in ITU-T Rec. P.910 (11/21): Subjective video
       quality assessment methods for multimedia applications.	Available  PDF
       at  <https://www.itu.int/rec/T-REC-P.910-202111-S/en>.	Note that this
       is  a  legacy  implementation  that   corresponds   to	a   superseded
       recommendation.	 Refer	to  ITU-T  Rec.	 P.910	(07/22) for the latest
       version: <https://www.itu.int/rec/T-REC-P.910-202207-I/en>

       It accepts the following option:

       print_summary
	   If set to 1, Summary statistics will be  printed  to	 the  console.
	   Default 0.

       Examples

       •   To calculate SI/TI metrics and print summary:

		   ffmpeg -i input.mp4 -vf siti=print_summary=1 -f null -

   smartblur
       Blur the input video without impacting the outlines.

       It accepts the following options:

       luma_radius, lr
	   Set the luma radius. The option value must be a float number in the
	   range  [0.1,5.0] that specifies the variance of the gaussian filter
	   used to blur the image (slower if larger). Default value is 1.0.

       luma_strength, ls
	   Set the luma strength. The option value must be a float  number  in
	   the range [-1.0,1.0] that configures the blurring. A value included
	   in  [0.0,1.0]  will	blur  the  image  whereas  a value included in
	   [-1.0,0.0] will sharpen the image. Default value is 1.0.

       luma_threshold, lt
	   Set the luma threshold used as a coefficient to determine whether a
	   pixel should be blurred or not. The option value must be an integer
	   in the range [-30,30]. A value of 0 will filter all	the  image,  a
	   value  included  in	[0,30]	will  filter  flat  areas  and a value
	   included in [-30,0] will filter edges. Default value is 0.

       chroma_radius, cr
	   Set the chroma radius. The option value must be a float  number  in
	   the	range  [0.1,5.0]  that	specifies the variance of the gaussian
	   filter used to blur the image (slower if larger). Default value  is
	   luma_radius.

       chroma_strength, cs
	   Set the chroma strength. The option value must be a float number in
	   the range [-1.0,1.0] that configures the blurring. A value included
	   in  [0.0,1.0]  will	blur  the  image  whereas  a value included in
	   [-1.0,0.0] will sharpen the image. Default value is luma_strength.

       chroma_threshold, ct
	   Set the chroma threshold used as a coefficient to determine whether
	   a pixel should be blurred or not.  The  option  value  must	be  an
	   integer  in	the  range  [-30,30]. A value of 0 will filter all the
	   image, a value included in [0,30] will  filter  flat	 areas	and  a
	   value  included  in	[-30,0]	 will  filter  edges. Default value is
	   luma_threshold.

       If a chroma option is not explicitly set, the corresponding luma	 value
       is set.

   sobel
       Apply sobel operator to input video stream.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       scale
	   Set value which will be multiplied with filtered result.

       delta
	   Set value which will be added to filtered result.

       Commands

       This filter supports the all above options as commands.

   spp
       Apply a simple postprocessing filter that compresses  and  decompresses
       the image at several (or - in the case of quality level 6 - all) shifts
       and average the results.

       The filter accepts the following options:

       quality
	   Set	 quality.  This	 option	 defines  the  number  of  levels  for
	   averaging. It accepts an integer in the range 0-6. If set to 0, the
	   filter will have no effect. A value of 6 means the higher  quality.
	   For	each  increment	 of  that value the speed drops by a factor of
	   approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If	not  set,  the	filter
	   will use the QP from the video stream (if available).

       mode
	   Set thresholding mode. Available modes are:

	   hard
	       Set hard thresholding (default).

	   soft
	       Set  soft  thresholding	(better	 de-ringing effect, but likely
	       blurrier).

       use_bframe_qp
	   Enable the use of the QP from the B-Frames if set to 1. Using  this
	   option  may	cause flicker since the B-Frames have often larger QP.
	   Default is 0 (not enabled).

       Commands

       This filter supports the following commands:

       quality, level
	   Set quality level. The value "max" can be used to set  the  maximum
	   level, currently 6.

   sr
       Scale  the  input by applying one of the super-resolution methods based
       on convolutional neural networks. Supported models:

       •   Super-Resolution Convolutional Neural Network model	(SRCNN).   See
	   <https://arxiv.org/abs/1501.00092>.

       •   Efficient  Sub-Pixel	 Convolutional	Neural	Network model (ESPCN).
	   See <https://arxiv.org/abs/1609.05158>.

       Training scripts as well as scripts for model file (.pb) saving can  be
       found	 at	<https://github.com/XueweiMeng/sr/tree/sr_dnn_native>.
       Original		       repository		 is		    at
       <https://github.com/HighVoltageRocknRoll/sr.git>.

       The filter accepts the following options:

       dnn_backend
	   Specify  which  DNN backend to use for model loading and execution.
	   This option accepts the following values:

	   tensorflow
	       TensorFlow backend. To enable this backend you need to  install
	       the	 TensorFlow	  for	    C	    library	  (see
	       <https://www.tensorflow.org/install/lang_c>)   and    configure
	       FFmpeg with "--enable-libtensorflow"

       model
	   Set	path  to  model	 file  specifying network architecture and its
	   parameters.	 Note  that  different	backends  use  different  file
	   formats.  TensorFlow,  OpenVINO backend can load files for only its
	   format.

       scale_factor
	   Set scale factor for SRCNN model. Allowed values are 2,  3  and  4.
	   Default  value  is  2.  Scale  factor is necessary for SRCNN model,
	   because it accepts input  upscaled  using  bicubic  upscaling  with
	   proper scale factor.

       To  get	full  functionality  (such as async execution), please use the
       dnn_processing filter.

   ssim
       Obtain the  SSIM	 (Structural  SImilarity  Metric)  between  two	 input
       videos.

       This  filter  takes  in	input  two  input  videos,  the first input is
       considered the "main" source and is passed unchanged to the output. The
       second input is used as a "reference" video for computing the SSIM.

       Both video inputs must have the same resolution and  pixel  format  for
       this  filter  to	 work correctly. Also it assumes that both inputs have
       the same number of frames, which are compared one by one.

       The filter stores the calculated SSIM of each frame.

       The description of the accepted parameters follows.

       stats_file, f
	   If specified the filter will use the named file to save the SSIM of
	   each individual frame. When filename equals "-" the data is sent to
	   standard output.

       The file printed if stats_file is  selected,  contains  a  sequence  of
       key/value  pairs	 of  the  form	key:value  for each compared couple of
       frames.

       A description of each shown parameter follows:

       n   sequential number of the input frame, starting from 1

       Y, U, V, R, G, B
	   SSIM of the compared frames for  the	 component  specified  by  the
	   suffix.

       All SSIM of the compared frames for the whole frame.

       dB  Same as above but in dB representation.

       This filter also supports the framesync options.

       Examples

       •   For example:

		   movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
		   [main][ref] ssim="stats_file=stats.log" [out]

	   On this example the input file being processed is compared with the
	   reference  file ref_movie.mpg. The SSIM of each individual frame is
	   stored in stats.log.

       •   Another example with both psnr and ssim at same time:

		   ffmpeg -i main.mpg -i ref.mpg -lavfi	 "ssim;[0:v][1:v]psnr" -f null -

       •   Another example with different containers:

		   ffmpeg -i main.mpg -i ref.mkv -lavfi	 "[0:v]settb=AVTB,setpts=PTS-STARTPTS[main];[1:v]settb=AVTB,setpts=PTS-STARTPTS[ref];[main][ref]ssim" -f null -

   stereo3d
       Convert between different stereoscopic image formats.

       The filters accept the following options:

       in  Set stereoscopic image format of input.

	   Available values for input image formats are:

	   sbsl
	       side by side parallel (left eye left, right eye right)

	   sbsr
	       side by side crosseye (right eye left, left eye right)

	   sbs2l
	       side by side parallel with  half	 width	resolution  (left  eye
	       left, right eye right)

	   sbs2r
	       side  by	 side  crosseye	 with half width resolution (right eye
	       left, left eye right)

	   abl
	   tbl above-below (left eye above, right eye below)

	   abr
	   tbr above-below (right eye above, left eye below)

	   ab2l
	   tb2l
	       above-below with half height resolution (left eye above,	 right
	       eye below)

	   ab2r
	   tb2r
	       above-below  with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	   irl interleaved rows (left eye has top row,	right  eye  starts  on
	       next row)

	   irr interleaved  rows  (right  eye  has top row, left eye starts on
	       next row)

	   icl interleaved columns, left eye first

	   icr interleaved columns, right eye first

	       Default value is sbsl.

       out Set stereoscopic image format of output.

	   sbsl
	       side by side parallel (left eye left, right eye right)

	   sbsr
	       side by side crosseye (right eye left, left eye right)

	   sbs2l
	       side by side parallel with  half	 width	resolution  (left  eye
	       left, right eye right)

	   sbs2r
	       side  by	 side  crosseye	 with half width resolution (right eye
	       left, left eye right)

	   abl
	   tbl above-below (left eye above, right eye below)

	   abr
	   tbr above-below (right eye above, left eye below)

	   ab2l
	   tb2l
	       above-below with half height resolution (left eye above,	 right
	       eye below)

	   ab2r
	   tb2r
	       above-below  with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	   irl interleaved rows (left eye has top row,	right  eye  starts  on
	       next row)

	   irr interleaved  rows  (right  eye  has top row, left eye starts on
	       next row)

	   arbg
	       anaglyph red/blue gray (red filter on left eye, blue filter  on
	       right eye)

	   argg
	       anaglyph	 red/green  gray (red filter on left eye, green filter
	       on right eye)

	   arcg
	       anaglyph red/cyan gray (red filter on left eye, cyan filter  on
	       right eye)

	   arch
	       anaglyph	 red/cyan  half	 colored (red filter on left eye, cyan
	       filter on right eye)

	   arcc
	       anaglyph red/cyan color (red filter on left eye, cyan filter on
	       right eye)

	   arcd
	       anaglyph	 red/cyan  color  optimized  with  the	least  squares
	       projection  of  dubois  (red filter on left eye, cyan filter on
	       right eye)

	   agmg
	       anaglyph green/magenta gray (green filter on left eye,  magenta
	       filter on right eye)

	   agmh
	       anaglyph	 green/magenta half colored (green filter on left eye,
	       magenta filter on right eye)

	   agmc
	       anaglyph green/magenta  colored	(green	filter	on  left  eye,
	       magenta filter on right eye)

	   agmd
	       anaglyph	 green/magenta	color optimized with the least squares
	       projection of dubois (green filter on left eye, magenta	filter
	       on right eye)

	   aybg
	       anaglyph	 yellow/blue  gray  (yellow  filter  on left eye, blue
	       filter on right eye)

	   aybh
	       anaglyph yellow/blue half colored (yellow filter on  left  eye,
	       blue filter on right eye)

	   aybc
	       anaglyph	 yellow/blue  colored (yellow filter on left eye, blue
	       filter on right eye)

	   aybd
	       anaglyph yellow/blue color optimized  with  the	least  squares
	       projection of dubois (yellow filter on left eye, blue filter on
	       right eye)

	   ml  mono output (left eye only)

	   mr  mono output (right eye only)

	   chl checkerboard, left eye first

	   chr checkerboard, right eye first

	   icl interleaved columns, left eye first

	   icr interleaved columns, right eye first

	   hdmi
	       HDMI frame pack

	   Default value is arcd.

       Examples

       •   Convert  input  video  from	side  by  side	parallel  to  anaglyph
	   yellow/blue dubois:

		   stereo3d=sbsl:aybd

       •   Convert input video from above below (left  eye  above,  right  eye
	   below) to side by side crosseye.

		   stereo3d=abl:sbsr

   streamselect, astreamselect
       Select video or audio streams.

       The filter accepts the following options:

       inputs
	   Set number of inputs. Default is 2.

       map Set input indexes to remap to outputs.

       Commands

       The  "streamselect"  and	 "astreamselect" filter supports the following
       commands:

       map Set input indexes to remap to outputs.

       Examples

       •   Select first 5 seconds 1st stream and rest of time 2nd stream:

		   sendcmd='5.0 streamselect map 1',streamselect=inputs=2:map=0

       •   Same as above, but for audio:

		   asendcmd='5.0 astreamselect map 1',astreamselect=inputs=2:map=0

   subtitles
       Draw subtitles on top of input video using the libass library.

       To enable compilation of this filter you need to configure FFmpeg  with
       "--enable-libass".  This	 filter	 also requires a build with libavcodec
       and libavformat to convert the passed subtitles file to	ASS  (Advanced
       Substation Alpha) subtitles format.

       The filter accepts the following options:

       filename, f
	   Set	the  filename  of  the	subtitle  file	to  read.  It  must be
	   specified.

       original_size
	   Specify the size of the original video, the video for which the ASS
	   file was composed. For the syntax of this option, check the	"Video
	   size"  section  in  the ffmpeg-utils manual.	 Due to a misdesign in
	   ASS aspect ratio arithmetic, this is necessary to  correctly	 scale
	   the fonts if the aspect ratio has been changed.

       fontsdir
	   Set	a  directory  path  containing	fonts  that can be used by the
	   filter.  These fonts will be used in addition to whatever the  font
	   provider uses.

       alpha
	   Process alpha channel, by default alpha channel is untouched.

       charenc
	   Set	subtitles  input  character encoding. "subtitles" filter only.
	   Only useful if not UTF-8.

       stream_index, si
	   Set subtitles stream index. "subtitles" filter only.

       force_style
	   Override default style or script info parameters of the  subtitles.
	   It accepts a string containing ASS style format "KEY=VALUE" couples
	   separated by ",".

       wrap_unicode
	   Break  lines	 according  to	the  Unicode  Line Breaking Algorithm.
	   Availability	 requires  at  least   libass	release	  0.17.0   (or
	   LIBASS_VERSION  0x01600010),	 and  libass must have been built with
	   libunibreak.

	   The option is enabled by default except for native ASS.

       If the first key is not specified, it is assumed that the  first	 value
       specifies the filename.

       For  example, to render the file sub.srt on top of the input video, use
       the command:

	       subtitles=sub.srt

       which is equivalent to:

	       subtitles=filename=sub.srt

       To render the default subtitles stream from file video.mkv, use:

	       subtitles=video.mkv

       To render the second subtitles stream from that file, use:

	       subtitles=video.mkv:si=1

       To make the subtitles stream from sub.srt  appear  in  80%  transparent
       blue "DejaVu Serif", use:

	       subtitles=sub.srt:force_style='Fontname=DejaVu Serif,PrimaryColour=&HCCFF0000'

   super2xsai
       Scale  the  input  by  2x  and  smooth  using the Super2xSaI (Scale and
       Interpolate) pixel art scaling algorithm.

       Useful for enlarging pixel art images without reducing sharpness.

   swaprect
       Swap two rectangular objects in video.

       This filter accepts the following options:

       w   Set object width.

       h   Set object height.

       x1  Set 1st rect x coordinate.

       y1  Set 1st rect y coordinate.

       x2  Set 2nd rect x coordinate.

       y2  Set 2nd rect y coordinate.

	   All expressions are evaluated once for each frame.

       The all options are expressions containing the following constants:

       w
       h   The input width and height.

       a   same as w / h

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (w / h) * sar

       n   The number of the input frame, starting from 0.

       t   The timestamp expressed in seconds. It's NAN if the input timestamp
	   is unknown.

       pos the position in the file  of	 the  input  frame,  NAN  if  unknown;
	   deprecated, do not use

       Commands

       This filter supports the all above options as commands.

   swapuv
       Swap U & V plane.

   tblend
       Blend successive video frames.

       See blend

   telecine
       Apply telecine process to the video.

       This filter accepts the following options:

       first_field
	   top, t
	       top field first

	   bottom, b
	       bottom field first The default value is "top".

       pattern
	   A  string  of numbers representing the pulldown pattern you wish to
	   apply.  The default value is 23.

	       Some typical patterns:

	       NTSC output (30i):
	       27.5p: 32222
	       24p: 23 (classic)
	       24p: 2332 (preferred)
	       20p: 33
	       18p: 334
	       16p: 3444

	       PAL output (25i):
	       27.5p: 12222
	       24p: 222222222223 ("Euro pulldown")
	       16.67p: 33
	       16p: 33333334

   thistogram
       Compute and draw a color distribution histogram	for  the  input	 video
       across time.

       Unlike  histogram  video	 filter	 which	only shows histogram of single
       input frame at certain time, this filter shows also past histograms  of
       number of frames defined by "width" option.

       The  computed  histogram	 is  a	representation	of the color component
       distribution in an image.

       The filter accepts the following options:

       width, w
	   Set width of single color component output.	Default	 value	is  0.
	   Value  of 0 means width will be picked from input video.  This also
	   set number of passed histograms to  keep.   Allowed	range  is  [0,
	   8192].

       display_mode, d
	   Set display mode.  It accepts the following values:

	   stack
	       Per color component graphs are placed below each other.

	   parade
	       Per color component graphs are placed side by side.

	   overlay
	       Presents	 information identical to that in the "parade", except
	       that the graphs representing color components are  superimposed
	       directly over one another.

	   Default is "stack".

       levels_mode, m
	   Set	mode.  Can  be	either "linear", or "logarithmic".  Default is
	   "linear".

       components, c
	   Set what color components to display.  Default is 7.

       bgopacity, b
	   Set background opacity. Default is 0.9.

       envelope, e
	   Show envelope. Default is disabled.

       ecolor, ec
	   Set envelope color. Default is "gold".

       slide
	   Set slide mode.

	   Available values for slide is:

	   frame
	       Draw new frame when right border is reached.

	   replace
	       Replace old columns with new ones.

	   scroll
	       Scroll from right to left.

	   rscroll
	       Scroll from left to right.

	   picture
	       Draw single picture.

	   Default is "replace".

   threshold
       Apply threshold effect to video stream.

       This filter needs four video streams to	perform	 thresholding.	 First
       stream  is stream we are filtering.  Second stream is holding threshold
       values, third stream is holding min values, and last, fourth stream  is
       holding max values.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

       For example if first  stream  pixel's  component	 value	is  less  then
       threshold  value	 of  pixel  component from 2nd threshold stream, third
       stream value will picked, otherwise fourth stream pixel component value
       will be picked.

       Using  color  source  filter  one  can	perform	  various   types   of
       thresholding:

       Commands

       This filter supports the all options as commands.

       Examples

       •   Binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=black -f lavfi -i color=white -lavfi threshold output.avi

       •   Inverted binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -f lavfi -i color=black -lavfi threshold output.avi

       •   Truncate binary threshold, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=gray -lavfi threshold output.avi

       •   Threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -f lavfi -i color=white -i 320x240.avi -lavfi threshold output.avi

       •   Inverted threshold to zero, using gray color as threshold:

		   ffmpeg -i 320x240.avi -f lavfi -i color=gray -i 320x240.avi -f lavfi -i color=white -lavfi threshold output.avi

   thumbnail
       Select the most representative frame in a given sequence of consecutive
       frames.

       The filter accepts the following options:

       n   Set	the  frames  batch  size to analyze; in a set of n frames, the
	   filter will pick one of them, and then handle the next batch	 of  n
	   frames until the end. Default is 100.

       log Set	the  log  level	 to  display  picked  frame stats.  Default is
	   "info".

       Since the filter keeps track of the whole frames sequence, a  bigger  n
       value  will  result  in	a  higher memory usage, so a high value is not
       recommended.

       Examples

       •   Extract one picture each 50 frames:

		   thumbnail=50

       •   Complete example of a thumbnail creation with ffmpeg:

		   ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

   tile
       Tile several successive frames together.

       The untile filter can do the reverse.

       The filter accepts the following options:

       layout
	   Set the grid	 size  in  the	form  "COLUMNSxROWS".  Range  is  upto
	   UINT_MAX cells.  Default is "6x5".

       nb_frames
	   Set	the  maximum  number of frames to render in the given area. It
	   must be less than or equal to wxh. The default value is 0,  meaning
	   all the area will be used.

       margin
	   Set	the outer border margin in pixels. Range is 0 to 1024. Default
	   is 0.

       padding
	   Set the inner border thickness (i.e. the number of  pixels  between
	   frames).   For  more	 advanced  padding  options  (such  as	having
	   different values for the edges), refer to  the  pad	video  filter.
	   Range is 0 to 1024. Default is 0.

       color
	   Specify  the	 color	of  the	 unused	 area.	For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.   The
	   default value of color is "black".

       overlap
	   Set	the number of frames to overlap when tiling several successive
	   frames together.  The value must be between 0 and  nb_frames	 -  1.
	   Default is 0.

       init_padding
	   Set	the  number  of frames to initially be empty before displaying
	   first output frame.	This controls how  soon	 will  one  get	 first
	   output  frame.   The	 value	must  be  between 0 and nb_frames - 1.
	   Default is 0.

       Examples

       •   Produce 8x8 PNG tiles of all keyframes  (-skip_frame	 nokey)	 in  a
	   movie:

		   ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png

	   The	-vsync	0 is necessary to prevent ffmpeg from duplicating each
	   output frame to accommodate the originally detected frame rate.

       •   Display 5 pictures in an  area  of  "3x2"  frames,  with  7	pixels
	   between  them, and 2 pixels of initial margin, using mixed flat and
	   named options:

		   tile=3x2:nb_frames=5:padding=7:margin=2

   tinterlace
       Perform various types of temporal field interlacing.

       Frames are counted starting  from  1,  so  the  first  input  frame  is
       considered odd.

       The filter accepts the following options:

       mode
	   Specify  the	 mode  of  the	interlacing.  This  option can also be
	   specified as a value alone. See below for a list of values for this
	   option.

	   Available values are:

	   merge, 0
	       Move odd frames into the	 upper	field,	even  into  the	 lower
	       field, generating a double height frame at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444

	   drop_even, 1
	       Only  output  odd frames, even frames are dropped, generating a
	       frame with unchanged height at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111			       33333
		       11111			       33333
		       11111			       33333
		       11111			       33333

	   drop_odd, 2
	       Only output even frames, odd frames are dropped,	 generating  a
	       frame with unchanged height at half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
				       22222			       44444
				       22222			       44444
				       22222			       44444
				       22222			       44444

	   pad, 3
	       Expand  each frame to full height, but pad alternate lines with
	       black, generating a frame with double height at the same	 input
	       frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444
		       11111	       .....	       33333	       .....
		       .....	       22222	       .....	       44444

	   interleave_top, 4
	       Interleave the upper field from odd frames with the lower field
	       from  even  frames, generating a frame with unchanged height at
	       half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-

		       Output:
		       11111			       33333
		       22222			       44444
		       11111			       33333
		       22222			       44444

	   interleave_bottom, 5
	       Interleave the lower field from odd frames with the upper field
	       from even frames, generating a frame with unchanged  height  at
	       half frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444
		       11111	       22222<-	       33333	       44444<-
		       11111<-	       22222	       33333<-	       44444

		       Output:
		       22222			       44444
		       11111			       33333
		       22222			       44444
		       11111			       33333

	   interlacex2, 6
	       Double  frame  rate  with unchanged height. Frames are inserted
	       each containing the second temporal  field  from	 the  previous
	       input  frame  and  the first temporal field from the next input
	       frame. This mode relies on the top_field_first flag. Useful for
	       interlaced video displays with no field synchronisation.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
			11111		22222		33333		44444
		       11111	       22222	       33333	       44444
			11111		22222		33333		44444

		       Output:
		       11111   22222   22222   33333   33333   44444   44444
			11111	11111	22222	22222	33333	33333	44444
		       11111   22222   22222   33333   33333   44444   44444
			11111	11111	22222	22222	33333	33333	44444

	   mergex2, 7
	       Move odd frames into the	 upper	field,	even  into  the	 lower
	       field, generating a double height frame at same frame rate.

			------> time
		       Input:
		       Frame 1	       Frame 2	       Frame 3	       Frame 4

		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444
		       11111	       22222	       33333	       44444

		       Output:
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444
		       11111	       33333	       33333	       55555
		       22222	       22222	       44444	       44444

	   Numeric  values  are	 deprecated  but  are  accepted	 for  backward
	   compatibility reasons.

	   Default mode is "merge".

       flags
	   Specify flags influencing the filter process.

	   Available value for flags is:

	   low_pass_filter, vlpf
	       Enable  linear  vertical	 low-pass  filtering  in  the  filter.
	       Vertical	 low-pass  filtering  is  required  when  creating  an
	       interlaced destination from a progressive source which contains
	       high-frequency vertical detail. Filtering will reduce interlace
	       'twitter' and Moire patterning.

	   complex_filter, cvlpf
	       Enable complex vertical low-pass filtering.  This will slightly
	       less reduce interlace 'twitter' and Moire patterning but better
	       retain detail and subjective sharpness impression.

	   bypass_il
	       Bypass already interlaced frames, only adjust the frame rate.

	   Vertical low-pass filtering and bypassing already interlaced frames
	   can only be enabled for mode interleave_top and interleave_bottom.

   tmedian
       Pick median pixels from several successive input video frames.

       The filter accepts the following options:

       radius
	   Set radius of median filter.	 Default is 1. Allowed range is from 1
	   to 127.

       planes
	   Set which planes to filter. Default	value  is  15,	by  which  all
	   planes are processed.

       percentile
	   Set	median percentile. Default value is 0.5.  Default value of 0.5
	   will pick always median values, while 0 will pick  minimum  values,
	   and 1 maximum values.

       Commands

       This  filter  supports  all above options as commands, excluding option
       "radius".

   tmidequalizer
       Apply Temporal Midway Video Equalization effect.

       Midway Video Equalization adjusts a sequence of video  frames  to  have
       the  same  histograms,  while  maintaining  their  dynamics  as much as
       possible. It's useful for e.g. matching exposures from a	 video	frames
       sequence.

       This filter accepts the following option:

       radius
	   Set filtering radius. Default is 5. Allowed range is from 1 to 127.

       sigma
	   Set	filtering  sigma.  Default  is	0.5. This controls strength of
	   filtering.  Setting this option to 0 effectively does nothing.

       planes
	   Set which planes to process. Default is 15, which is all  available
	   planes.

   tmix
       Mix successive video frames.

       A description of the accepted options follows.

       frames
	   The number of successive frames to mix. If unspecified, it defaults
	   to 3.

       weights
	   Specify weight of each input video frame.  Each weight is separated
	   by  space.  If  number  of weights is smaller than number of frames
	   last specified weight will be used for all remaining unset weights.

       scale
	   Specify scale, if it is set it will be multiplied with sum of  each
	   weight multiplied with pixel values to give final destination pixel
	   value. By default scale is auto scaled to sum of weights.

       planes
	   Set which planes to filter. Default is all. Allowed range is from 0
	   to 15.

       Examples

       •   Average 7 successive frames:

		   tmix=frames=7:weights="1 1 1 1 1 1 1"

       •   Apply simple temporal convolution:

		   tmix=frames=3:weights="-1 3 -1"

       •   Similar as above but only showing temporal differences:

		   tmix=frames=3:weights="-1 2 -1":scale=1

       Commands

       This filter supports the following commands:

       weights
       scale
       planes
	   Syntax is same as option with same name.

   tonemap
       Tone map colors from different dynamic ranges.

       This  filter  expects  data  in	single precision floating point, as it
       needs to operate on  (and  can  output)	out-of-range  values.  Another
       filter,	such  as zscale, is needed to convert the resulting frame to a
       usable format.

       The tonemapping algorithms implemented only work on  linear  light,  so
       input  data  should  be	linearized  beforehand (and possibly correctly
       tagged).

	       ffmpeg -i INPUT -vf zscale=transfer=linear,tonemap=clip,zscale=transfer=bt709,format=yuv420p OUTPUT

       Options

       The filter accepts the following options.

       tonemap
	   Set the tone map algorithm to use.

	   Possible values are:

	   none
	       Do not apply any tone map, only desaturate overbright pixels.

	   clip
	       Hard-clip any out-of-range values. Use  it  for	perfect	 color
	       accuracy	 for  in-range	values,	 while distorting out-of-range
	       values.

	   linear
	       Stretch the entire reference gamut to a linear multiple of  the
	       display.

	   gamma
	       Fit a logarithmic transfer between the tone curves.

	   reinhard
	       Preserve	 overall  image	 brightness with a simple curve, using
	       nonlinear contrast, which results  in  flattening  details  and
	       degrading color accuracy.

	   hable
	       Preserve	 both dark and bright details better than reinhard, at
	       the cost of slightly darkening everything. Use it  when	detail
	       preservation  is	 more  important  than	color  and  brightness
	       accuracy.

	   mobius
	       Smoothly map out-of-range values, while retaining contrast  and
	       colors  for  in-range material as much as possible. Use it when
	       color accuracy is more important than detail preservation.

	   Default is none.

       param
	   Tune the tone mapping algorithm.

	   This affects the following algorithms:

	   none
	       Ignored.

	   linear
	       Specifies the scale factor to use while stretching.  Default to
	       1.0.

	   gamma
	       Specifies the exponent of the function.	Default to 1.8.

	   clip
	       Specify an extra linear coefficient to multiply into the signal
	       before clipping.	 Default to 1.0.

	   reinhard
	       Specify the local contrast coefficient  at  the	display	 peak.
	       Default	to 0.5, which means that in-gamut values will be about
	       half as bright as when clipping.

	   hable
	       Ignored.

	   mobius
	       Specify the transition point from linear to  mobius  transform.
	       Every  value  below  this point is guaranteed to be mapped 1:1.
	       The higher the value, the more accurate the result will be,  at
	       the  cost  of losing bright details.  Default to 0.3, which due
	       to the steep initial  slope  still  preserves  in-range	colors
	       fairly accurately.

       desat
	   Apply  desaturation	for  highlights	 that  exceed  this  level  of
	   brightness. The higher the parameter, the  more  color  information
	   will be preserved. This setting helps prevent unnaturally blown-out
	   colors  for	super-highlights,  by  (smoothly)  turning  into white
	   instead. This makes images  feel  more  natural,  at	 the  cost  of
	   reducing information about out-of-range colors.

	   The	default	 of  2.0 is somewhat conservative and will mostly just
	   apply to skies or  directly	sunlit	surfaces.  A  setting  of  0.0
	   disables this option.

	   This	 option	 works	only  if the input frame has a supported color
	   tag.

       peak
	   Override signal/nominal/reference peak with this value. Useful when
	   the embedded peak information in display metadata is	 not  reliable
	   or when tone mapping from a lower range to a higher range.

   tpad
       Temporarily pad video frames.

       The filter accepts the following options:

       start
	   Specify  number  of delay frames before input video stream. Default
	   is 0.

       stop
	   Specify number of padding frames after input video stream.  Set  to
	   -1 to pad indefinitely. Default is 0.

       start_mode
	   Set kind of frames added to beginning of stream.  Can be either add
	   or  clone.	With  add frames of solid-color are added.  With clone
	   frames are clones of first frame.  Default is add.

       stop_mode
	   Set kind of frames added to end of stream.  Can be  either  add  or
	   clone.   With  add  frames  of  solid-color	are added.  With clone
	   frames are clones of last frame.  Default is add.

       start_duration, stop_duration
	   Specify the duration of the start/stop delay. See the Time duration
	   section in the ffmpeg-utils(1)  manual  for	the  accepted  syntax.
	   These options override start and stop. Default is 0.

       color
	   Specify  the	 color	of  the	 padded	 area.	For the syntax of this
	   option, check the "Color" section in the ffmpeg-utils manual.

	   The default value of color is "black".

   transpose
       Transpose rows with columns in the input video and optionally flip it.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   0, 4, cclock_flip
	       Rotate by  90  degrees  counterclockwise	 and  vertically  flip
	       (default), that is:

		       L.R     L.l
		       . . ->  . .
		       l.r     R.r

	   1, 5, clock
	       Rotate by 90 degrees clockwise, that is:

		       L.R     l.L
		       . . ->  . .
		       l.r     r.R

	   2, 6, cclock
	       Rotate by 90 degrees counterclockwise, that is:

		       L.R     R.r
		       . . ->  . .
		       l.r     L.l

	   3, 7, clock_flip
	       Rotate by 90 degrees clockwise and vertically flip, that is:

		       L.R     r.R
		       . . ->  . .
		       l.r     l.L

	   For values between 4-7, the transposition is only done if the input
	   video  geometry  is	portrait  and  not landscape. These values are
	   deprecated, the "passthrough" option should be used instead.

	   Numerical values are deprecated, and should be dropped in favor  of
	   symbolic constants.

       passthrough
	   Do  not  apply  the transposition if the input geometry matches the
	   one specified by the specified  value.  It  accepts	the  following
	   values:

	   none
	       Always apply transposition.

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

	   Default value is "none".

       For  example  to	 rotate	 by 90 degrees clockwise and preserve portrait
       layout:

	       transpose=dir=1:passthrough=portrait

       The command above can also be specified as:

	       transpose=1:portrait

   transpose_npp
       Transpose rows with columns in the input video and optionally flip  it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate  by  90  degrees	counterclockwise  and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

       passthrough
	   Do not apply the transposition if the input	geometry  matches  the
	   one	specified  by  the  specified  value. It accepts the following
	   values:

	   none
	       Always apply transposition. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

   trim
       Trim the input so that the output contains one  continuous  subpart  of
       the input.

       It accepts the following parameters:

       start
	   Specify  the	 time of the start of the kept section, i.e. the frame
	   with the timestamp start will be the first frame in the output.

       end Specify the time of the first frame that will be dropped, i.e.  the
	   frame  immediately preceding the one with the timestamp end will be
	   the last frame in the output.

       start_pts
	   This is the same as	start,	except	this  option  sets  the	 start
	   timestamp in timebase units instead of seconds.

       end_pts
	   This	 is the same as end, except this option sets the end timestamp
	   in timebase units instead of seconds.

       duration
	   The maximum duration of the output in seconds.

       start_frame
	   The number of the first frame that should be passed to the output.

       end_frame
	   The number of the first frame that should be dropped.

       start, end, and duration are expressed as time duration specifications;
       see the Time duration section in the  ffmpeg-utils(1)  manual  for  the
       accepted syntax.

       Note  that the first two sets of the start/end options and the duration
       option look at the frame timestamp, while the  _frame  variants	simply
       count  the  frames  that	 pass  through the filter. Also note that this
       filter does not modify the timestamps.  If  you	wish  for  the	output
       timestamps  to  start  at  zero,	 insert a setpts filter after the trim
       filter.

       If multiple start or end options are  set,  this	 filter	 tries	to  be
       greedy and keep all the frames that match at least one of the specified
       constraints.  To keep only the part that matches all the constraints at
       once, chain multiple trim filters.

       The defaults are such that all the input is kept. So it is possible  to
       set  e.g.   just the end values to keep everything before the specified
       time.

       Examples:

       •   Drop everything except the second minute of input:

		   ffmpeg -i INPUT -vf trim=60:120

       •   Keep only the first second:

		   ffmpeg -i INPUT -vf trim=duration=1

   unpremultiply
       Apply alpha unpremultiply effect to  input  video  stream  using	 first
       plane of second stream as alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
	   Set	which  planes  will  be	 processed, unprocessed planes will be
	   copied.  By default value 0xf, all planes will be processed.

	   If the format has 1 or 2 components, then luma is bit  0.   If  the
	   format has 3 or 4 components: for RGB formats bit 0 is green, bit 1
	   is  blue  and bit 2 is red; for YUV formats bit 0 is luma, bit 1 is
	   chroma-U and bit 2 is chroma-V.  If present, the alpha  channel  is
	   always the last bit.

       inplace
	   Do  not  require  2nd input for processing, instead use alpha plane
	   from input stream.

   unsharp
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
	   Set the luma matrix horizontal size. It  must  be  an  odd  integer
	   between 3 and 23. The default value is 5.

       luma_msize_y, ly
	   Set	the  luma  matrix  vertical  size.  It	must be an odd integer
	   between 3 and 23. The default value is 5.

       luma_amount, la
	   Set the luma effect strength. It must be a floating	point  number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative  values  will  blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 1.0.

       chroma_msize_x, cx
	   Set the chroma matrix horizontal size. It must be  an  odd  integer
	   between 3 and 23. The default value is 5.

       chroma_msize_y, cy
	   Set	the  chroma  matrix  vertical  size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video,  while  positive	values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 0.0.

       alpha_msize_x, ax
	   Set	the  alpha  matrix  horizontal size. It must be an odd integer
	   between 3 and 23. The default value is 5.

       alpha_msize_y, ay
	   Set the alpha matrix vertical size.	It  must  be  an  odd  integer
	   between 3 and 23. The default value is 5.

       alpha_amount, aa
	   Set	the alpha effect strength. It must be a floating point number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video,  while  positive	values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 0.0.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       •   Apply strong luma sharpen effect:

		   unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

       •   Apply a strong blur of both luma and chroma parameters:

		   unsharp=7:7:-2:7:7:-2

   untile
       Decompose a video made of tiled images into the individual images.

       The frame rate of the output video is the frame rate of the input video
       multiplied by the number of tiles.

       This filter does the reverse of tile.

       The filter accepts the following options:

       layout
	   Set	the  grid size (i.e. the number of lines and columns). For the
	   syntax of this option,  check  the  "Video  size"  section  in  the
	   ffmpeg-utils manual.

       Examples

       •   Produce  a 1-second video from a still image file made of 25 frames
	   stacked vertically, like an analogic film reel:

		   ffmpeg -r 1 -i image.jpg -vf untile=1x25 movie.mkv

   uspp
       Apply ultra  slow/simple	 postprocessing	 filter	 that  compresses  and
       decompresses  the image at several (or - in the case of quality level 8
       - all) shifts and average the results.

       The way this differs from the behavior of spp  is  that	uspp  actually
       encodes	&  decodes  each case with libavcodec Snow, whereas spp uses a
       simplified intra only 8x8 DCT similar to MJPEG.

       This filter is only available in ffmpeg version 4.4 or earlier.

       The filter accepts the following options:

       quality
	   Set	quality.  This	option	defines	 the  number  of  levels   for
	   averaging. It accepts an integer in the range 0-8. If set to 0, the
	   filter  will have no effect. A value of 8 means the higher quality.
	   For each increment of that value the speed drops  by	 a  factor  of
	   approximately 2.  Default value is 3.

       qp  Force  a  constant  quantization  parameter. If not set, the filter
	   will use the QP from the video stream (if available).

       codec
	   Use specified codec instead of snow.

   v360
       Convert 360 videos between various formats.

       The filter accepts the following options:

       input
       output
	   Set format of the input/output video.

	   Available formats:

	   e
	   equirect
	       Equirectangular projection.

	   c3x2
	   c6x1
	   c1x6
	       Cubemap with 3x2/6x1/1x6 layout.

	       Format specific options:

	       in_pad
	       out_pad
		   Set padding proportion for the input/output cubemap. Values
		   in decimals.

		   Example values:

		   0   No padding.

		   0.01
		       1% of face is  padding.	For  example,  with  1920x1280
		       resolution face size would be 640x640 and padding would
		       be 3 pixels from each side. (640 * 0.01 = 6 pixels)

		   Default value is @samp{0}.  Maximum value is @samp{0.1}.

	       fin_pad
	       fout_pad
		   Set	fixed  padding for the input/output cubemap. Values in
		   pixels.

		   Default  value  is  @samp{0}.  If  greater  than  zero   it
		   overrides other padding options.

	       in_forder
	       out_forder
		   Set order of faces for the input/output cubemap. Choose one
		   direction for each position.

		   Designation of directions:

		   r   right

		   l   left

		   u   up

		   d   down

		   f   forward

		   b   back

		   Default value is @samp{rludfb}.

	       in_frot
	       out_frot
		   Set	rotation of faces for the input/output cubemap. Choose
		   one angle for each position.

		   Designation of angles:

		   0   0 degrees clockwise

		   1   90 degrees clockwise

		   2   180 degrees clockwise

		   3   270 degrees clockwise

		   Default value is @samp{000000}.

	   eac Equi-Angular Cubemap.

	   flat
	   gnomonic
	   rectilinear
	       Regular video.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   dfisheye
	       Dual fisheye.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   barrel
	   fb
	   barrelsplit
	       Facebook's 360 formats.

	   sg  Stereographic format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   mercator
	       Mercator format.

	   ball
	       Ball format, gives significant distortion toward the back.

	   hammer
	       Hammer-Aitoff map projection format.

	   sinusoidal
	       Sinusoidal map projection format.

	   fisheye
	       Fisheye projection.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   pannini
	       Pannini projection.

	       Format specific options:

	       h_fov
		   Set output pannini parameter.

	       ih_fov
		   Set input pannini parameter.

	   cylindrical
	       Cylindrical projection.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   perspective
	       Perspective projection. (output only)

	       Format specific options:

	       v_fov
		   Set perspective parameter.

	   tetrahedron
	       Tetrahedron projection.

	   tsp Truncated square pyramid projection.

	   he
	   hequirect
	       Half equirectangular projection.

	   equisolid
	       Equisolid format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   og  Orthographic format.

	       Format specific options:

	       h_fov
	       v_fov
	       d_fov
		   Set	output	horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	       ih_fov
	       iv_fov
	       id_fov
		   Set	input  horizontal/vertical/diagonal  field  of	 view.
		   Values in degrees.

		   If  diagonal	 field	of view is set it overrides horizontal
		   and vertical field of view.

	   octahedron
	       Octahedron projection.

	   cylindricalea
	       Cylindrical Equal Area projection.

       interp
	   Set interpolation method.Note: more complex	interpolation  methods
	   require much more memory to run.

	   Available methods:

	   near
	   nearest
	       Nearest neighbour.

	   line
	   linear
	       Bilinear interpolation.

	   lagrange9
	       Lagrange9 interpolation.

	   cube
	   cubic
	       Bicubic interpolation.

	   lanc
	   lanczos
	       Lanczos interpolation.

	   sp16
	   spline16
	       Spline16 interpolation.

	   gauss
	   gaussian
	       Gaussian interpolation.

	   mitchell
	       Mitchell interpolation.

	   Default value is @samp{line}.

       w
       h   Set the output video resolution.

	   Default resolution depends on formats.

       in_stereo
       out_stereo
	   Set the input/output stereo format.

	   2d  2D mono

	   sbs Side by side

	   tb  Top bottom

	   Default value is @samp{2d} for input and output format.

       yaw
       pitch
       roll
	   Set rotation for the output video. Values in degrees.

       rorder
	   Set	rotation  order for the output video. Choose one item for each
	   position.

	   y, Y
	       yaw

	   p, P
	       pitch

	   r, R
	       roll

	   Default value is @samp{ypr}.

       h_flip
       v_flip
       d_flip
	   Flip	      the	 output	       video	    horizontally(swaps
	   left-right)/vertically(swaps up-down)/in-depth(swaps back-forward).
	   Boolean values.

       ih_flip
       iv_flip
	   Set	if  input  video  is  flipped horizontally/vertically. Boolean
	   values.

       in_trans
	   Set if  input  video	 is  transposed.  Boolean  value,  by  default
	   disabled.

       out_trans
	   Set	if  output  video  needs  to  be transposed. Boolean value, by
	   default disabled.

       h_offset
       v_offset
	   Set output horizontal/vertical off-axis offset. Default is  set  to
	   0.  Allowed range is from -1 to 1.

       alpha_mask
	   Build  mask	in alpha plane for all unmapped pixels by marking them
	   fully transparent. Boolean value, by default disabled.

       reset_rot
	   Reset rotation of output video. Boolean value, by default disabled.

       Examples

       •   Convert equirectangular video to cubemap with  3x2  layout  and  1%
	   padding using bicubic interpolation:

		   ffmpeg -i input.mkv -vf v360=e:c3x2:cubic:out_pad=0.01 output.mkv

       •   Extract back view of Equi-Angular Cubemap:

		   ffmpeg -i input.mkv -vf v360=eac:flat:yaw=180 output.mkv

       •   Convert transposed and horizontally flipped Equi-Angular Cubemap in
	   side-by-side	 stereo	 format	 to  equirectangular top-bottom stereo
	   format:

		   v360=eac:equirect:in_stereo=sbs:in_trans=1:ih_flip=1:out_stereo=tb

       Commands

       This filter supports subset of above options as commands.

   vaguedenoiser
       Apply a wavelet based denoiser.

       It transforms each frame from the video input into the wavelet  domain,
       using  Cohen-Daubechies-Feauveau 9/7. Then it applies some filtering to
       the obtained coefficients. It does an inverse wavelet transform	after.
       Due  to	wavelet properties, it should give a nice smoothed result, and
       reduced noise, without blurring picture features.

       This filter accepts the following options:

       threshold
	   The filtering strength. The higher, the  more  filtered  the	 video
	   will	 be.   Hard  thresholding can use a higher threshold than soft
	   thresholding before the video looks overfiltered. Default value  is
	   2.

       method
	   The filtering method the filter will use.

	   It accepts the following values:

	   hard
	       All values under the threshold will be zeroed.

	   soft
	       All values under the threshold will be zeroed. All values above
	       will be reduced by the threshold.

	   garrote
	       Scales  or nullifies coefficients - intermediary between (more)
	       soft and (less) hard thresholding.

	   Default is garrote.

       nsteps
	   Number of times, the wavelet will decompose	the  picture.  Picture
	   can't  be  decomposed beyond a particular point (typically, 8 for a
	   640x480 frame - as 2^9 = 512 >  480).  Valid	 values	 are  integers
	   between 1 and 32. Default value is 6.

       percent
	   Partial  of full denoising (limited coefficients shrinking), from 0
	   to 100. Default value is 85.

       planes
	   A list of  the  planes  to  process.	 By  default  all  planes  are
	   processed.

       type
	   The threshold type the filter will use.

	   It accepts the following values:

	   universal
	       Threshold used is same for all decompositions.

	   bayes
	       Threshold used depends also on each decomposition coefficients.

	   Default is universal.

   varblur
       Apply  variable	blur  filter  by  using	 2nd  video stream to set blur
       radius.	The 2nd stream must have the same dimensions.

       This filter accepts the following options:

       min_r
	   Set min allowed radius. Allowed range is from 0 to 254. Default  is
	   0.

       max_r
	   Set	max allowed radius. Allowed range is from 1 to 255. Default is
	   8.

       planes
	   Set which planes to process. By default, all are used.

       The "varblur" filter also supports the framesync options.

       Commands

       This filter supports all the above options as commands.

   vectorscope
       Display 2 color component values in the two dimensional graph (which is
       called a vectorscope).

       This filter accepts the following options:

       mode, m
	   Set vectorscope mode.

	   It accepts the following values:

	   gray
	   tint
	       Gray values are displayed on  graph,  higher  brightness	 means
	       more  pixels  have  same	 component  color value on location in
	       graph. This is the default mode.

	   color
	       Gray values are displayed on graph. Surrounding	pixels	values
	       which are not present in video frame are drawn in gradient of 2
	       color  components  which are set by option "x" and "y". The 3rd
	       color component is static.

	   color2
	       Actual color components	values	present	 in  video  frame  are
	       displayed on graph.

	   color3
	       Similar	as  color2 but higher frequency of same values "x" and
	       "y" on graph increases value of another color component,	 which
	       is luminance by default values of "x" and "y".

	   color4
	       Actual colors present in video frame are displayed on graph. If
	       two  different  colors map to same position on graph then color
	       with higher value of component not present in graph is picked.

	   color5
	       Gray values are displayed on graph. Similar to "color" but with
	       3rd color component picked from radial gradient.

       x   Set which color component will be represented on X-axis. Default is
	   1.

       y   Set which color component will be represented on Y-axis. Default is
	   2.

       intensity, i
	   Set intensity, used by modes: gray, color, color3  and  color5  for
	   increasing brightness of color component which represents frequency
	   of (X, Y) location in graph.

       envelope, e
	   none
	       No envelope, this is default.

	   instant
	       Instant	envelope,  even	 darkest  single pixel will be clearly
	       highlighted.

	   peak
	       Hold maximum and minimum values presented in graph  over	 time.
	       This  way  you  can  still  spot	 out  of  range values without
	       constantly looking at vectorscope.

	   peak+instant
	       Peak and instant envelope combined together.

       graticule, g
	   Set what kind of graticule to draw.

	   none
	   green
	   color
	   invert
       opacity, o
	   Set graticule opacity.

       flags, f
	   Set graticule flags.

	   white
	       Draw graticule for white point.

	   black
	       Draw graticule for black point.

	   name
	       Draw color points short names.

       bgopacity, b
	   Set background opacity.

       lthreshold, l
	   Set low threshold for color component not represented  on  X	 or  Y
	   axis.   Values lower than this value will be ignored. Default is 0.
	   Note this value is multiplied with actual max  possible  value  one
	   pixel  component  can  have.	 So  for 8-bit input and low threshold
	   value of 0.1 actual threshold is 0.1 * 255 = 25.

       hthreshold, h
	   Set high threshold for color component not represented on  X	 or  Y
	   axis.  Values higher than this value will be ignored. Default is 1.
	   Note	 this  value  is multiplied with actual max possible value one
	   pixel component can have. So for 8-bit  input  and  high  threshold
	   value of 0.9 actual threshold is 0.9 * 255 = 230.

       colorspace, c
	   Set what kind of colorspace to use when drawing graticule.

	   auto
	   601
	   709

	   Default is auto.

       tint0, t0
       tint1, t1
	   Set	color  tint  for  gray/tint  vectorscope mode. By default both
	   options are zero.  This means no tint, and output will remain gray.

   vidstabdetect
       Analyze	video  stabilization/deshaking.	 Perform  pass	1  of  2,  see
       vidstabtransform for pass 2.

       This  filter  generates	a  file with relative translation and rotation
       transform information about subsequent frames, which is	then  used  by
       the vidstabtransform filter.

       To  enable compilation of this filter you need to configure FFmpeg with
       "--enable-libvidstab".

       This filter accepts the following options:

       result
	   Set the path to the file used to write the transforms  information.
	   Default value is transforms.trf.

       shakiness
	   Set	how shaky the video is and how quick the camera is. It accepts
	   an integer in the range 1-10, a value of 1 means little  shakiness,
	   a value of 10 means strong shakiness. Default value is 5.

       accuracy
	   Set	the  accuracy  of the detection process. It must be a value in
	   the range 1-15. A value of 1 means low  accuracy,  a	 value	of  15
	   means high accuracy. Default value is 15.

       stepsize
	   Set	stepsize  of  the search process. The region around minimum is
	   scanned with 1 pixel resolution. Default value is 6.

       mincontrast
	   Set minimum contrast. Below this value a local measurement field is
	   discarded. Must be a floating point value in the range 0-1. Default
	   value is 0.3.

       tripod
	   Set reference frame number for tripod mode.

	   If enabled, the motion of the frames is  compared  to  a  reference
	   frame  in  the filtered stream, identified by the specified number.
	   The idea is to compensate all movements in  a  more-or-less	static
	   scene and keep the camera view absolutely still.

	   If  set  to 0, it is disabled. The frames are counted starting from
	   1.

       show
	   Show fields and transforms in the resulting frames. It  accepts  an
	   integer  in	the  range 0-2. Default value is 0, which disables any
	   visualization.

       fileformat
	   Format for the transforms data  file	 to  be	 written.   Acceptable
	   values are

	   ascii
	       Human-readable plain text

	   binary
	       Binary format, roughly 40% smaller than "ascii". (default)

       Examples

       •   Use default values:

		   vidstabdetect

       •   Analyze   strongly	shaky  movie  and  put	the  results  in  file
	   mytransforms.trf:

		   vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

       •   Visualize the result of internal transformations in	the  resulting
	   video:

		   vidstabdetect=show=1

       •   Analyze a video with medium shakiness using ffmpeg:

		   ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

   vidstabtransform
       Video  stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
       1.

       Read  a	file  with  transform	information   for   each   frame   and
       apply/compensate	 them. Together with the vidstabdetect filter this can
       be	used	   to	    deshake	  videos.	See	  also
       <http://public.hronopik.de/vid.stab>.  It  is important to also use the
       unsharp filter, see below.

       To enable compilation of this filter you need to configure FFmpeg  with
       "--enable-libvidstab".

       Options

       input
	   Set	path to the file used to read the transforms. Default value is
	   transforms.trf.

       smoothing
	   Set the number of frames (value*2 + 1) used for  lowpass  filtering
	   the camera movements. Default value is 10.

	   For example a number of 10 means that 21 frames are used (10 in the
	   past	 and  10 in the future) to smoothen the motion in the video. A
	   larger value leads to a smoother video, but limits the acceleration
	   of the camera (pan/tilt movements). 0 is a  special	case  where  a
	   static camera is simulated.

       optalgo
	   Set the camera path optimization algorithm.

	   Accepted values are:

	   gauss
	       gaussian kernel low-pass filter on camera motion (default)

	   avg averaging on transformations

       maxshift
	   Set	maximal number of pixels to translate frames. Default value is
	   -1, meaning no limit.

       maxangle
	   Set maximal angle in	 radians  (degree*PI/180)  to  rotate  frames.
	   Default value is -1, meaning no limit.

       crop
	   Specify  how	 to  deal  with	 borders  that	may  be visible due to
	   movement compensation.

	   Available values are:

	   keep
	       keep image information from previous frame (default)

	   black
	       fill the border black

       invert
	   Invert transforms if set to 1. Default value is 0.

       relative
	   Consider transforms as relative to previous	frame  if  set	to  1,
	   absolute if set to 0. Default value is 0.

       zoom
	   Set	percentage  to zoom. A positive value will result in a zoom-in
	   effect, a negative value in a zoom-out effect. Default value	 is  0
	   (no zoom).

       optzoom
	   Set optimal zooming to avoid borders.

	   Accepted values are:

	   0   disabled

	   1   optimal	static	zoom  value  is	 determined  (only very strong
	       movements will lead to visible borders) (default)

	   2   optimal adaptive zoom value is determined (no borders  will  be
	       visible), see zoomspeed

	   Note	 that  the  value given at zoom is added to the one calculated
	   here.

       zoomspeed
	   Set percent to zoom maximally each frame (enabled when  optzoom  is
	   set to 2). Range is from 0 to 5, default value is 0.25.

       interpol
	   Specify type of interpolation.

	   Available values are:

	   no  no interpolation

	   linear
	       linear only horizontal

	   bilinear
	       linear in both directions (default)

	   bicubic
	       cubic in both directions (slow)

       tripod
	   Enable  virtual  tripod  mode  if  set to 1, which is equivalent to
	   "relative=0:smoothing=0". Default value is 0.

	   Use also "tripod" option of vidstabdetect.

       debug
	   Increase log verbosity if  set  to  1.  Also	 the  detected	global
	   motions  are	 written  to  the  temporary  file global_motions.trf.
	   Default value is 0.

       Examples

       •   Use ffmpeg for a typical stabilization with default values:

		   ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

	   Note the use of the unsharp filter which is always recommended.

       •   Zoom in a bit more and load transform data from a given file:

		   vidstabtransform=zoom=5:input="mytransforms.trf"

       •   Smoothen the video even more:

		   vidstabtransform=smoothing=30

   vflip
       Flip the input video vertically.

       For example, to vertically flip a video with ffmpeg:

	       ffmpeg -i in.avi -vf "vflip" out.avi

   vfrdet
       Detect variable frame rate video.

       This filter tries to detect if the input is variable or constant	 frame
       rate.

       At  end	it  will  output  number of frames detected as having variable
       delta pts, and ones with constant delta pts.  If there was frames  with
       variable	 delta,	 than  it  will	 also  show min, max and average delta
       encountered.

   vibrance
       Boost or alter saturation.

       The filter accepts the following options:

       intensity
	   Set strength of boost if positive value or  strength	 of  alter  if
	   negative value.  Default is 0. Allowed range is from -2 to 2.

       rbal
	   Set the red balance. Default is 1. Allowed range is from -10 to 10.

       gbal
	   Set	the  green balance. Default is 1. Allowed range is from -10 to
	   10.

       bbal
	   Set the blue balance. Default is 1. Allowed range is	 from  -10  to
	   10.

       rlum
	   Set the red luma coefficient.

       glum
	   Set the green luma coefficient.

       blum
	   Set the blue luma coefficient.

       alternate
	   If  "intensity"  is	negative  and  this  is	 set to 1, colors will
	   change, otherwise colors will be less saturated, more towards gray.

       Commands

       This filter supports the all above options as commands.

   vif
       Obtain the average VIF (Visual Information Fidelity) between two	 input
       videos.

       This filter takes two input videos.

       Both  input  videos  must have the same resolution and pixel format for
       this filter to work correctly. Also it assumes that  both  inputs  have
       the same number of frames, which are compared one by one.

       The obtained average VIF score is printed through the logging system.

       The filter stores the calculated VIF score of each frame.

       This filter also supports the framesync options.

       In  the	below  example	the  input  file  main.mpg  being processed is
       compared with the reference file ref.mpg.

	       ffmpeg -i main.mpg -i ref.mpg -lavfi vif -f null -

   vignette
       Make or reverse a natural vignetting effect.

       The filter accepts the following options:

       angle, a
	   Set lens angle expression as a number of radians.

	   The value is clipped in the "[0,PI/2]" range.

	   Default value: "PI/5"

       x0
       y0  Set center coordinates expressions. Respectively "w/2" and "h/2" by
	   default.

       mode
	   Set forward/backward mode.

	   Available modes are:

	   forward
	       The larger the distance from the central point, the darker  the
	       image becomes.

	   backward
	       The  larger  the	 distance from the central point, the brighter
	       the image becomes.  This can be	used  to  reverse  a  vignette
	       effect,	though	there is no automatic detection to extract the
	       lens angle and other settings (yet). It can  also  be  used  to
	       create a burning effect.

	   Default value is forward.

       eval
	   Set evaluation mode for the expressions (angle, x0, y0).

	   It accepts the following values:

	   init
	       Evaluate	   expressions	  only	  once	 during	  the	filter
	       initialization.

	   frame
	       Evaluate expressions for	 each  incoming	 frame.	 This  is  way
	       slower  than the init mode since it requires all the scalers to
	       be re-computed, but it allows advanced dynamic expressions.

	   Default value is init.

       dither
	   Set dithering to reduce the circular banding effects. Default is  1
	   (enabled).

       aspect
	   Set vignette aspect. This setting allows one to adjust the shape of
	   the vignette.  Setting this value to the SAR of the input will make
	   a rectangular vignetting following the dimensions of the video.

	   Default is "1/1".

       Expressions

       The alpha, x0 and y0 expressions can contain the following parameters.

       w
       h   input width and height

       n   the number of input frame, starting from 0

       pts the	PTS (Presentation TimeStamp) time of the filtered video frame,
	   expressed in TB units, NAN if undefined

       r   frame rate of the input video, NAN  if  the	input  frame  rate  is
	   unknown

       t   the	PTS  (Presentation  TimeStamp)	of  the	 filtered video frame,
	   expressed in seconds, NAN if undefined

       tb  time base of the input video

       Examples

       •   Apply simple strong vignetting effect:

		   vignette=PI/4

       •   Make a flickering vignetting:

		   vignette='PI/4+random(1)*PI/50':eval=frame

   vmafmotion
       Obtain the average VMAF motion score of a video.	  It  is  one  of  the
       component metrics of VMAF.

       The  obtained  average  motion  score  is  printed  through the logging
       system.

       The filter accepts the following options:

       stats_file
	   If specified, the filter will use the named file to save the motion
	   score of each frame with  respect  to  the  previous	 frame.	  When
	   filename equals "-" the data is sent to standard output.

       Example:

	       ffmpeg -i ref.mpg -vf vmafmotion -f null -

   vstack
       Stack input videos vertically.

       All streams must be of same pixel format and of same width.

       Note  that  this	 filter is faster than using overlay and pad filter to
       create same output.

       The filter accepts the following options:

       inputs
	   Set number of input streams. Default is 2.

       shortest
	   If set to 1, force the output to terminate when the shortest	 input
	   terminates. Default value is 0.

   w3fdif
       Deinterlace  the	 input	video  ("w3fdif"  stands  for  "Weston 3 Field
       Deinterlacing Filter").

       Based on the process described  by  Martin  Weston  for	BBC  R&D,  and
       implemented   based  on	the  de-interlace  algorithm  written  by  Jim
       Easterbrook for BBC R&D, the Weston 3 field deinterlacing  filter  uses
       filter coefficients calculated by BBC R&D.

       This  filter  uses field-dominance information in frame to decide which
       of each pair of fields to place first in the output.   If  it  gets  it
       wrong use setfield filter before "w3fdif" filter.

       There  are  two	sets  of  filter  coefficients, so called "simple" and
       "complex". Which set of filter coefficients  is	used  can  be  set  by
       passing an optional parameter:

       filter
	   Set	the  interlacing  filter  coefficients.	 Accepts  one  of  the
	   following values:

	   simple
	       Simple filter coefficient set.

	   complex
	       More-complex filter coefficient set.

	   Default value is complex.

       mode
	   The interlacing mode to adopt. It  accepts  one  of	the  following
	   values:

	   frame
	       Output one frame for each frame.

	   field
	       Output one frame for each field.

	   The default value is "field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   tff Assume the top field is first.

	   bff Assume the bottom field is first.

	   auto
	       Enable automatic detection of field parity.

	   The	default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first  will  be
	   assumed.

       deint
	   Specify  which  frames to deinterlace. Accepts one of the following
	   values:

	   all Deinterlace all frames,

	   interlaced
	       Only deinterlace frames marked as interlaced.

	   Default value is all.

       Commands

       This filter supports same commands as options.

   waveform
       Video waveform monitor.

       The waveform monitor plots color component intensity. By	 default  luma
       only.  Each column of the waveform corresponds to a column of pixels in
       the source video.

       It accepts the following options:

       mode, m
	   Can be either "row", or "column".  Default  is  "column".   In  row
	   mode, the graph on the left side represents color component value 0
	   and	the right side represents value = 255. In column mode, the top
	   side	 represents  color  component  value  =	 0  and	 bottom	  side
	   represents value = 255.

       intensity, i
	   Set	intensity.  Smaller  values  are  useful  to find out how many
	   values  of  the  same  luminance  are  distributed	across	 input
	   rows/columns.  Default value is 0.04. Allowed range is [0, 1].

       mirror, r
	   Set	mirroring  mode.  0  means  unmirrored,	 1 means mirrored.  In
	   mirrored mode, higher values will be represented on the  left  side
	   for	"row"  mode  and  at  the  top for "column" mode. Default is 1
	   (mirrored).

       display, d
	   Set display mode.  It accepts the following values:

	   overlay
	       Presents information identical to that in the "parade",	except
	       that  the graphs representing color components are superimposed
	       directly over one another.

	       This display mode makes it easier to spot relative  differences
	       or  similarities	 in  overlapping areas of the color components
	       that are supposed to be	identical,  such  as  neutral  whites,
	       grays, or blacks.

	   stack
	       Display separate graph for the color components side by side in
	       "row" mode or one below the other in "column" mode.

	   parade
	       Display separate graph for the color components side by side in
	       "column" mode or one below the other in "row" mode.

	       Using  this  display  mode makes it easy to spot color casts in
	       the highlights and  shadows  of	an  image,  by	comparing  the
	       contours	 of  the  top  and the bottom graphs of each waveform.
	       Since whites, grays, and blacks are  characterized  by  exactly
	       equal  amounts  of  red,	 green, and blue, neutral areas of the
	       picture	should	display	 three	waveforms  of  roughly	 equal
	       width/height.  If  not,	the  correction	 is easy to perform by
	       making level adjustments the three waveforms.

	   Default is "stack".

       components, c
	   Set which color components to display. Default is  1,  which	 means
	   only	 luma or red color component if input is in RGB colorspace. If
	   is set for example to 7 it will display all 3 (if) available	 color
	   components.

       envelope, e
	   none
	       No envelope, this is default.

	   instant
	       Instant envelope, minimum and maximum values presented in graph
	       will be easily visible even with small "step" value.

	   peak
	       Hold minimum and maximum values presented in graph across time.
	       This  way  you  can  still  spot	 out  of  range values without
	       constantly looking at waveforms.

	   peak+instant
	       Peak and instant envelope combined together.

       filter, f
	   lowpass
	       No filtering, this is default.

	   flat
	       Luma and chroma combined together.

	   aflat
	       Similar as above, but shows difference  between	blue  and  red
	       chroma.

	   xflat
	       Similar as above, but use different colors.

	   yflat
	       Similar as above, but again with different colors.

	   chroma
	       Displays only chroma.

	   color
	       Displays actual color value on waveform.

	   acolor
	       Similar	as  above,  but	 with luma showing frequency of chroma
	       values.

       graticule, g
	   Set which graticule to display.

	   none
	       Do not display graticule.

	   green
	       Display green graticule showing legal broadcast ranges.

	   orange
	       Display orange graticule showing legal broadcast ranges.

	   invert
	       Display invert graticule showing legal broadcast ranges.

       opacity, o
	   Set graticule opacity.

       flags, fl
	   Set graticule flags.

	   numbers
	       Draw numbers above lines. By default enabled.

	   dots
	       Draw dots instead of lines.

       scale, s
	   Set scale used for displaying graticule.

	   digital
	   millivolts
	   ire

	   Default is digital.

       bgopacity, b
	   Set background opacity.

       tint0, t0
       tint1, t1
	   Set tint for output.	  Only	used  with  lowpass  filter  and  when
	   display is not overlay and input pixel formats are not RGB.

       fitmode, fm
	   Set	sample	aspect	ratio  of video output frames.	Can be used to
	   configure waveform so it  is	 not  streched	too  much  in  one  of
	   directions.

	   none
	       Set sample aspect ration to 1/1.

	   size
	       Set sample aspect ratio to match input size of video

	   Default is none.

       input
	   Set	input  formats	for  filter  to	 pick  from.   Can be all, for
	   selecting from all available formats, or first, for selecting first
	   available format.  Default is first.

   weave, doubleweave
       The  "weave"  takes  a  field-based  video  input  and  join  each  two
       sequential fields into single frame, producing a new double height clip
       with half the frame rate and half the frame count.

       The  "doubleweave" works same as "weave" but without halving frame rate
       and frame count.

       It accepts the following option:

       first_field
	   Set first field. Available values are:

	   top, t
	       Set the frame as top-field-first.

	   bottom, b
	       Set the frame as bottom-field-first.

       Examples

       •   Interlace video using select and separatefields filter:

		   separatefields,select=eq(mod(n,4),0)+eq(mod(n,4),3),weave

   xbr
       Apply the xBR high-quality magnification filter which is	 designed  for
       pixel   art.   It   follows   a	 set   of  edge-detection  rules,  see
       <https://forums.libretro.com/t/xbr-algorithm-tutorial/123>.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "2xBR", 3 for  "3xBR"  and	4  for
	   "4xBR".  Default is 3.

   xcorrelate
       Apply normalized cross-correlation between first and second input video
       stream.

       Second  input  video  stream  dimensions must be lower than first input
       video stream.

       The filter accepts the following options:

       planes
	   Set which planes to process.

       secondary
	   Set which secondary video frames  will  be  processed  from	second
	   input video stream, can be first or all. Default is all.

       The "xcorrelate" filter also supports the framesync options.

   xfade
       Apply  cross  fade  from	 one input video stream to another input video
       stream.	The cross fade is applied for specified duration.

       Both inputs must be constant frame-rate and have the  same  resolution,
       pixel format, frame rate and timebase.

       The filter accepts the following options:

       transition
	   Set one of available transition effects:

	   custom
	   fade
	   wipeleft
	   wiperight
	   wipeup
	   wipedown
	   slideleft
	   slideright
	   slideup
	   slidedown
	   circlecrop
	   rectcrop
	   distance
	   fadeblack
	   fadewhite
	   radial
	   smoothleft
	   smoothright
	   smoothup
	   smoothdown
	   circleopen
	   circleclose
	   vertopen
	   vertclose
	   horzopen
	   horzclose
	   dissolve
	   pixelize
	   diagtl
	   diagtr
	   diagbl
	   diagbr
	   hlslice
	   hrslice
	   vuslice
	   vdslice
	   hblur
	   fadegrays
	   wipetl
	   wipetr
	   wipebl
	   wipebr
	   squeezeh
	   squeezev
	   zoomin
	   fadefast
	   fadeslow
	   hlwind
	   hrwind
	   vuwind
	   vdwind
	   coverleft
	   coverright
	   coverup
	   coverdown
	   revealleft
	   revealright
	   revealup
	   revealdown

	   Default transition effect is fade.

       duration
	   Set	cross  fade  duration  in  seconds.  Range is 0 to 60 seconds.
	   Default duration is 1 second.

       offset
	   Set cross fade start relative to first  input  stream  in  seconds.
	   Default offset is 0.

       expr
	   Set expression for custom transition effect.

	   The expressions can use the following variables and functions:

	   X
	   Y   The coordinates of the current sample.

	   W
	   H   The width and height of the image.

	   P   Progress of transition effect.

	   PLANE
	       Currently processed plane.

	   A   Return value of first input at current location and plane.

	   B   Return value of second input at current location and plane.

	   a0(x, y)
	   a1(x, y)
	   a2(x, y)
	   a3(x, y)
	       Return  the  value  of  the  pixel  at  location	 (x,y)	of the
	       first/second/third/fourth component of first input.

	   b0(x, y)
	   b1(x, y)
	   b2(x, y)
	   b3(x, y)
	       Return the  value  of  the  pixel  at  location	(x,y)  of  the
	       first/second/third/fourth component of second input.

       Examples

       •   Cross  fade	from one input video to another input video, with fade
	   transition and duration of transition  of  2	 seconds  starting  at
	   offset of 5 seconds:

		   ffmpeg -i first.mp4 -i second.mp4 -filter_complex xfade=transition=fade:duration=2:offset=5 output.mp4

   xmedian
       Pick median pixels from several input videos.

       The filter accepts the following options:

       inputs
	   Set	number	of  inputs.   Default is 3. Allowed range is from 3 to
	   255.	 If number of inputs is even number, than result will be  mean
	   value between two median values.

       planes
	   Set	which  planes  to  filter.  Default  value is 15, by which all
	   planes are processed.

       percentile
	   Set median percentile. Default value is 0.5.	 Default value of  0.5
	   will	 pick  always median values, while 0 will pick minimum values,
	   and 1 maximum values.

       Commands

       This filter supports all above options as  commands,  excluding	option
       "inputs".

   xstack
       Stack video inputs into custom layout.

       All streams must be of same pixel format.

       The filter accepts the following options:

       inputs
	   Set number of input streams. Default is 2.

       layout
	   Specify  layout of inputs.  This option requires the desired layout
	   configuration to be explicitly set by the user.  This sets position
	   of each video input in output. Each input is separated by '|'.  The
	   first  number  represents  the  column,  and	 the   second	number
	   represents  the  row.  Numbers start at 0 and are separated by '_'.
	   Optionally one can use wX and hX, where X is video input from which
	   to take  width  or  height.	 Multiple  values  can	be  used  when
	   separated by '+'. In such case values are summed together.

	   Note	 that if inputs are of different sizes gaps may appear, as not
	   all of the output video frame will be filled. Similarly, videos can
	   overlap each other if their position doesn't leave enough space for
	   the full frame of adjoining videos.

	   For 2  inputs,  a  default  layout  of  "0_0|w0_0"  (equivalent  to
	   "grid=2x1")	is set. In all other cases, a layout or a grid must be
	   set by the user. Either "grid" or "layout" can be  specified	 at  a
	   time.  Specifying both will result in an error.

       grid
	   Specify a fixed size grid of inputs.	 This option is used to create
	   a  fixed  size  grid of the input streams. Set the grid size in the
	   form "COLUMNSxROWS". There must be "ROWS * COLUMNS"	input  streams
	   and	they will be arranged as a grid with "ROWS" rows and "COLUMNS"
	   columns. When using this option, each input	stream	within	a  row
	   must	 have  the  same  height  and  all the rows must have the same
	   width.

	   If "grid" is set, then "inputs" option is ignored and is implicitly
	   set to "ROWS * COLUMNS".

	   For	2  inputs,  a	default	  grid	 of   "2x1"   (equivalent   to
	   "layout=0_0|w0_0")  is  set. In all other cases, a layout or a grid
	   must be set by the user. Either "grid" or "layout" can be specified
	   at a time.  Specifying both will result in an error.

       shortest
	   If set to 1, force the output to terminate when the shortest	 input
	   terminates. Default value is 0.

       fill
	   If  set  to valid color, all unused pixels will be filled with that
	   color.  By default fill is set to none, so it is disabled.

       Examples

       •   Display 4 inputs into 2x2 grid.

	   Layout:

		   input1(0, 0)	 | input3(w0, 0)
		   input2(0, h0) | input4(w0, h0)

		   xstack=inputs=4:layout=0_0|0_h0|w0_0|w0_h0

	   Note that if inputs are of different sizes, gaps  or	 overlaps  may
	   occur.

       •   Display 4 inputs into 1x4 grid.

	   Layout:

		   input1(0, 0)
		   input2(0, h0)
		   input3(0, h0+h1)
		   input4(0, h0+h1+h2)

		   xstack=inputs=4:layout=0_0|0_h0|0_h0+h1|0_h0+h1+h2

	   Note	 that  if  inputs  are	of different widths, unused space will
	   appear.

       •   Display 9 inputs into 3x3 grid.

	   Layout:

		   input1(0, 0)	      | input4(w0, 0)	   | input7(w0+w3, 0)
		   input2(0, h0)      | input5(w0, h0)	   | input8(w0+w3, h0)
		   input3(0, h0+h1)   | input6(w0, h0+h1)  | input9(w0+w3, h0+h1)

		   xstack=inputs=9:layout=0_0|0_h0|0_h0+h1|w0_0|w0_h0|w0_h0+h1|w0+w3_0|w0+w3_h0|w0+w3_h0+h1

	   Note that if inputs are of different sizes, gaps  or	 overlaps  may
	   occur.

       •   Display 16 inputs into 4x4 grid.

	   Layout:

		   input1(0, 0)	      | input5(w0, 0)	    | input9 (w0+w4, 0)	      | input13(w0+w4+w8, 0)
		   input2(0, h0)      | input6(w0, h0)	    | input10(w0+w4, h0)      | input14(w0+w4+w8, h0)
		   input3(0, h0+h1)   | input7(w0, h0+h1)   | input11(w0+w4, h0+h1)   | input15(w0+w4+w8, h0+h1)
		   input4(0, h0+h1+h2)| input8(w0, h0+h1+h2)| input12(w0+w4, h0+h1+h2)| input16(w0+w4+w8, h0+h1+h2)

		   xstack=inputs=16:layout=0_0|0_h0|0_h0+h1|0_h0+h1+h2|w0_0|w0_h0|w0_h0+h1|w0_h0+h1+h2|w0+w4_0|
		   w0+w4_h0|w0+w4_h0+h1|w0+w4_h0+h1+h2|w0+w4+w8_0|w0+w4+w8_h0|w0+w4+w8_h0+h1|w0+w4+w8_h0+h1+h2

	   Note	 that  if  inputs are of different sizes, gaps or overlaps may
	   occur.

   yadif
       Deinterlace the input video ("yadif" means "yet	another	 deinterlacing
       filter").

       It accepts the following parameters:

       mode
	   The	interlacing  mode  to  adopt.  It accepts one of the following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   2, send_frame_nospatial
	       Like "send_frame", but it skips the spatial interlacing check.

	   3, send_field_nospatial
	       Like "send_field", but it skips the spatial interlacing check.

	   The default value is "send_frame".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The default value is "auto".	 If the interlacing is unknown or  the
	   decoder  does  not export this information, top field first will be
	   assumed.

       deint
	   Specify which frames to deinterlace. Accepts one of	the  following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   yadif_cuda
       Deinterlace  the input video using the yadif algorithm, but implemented
       in CUDA so that it can work as part of a GPU accelerated pipeline  with
       nvdec and/or nvenc.

       It accepts the following parameters:

       mode
	   The	interlacing  mode  to  adopt.  It accepts one of the following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   2, send_frame_nospatial
	       Like "send_frame", but it skips the spatial interlacing check.

	   3, send_field_nospatial
	       Like "send_field", but it skips the spatial interlacing check.

	   The default value is "send_frame".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The default value is "auto".	 If the interlacing is unknown or  the
	   decoder  does  not export this information, top field first will be
	   assumed.

       deint
	   Specify which frames to deinterlace. Accepts one of	the  following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   yaepblur
       Apply blur filter while preserving edges ("yaepblur" means "yet another
       edge  preserving	 blur  filter").  The algorithm is described in "J. S.
       Lee, Digital image enhancement and noise	 filtering  by	use  of	 local
       statistics, IEEE Trans. Pattern Anal. Mach. Intell. PAMI-2, 1980."

       It accepts the following parameters:

       radius, r
	   Set the window radius. Default value is 3.

       planes, p
	   Set which planes to filter. Default is only the first plane.

       sigma, s
	   Set blur strength. Default value is 128.

       Commands

       This filter supports same commands as options.

   zoompan
       Apply Zoom & Pan effect.

       This filter accepts the following options:

       zoom, z
	   Set the zoom expression. Range is 1-10. Default is 1.

       x
       y   Set the x and y expression. Default is 0.

       d   Set the duration expression in number of frames.  This sets for how
	   many	 number	 of  frames  effect  will last for single input image.
	   Default is 90.

       s   Set the output image size, default is 'hd720'.

       fps Set the output frame rate, default is '25'.

       Each expression can contain the following constants:

       in_w, iw
	   Input width.

       in_h, ih
	   Input height.

       out_w, ow
	   Output width.

       out_h, oh
	   Output height.

       in  Input frame count.

       on  Output frame count.

       in_time, it
	   The input timestamp expressed in seconds. It's  NAN	if  the	 input
	   timestamp is unknown.

       out_time, time, ot
	   The output timestamp expressed in seconds.

       x
       y   Last	 calculated  'x'  and 'y' position from 'x' and 'y' expression
	   for current input frame.

       px
       py  'x' and 'y' of last output frame of previous input frame or 0  when
	   there was not yet such frame (first input frame).

       zoom
	   Last calculated zoom from 'z' expression for current input frame.

       pzoom
	   Last calculated zoom of last output frame of previous input frame.

       duration
	   Number  of  output  frames for current input frame. Calculated from
	   'd' expression for each input frame.

       pduration
	   number of output frames created for previous input frame

       a   Rational number: input width / input height

       sar sample aspect ratio

       dar display aspect ratio

       Examples

       •   Zoom in up to 1.5x and pan at same time to some spot near center of
	   picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='if(gte(zoom,1.5),x,x+1/a)':y='if(gte(zoom,1.5),y,y+1)':s=640x360

       •   Zoom in up to 1.5x and pan always at center of picture:

		   zoompan=z='min(zoom+0.0015,1.5)':d=700:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       •   Same as above but without pausing:

		   zoompan=z='min(max(zoom,pzoom)+0.0015,1.5)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

       •   Zoom in 2x into center of picture only for the first second of  the
	   input video:

		   zoompan=z='if(between(in_time,0,1),2,1)':d=1:x='iw/2-(iw/zoom/2)':y='ih/2-(ih/zoom/2)'

   zscale
       Scale   (resize)	  the	input	video,	 using	 the   z.lib  library:
       <https://github.com/sekrit-twc/zimg>. To	 enable	 compilation  of  this
       filter, you need to configure FFmpeg with "--enable-libzimg".

       The zscale filter forces the output display aspect ratio to be the same
       as the input, by changing the output sample aspect ratio.

       If the input image format is different from the format requested by the
       next  filter, the zscale filter will convert the input to the requested
       format.

       Options

       The filter accepts the following options.

       width, w
       height, h
	   Set the output video dimension expression.  Default	value  is  the
	   input dimension.

	   If  the  width  or  w  value	 is 0, the input width is used for the
	   output. If the height or h value is 0, the input height is used for
	   the output.

	   If one and only one of the values is -n with n  >=  1,  the	zscale
	   filter  will	 use  a	 value	that maintains the aspect ratio of the
	   input image, calculated from the other specified  dimension.	 After
	   that	 it  will, however, make sure that the calculated dimension is
	   divisible by n and adjust the value if necessary.

	   If both values are -n with n >= 1, the behavior will	 be  identical
	   to both values being set to 0 as previously detailed.

	   See	below  for  the	 list  of  accepted  constants	for use in the
	   dimension expression.

       size, s
	   Set the video size. For the syntax of this option, check the "Video
	   size" section in the ffmpeg-utils manual.

       dither, d
	   Set the dither type.

	   Possible values are:

	   none
	   ordered
	   random
	   error_diffusion

	   Default is none.

       filter, f
	   Set the resize filter type.

	   Possible values are:

	   point
	   bilinear
	   bicubic
	   spline16
	   spline36
	   lanczos

	   Default is bilinear.

       range, r
	   Set the color range.

	   Possible values are:

	   input
	   limited
	   full

	   Default is same as input.

       primaries, p
	   Set the color primaries.

	   Possible values are:

	   input
	   709
	   unspecified
	   170m
	   240m
	   2020

	   Default is same as input.

       transfer, t
	   Set the transfer characteristics.

	   Possible values are:

	   input
	   709
	   unspecified
	   601
	   linear
	   2020_10
	   2020_12
	   smpte2084
	   iec61966-2-1
	   arib-std-b67

	   Default is same as input.

       matrix, m
	   Set the colorspace matrix.

	   Possible value are:

	   input
	   709
	   unspecified
	   470bg
	   170m
	   2020_ncl
	   2020_cl

	   Default is same as input.

       rangein, rin
	   Set the input color range.

	   Possible values are:

	   input
	   limited
	   full

	   Default is same as input.

       primariesin, pin
	   Set the input color primaries.

	   Possible values are:

	   input
	   709
	   unspecified
	   170m
	   240m
	   2020

	   Default is same as input.

       transferin, tin
	   Set the input transfer characteristics.

	   Possible values are:

	   input
	   709
	   unspecified
	   601
	   linear
	   2020_10
	   2020_12

	   Default is same as input.

       matrixin, min
	   Set the input colorspace matrix.

	   Possible value are:

	   input
	   709
	   unspecified
	   470bg
	   170m
	   2020_ncl
	   2020_cl
       chromal, c
	   Set the output chroma location.

	   Possible values are:

	   input
	   left
	   center
	   topleft
	   top
	   bottomleft
	   bottom
       chromalin, cin
	   Set the input chroma location.

	   Possible values are:

	   input
	   left
	   center
	   topleft
	   top
	   bottomleft
	   bottom
       npl Set the nominal peak luminance.

       param_a
	   Parameter A for scaling filters. Parameter "b" for bicubic, and the
	   number of filter taps for lanczos.

       param_b
	   Parameter B for scaling filters. Parameter "c" for bicubic.

       The values of the w  and	 h  options  are  expressions  containing  the
       following constants:

       in_w
       in_h
	   The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
	   horizontal  and vertical input chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
	   horizontal and vertical output chroma subsample values. For example
	   for the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Commands

       This filter supports the following commands:

       width, w
       height, h
	   Set the output video dimension expression.  The command accepts the
	   same syntax of the corresponding option.

	   If the specified expression is not valid, it is kept at its current
	   value.

OPENCL VIDEO FILTERS
       Below is a description of the currently available OpenCL video filters.

       To enable compilation of these filters you  need	 to  configure	FFmpeg
       with "--enable-opencl".

       Running OpenCL filters requires you to initialize a hardware device and
       to pass that device to all filters in any filter graph.

       -init_hw_device opencl[=name][:device[,key=value...]]
	   Initialise  a new hardware device of type opencl called name, using
	   the given device parameters.

       -filter_hw_device name
	   Pass the hardware device called name to all filters in  any	filter
	   graph.

       For	     more	    detailed	      information	   see
       <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

       •   Example of choosing the first device on  the	 second	 platform  and
	   running avgblur_opencl filter with default parameters on it.

		   -init_hw_device opencl=gpu:1.0 -filter_hw_device gpu -i INPUT -vf "hwupload, avgblur_opencl, hwdownload" OUTPUT

       Since  OpenCL  filters  are  not	 able  to  access frame data in normal
       memory, all frame data  needs  to  be  uploaded(hwupload)  to  hardware
       surfaces connected to the appropriate device before being used and then
       downloaded(hwdownload)  back  to normal memory. Note that hwupload will
       upload to a surface with the same layout as the software frame,	so  it
       may  be	necessary to add a format filter immediately before to get the
       input into the right format and hwdownload does not support all formats
       on the output - it may be necessary  to	insert	an  additional	format
       filter  immediately  following  in  the	graph  to  get the output in a
       supported format.

   avgblur_opencl
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.	Range is "[1, 1024]" and default value
	   is 1.

       planes
	   Set which planes to filter. Default value  is  0xf,	by  which  all
	   planes are processed.

       sizeY
	   Set vertical radius size. Range is "[1, 1024]" and default value is
	   0. If zero, "sizeX" value will be used.

       Example

       •   Apply  average  blur filter with horizontal and vertical size of 3,
	   setting each pixel of the output to the average value  of  the  7x7
	   region  centered on it in the input. For pixels on the edges of the
	   image, the region does not extend beyond the image boundaries,  and
	   so out-of-range coordinates are not used in the calculations.

		   -i INPUT -vf "hwupload, avgblur_opencl=3, hwdownload" OUTPUT

   boxblur_opencl
       Apply a boxblur algorithm to the input video.

       It accepts the following parameters:

       luma_radius, lr
       luma_power, lp
       chroma_radius, cr
       chroma_power, cp
       alpha_radius, ar
       alpha_power, ap

       A description of the accepted options follows.

       luma_radius, lr
       chroma_radius, cr
       alpha_radius, ar
	   Set	an  expression	for the box radius in pixels used for blurring
	   the corresponding input plane.

	   The radius value must be a non-negative number,  and	 must  not  be
	   greater  than the value of the expression "min(w,h)/2" for the luma
	   and alpha planes, and of "min(cw,ch)/2" for the chroma planes.

	   Default  value  for	luma_radius  is	  "2".	 If   not   specified,
	   chroma_radius  and  alpha_radius default to the corresponding value
	   set for luma_radius.

	   The expressions can contain the following constants:

	   w
	   h   The input width and height in pixels.

	   cw
	   ch  The input chroma image width and height in pixels.

	   hsub
	   vsub
	       The  horizontal	and  vertical  chroma  subsample  values.  For
	       example,	 for the pixel format "yuv422p", hsub is 2 and vsub is
	       1.

       luma_power, lp
       chroma_power, cp
       alpha_power, ap
	   Specify how many  times  the	 boxblur  filter  is  applied  to  the
	   corresponding plane.

	   Default  value  for luma_power is 2. If not specified, chroma_power
	   and	alpha_power  default  to  the  corresponding  value  set   for
	   luma_power.

	   A value of 0 will disable the effect.

       Examples

       Apply  boxblur  filter, setting each pixel of the output to the average
       value of box-radiuses luma_radius, chroma_radius, alpha_radius for each
       plane respectively. The filter  will  apply  luma_power,	 chroma_power,
       alpha_power times onto the corresponding plane. For pixels on the edges
       of  the	image, the radius does not extend beyond the image boundaries,
       and so out-of-range coordinates are not used in the calculations.

       •   Apply a boxblur filter with the luma, chroma, and alpha radius  set
	   to  2  and  luma, chroma, and alpha power set to 3. The filter will
	   run 3 times with box-radius set to 2 for every plane of the image.

		   -i INPUT -vf "hwupload, boxblur_opencl=luma_radius=2:luma_power=3, hwdownload" OUTPUT
		   -i INPUT -vf "hwupload, boxblur_opencl=2:3, hwdownload" OUTPUT

       •   Apply a boxblur filter with luma radius set to 2, luma_power to  1,
	   chroma_radius  to  4,  chroma_power	to  5,	alpha_radius  to 3 and
	   alpha_power to 7.

	   For the luma plane, a 2x2 box radius will be run once.

	   For the chroma plane, a 4x4 box radius will be run 5 times.

	   For the alpha plane, a 3x3 box radius will be run 7 times.

		   -i INPUT -vf "hwupload, boxblur_opencl=2:1:4:5:3:7, hwdownload" OUTPUT

   colorkey_opencl
       RGB colorspace color keying.

       The filter accepts the following options:

       color
	   The color which will be replaced with transparency.

       similarity
	   Similarity percentage with the key color.

	   0.01	 matches  only	the  exact  key	 color,	 while	 1.0   matches
	   everything.

       blend
	   Blend percentage.

	   0.0	makes  pixels  either fully transparent, or not transparent at
	   all.

	   Higher values result in  semi-transparent  pixels,  with  a	higher
	   transparency the more similar the pixels color is to the key color.

       Examples

       •   Make	 every	semi-green  pixel  in  the input transparent with some
	   slight blending:

		   -i INPUT -vf "hwupload, colorkey_opencl=green:0.3:0.1, hwdownload" OUTPUT

   convolution_opencl
       Apply convolution of 3x3, 5x5, 7x7 matrix.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set matrix for each plane.  Matrix is  sequence  of	9,  25	or  49
	   signed  numbers.   Default value for each plane is "0 0 0 0 1 0 0 0
	   0".

       0rdiv
       1rdiv
       2rdiv
       3rdiv
	   Set multiplier for calculated value for each plane.	If unset or 0,
	   it will be sum of all matrix elements.  The option value must be  a
	   float number greater or equal to 0.0. Default value is 1.0.

       0bias
       1bias
       2bias
       3bias
	   Set	bias  for each plane. This value is added to the result of the
	   multiplication.  Useful for making the overall  image  brighter  or
	   darker.   The  option value must be a float number greater or equal
	   to 0.0. Default value is 0.0.

       Examples

       •   Apply sharpen:

		   -i INPUT -vf "hwupload, convolution_opencl=0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0, hwdownload" OUTPUT

       •   Apply blur:

		   -i INPUT -vf "hwupload, convolution_opencl=1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9, hwdownload" OUTPUT

       •   Apply edge enhance:

		   -i INPUT -vf "hwupload, convolution_opencl=0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128, hwdownload" OUTPUT

       •   Apply edge detect:

		   -i INPUT -vf "hwupload, convolution_opencl=0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128, hwdownload" OUTPUT

       •   Apply laplacian edge detector which includes diagonals:

		   -i INPUT -vf "hwupload, convolution_opencl=1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0, hwdownload" OUTPUT

       •   Apply emboss:

		   -i INPUT -vf "hwupload, convolution_opencl=-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2, hwdownload" OUTPUT

   erosion_opencl
       Apply erosion effect to the video.

       This filter replaces the pixel by the local(3x3) minimum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit the maximum change for each plane. Range is "[0, 65535]"  and
	   default value is 65535.  If 0, plane will remain unchanged.

       coordinates
	   Flag	 which	specifies  the pixel to refer to.  Range is "[0, 255]"
	   and default value is 255, i.e. all eight pixels are used.

	   Flags to local 3x3 coordinates region centered on "x":

	       1 2 3

	       4 x 5

	       6 7 8

       Example

       •   Apply erosion filter with threshold0 set to 30, threshold1 set  40,
	   threshold2 set to 50 and coordinates set to 231, setting each pixel
	   of the output to the local minimum between pixels: 1, 2, 3, 6, 7, 8
	   of  the  3x3	 region centered on it in the input. If the difference
	   between input pixel and local minimum is more then threshold of the
	   corresponding plane, output pixel will be  set  to  input  pixel  -
	   threshold of corresponding plane.

		   -i INPUT -vf "hwupload, erosion_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

   deshake_opencl
       Feature-point based video stabilization filter.

       The filter accepts the following options:

       tripod
	   Simulates  a	 tripod	 by  preventing any camera movement whatsoever
	   from the original frame. Defaults to 0.

       debug
	   Whether or not additional debug info should be displayed,  both  in
	   the processed output and in the console.

	   Note	 that  in order to see console debug output you will also need
	   to pass "-v verbose" to ffmpeg.

	   Viewing point matches in the output video is only supported for RGB
	   input.

	   Defaults to 0.

       adaptive_crop
	   Whether or not to do a tiny bit of cropping at the borders  to  cut
	   down on the amount of mirrored pixels.

	   Defaults to 1.

       refine_features
	   Whether  or	not  feature  points  should be refined at a sub-pixel
	   level.

	   This can be turned off for a slight performance gain at the cost of
	   precision.

	   Defaults to 1.

       smooth_strength
	   The strength of the smoothing applied to the camera path  from  0.0
	   to 1.0.

	   1.0	is  the maximum smoothing strength while values less than that
	   result in less smoothing.

	   0.0 causes the filter to adaptively choose a smoothing strength  on
	   a per-frame basis.

	   Defaults to 0.0.

       smooth_window_multiplier
	   Controls  the  size	of  the smoothing window (the number of frames
	   buffered to determine motion information from).

	   The size of the smoothing window is determined by  multiplying  the
	   framerate of the video by this number.

	   Acceptable values range from 0.1 to 10.0.

	   Larger  values  increase  the  amount  of motion data available for
	   determining how to smooth the camera	 path,	potentially  improving
	   smoothness, but also increase latency and memory usage.

	   Defaults to 2.0.

       Examples

       •   Stabilize a video with a fixed, medium smoothing strength:

		   -i INPUT -vf "hwupload, deshake_opencl=smooth_strength=0.5, hwdownload" OUTPUT

       •   Stabilize  a	 video with debugging (both in console and in rendered
	   video):

		   -i INPUT -filter_complex "[0:v]format=rgba, hwupload, deshake_opencl=debug=1, hwdownload, format=rgba, format=yuv420p" -v verbose OUTPUT

   dilation_opencl
       Apply dilation effect to the video.

       This filter replaces the pixel by the local(3x3) maximum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
	   Limit the maximum change for each plane. Range is "[0, 65535]"  and
	   default value is 65535.  If 0, plane will remain unchanged.

       coordinates
	   Flag	 which	specifies  the pixel to refer to.  Range is "[0, 255]"
	   and default value is 255, i.e. all eight pixels are used.

	   Flags to local 3x3 coordinates region centered on "x":

	       1 2 3

	       4 x 5

	       6 7 8

       Example

       •   Apply dilation filter with threshold0 set to 30, threshold1 set 40,
	   threshold2 set to 50 and coordinates set to 231, setting each pixel
	   of the output to the local maximum between pixels: 1, 2, 3, 6, 7, 8
	   of the 3x3 region centered on it in the input.  If  the  difference
	   between input pixel and local maximum is more then threshold of the
	   corresponding  plane,  output  pixel	 will  be set to input pixel +
	   threshold of corresponding plane.

		   -i INPUT -vf "hwupload, dilation_opencl=30:40:50:coordinates=231, hwdownload" OUTPUT

   nlmeans_opencl
       Non-local Means denoise filter through OpenCL, this filter accepts same
       options as nlmeans.

   overlay_opencl
       Overlay one video on top of another.

       It takes two inputs and has one output. The first input is  the	"main"
       video on which the second input is overlaid.  This filter requires same
       memory layout for all the inputs. So, format conversion may be needed.

       The filter accepts the following options:

       x   Set	the  x	coordinate  of	the  overlaid video on the main video.
	   Default value is 0.

       y   Set the y coordinate of the	overlaid  video	 on  the  main	video.
	   Default value is 0.

       Examples

       •   Overlay  an	image  LOGO at the top-left corner of the INPUT video.
	   Both inputs are yuv420p format.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

       •   The inputs have same memory layout for color channels , the overlay
	   has additional alpha plane, like INPUT is yuv420p, and the LOGO  is
	   yuva420p.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_opencl, hwdownload" OUTPUT

   pad_opencl
       Add  paddings  to  the input image, and place the original input at the
       provided x, y coordinates.

       It accepts the following options:

       width, w
       height, h
	   Specify an expression for the size of the  output  image  with  the
	   paddings  added.  If	 the  value  for  width	 or  height  is 0, the
	   corresponding input size is used for the output.

	   The width expression can reference the  value  set  by  the	height
	   expression, and vice versa.

	   The default value of width and height is 0.

       x
       y   Specify  the	 offsets to place the input image at within the padded
	   area, with respect to the top/left border of the output image.

	   The x expression can reference the value set by the	y  expression,
	   and vice versa.

	   The default value of x and y is 0.

	   If  x or y evaluate to a negative number, they'll be changed so the
	   input image is centered on the padded area.

       color
	   Specify the color of the  padded  area.  For	 the  syntax  of  this
	   option, check the "Color" section in the ffmpeg-utils manual.

       aspect
	   Pad to an aspect instead to a resolution.

       The  value  for	the  width,  height,  x, and y options are expressions
       containing the following constants:

       in_w
       in_h
	   The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
	   The output width and height (the  size  of  the  padded  area),  as
	   specified by the width and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN
	   if not yet specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

   prewitt_opencl
       Apply		    the		       Prewitt		      operator
       (<https://en.wikipedia.org/wiki/Prewitt_operator>)   to	 input	 video
       stream.

       The filter accepts the following option:

       planes
	   Set	which  planes  to  filter.  Default value is 0xf, by which all
	   planes are processed.

       scale
	   Set value which will be multiplied with filtered result.  Range  is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set	value  which  will  be	added  to  filtered  result.  Range is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       •   Apply the Prewitt operator with scale set to 2 and delta set to 10.

		   -i INPUT -vf "hwupload, prewitt_opencl=scale=2:delta=10, hwdownload" OUTPUT

   program_opencl
       Filter video using an OpenCL program.

       source
	   OpenCL program source file.

       kernel
	   Kernel name in program.

       inputs
	   Number of inputs to the filter.  Defaults to 1.

       size, s
	   Size of output frames.  Defaults to the same as the first input.

       The "program_opencl" filter also supports the framesync options.

       The program source file must contain a kernel function with  the	 given
       name, which will be run once for each plane of the output.  Each run on
       a  plane	 gets  enqueued as a separate 2D global NDRange with one work-
       item for each pixel to be generated.  The global	 ID  offset  for  each
       work-item  is  therefore	 the coordinates of a pixel in the destination
       image.

       The kernel function needs to take the following arguments:

       •   Destination image, __write_only image2d_t.

	   This image will become the output; the kernel should write  all  of
	   it.

       •   Frame index, unsigned int.

	   This is a counter starting from zero and increasing by one for each
	   frame.

       •   Source images, __read_only image2d_t.

	   These  are  the  most  recent images on each input.	The kernel may
	   read from them to generate the output, but they  can't  be  written
	   to.

       Example programs:

       •   Copy	 the  input to the output (output must be the same size as the
	   input).

		   __kernel void copy(__write_only image2d_t destination,
				      unsigned int index,
				      __read_only  image2d_t source)
		   {
		       const sampler_t sampler = CLK_NORMALIZED_COORDS_FALSE;

		       int2 location = (int2)(get_global_id(0), get_global_id(1));

		       float4 value = read_imagef(source, sampler, location);

		       write_imagef(destination, location, value);
		   }

       •   Apply a simple transformation, rotating  the	 input	by  an	amount
	   increasing  with  the  index	 counter.   Pixel  values are linearly
	   interpolated by the sampler, and the output need not have the  same
	   dimensions as the input.

		   __kernel void rotate_image(__write_only image2d_t dst,
					      unsigned int index,
					      __read_only  image2d_t src)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);

		       float angle = (float)index / 100.0f;

		       float2 dst_dim = convert_float2(get_image_dim(dst));
		       float2 src_dim = convert_float2(get_image_dim(src));

		       float2 dst_cen = dst_dim / 2.0f;
		       float2 src_cen = src_dim / 2.0f;

		       int2   dst_loc = (int2)(get_global_id(0), get_global_id(1));

		       float2 dst_pos = convert_float2(dst_loc) - dst_cen;
		       float2 src_pos = {
			   cos(angle) * dst_pos.x - sin(angle) * dst_pos.y,
			   sin(angle) * dst_pos.x + cos(angle) * dst_pos.y
		       };
		       src_pos = src_pos * src_dim / dst_dim;

		       float2 src_loc = src_pos + src_cen;

		       if (src_loc.x < 0.0f	 || src_loc.y < 0.0f ||
			   src_loc.x > src_dim.x || src_loc.y > src_dim.y)
			   write_imagef(dst, dst_loc, 0.5f);
		       else
			   write_imagef(dst, dst_loc, read_imagef(src, sampler, src_loc));
		   }

       •   Blend  two  inputs  together,  with	the  amount of each input used
	   varying with the index counter.

		   __kernel void blend_images(__write_only image2d_t dst,
					      unsigned int index,
					      __read_only  image2d_t src1,
					      __read_only  image2d_t src2)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);

		       float blend = (cos((float)index / 50.0f) + 1.0f) / 2.0f;

		       int2  dst_loc = (int2)(get_global_id(0), get_global_id(1));
		       int2 src1_loc = dst_loc * get_image_dim(src1) / get_image_dim(dst);
		       int2 src2_loc = dst_loc * get_image_dim(src2) / get_image_dim(dst);

		       float4 val1 = read_imagef(src1, sampler, src1_loc);
		       float4 val2 = read_imagef(src2, sampler, src2_loc);

		       write_imagef(dst, dst_loc, val1 * blend + val2 * (1.0f - blend));
		   }

   remap_opencl
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x,  y)
       position where x = Xmap(X, Y) and y = Ymap(X, Y). If mapping values are
       out of range, zero value for pixel will be used for destination pixel.

       Xmap  and  Ymap	input video streams must be of same dimensions. Output
       video stream will have Xmap/Ymap video  stream  dimensions.   Xmap  and
       Ymap input video streams are 32bit float pixel format, single channel.

       interp
	   Specify interpolation used for remapping of pixels.	Allowed values
	   are "near" and "linear".  Default value is "linear".

       fill
	   Specify  the	 color	of the unmapped pixels. For the syntax of this
	   option, check the  "Color"  section	in  the	 ffmpeg-utils  manual.
	   Default color is "black".

   roberts_opencl
       Apply	       the	     Roberts	       cross	      operator
       (<https://en.wikipedia.org/wiki/Roberts_cross>) to input video stream.

       The filter accepts the following option:

       planes
	   Set which planes to filter. Default value  is  0xf,	by  which  all
	   planes are processed.

       scale
	   Set	value which will be multiplied with filtered result.  Range is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set value which  will  be  added  to	 filtered  result.   Range  is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       •   Apply  the Roberts cross operator with scale set to 2 and delta set
	   to 10

		   -i INPUT -vf "hwupload, roberts_opencl=scale=2:delta=10, hwdownload" OUTPUT

   sobel_opencl
       Apply		    the			Sobel		      operator
       (<https://en.wikipedia.org/wiki/Sobel_operator>) to input video stream.

       The filter accepts the following option:

       planes
	   Set	which  planes  to  filter.  Default value is 0xf, by which all
	   planes are processed.

       scale
	   Set value which will be multiplied with filtered result.  Range  is
	   "[0.0, 65535]" and default value is 1.0.

       delta
	   Set	value  which  will  be	added  to  filtered  result.  Range is
	   "[-65535, 65535]" and default value is 0.0.

       Example

       •   Apply sobel operator with scale set to 2 and delta set to 10

		   -i INPUT -vf "hwupload, sobel_opencl=scale=2:delta=10, hwdownload" OUTPUT

   tonemap_opencl
       Perform HDR(PQ/HLG) to SDR conversion with tone-mapping.

       It accepts the following parameters:

       tonemap
	   Specify the tone-mapping operator  to  be  used.  Same  as  tonemap
	   option in tonemap.

       param
	   Tune the tone mapping algorithm. same as param option in tonemap.

       desat
	   Apply  desaturation	for  highlights	 that  exceed  this  level  of
	   brightness. The higher the parameter, the  more  color  information
	   will be preserved. This setting helps prevent unnaturally blown-out
	   colors  for	super-highlights,  by  (smoothly)  turning  into white
	   instead. This makes images  feel  more  natural,  at	 the  cost  of
	   reducing information about out-of-range colors.

	   The	default	 value	is  0.5,  and  the  algorithm here is a little
	   different from the cpu version tonemap currently. A setting of  0.0
	   disables this option.

       threshold
	   The	tonemapping algorithm parameters is fine-tuned per each scene.
	   And a threshold is used to detect whether the scene has changed  or
	   not.	 If  the distance between the current frame average brightness
	   and the current running average exceeds a threshold value, we would
	   re-calculate scene average and peak brightness.  The default	 value
	   is 0.2.

       format
	   Specify the output pixel format.

	   Currently supported formats are:

	   p010
	   nv12
       range, r
	   Set the output color range.

	   Possible values are:

	   tv/mpeg
	   pc/jpeg

	   Default is same as input.

       primaries, p
	   Set the output color primaries.

	   Possible values are:

	   bt709
	   bt2020

	   Default is same as input.

       transfer, t
	   Set the output transfer characteristics.

	   Possible values are:

	   bt709
	   bt2020

	   Default is bt709.

       matrix, m
	   Set the output colorspace matrix.

	   Possible value are:

	   bt709
	   bt2020

	   Default is same as input.

       Example

       •   Convert  HDR(PQ/HLG)	 video	to bt2020-transfer-characteristic p010
	   format using linear operator.

		   -i INPUT -vf "format=p010,hwupload,tonemap_opencl=t=bt2020:tonemap=linear:format=p010,hwdownload,format=p010" OUTPUT

   unsharp_opencl
       Sharpen or blur the input video.

       It accepts the following parameters:

       luma_msize_x, lx
	   Set the luma matrix	horizontal  size.   Range  is  "[1,  23]"  and
	   default value is 5.

       luma_msize_y, ly
	   Set	the luma matrix vertical size.	Range is "[1, 23]" and default
	   value is 5.

       luma_amount, la
	   Set the luma effect strength.  Range is  "[-10,  10]"  and  default
	   value is 1.0.

	   Negative  values  will  blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

       chroma_msize_x, cx
	   Set the chroma matrix horizontal size.   Range  is  "[1,  23]"  and
	   default value is 5.

       chroma_msize_y, cy
	   Set	the  chroma  matrix  vertical  size.   Range  is "[1, 23]" and
	   default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength.  Range is "[-10, 10]"  and  default
	   value is 0.0.

	   Negative  values  will  blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       •   Apply strong luma sharpen effect:

		   -i INPUT -vf "hwupload, unsharp_opencl=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5, hwdownload" OUTPUT

       •   Apply a strong blur of both luma and chroma parameters:

		   -i INPUT -vf "hwupload, unsharp_opencl=7:7:-2:7:7:-2, hwdownload" OUTPUT

   xfade_opencl
       Cross fade two videos with custom transition effect by using OpenCL.

       It accepts the following options:

       transition
	   Set one of possible transition effects.

	   custom
	       Select  custom  transition  effect,   the   actual   transition
	       description will be picked from source and kernel options.

	   fade
	   wipeleft
	   wiperight
	   wipeup
	   wipedown
	   slideleft
	   slideright
	   slideup
	   slidedown
	       Default transition is fade.

       source
	   OpenCL program source file for custom transition.

       kernel
	   Set name of kernel to use for custom transition from program source
	   file.

       duration
	   Set duration of video transition.

       offset
	   Set time of start of transition relative to first video.

       The  program  source file must contain a kernel function with the given
       name, which will be run once for each plane of the output.  Each run on
       a plane gets enqueued as a separate 2D global NDRange  with  one	 work-
       item  for  each	pixel  to be generated.	 The global ID offset for each
       work-item is therefore the coordinates of a pixel  in  the  destination
       image.

       The kernel function needs to take the following arguments:

       •   Destination image, __write_only image2d_t.

	   This	 image	will become the output; the kernel should write all of
	   it.

       •   First Source image, __read_only image2d_t.	Second	Source	image,
	   __read_only image2d_t.

	   These  are  the  most  recent images on each input.	The kernel may
	   read from them to generate the output, but they  can't  be  written
	   to.

       •   Transition  progress,  float.  This value is always between 0 and 1
	   inclusive.

       Example programs:

       •   Apply dots curtain transition effect:

		   __kernel void blend_images(__write_only image2d_t dst,
					      __read_only  image2d_t src1,
					      __read_only  image2d_t src2,
					      float progress)
		   {
		       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
						  CLK_FILTER_LINEAR);
		       int2  p = (int2)(get_global_id(0), get_global_id(1));
		       float2 rp = (float2)(get_global_id(0), get_global_id(1));
		       float2 dim = (float2)(get_image_dim(src1).x, get_image_dim(src1).y);
		       rp = rp / dim;

		       float2 dots = (float2)(20.0, 20.0);
		       float2 center = (float2)(0,0);
		       float2 unused;

		       float4 val1 = read_imagef(src1, sampler, p);
		       float4 val2 = read_imagef(src2, sampler, p);
		       bool next = distance(fract(rp * dots, &unused), (float2)(0.5, 0.5)) < (progress / distance(rp, center));

		       write_imagef(dst, p, next ? val1 : val2);
		   }

VAAPI VIDEO FILTERS
       VAAPI Video filters are usually	used  with  VAAPI  decoder  and	 VAAPI
       encoder. Below is a description of VAAPI video filters.

       To  enable  compilation	of  these filters you need to configure FFmpeg
       with "--enable-vaapi".

       To use vaapi filters, you need to setup the vaapi device correctly. For
       more		  information,		     please		  read
       <https://trac.ffmpeg.org/wiki/Hardware/VAAPI>

   overlay_vaapi
       Overlay one video on the top of another.

       It  takes  two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.

       The filter accepts the following options:

       x
       y   Set expressions for the x and y coordinates of the  overlaid	 video
	   on the main video.

	   Default value is "0" for both expressions.

       w
       h   Set	expressions for the width and height the overlaid video on the
	   main video.

	   Default     values	  are	  'overlay_iw'	   for	   'w'	   and
	   'overlay_ih*w/overlay_iw' for 'h'.

	   The expressions can contain the following parameters:

	   main_w, W
	   main_h, H
	       The main input width and height.

	   overlay_iw
	   overlay_ih
	       The overlay input width and height.

	   overlay_w, w
	   overlay_h, h
	       The overlay output width and height.

	   overlay_x, x
	   overlay_y, y
	       Position of the overlay layer inside of main

       alpha
	   Set	transparency  of  overlaid video. Allowed range is 0.0 to 1.0.
	   Higher value means lower transparency.  Default value is 1.0.

       eof_action
	   See framesync.

       shortest
	   See framesync.

       repeatlast
	   See framesync.

       This filter also supports the framesync options.

       Examples

       •   Overlay an image LOGO at the top-left corner of  the	 INPUT	video.
	   Both inputs for this filter are yuv420p format.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuv420p, hwupload[b], [a][b]overlay_vaapi" OUTPUT

       •   Overlay  an	image  LOGO at the offset (200, 100) from the top-left
	   corner of the INPUT video.  The inputs have same memory layout  for
	   color  channels, the overlay has additional alpha plane, like INPUT
	   is yuv420p, and the LOGO is yuva420p.

		   -i INPUT -i LOGO -filter_complex "[0:v]hwupload[a], [1:v]format=yuva420p, hwupload[b], [a][b]overlay_vaapi=x=200:y=100:w=400:h=300:alpha=1.0, hwdownload, format=nv12" OUTPUT

   tonemap_vaapi
       Perform	HDR(High  Dynamic  Range)  to  SDR(Standard   Dynamic	Range)
       conversion  with	 tone-mapping.	 It  maps  the	dynamic range of HDR10
       content to the SDR content.  It currently only accepts HDR10 as input.

       It accepts the following parameters:

       format
	   Specify the output pixel format.

	   Currently supported formats are:

	   p010
	   nv12

	   Default is nv12.

       primaries, p
	   Set the output color primaries.

	   Default is same as input.

       transfer, t
	   Set the output transfer characteristics.

	   Default is bt709.

       matrix, m
	   Set the output colorspace matrix.

	   Default is same as input.

       Example

       •   Convert HDR(HDR10)  video  to  bt2020-transfer-characteristic  p010
	   format

		   tonemap_vaapi=format=p010:t=bt2020-10

   hstack_vaapi
       Stack input videos horizontally.

       This  is the VA-API variant of the hstack filter, each input stream may
       have different height, this filter will scale down/up each input stream
       while keeping the orignal aspect.

       It accepts the following options:

       inputs
	   See hstack.

       shortest
	   See hstack.

       height
	   Set height of output. If set to 0, this filter will set  height  of
	   output to height of the first input stream. Default value is 0.

   vstack_vaapi
       Stack input videos vertically.

       This  is the VA-API variant of the vstack filter, each input stream may
       have different width, this filter will scale down/up each input	stream
       while keeping the orignal aspect.

       It accepts the following options:

       inputs
	   See vstack.

       shortest
	   See vstack.

       width
	   Set	width  of  output.  If set to 0, this filter will set width of
	   output to width of the first input stream. Default value is 0.

   xstack_vaapi
       Stack video inputs into custom layout.

       This is the VA-API variant of the xstack filter,	 each input stream may
       have different size, this filter will scale down/up each	 input	stream
       to the given output size, or the size of the first input stream.

       It accepts the following options:

       inputs
	   See xstack.

       shortest
	   See xstack.

       layout
	   See	xstack.	 Moreover, this permits the user to supply output size
	   for each input stream.

		   xstack_vaapi=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

       grid
	   See xstack.

       grid_tile_size
	   Set output size for each input stream when grid  is	set.  If  this
	   option  is  not set, this filter will set output size by default to
	   the size of the first input stream. For the syntax of this  option,
	   check the "Video size" section in the ffmpeg-utils manual.

       fill
	   See xstack.

VULKAN VIDEO FILTERS
       Below is a description of the currently available Vulkan video filters.

       To  enable  compilation	of  these filters you need to configure FFmpeg
       with   "--enable-vulkan"	  and	 either	   "--enable-libglslang"    or
       "--enable-libshaderc".

       Running Vulkan filters requires you to initialize a hardware device and
       to pass that device to all filters in any filter graph.

       -init_hw_device vulkan[=name][:device[,key=value...]]
	   Initialise  a new hardware device of type vulkan called name, using
	   the given device parameters and options in key=value. The following
	   options are supported:

	   debug
	       Switches validation layers on if set to 1.

	   linear_images
	       Allocates linear images. Does not apply to decoding.

	   disable_multiplane
	       Disables multiplane images. Does not apply to decoding.

       -filter_hw_device name
	   Pass the hardware device called name to all filters in  any	filter
	   graph.

       For	     more	    detailed	      information	   see
       <https://www.ffmpeg.org/ffmpeg.html#Advanced-Video-options>

       •   Example of choosing the first  device  and  running	nlmeans_vulkan
	   filter with default parameters on it.

		   -init_hw_device vulkan=vk:0 -filter_hw_device vk -i INPUT -vf "hwupload,nlmeans_vulkan,hwdownload" OUTPUT

       As  Vulkan  filters are not able to access frame data in normal memory,
       all frame data needs to be uploaded  (hwupload)	to  hardware  surfaces
       connected  to  the  appropriate	device	before	being  used  and  then
       downloaded (hwdownload) back to normal memory. Note that hwupload  will
       upload to a frame with the same layout as the software frame, so it may
       be necessary to add a format filter immediately before to get the input
       into  the  right	 format and hwdownload does not support all formats on
       the output - it is usually necessary to	insert	an  additional	format
       filter  immediately  following  in  the	graph  to  get the output in a
       supported format.

   avgblur_vulkan
       Apply an average blur filter, implemented on the GPU using Vulkan.

       The filter accepts the following options:

       sizeX
	   Set horizontal radius size.	Range is "[1, 32]" and	default	 value
	   is 3.

       sizeY
	   Set	vertical  radius size. Range is "[1, 32]" and default value is
	   3.

       planes
	   Set which planes to filter. Default value  is  0xf,	by  which  all
	   planes are processed.

   blend_vulkan
       Blend two Vulkan frames into each other.

       The  "blend" filter takes two input streams and outputs one stream, the
       first input is the "top" layer and second input is "bottom" layer.   By
       default, the output terminates when the longest input terminates.

       A description of the accepted options follows.

       c0_mode
       c1_mode
       c2_mode
       c3_mode
       all_mode
	   Set blend mode for specific pixel component or all pixel components
	   in case of all_mode. Default value is "normal".

	   Available values for component modes are:

	   normal
	   multiply

   bwdif_vulkan
       Deinterlacer   using  bwdif,  the  "Bob	Weaver	Deinterlacing  Filter"
       algorithm, implemented on the GPU using Vulkan.

       It accepts the following parameters:

       mode
	   The interlacing mode to adopt. It  accepts  one  of	the  following
	   values:

	   0, send_frame
	       Output one frame for each frame.

	   1, send_field
	       Output one frame for each field.

	   The default value is "send_field".

       parity
	   The picture field parity assumed for the input interlaced video. It
	   accepts one of the following values:

	   0, tff
	       Assume the top field is first.

	   1, bff
	       Assume the bottom field is first.

	   -1, auto
	       Enable automatic detection of field parity.

	   The	default value is "auto".  If the interlacing is unknown or the
	   decoder does not export this information, top field first  will  be
	   assumed.

       deint
	   Specify  which  frames to deinterlace. Accepts one of the following
	   values:

	   0, all
	       Deinterlace all frames.

	   1, interlaced
	       Only deinterlace frames marked as interlaced.

	   The default value is "all".

   chromaber_vulkan
       Apply an effect that emulates chromatic aberration. Works best with RGB
       inputs, but provides a similar effect with YCbCr inputs too.

       dist_x
	   Horizontal displacement multiplier. Each  chroma  pixel's  position
	   will	 be multiplied by this amount, starting from the center of the
	   image. Default is 0.

       dist_y
	   Similarly, this sets the vertical displacement multiplier.  Default
	   is 0.

   color_vulkan
       Video  source that creates a Vulkan frame of a solid color.  Useful for
       benchmarking, or overlaying.

       It accepts the following parameters:

       color
	   The color to use. Either a  name,  or  a  hexadecimal  value.   The
	   default value is "black".

       size
	   The size of the output frame. Default value is "1920x1080".

       rate
	   The framerate to output at. Default value is 60 frames per second.

       duration
	   The video duration. Default value is -0.000001.

       sar The video signal aspect ratio. Default value is "1/1".

       format
	   The	pixel  format  of  the	output Vulkan frames. Default value is
	   "yuv444p".

       out_range
	   Set the output YCbCr sample range.

	   This allows the autodetected value to  be  overridden  as  well  as
	   allows forcing a specific value used for the output and encoder. If
	   not	specified,  the	 range	depends	 on the pixel format. Possible
	   values:

	   auto/unknown
	       Choose automatically.

	   jpeg/full/pc
	       Set full range (0-255 in case of 8-bit luma).

	   mpeg/limited/tv
	       Set "MPEG" range (16-235 in case of 8-bit luma).

   vflip_vulkan
       Flips an image vertically.

   hflip_vulkan
       Flips an image horizontally.

   flip_vulkan
       Flips an image along both the vertical and horizontal axis.

   gblur_vulkan
       Apply Gaussian blur filter on Vulkan frames.

       The filter accepts the following options:

       sigma
	   Set horizontal sigma, standard deviation of Gaussian blur.  Default
	   is 0.5.

       sigmaV
	   Set	vertical  sigma,  if  negative	it  will  be  same as "sigma".
	   Default is -1.

       planes
	   Set which planes to filter. By default all planes are filtered.

       size
	   Set the kernel size along the horizontal axis. Default is 19.

       sizeV
	   Set the kernel size along the vertical axis. Default	 is  0,	 which
	   sets to use the same value as size.

   nlmeans_vulkan
       Denoise	frames using Non-Local Means algorithm, implemented on the GPU
       using  Vulkan.	Supports  more	 pixel	 formats   than	  nlmeans   or
       nlmeans_opencl, including alpha channel support.

       The filter accepts the following options.

       s   Set	denoising strength for all components. Default is 1.0. Must be
	   in range [1.0, 100.0].

       p   Set patch size for all planes. Default is 7. Must be odd number  in
	   range [0, 99].

       r   Set	research  size. Default is 15. Must be odd number in range [0,
	   99].

       t   Set parallelism. Default is 36. Must be a number in the  range  [1,
	   168].   Larger  values may speed up processing, at the cost of more
	   VRAM.  Lower values will slow it down, reducing VRAM	 usage.	  Only
	   supported on GPUs with atomic float operations (RDNA3+, Ampere+).

       s0
       s1
       s2
       s3  Set	denoising  strength  for  a  specific component. Default is 1,
	   equal to s.	Must be odd number in range [1, 100].

       p0
       p1
       p2
       p3  Set patch size for a specific component. Default is 7, equal to  p.
	   Must be odd number in range [0, 99].

   overlay_vulkan
       Overlay one video on top of another.

       It  takes  two inputs and has one output. The first input is the "main"
       video on which the second input is overlaid.  This filter requires  all
       inputs  to  use	the  same  pixel  format. So, format conversion may be
       needed.

       The filter accepts the following options:

       x   Set the x coordinate of the	overlaid  video	 on  the  main	video.
	   Default value is 0.

       y   Set	the  y	coordinate  of	the  overlaid video on the main video.
	   Default value is 0.

   transpose_vt
       Transpose rows with columns in the input video and optionally flip  it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate  by  90  degrees	counterclockwise  and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

	   hflip
	       Flip the input video horizontally.

	   vflip
	       Flip the input video vertically.

       passthrough
	   Do not apply the transposition if the input	geometry  matches  the
	   one	specified  by  the  specified  value. It accepts the following
	   values:

	   none
	       Always apply transposition. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

   transpose_vulkan
       Transpose rows with columns in the input video and optionally flip  it.
       For more in depth examples see the transpose video filter, which shares
       mostly the same options.

       It accepts the following parameters:

       dir Specify the transposition direction.

	   Can assume the following values:

	   cclock_flip
	       Rotate  by  90  degrees	counterclockwise  and vertically flip.
	       (default)

	   clock
	       Rotate by 90 degrees clockwise.

	   cclock
	       Rotate by 90 degrees counterclockwise.

	   clock_flip
	       Rotate by 90 degrees clockwise and vertically flip.

       passthrough
	   Do not apply the transposition if the input	geometry  matches  the
	   one	specified  by  the  specified  value. It accepts the following
	   values:

	   none
	       Always apply transposition. (default)

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

QSV VIDEO FILTERS
       Below is a description of the currently available QSV video filters.

       To enable compilation of these filters you  need	 to  configure	FFmpeg
       with "--enable-libmfx" or "--enable-libvpl".

       To  use	QSV  filters,  you need to setup the QSV device correctly. For
       more		  information,		     please		  read
       <https://trac.ffmpeg.org/wiki/Hardware/QuickSync>

   hstack_qsv
       Stack input videos horizontally.

       This  is	 the  QSV  variant of the hstack filter, each input stream may
       have different height, this filter will scale down/up each input stream
       while keeping the orignal aspect.

       It accepts the following options:

       inputs
	   See hstack.

       shortest
	   See hstack.

       height
	   Set height of output. If set to 0, this filter will set  height  of
	   output to height of the first input stream. Default value is 0.

   vstack_qsv
       Stack input videos vertically.

       This  is	 the  QSV  variant of the vstack filter, each input stream may
       have different width, this filter will scale down/up each input	stream
       while keeping the orignal aspect.

       It accepts the following options:

       inputs
	   See vstack.

       shortest
	   See vstack.

       width
	   Set	width  of  output.  If set to 0, this filter will set width of
	   output to width of the first input stream. Default value is 0.

   xstack_qsv
       Stack video inputs into custom layout.

       This is the QSV variant of the xstack filter.

       It accepts the following options:

       inputs
	   See xstack.

       shortest
	   See xstack.

       layout
	   See xstack.	Moreover, this permits the user to supply output  size
	   for each input stream.

		   xstack_qsv=inputs=4:layout=0_0_1920x1080|0_h0_1920x1080|w0_0_1920x1080|w0_h0_1920x1080

       grid
	   See xstack.

       grid_tile_size
	   Set	output	size  for  each input stream when grid is set. If this
	   option is not set, this filter will set output size by  default  to
	   the	size of the first input stream. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       fill
	   See xstack.

VIDEO SOURCES
       Below is a description of the currently available video sources.

   buffer
       Buffer video frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic  use,	in  particular
       through the interface defined in libavfilter/buffersrc.h.

       It accepts the following parameters:

       video_size
	   Specify  the	 size (width and height) of the buffered video frames.
	   For the syntax of this option, check the "Video  size"  section  in
	   the ffmpeg-utils manual.

       width
	   The input video width.

       height
	   The input video height.

       pix_fmt
	   A  string  representing  the	 pixel	format	of  the buffered video
	   frames.  It may be a number corresponding to a pixel format,	 or  a
	   pixel format name.

       time_base
	   Specify  the	 timebase  assumed  by	the timestamps of the buffered
	   frames.

       frame_rate
	   Specify the frame rate expected for the video stream.

       pixel_aspect, sar
	   The sample (pixel) aspect ratio of the input video.

       hw_frames_ctx
	   When using a hardware pixel format, this should be a	 reference  to
	   an AVHWFramesContext describing input frames.

       For example:

	       buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

       will  instruct  the source to accept video frames with size 320x240 and
       with format "yuv410p", assuming 1/24 as	the  timestamps	 timebase  and
       square  pixels  (1:1 sample aspect ratio).  Since the pixel format with
       name  "yuv410p"	corresponds  to	 the  number   6   (check   the	  enum
       AVPixelFormat   definition   in	 libavutil/pixfmt.h),	this   example
       corresponds to:

	       buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

       Alternatively, the options can be specified as a flat string, but  this
       syntax is deprecated:

       width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den

   cellauto
       Create a pattern generated by an elementary cellular automaton.

       The  initial state of the cellular automaton can be defined through the
       filename and pattern options. If such  options  are  not	 specified  an
       initial state is created randomly.

       At  each	 new frame a new row in the video is filled with the result of
       the cellular automaton next generation. The  behavior  when  the	 whole
       frame is filled is defined by the scroll option.

       This source accepts the following options:

       filename, f
	   Read	 the  initial cellular automaton state, i.e. the starting row,
	   from	 the  specified	 file.	 In  the  file,	 each	non-whitespace
	   character is considered an alive cell, a newline will terminate the
	   row, and further characters in the file will be ignored.

       pattern, p
	   Read	 the  initial cellular automaton state, i.e. the starting row,
	   from the specified string.

	   Each non-whitespace character in the string is considered an	 alive
	   cell,  a  newline will terminate the row, and further characters in
	   the string will be ignored.

       rate, r
	   Set the video rate, that is the  number  of	frames	generated  per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set	the  random fill ratio for the initial cellular automaton row.
	   It is a floating point number value ranging from 0 to  1,  defaults
	   to 1/PHI.

	   This option is ignored when a file or a pattern is specified.

       random_seed, seed
	   Set	the  seed  for	filling	 randomly  the initial row, must be an
	   integer included between 0 and UINT32_MAX. If not specified, or  if
	   explicitly set to -1, the filter will try to use a good random seed
	   on a best effort basis.

       rule
	   Set	the  cellular automaton rule, it is a number ranging from 0 to
	   255.	 Default value is 110.

       size, s
	   Set the size of the output video. For the syntax  of	 this  option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If  filename or pattern is specified, the size is set by default to
	   the width of the specified initial state row, and the height is set
	   to width * PHI.

	   If size is set, it must contain the width of the specified  pattern
	   string,  and	 the  specified pattern will be centered in the larger
	   row.

	   If a filename or a pattern string is not specified, the size	 value
	   defaults  to	 "320x518"  (used  for	a  randomly  generated initial
	   state).

       scroll
	   If set to 1, scroll the output upward when  all  the	 rows  in  the
	   output have been already filled. If set to 0, the new generated row
	   will	 be  written  over  the	 top  row just after the bottom row is
	   filled.  Defaults to 1.

       start_full, full
	   If set to 1, completely fill the output with generated rows	before
	   outputting  the  first  frame.   This  is the default behavior, for
	   disabling set the value to 0.

       stitch
	   If set to 1, stitch the left and right row edges together.  This is
	   the default behavior, for disabling set the value to 0.

       Examples

       •   Read the initial state from pattern, and specify an output of  size
	   200x400.

		   cellauto=f=pattern:s=200x400

       •   Generate  a	random	initial	 row with a width of 200 cells, with a
	   fill ratio of 2/3:

		   cellauto=ratio=2/3:s=200x200

       •   Create a pattern generated by rule 18 starting by  a	 single	 alive
	   cell centered on an initial row with width 100:

		   cellauto=p=@s=100x400:full=0:rule=18

       •   Specify a more elaborated initial pattern:

		   cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

   coreimagesrc
       Video source generated on GPU using Apple's CoreImage API on OSX.

       This  video  source  is	a  specialized	version of the coreimage video
       filter.	Use a core image generator at the  beginning  of  the  applied
       filterchain to generate the content.

       The coreimagesrc video source accepts the following options:

       list_generators
	   List	 all  available	 generators  along  with  all their respective
	   options as well as possible minimum and maximum values  along  with
	   the default values.

		   list_generators=true

       size, s
	   Specify  the	 size  of  the	sourced	 video. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   The default value is "320x240".

       rate, r
	   Specify  the	 frame	rate  of  the  sourced video, as the number of
	   frames generated per second. It has to be a string  in  the	format
	   frame_rate_num/frame_rate_den,  an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default	 value
	   is "25".

       sar Set the sample aspect ratio of the sourced video.

       duration, d
	   Set	the  duration  of  the	sourced	 video.	 See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the	 video
	   is supposed to be generated forever.

       Additionally,  all  options of the coreimage video filter are accepted.
       A complete filterchain can  be  used  for  further  processing  of  the
       generated  input without CPU-HOST transfer. See coreimage documentation
       and examples for details.

       Examples

       •   Use CIQRCodeGenerator to create a QR code for the FFmpeg  homepage,
	   given  as  complete	and  escaped command-line for Apple's standard
	   bash shell:

		   ffmpeg -f lavfi -i coreimagesrc=s=100x100:filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

	   This example is equivalent  to  the	QRCode	example	 of  coreimage
	   without the need for a nullsrc video source.

   ddagrab
       Captures the Windows Desktop via Desktop Duplication API.

       The  filter  exclusively	 returns  D3D11	 Hardware  Frames,  for on-gpu
       encoding or processing. So an explicit hwdownload  is  needed  for  any
       kind of software processing.

       It accepts the following options:

       output_idx
	   DXGI Output Index to capture.

	   Usually corresponds to the index Windows has given the screen minus
	   one, so it's starting at 0.

	   Defaults to output 0.

       draw_mouse
	   Whether to draw the mouse cursor.

	   Defaults to true.

	   Only affects hardware cursors. If a game or application renders its
	   own cursor, it'll always be captured.

       framerate
	   Framerate at which the desktop will be captured.

	   Defaults to 30 FPS.

       video_size
	   Specify the size of the captured video.

	   Defaults to the full size of the screen.

	   Cropped from the bottom/right if smaller than screen size.

       offset_x
	   Horizontal offset of the captured video.

       offset_y
	   Vertical offset of the captured video.

       output_fmt
	   Desired filter output format.  Defaults to 8 Bit BGRA.

	   It accepts the following values:

	   auto
	       Passes all supported output formats to DDA and returns what DDA
	       decides to use.

	   8bit
	   bgra
	       8  Bit  formats	always	work,  and DDA will convert to them if
	       neccesary.

	   10bit
	   x2bgr10
	       Filter initialization will fail if 10 bit format	 is  requested
	       but unavailable.

       Examples

       Capture primary screen and encode using nvenc:

	       ffmpeg -f lavfi -i ddagrab -c:v h264_nvenc -cq 18 output.mp4

       You  can	 also skip the lavfi device and directly use the filter.  Also
       demonstrates downloading the frame and encoding with libx264.  Explicit
       output format specification is required in this case:

	       ffmpeg -filter_complex ddagrab=output_idx=1:framerate=60,hwdownload,format=bgra -c:v libx264 -crf 18 output.mp4

       If you want to capture only a subsection of the desktop,	 this  can  be
       achieved by specifying a smaller size and its offsets into the screen:

	       ddagrab=video_size=800x600:offset_x=100:offset_y=100

   gradients
       Generate several gradients.

       size, s
	   Set	frame  size.  For  the syntax of this option, check the "Video
	   size"  section  in  the  ffmpeg-utils  manual.  Default  value   is
	   "640x480".

       rate, r
	   Set	frame  rate, expressed as number of frames per second. Default
	   value is "25".

       c0, c1, c2, c3, c4, c5, c6, c7
	   Set 8 colors. Default values for colors is to pick random one.

       x0, y0, y0, y1
	   Set gradient line source and destination points. If negative or out
	   of range, random ones are picked.

       nb_colors, n
	   Set number of colors to use at once. Allowed range is from 2 to  8.
	   Default value is 2.

       seed
	   Set seed for picking gradient line points.

       duration, d
	   Set	the  duration  of  the	sourced	 video.	 See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the	 video
	   is supposed to be generated forever.

       speed
	   Set speed of gradients rotation.

       type, t
	   Set type of gradients, can be "linear" or "radial" or "circular" or
	   "spiral".

   mandelbrot
       Generate	 a  Mandelbrot set fractal, and progressively zoom towards the
       point specified with start_x and start_y.

       This source accepts the following options:

       end_pts
	   Set the terminal pts value. Default value is 400.

       end_scale
	   Set the terminal scale value.  Must	be  a  floating	 point	value.
	   Default value is 0.3.

       inner
	   Set the inner coloring mode, that is the algorithm used to draw the
	   Mandelbrot fractal internal region.

	   It shall assume one of the following values:

	   black
	       Set black mode.

	   convergence
	       Show time until convergence.

	   mincol
	       Set  color  based  on  point  closest  to  the  origin  of  the
	       iterations.

	   period
	       Set period mode.

	   Default value is mincol.

       bailout
	   Set the bailout value. Default value is 10.0.

       maxiter
	   Set the maximum of iterations performed by the rendering algorithm.
	   Default value is 7189.

       outer
	   Set outer coloring mode.  It shall assume one of following values:

	   iteration_count
	       Set iteration count mode.

	   normalized_iteration_count
	       set normalized iteration count mode.

	   Default value is normalized_iteration_count.

       rate, r
	   Set frame rate, expressed as number of frames per  second.  Default
	   value is "25".

       size, s
	   Set	frame  size.  For  the syntax of this option, check the "Video
	   size"  section  in  the  ffmpeg-utils  manual.  Default  value   is
	   "640x480".

       start_scale
	   Set the initial scale value. Default value is 3.0.

       start_x
	   Set	the initial x position. Must be a floating point value between
	   -100	       and	  100.	      Default	      value	    is
	   -0.743643887037158704752191506114774.

       start_y
	   Set	the initial y position. Must be a floating point value between
	   -100	       and	  100.	      Default	      value	    is
	   -0.131825904205311970493132056385139.

   mptestsrc
       Generate	 various  test	patterns,  as  generated  by  the MPlayer test
       filter.

       The size of the generated video is fixed, and is 256x256.  This	source
       is useful in particular for testing encoding features.

       This source accepts the following options:

       rate, r
	   Specify  the	 frame	rate  of  the  sourced video, as the number of
	   frames generated per second. It has to be a string  in  the	format
	   frame_rate_num/frame_rate_den,  an integer number, a floating point
	   number or a valid video frame rate abbreviation. The default	 value
	   is "25".

       duration, d
	   Set	the  duration  of  the	sourced	 video.	 See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the	 video
	   is supposed to be generated forever.

       test, t
	   Set	the number or the name of the test to perform. Supported tests
	   are:

	   dc_luma
	   dc_chroma
	   freq_luma
	   freq_chroma
	   amp_luma
	   amp_chroma
	   cbp
	   mv
	   ring1
	   ring2
	   all
	   max_frames, m
	       Set the maximum number  of  frames  generated  for  each	 test,
	       default value is 30.

	   Default  value  is  "all", which will cycle through the list of all
	   tests.

       Some examples:

	       mptestsrc=t=dc_luma

       will generate a "dc_luma" test pattern.

   frei0r_src
       Provide a frei0r source.

       To enable compilation of this filter you need  to  install  the	frei0r
       header and configure FFmpeg with "--enable-frei0r".

       This source accepts the following parameters:

       size
	   The	size  of the video to generate. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.

       framerate
	   The framerate of the generated video. It may be  a  string  of  the
	   form num/den or a frame rate abbreviation.

       filter_name
	   The	name  to  the  frei0r  source  to  load.  For more information
	   regarding frei0r and how to set the	parameters,  read  the	frei0r
	   section in the video filters documentation.

       filter_params
	   A '|'-separated list of parameters to pass to the frei0r source.

       For example, to generate a frei0r partik0l source with size 200x200 and
       frame rate 10 which is overlaid on the overlay filter main input:

	       frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

   life
       Generate a life pattern.

       This source is based on a generalization of John Conway's life game.

       The  sourced input represents a life grid, each pixel represents a cell
       which can be in one of two possible states, alive or dead.  Every  cell
       interacts  with	its  eight  neighbours,	 which	are the cells that are
       horizontally, vertically, or diagonally adjacent.

       At each interaction the grid evolves according  to  the	adopted	 rule,
       which  specifies	 the  number of neighbor alive cells which will make a
       cell stay alive or born. The rule option allows one to specify the rule
       to adopt.

       This source accepts the following options:

       filename, f
	   Set the file from which to read the	initial	 grid  state.  In  the
	   file,  each	non-whitespace	character is considered an alive cell,
	   and newline is used to delimit the end of each row.

	   If this option is not specified,  the  initial  grid	 is  generated
	   randomly.

       rate, r
	   Set	the  video  rate,  that	 is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial  random  grid.  It	 is  a
	   floating point number value ranging from 0 to 1, defaults to 1/PHI.
	   It is ignored when a file is specified.

       random_seed, seed
	   Set	the  seed  for	filling	 the  initial  random grid, must be an
	   integer included between 0 and UINT32_MAX. If not specified, or  if
	   explicitly set to -1, the filter will try to use a good random seed
	   on a best effort basis.

       rule
	   Set the life rule.

	   A rule can be specified with a code of the kind "SNS/BNB", where NS
	   and	NB are sequences of numbers in the range 0-8, NS specifies the
	   number of alive neighbor cells which make a live cell  stay	alive,
	   and NB the number of alive neighbor cells which make a dead cell to
	   become alive (i.e. to "born").  "s" and "b" can be used in place of
	   "S" and "B", respectively.

	   Alternatively  a rule can be specified by an 18-bits integer. The 9
	   high order bits are used to encode the next cell  state  if	it  is
	   alive  for  each number of neighbor alive cells, the low order bits
	   specify the rule for "borning" new cells. Higher order bits	encode
	   for	an  higher  number  of neighbor cells.	For example the number
	   6153 = "(12<<9)+9" specifies a stay alive rule of  12  and  a  born
	   rule of 9, which corresponds to "S23/B03".

	   Default  value  is "S23/B3", which is the original Conway's game of
	   life rule, and will keep a cell alive if it has  2  or  3  neighbor
	   alive  cells,  and  will  born  a new cell if there are three alive
	   cells around a dead cell.

       size, s
	   Set the size of the output video. For the syntax  of	 this  option,
	   check the "Video size" section in the ffmpeg-utils manual.

	   If  filename	 is  specified, the size is set by default to the same
	   size of the input file. If size is set, it must  contain  the  size
	   specified  in  the input file, and the initial grid defined in that
	   file is centered in the larger resulting area.

	   If a	 filename  is  not  specified,	the  size  value  defaults  to
	   "320x240" (used for a randomly generated initial grid).

       stitch
	   If set to 1, stitch the left and right grid edges together, and the
	   top and bottom edges also. Defaults to 1.

       mold
	   Set	cell  mold speed. If set, a dead cell will go from death_color
	   to mold_color with a step of mold. mold can have a value from 0  to
	   255.

       life_color
	   Set the color of living (or new born) cells.

       death_color
	   Set	the  color  of	dead  cells. If mold is set, this is the first
	   color used to represent a dead cell.

       mold_color
	   Set mold color, for definitely dead and moldy cells.

	   For the syntax of these 3 color options, check the "Color"  section
	   in the ffmpeg-utils manual.

       Examples

       •   Read	 a  grid from pattern, and center it on a grid of size 300x300
	   pixels:

		   life=f=pattern:s=300x300

       •   Generate a random grid of size 200x200, with a fill ratio of 2/3:

		   life=ratio=2/3:s=200x200

       •   Specify a custom rule for evolving a randomly generated grid:

		   life=rule=S14/B34

       •   Full example with slow death effect (mold) using ffplay:

		   ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16

   allrgb, allyuv, color,  colorchart,	colorspectrum,	haldclutsrc,  nullsrc,
       pal75bars,  pal100bars,	rgbtestsrc,  smptebars,	 smptehdbars, testsrc,
       testsrc2, yuvtestsrc
       The "allrgb" source returns frames of size 4096x4096 of all rgb colors.

       The "allyuv" source returns frames of size 4096x4096 of all yuv colors.

       The "color" source provides an uniformly colored input.

       The "colorchart" source provides a colors checker chart.

       The "colorspectrum" source provides a color spectrum input.

       The "haldclutsrc" source provides  an  identity	Hald  CLUT.  See  also
       haldclut filter.

       The  "nullsrc"  source  returns	unprocessed video frames. It is mainly
       useful to be employed in analysis / debugging tools, or as  the	source
       for filters which ignore the input data.

       The "pal75bars" source generates a color bars pattern, based on EBU PAL
       recommendations with 75% color levels.

       The  "pal100bars"  source  generates a color bars pattern, based on EBU
       PAL recommendations with 100% color levels.

       The "rgbtestsrc" source	generates  an  RGB  test  pattern  useful  for
       detecting  RGB  vs  BGR	issues.	 You  should see a red, green and blue
       stripe from top to bottom.

       The "smptebars" source generates a color bars  pattern,	based  on  the
       SMPTE Engineering Guideline EG 1-1990.

       The  "smptehdbars"  source generates a color bars pattern, based on the
       SMPTE RP 219-2002.

       The "testsrc" source generates a test video pattern,  showing  a	 color
       pattern,	 a scrolling gradient and a timestamp. This is mainly intended
       for testing purposes.

       The "testsrc2" source is similar to testsrc, but	 supports  more	 pixel
       formats	instead	 of just "rgb24". This allows using it as an input for
       other tests without requiring a format conversion.

       The "yuvtestsrc" source generates an YUV test pattern. You should see a
       y, cb and cr stripe from top to bottom.

       The sources accept the following parameters:

       level
	   Specify  the	 level	of  the	 Hald  CLUT,  only  available  in  the
	   "haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
	   by  "N*N*N"	pixels	to  be	used  as identity matrix for 3D lookup
	   tables. Each component is coded on a "1/(N*N)" scale.

       color, c
	   Specify the color of the source,  only  available  in  the  "color"
	   source. For the syntax of this option, check the "Color" section in
	   the ffmpeg-utils manual.

       size, s
	   Specify  the	 size  of  the	sourced	 video. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   The default value is "320x240".

	   This	 option	 is  not  available  with  the "allrgb", "allyuv", and
	   "haldclutsrc" filters.

       rate, r
	   Specify the frame rate of the  sourced  video,  as  the  number  of
	   frames  generated  per  second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a floating	 point
	   number  or a valid video frame rate abbreviation. The default value
	   is "25".

       duration, d
	   Set the duration of	the  sourced  video.  See  the	Time  duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If  not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

	   Since the frame rate is used as time base, all frames including the
	   last one will have their full duration. If the  specified  duration
	   is not a multiple of the frame duration, it will be rounded up.

       sar Set the sample aspect ratio of the sourced video.

       alpha
	   Specify  the	 alpha	(opacity) of the background, only available in
	   the	"testsrc2"  source.  The  value	 must  be  between  0	(fully
	   transparent) and 255 (fully opaque, the default).

       decimals, n
	   Set the number of decimals to show in the timestamp, only available
	   in the "testsrc" source.

	   The	displayed  timestamp  value  will  correspond  to the original
	   timestamp value multiplied by the power  of	10  of	the  specified
	   value. Default value is 0.

       type
	   Set	the  type  of  the  color  spectrum,  only  available  in  the
	   "colorspectrum" source. Can be one of the following:

	   black
	   white
	   all
       patch_size
	   Set patch size  of  single  color  patch,  only  available  in  the
	   "colorchart" source. Default is "64x64".

       preset
	   Set	colorchecker colors preset, only available in the "colorchart"
	   source.

	   Available values are:

	   reference
	   skintones

	   Default value is "reference".

       Examples

       •   Generate a video with a duration of 5.3 seconds, with size  176x144
	   and a frame rate of 10 frames per second:

		   testsrc=duration=5.3:size=qcif:rate=10

       •   The	following graph description will generate a red source with an
	   opacity of 0.2, with size "qcif" and a frame rate of 10 frames  per
	   second:

		   color=c=red@0.2:s=qcif:r=10

       •   If  the  input content is to be ignored, "nullsrc" can be used. The
	   following command generates noise in the luma  plane	 by  employing
	   the "geq" filter:

		   nullsrc=s=256x256, geq=random(1)*255:128:128

       Commands

       The "color" source supports the following commands:

       c, color
	   Set	the color of the created image. Accepts the same syntax of the
	   corresponding color option.

   openclsrc
       Generate video using an OpenCL program.

       source
	   OpenCL program source file.

       kernel
	   Kernel name in program.

       size, s
	   Size of frames to generate.	This must be set.

       format
	   Pixel format to use for the generated frames.  This must be set.

       rate, r
	   Number of frames generated every second.  Default value is '25'.

       For details of how the program loading works,  see  the	program_opencl
       filter.

       Example programs:

       •   Generate a colour ramp by setting pixel values from the position of
	   the	pixel in the output image.  (Note that this will work with all
	   pixel formats, but the generated output will not be the same.)

		   __kernel void ramp(__write_only image2d_t dst,
				      unsigned int index)
		   {
		       int2 loc = (int2)(get_global_id(0), get_global_id(1));

		       float4 val;
		       val.xy = val.zw = convert_float2(loc) / convert_float2(get_image_dim(dst));

		       write_imagef(dst, loc, val);
		   }

       •   Generate a Sierpinski carpet pattern, panning  by  a	 single	 pixel
	   each frame.

		   __kernel void sierpinski_carpet(__write_only image2d_t dst,
						   unsigned int index)
		   {
		       int2 loc = (int2)(get_global_id(0), get_global_id(1));

		       float4 value = 0.0f;
		       int x = loc.x + index;
		       int y = loc.y + index;
		       while (x > 0 || y > 0) {
			   if (x % 3 == 1 && y % 3 == 1) {
			       value = 1.0f;
			       break;
			   }
			   x /= 3;
			   y /= 3;
		       }

		       write_imagef(dst, loc, value);
		   }

   sierpinski
       Generate a Sierpinski carpet/triangle fractal, and randomly pan around.

       This source accepts the following options:

       size, s
	   Set	frame  size.  For  the syntax of this option, check the "Video
	   size"  section  in  the  ffmpeg-utils  manual.  Default  value   is
	   "640x480".

       rate, r
	   Set	frame  rate, expressed as number of frames per second. Default
	   value is "25".

       seed
	   Set seed which is used for random panning.

       jump
	   Set max jump for single pan destination. Allowed range is from 1 to
	   10000.

       type
	   Set fractal type, can be default "carpet" or "triangle".

   zoneplate
       Generate a zoneplate test video pattern.

       This source accepts the following options:

       size, s
	   Set frame size. For the syntax of this  option,  check  the	"Video
	   size"   section  in	the  ffmpeg-utils  manual.  Default  value  is
	   "320x240".

       rate, r
	   Set frame rate, expressed as number of frames per  second.  Default
	   value is "25".

       duration, d
	   Set	the  duration  of  the	sourced	 video.	 See the Time duration
	   section in the ffmpeg-utils(1) manual for the accepted syntax.

	   If not specified, or the expressed duration is negative, the	 video
	   is supposed to be generated forever.

       sar Set the sample aspect ratio of the sourced video.

       precision
	   Set	precision  in  bits  for  look-up table for sine calculations.
	   Default value is 10.	 Allowed range is from 4 to 16.

       xo  Set horizontal axis offset for output signal. Default value is 0.

       yo  Set vertical axis offset for output signal. Default value is 0.

       to  Set time axis offset for output signal. Default value is 0.

       k0  Set 0-order, constant added to signal phase. Default value is 0.

       kx  Set 1-order, phase factor multiplier for horizontal	axis.  Default
	   value is 0.

       ky  Set	1-order,  phase	 factor	 multiplier for vertical axis. Default
	   value is 0.

       kt  Set 1-order, phase factor multiplier for time axis.	Default	 value
	   is 0.

       kxt, kyt, kxy
	   Set	phase  factor  multipliers  for	 combination  of  spatial  and
	   temporal axis.  Default value is 0.

       kx2 Set 2-order, phase factor multiplier for horizontal	axis.  Default
	   value is 0.

       ky2 Set	2-order,  phase	 factor	 multiplier for vertical axis. Default
	   value is 0.

       kt2 Set 2-order, phase factor multiplier for time axis.	Default	 value
	   is 0.

       ku  Set	the  constant  added  to  final	 phase	to produce chroma-blue
	   component of signal.	 Default value is 0.

       kv  Set the  constant  added  to	 final	phase  to  produce  chroma-red
	   component of signal.	 Default value is 0.

       Commands

       This source supports the some above options as commands.

       Examples

       •   Generate horizontal color sine sweep:

		   zoneplate=ku=512:kv=0:kt2=0:kx2=256:s=wvga:xo=-426:kt=11

       •   Generate vertical color sine sweep:

		   zoneplate=ku=512:kv=0:kt2=0:ky2=156:s=wvga:yo=-240:kt=11

       •   Generate circular zone-plate:

		   zoneplate=ku=512:kv=100:kt2=0:ky2=256:kx2=556:s=wvga:yo=0:kt=11

VIDEO SINKS
       Below is a description of the currently available video sinks.

   buffersink
       Buffer  video  frames, and make them available to the end of the filter
       graph.

       This sink is  mainly  intended  for  programmatic  use,	in  particular
       through	the  interface	defined	 in  libavfilter/buffersink.h  or  the
       options system.

       It accepts a pointer to an AVBufferSinkContext structure, which defines
       the incoming buffers' formats, to be passed as the opaque parameter  to
       "avfilter_init_filter" for initialization.

   nullsink
       Null  video  sink:  do  absolutely  nothing with the input video. It is
       mainly useful as a template and for use in analysis / debugging tools.

MULTIMEDIA FILTERS
       Below is a description of the currently available multimedia filters.

   a3dscope
       Convert input audio to 3d scope video output.

       The filter accepts the following options:

       rate, r
	   Set frame rate, expressed as number of frames per  second.  Default
	   value is "25".

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "hd720".

       fov Set the camera field of view. Default is 90 degrees.	 Allowed range
	   is from 40 to 150.

       roll
	   Set the camera roll.

       pitch
	   Set the camera pitch.

       yaw Set the camera yaw.

       xzoom
	   Set the camera zoom on X-axis.

       yzoom
	   Set the camera zoom on Y-axis.

       zzoom
	   Set the camera zoom on Z-axis.

       xpos
	   Set the camera position on X-axis.

       ypos
	   Set the camera position on Y-axis.

       zpos
	   Set the camera position on Z-axis.

       length
	   Set the length of displayed audio waves in number of frames.

       Commands

       Filter supports the some above options as commands.

   abitscope
       Convert input audio to a video output, displaying the audio bit scope.

       The filter accepts the following options:

       rate, r
	   Set	frame  rate, expressed as number of frames per second. Default
	   value is "25".

       size, s
	   Specify the video size for the  output.  For	 the  syntax  of  this
	   option,  check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "1024x256".

       colors
	   Specify list of colors separated by space or by '|' which  will  be
	   used	 to  draw  channels.  Unrecognized  or	missing colors will be
	   replaced by white color.

       mode, m
	   Set output mode. Can be "bars" or "trace". Default is "bars".

   adrawgraph
       Draw a graph using input audio metadata.

       See drawgraph

   agraphmonitor
       See graphmonitor.

   ahistogram
       Convert input audio to a video output, displaying the volume histogram.

       The filter accepts the following options:

       dmode
	   Specify how histogram is calculated.

	   It accepts the following values:

	   single
	       Use single histogram for all channels.

	   separate
	       Use separate histogram for each channel.

	   Default is "single".

       rate, r
	   Set frame rate, expressed as number of frames per  second.  Default
	   value is "25".

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "hd720".

       scale
	   Set display scale.

	   It accepts the following values:

	   log logarithmic

	   sqrt
	       square root

	   cbrt
	       cubic root

	   lin linear

	   rlog
	       reverse logarithmic

	   Default is "log".

       ascale
	   Set amplitude scale.

	   It accepts the following values:

	   log logarithmic

	   lin linear

	   Default is "log".

       acount
	   Set	how  much  frames  to  accumulate in histogram.	 Default is 1.
	   Setting this to -1 accumulates all frames.

       rheight
	   Set histogram ratio of window height.

       slide
	   Set sonogram sliding.

	   It accepts the following values:

	   replace
	       replace old rows with new ones.

	   scroll
	       scroll from top to bottom.

	   Default is "replace".

       hmode
	   Set histogram mode.

	   It accepts the following values:

	   abs Use absolute values of samples.

	   sign
	       Use untouched values of samples.

	   Default is "abs".

   aphasemeter
       Measures	 phase	of  input  audio,  which  is  exported	 as   metadata
       "lavfi.aphasemeter.phase",  representing	 mean  phase  of current audio
       frame. A video output can also be produced and is enabled  by  default.
       The audio is passed through as first output.

       Audio  will  be	rematrixed  to	stereo	if  it has a different channel
       layout. Phase value is in range "[-1, 1]" where -1 means left and right
       channels are completely out of phase and 1 means channels are in phase.

       The filter accepts the following options,  all  related	to  its	 video
       output:

       rate, r
	   Set the output frame rate. Default value is 25.

       size, s
	   Set	the  video size for the output. For the syntax of this option,
	   check the "Video size" section in the ffmpeg-utils manual.  Default
	   value is "800x400".

       rc
       gc
       bc  Specify the red, green, blue contrast. Default values are 2, 7  and
	   1.  Allowed range is "[0, 255]".

       mpc Set	color which will be used for drawing median phase. If color is
	   "none" which is default, no median phase value will be drawn.

       video
	   Enable video output. Default is enabled.

       phasing detection

       The filter also detects out of  phase  and  mono	 sequences  in	stereo
       streams.	  It  logs  the sequence start, end and duration when it lasts
       longer or as long as the minimum set.

       The filter accepts the following options for this detection:

       phasing
	   Enable mono and out of phase detection. Default is disabled.

       tolerance, t
	   Set phase tolerance for mono detection, in amplitude ratio. Default
	   is 0.  Allowed range is "[0, 1]".

       angle, a
	   Set angle threshold for out of phase detection, in degree.  Default
	   is 170.  Allowed range is "[90, 180]".

       duration, d
	   Set	mono or out of phase duration until notification, expressed in
	   seconds. Default is 2.

       Examples

       •   Complete example with ffmpeg to detect 1 second of mono with	 0.001
	   phase tolerance:

		   ffmpeg -i stereo.wav -af aphasemeter=video=0:phasing=1:duration=1:tolerance=0.001 -f null -

   avectorscope
       Convert	input  audio  to a video output, representing the audio vector
       scope.

       The filter is used to measure the difference between channels of stereo
       audio stream. A monaural signal, consisting of identical left and right
       signal, results in straight vertical line.  Any	stereo	separation  is
       visible as a deviation from this line, creating a Lissajous figure.  If
       the  straight  (or  deviation from it) but horizontal line appears this
       indicates that the left and right channels are out of phase.

       The filter accepts the following options:

       mode, m
	   Set the vectorscope mode.

	   Available values are:

	   lissajous
	       Lissajous rotated by 45 degrees.

	   lissajous_xy
	       Same as above but not rotated.

	   polar
	       Shape resembling half of circle.

	   Default value is lissajous.

       size, s
	   Set the video size for the output. For the syntax of	 this  option,
	   check the "Video size" section in the ffmpeg-utils manual.  Default
	   value is "400x400".

       rate, r
	   Set the output frame rate. Default value is 25.

       rc
       gc
       bc
       ac  Specify the red, green, blue and alpha contrast. Default values are
	   40, 160, 80 and 255.	 Allowed range is "[0, 255]".

       rf
       gf
       bf
       af  Specify the red, green, blue and alpha fade. Default values are 15,
	   10, 5 and 5.	 Allowed range is "[0, 255]".

       zoom
	   Set	the  zoom  factor.  Default  value is 1. Allowed range is "[0,
	   10]".  Values lower than 1 will auto adjust zoom factor to  maximal
	   possible value.

       draw
	   Set the vectorscope drawing mode.

	   Available values are:

	   dot Draw dot for each sample.

	   line
	       Draw line between previous and current sample.

	   aaline
	       Draw anti-aliased line between previous and current sample.

	   Default value is dot.

       scale
	   Specify amplitude scale of audio samples.

	   Available values are:

	   lin Linear.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   log Logarithmic.

       swap
	   Swap left channel axis with right channel axis.

       mirror
	   Mirror axis.

	   none
	       No mirror.

	   x   Mirror only x axis.

	   y   Mirror only y axis.

	   xy  Mirror both axis.

       Examples

       •   Complete example using ffplay:

		   ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
				[a] avectorscope=zoom=1.3:rc=2:gc=200:bc=10:rf=1:gf=8:bf=7 [out0]'

       Commands

       This  filter  supports the all above options as commands except options
       "size" and "rate".

   bench, abench
       Benchmark part of a filtergraph.

       The filter accepts the following options:

       action
	   Start or stop a timer.

	   Available values are:

	   start
	       Get the current time, set it as frame metadata (using  the  key
	       "lavfi.bench.start_time"),  and	forward	 the frame to the next
	       filter.

	   stop
	       Get the current time  and  fetch	 the  "lavfi.bench.start_time"
	       metadata	 from  the  input  frame  metadata  to	get  the  time
	       difference. Time difference, average, maximum and minimum  time
	       (respectively  "t",  "avg",  "max" and "min") are then printed.
	       The timestamps are expressed in seconds.

       Examples

       •   Benchmark selectivecolor filter:

		   bench=start,selectivecolor=reds=-.2 .12 -.49,bench=stop

   concat
       Concatenate audio and video streams, joining them  together  one	 after
       the other.

       The  filter  works on segments of synchronized video and audio streams.
       All segments must have the same number of streams  of  each  type,  and
       that will also be the number of streams at output.

       The filter accepts the following options:

       n   Set the number of segments. Default is 2.

       v   Set	the number of output video streams, that is also the number of
	   video streams in each segment. Default is 1.

       a   Set the number of output audio streams, that is also the number  of
	   audio streams in each segment. Default is 0.

       unsafe
	   Activate  unsafe  mode:  do	not  fail if segments have a different
	   format.

       The filter has v+a  outputs:  first  v  video  outputs,	then  a	 audio
       outputs.

       There  are  nx(v+a)  inputs: first the inputs for the first segment, in
       the same order as the outputs, then the inputs for the second  segment,
       etc.

       Related	streams	 do  not  always  have	exactly the same duration, for
       various reasons including codec frame size  or  sloppy  authoring.  For
       that  reason,  related synchronized streams (e.g. a video and its audio
       track) should be concatenated at once. The concat filter will  use  the
       duration	 of  the longest stream in each segment (except the last one),
       and if necessary pad shorter audio streams with silence.

       For this filter to work correctly, all segments must start at timestamp
       0.

       All  corresponding  streams  must  have	the  same  parameters  in  all
       segments; the filtering system will automatically select a common pixel
       format  for  video streams, and a common sample format, sample rate and
       channel	layout	for  audio  streams,  but  other  settings,  such   as
       resolution, must be converted explicitly by the user.

       Different  frame rates are acceptable but will result in variable frame
       rate at output; be sure to configure the output file to handle it.

       Examples

       •   Concatenate an opening, an episode and an ending, all in  bilingual
	   version (video in stream 0, audio in streams 1 and 2):

		   ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \
		     '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2]
		      concat=n=3:v=1:a=2 [v] [a1] [a2]' \
		     -map '[v]' -map '[a1]' -map '[a2]' output.mkv

       •   Concatenate	two  parts, handling audio and video separately, using
	   the (a)movie sources, and adjusting the resolution:

		   movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ;
		   movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ;
		   [v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa]

	   Note that a desync will happen at the stitch if the audio and video
	   streams do not have exactly the same duration in the first file.

       Commands

       This filter supports the following commands:

       next
	   Close the current segment and step to the next one

   ebur128
       EBU R128 scanner filter. This filter takes an audio stream and analyzes
       its loudness level. By default, it logs a message  at  a	 frequency  of
       10Hz  with  the	Momentary  loudness  (identified  by  "M"), Short-term
       loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

       The filter can only analyze streams which have sample format is double-
       precision floating point. The input stream will be  converted  to  this
       specification,  if  needed.  Users  may	need  to insert aformat and/or
       aresample filters after this filter to obtain the original parameters.

       The filter also has a video output (see the video option) with  a  real
       time  graph to observe the loudness evolution. The graphic contains the
       logged message mentioned above, so it is not printed anymore when  this
       option  is  set,	 unless	 the verbose logging is set. The main graphing
       area contains the short-term loudness (3 seconds of analysis), and  the
       gauge  on  the  right is for the momentary loudness (400 milliseconds),
       but can optionally be configured to instead display short-term loudness
       (see gauge).

       The green area marks a  +/- 1LU target range around the target loudness
       (-23LUFS by default, unless modified through target).

       More  information  about	 the  Loudness	Recommendation	EBU  R128   on
       <http://tech.ebu.ch/loudness>.

       The filter accepts the following options:

       video
	   Activate  the  video	 output.  The audio stream is passed unchanged
	   whether this option is set or no. The  video	 stream	 will  be  the
	   first output stream if activated. Default is 0.

       size
	   Set	the  video size. This option is for video only. For the syntax
	   of this option, check the "Video size" section in the  ffmpeg-utils
	   manual.  Default and minimum resolution is "640x480".

       meter
	   Set	the EBU scale meter. Default is 9. Common values are 9 and 18,
	   respectively for EBU scale meter +9 and EBU scale  meter  +18.  Any
	   other integer value between this range is allowed.

       metadata
	   Set	metadata  injection.  If  set  to  1,  the audio input will be
	   segmented into 100ms output frames, each of them containing various
	   loudness information	 in  metadata.	 All  the  metadata  keys  are
	   prefixed with "lavfi.r128.".

	   Default is 0.

       framelog
	   Force the frame logging level.

	   Available values are:

	   quiet
	       logging disabled

	   info
	       information logging level

	   verbose
	       verbose logging level

	   By  default,	 the logging level is set to info. If the video or the
	   metadata options are set, it switches to verbose.

       peak
	   Set peak mode(s).

	   Available modes can be cumulated (the option	 is  a	"flag"	type).
	   Possible values are:

	   none
	       Disable any peak mode (default).

	   sample
	       Enable sample-peak mode.

	       Simple peak mode looking for the higher sample value. It logs a
	       message for sample-peak (identified by "SPK").

	   true
	       Enable true-peak mode.

	       If  enabled, the peak lookup is done on an over-sampled version
	       of the input stream for better peak accuracy. It logs a message
	       for true-peak.  (identified by "TPK") and true-peak  per	 frame
	       (identified  by	"FTPK").   This	 mode  requires	 a  build with
	       "libswresample".

       dualmono
	   Treat mono input files as "dual mono". If a mono file  is  intended
	   for	playback  on a stereo system, its EBU R128 measurement will be
	   perceptually	 incorrect.   If  set  to  "true",  this  option  will
	   compensate  for  this  effect.   Multi-channel  input files are not
	   affected by this option.

       panlaw
	   Set a specific pan law to be used for the measurement of dual  mono
	   files.   This  parameter  is	 optional,  and has a default value of
	   -3.01dB.

       target
	   Set a specific target level (in LUFS) used as relative zero in  the
	   visualization.   This parameter is optional and has a default value
	   of -23LUFS as specified by EBU R128.	 However,  material  published
	   online may prefer a level of -16LUFS (e.g. for use with podcasts or
	   video platforms).

       gauge
	   Set	the value displayed by the gauge. Valid values are "momentary"
	   and s "shortterm". By default the momentary value will be used, but
	   in certain scenarios it may be more useful  to  observe  the	 short
	   term value instead (e.g.  live mixing).

       scale
	   Sets	 the  display  scale  for  the	loudness. Valid parameters are
	   "absolute" (in LUFS) or "relative" (LU)  relative  to  the  target.
	   This	 only  affects the video output, not the summary or continuous
	   log output.

       integrated
	   Read-only exported value for measured integrated loudness, in LUFS.

       range
	   Read-only exported value for measured loudness range, in LU.

       lra_low
	   Read-only exported value for measured LRA low, in LUFS.

       lra_high
	   Read-only exported value for measured LRA high, in LUFS.

       sample_peak
	   Read-only exported value for measured sample peak, in dBFS.

       true_peak
	   Read-only exported value for measured true peak, in dBFS.

       Examples

       •   Real-time graph using ffplay, with a EBU scale meter +18:

		   ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]"

       •   Run an analysis with ffmpeg:

		   ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null -

   interleave, ainterleave
       Temporally interleave frames from several inputs.

       "interleave" works with video inputs, "ainterleave" with audio.

       These filters read frames from  several	inputs	and  send  the	oldest
       queued frame to the output.

       Input  streams  must  have well defined, monotonically increasing frame
       timestamp values.

       In order to submit one frame to output, these filters need  to  enqueue
       at  least  one  frame  for  each input, so they cannot work in case one
       input is not yet terminated and will not receive incoming frames.

       For example consider the case when one input is a "select" filter which
       always drops input frames. The "interleave" filter  will	 keep  reading
       from that input, but it will never be able to send new frames to output
       until the input sends an end-of-stream signal.

       Also, depending on inputs synchronization, the filters will drop frames
       in  case	 one  input  receives more frames than the other ones, and the
       queue is already filled.

       These filters accept the following options:

       nb_inputs, n
	   Set the number of different inputs, it is 2 by default.

       duration
	   How to determine the end-of-stream.

	   longest
	       The duration of the longest input. (default)

	   shortest
	       The duration of the shortest input.

	   first
	       The duration of the first input.

       Examples

       •   Interleave frames belonging to different streams using ffmpeg:

		   ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

       •   Add flickering blur effect:

		   select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

   latency, alatency
       Measure filtering latency.

       Report previous filter filtering latency,  delay	 in  number  of	 audio
       samples for audio filters or number of video frames for video filters.

       On end of input stream, filter will report min and max measured latency
       for previous running filter in filtergraph.

   metadata, ametadata
       Manipulate frame metadata.

       This filter accepts the following options:

       mode
	   Set mode of operation of the filter.

	   Can be one of the following:

	   select
	       If both "value" and "key" is set, select frames which have such
	       metadata.  If  only  "key"  is set, select every frame that has
	       such key in metadata.

	   add Add new metadata "key" and "value". If key is already available
	       do nothing.

	   modify
	       Modify value of already present key.

	   delete
	       If "value" is set, delete  only	keys  that  have  such	value.
	       Otherwise, delete key. If "key" is not set, delete all metadata
	       values in the frame.

	   print
	       Print  key and its value if metadata was found. If "key" is not
	       set print all metadata values available in frame.

       key Set key used with all modes. Must  be  set  for  all	 modes	except
	   "print" and "delete".

       value
	   Set metadata value which will be used. This option is mandatory for
	   "modify" and "add" mode.

       function
	   Which function to use when comparing metadata value and "value".

	   Can be one of following:

	   same_str
	       Values  are  interpreted	 as  strings, returns true if metadata
	       value is same as "value".

	   starts_with
	       Values are interpreted as strings,  returns  true  if  metadata
	       value starts with the "value" option string.

	   less
	       Values  are  interpreted	 as  floats,  returns true if metadata
	       value is less than "value".

	   equal
	       Values are interpreted as floats, returns true  if  "value"  is
	       equal with metadata value.

	   greater
	       Values  are  interpreted	 as  floats,  returns true if metadata
	       value is greater than "value".

	   expr
	       Values are interpreted as floats, returns  true	if  expression
	       from option "expr" evaluates to true.

	   ends_with
	       Values  are  interpreted	 as  strings, returns true if metadata
	       value ends with the "value" option string.

       expr
	   Set expression which is used when "function" is set to "expr".  The
	   expression is evaluated through the eval API and  can  contain  the
	   following constants:

	   VALUE1, FRAMEVAL
	       Float representation of "value" from metadata key.

	   VALUE2, USERVAL
	       Float  representation of "value" as supplied by user in "value"
	       option.

       file
	   If specified in "print" mode, output is written to the named	 file.
	   Instead  of	plain  filename	 any  writable	url  can be specified.
	   Filename ``-'' is a shorthand for standard output. If "file" option
	   is not set, output is written to the log with AV_LOG_INFO loglevel.

       direct
	   Reduces buffering in print mode when output is written to a URL set
	   using file.

       Examples

       •   Print    all	   metadata    values	 for	frames	  with	   key
	   "lavfi.signalstats.YDIF" with values between 0 and 1.

		   signalstats,metadata=print:key=lavfi.signalstats.YDIF:value=0:function=expr:expr='between(VALUE1,0,1)'

       •   Print silencedetect output to file metadata.txt.

		   silencedetect,ametadata=mode=print:file=metadata.txt

       •   Direct all metadata to a pipe with file descriptor 4.

		   metadata=mode=print:file='pipe\:4'

   perms, aperms
       Set read/write permissions for the output frames.

       These filters are mainly aimed at developers to test direct path in the
       following filter in the filtergraph.

       The filters accept the following options:

       mode
	   Select the permissions mode.

	   It accepts the following values:

	   none
	       Do nothing. This is the default.

	   ro  Set all the output frames read-only.

	   rw  Set all the output frames directly writable.

	   toggle
	       Make  the  frame	 read-only  if writable, and writable if read-
	       only.

	   random
	       Set each output frame read-only or writable randomly.

       seed
	   Set the seed for the random	mode,  must  be	 an  integer  included
	   between  0 and "UINT32_MAX". If not specified, or if explicitly set
	   to -1, the filter will try to use a good  random  seed  on  a  best
	   effort basis.

       Note: in case of auto-inserted filter between the permission filter and
       the  following one, the permission might not be received as expected in
       that following filter. Inserting a format or aformat filter before  the
       perms/aperms filter can avoid this problem.

   realtime, arealtime
       Slow down filtering to match real time approximately.

       These filters will pause the filtering for a variable amount of time to
       match  the  output rate with the input timestamps.  They are similar to
       the re option to "ffmpeg".

       They accept the following options:

       limit
	   Time limit for the pauses. Any  pause  longer  than	that  will  be
	   considered  a  timestamp discontinuity and reset the timer. Default
	   is 2 seconds.

       speed
	   Speed factor for processing. The value must be a float larger  than
	   zero.   Values  larger than 1.0 will result in faster than realtime
	   processing,	smaller	 will  slow  processing	 down.	The  limit  is
	   automatically adapted accordingly. Default is 1.0.

	   A  processing  speed	 faster	 than  what  is possible without these
	   filters cannot be achieved.

       Commands

       Both filters supports the all above options as commands.

   segment, asegment
       Split single input stream into multiple streams.

       This filter does opposite of concat filters.

       "segment" works on video frames, "asegment" on audio samples.

       This filter accepts the following options:

       timestamps
	   Timestamps of output segments separated by '|'. The	first  segment
	   will	 run  from the beginning of the input stream. The last segment
	   will run until the end of the input stream

       frames, samples
	   Exact frame/sample count to split the segments.

       In all cases, prefixing an each segment with '+' will make it  relative
       to the previous segment.

       Examples

       •   Split  input audio stream into three output audio streams, starting
	   at start of input audio stream and storing that in 1st output audio
	   stream, then following at 60th  second  and	storing	 than  in  2nd
	   output  audio  stream,  and	last after 150th second of input audio
	   stream store in 3rd output audio stream:

		   asegment=timestamps="60|150"

   select, aselect
       Select frames to pass in output.

       This filter accepts the following options:

       expr, e
	   Set expression, which is evaluated for each input frame.

	   If the expression is evaluated to zero, the frame is discarded.

	   If the evaluation result is negative or NaN, the frame is  sent  to
	   the	first  output;	otherwise  it is sent to the output with index
	   "ceil(val)-1", assuming that the input index starts from 0.

	   For example a value of 1.2 corresponds to  the  output  with	 index
	   "ceil(1.2)-1 = 2-1 = 1", that is the second output.

       outputs, n
	   Set the number of outputs. The output to which to send the selected
	   frame is based on the result of the evaluation. Default value is 1.

       The expression can contain the following constants:

       n   The (sequential) number of the filtered frame, starting from 0.

       selected_n
	   The (sequential) number of the selected frame, starting from 0.

       prev_selected_n
	   The	sequential  number  of	the  last  selected frame. It's NAN if
	   undefined.

       TB  The timebase of the input timestamps.

       pts The PTS (Presentation TimeStamp) of the filtered  frame,  expressed
	   in TB units. It's NAN if undefined.

       t   The	PTS  of	 the filtered frame, expressed in seconds. It's NAN if
	   undefined.

       prev_pts
	   The PTS of the previously filtered frame. It's NAN if undefined.

       prev_selected_pts
	   The PTS  of	the  last  previously  filtered	 frame.	 It's  NAN  if
	   undefined.

       prev_selected_t
	   The	PTS  of	 the  last  previously	selected  frame,  expressed in
	   seconds. It's NAN if undefined.

       start_pts
	   The first PTS in the stream which is not NAN. It remains NAN if not
	   found.

       start_t
	   The first PTS, in seconds, in the  stream  which  is	 not  NAN.  It
	   remains NAN if not found.

       pict_type (video only)
	   The	type of the filtered frame. It can assume one of the following
	   values:

	   I
	   P
	   B
	   S
	   SI
	   SP
	   BI
       interlace_type (video only)
	   The frame interlace type.  It  can  assume  one  of	the  following
	   values:

	   PROGRESSIVE
	       The frame is progressive (not interlaced).

	   TOPFIRST
	       The frame is top-field-first.

	   BOTTOMFIRST
	       The frame is bottom-field-first.

       consumed_sample_n (audio only)
	   the number of selected samples before the current frame

       samples_n (audio only)
	   the number of samples in the current frame

       sample_rate (audio only)
	   the input sample rate

       key This is 1 if the filtered frame is a key-frame, 0 otherwise.

       pos the	position  in  the  file	 of  the  filtered  frame,  -1	if the
	   information	is  not	 available   (e.g.   for   synthetic   video);
	   deprecated, do not use

       scene (video only)
	   value between 0 and 1 to indicate a new scene; a low value reflects
	   a  low  probability for the current frame to introduce a new scene,
	   while a higher value means the current frame is more likely	to  be
	   one (see the example below)

       concatdec_select
	   The	concat	demuxer can select only part of a concat input file by
	   setting an inpoint and an outpoint, but the output packets may  not
	   be  entirely	 contained  in	the  selected  interval. By using this
	   variable, it is possible to skip frames  generated  by  the	concat
	   demuxer which are not exactly contained in the selected interval.

	   This	   works    by	 comparing   the   frame   pts	 against   the
	   lavf.concat.start_time and the lavf.concat.duration packet metadata
	   values which are also present in the decoded frames.

	   The concatdec_select variable is -1 if the frame pts	 is  at	 least
	   start_time and either the duration metadata is missing or the frame
	   pts is less than start_time + duration, 0 otherwise, and NaN if the
	   start_time metadata is missing.

	   That	 basically means that an input frame is selected if its pts is
	   within the interval set by the concat demuxer.

       The default value of the select expression is "1".

       Examples

       •   Select all frames in input:

		   select

	   The example above is the same as:

		   select=1

       •   Skip all frames:

		   select=0

       •   Select only I-frames:

		   select='eq(pict_type\,I)'

       •   Select one frame every 100:

		   select='not(mod(n\,100))'

       •   Select only frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)

       •   Select only I-frames contained in the 10-20 time interval:

		   select=between(t\,10\,20)*eq(pict_type\,I)

       •   Select frames with a minimum distance of 10 seconds:

		   select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)'

       •   Use aselect to select only audio frames with samples number > 100:

		   aselect='gt(samples_n\,100)'

       •   Create a mosaic of the first scenes:

		   ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png

	   Comparing scene against a value between 0.3 and 0.5 is generally  a
	   sane choice.

       •   Send even and odd frames to separate outputs, and compose them:

		   select=n=2:e='mod(n, 2)+1' [odd][even]; [odd] pad=h=2*ih [tmp]; [tmp][even] overlay=y=h

       •   Select  useful frames from an ffconcat file which is using inpoints
	   and outpoints but where the source files are not intra frame only.

		   ffmpeg -copyts -vsync 0 -segment_time_metadata 1 -i input.ffconcat -vf select=concatdec_select -af aselect=concatdec_select output.avi

   sendcmd, asendcmd
       Send commands to filters in the filtergraph.

       These filters read  commands  to	 be  sent  to  other  filters  in  the
       filtergraph.

       "sendcmd"  must	be inserted between two video filters, "asendcmd" must
       be inserted between two audio filters, but apart from that they act the
       same way.

       The specification of commands can be provided in the  filter  arguments
       with  the  commands  option,  or	 in  a	file specified by the filename
       option.

       These filters accept the following options:

       commands, c
	   Set the commands to be read and sent to the other filters.

       filename, f
	   Set the filename of the commands to be read and sent to  the	 other
	   filters.

       Commands syntax

       A   commands   description   consists   of   a	sequence  of  interval
       specifications, comprising a list of commands to	 be  executed  when  a
       particular  event  related to that interval occurs. The occurring event
       is typically the current frame time entering or leaving	a  given  time
       interval.

       An interval is specified by the following syntax:

	       <START>[-<END>] <COMMANDS>;

       The  time  interval  is	specified  by the START and END times.	END is
       optional and defaults to the maximum time.

       The current frame time is considered within the specified  interval  if
       it  is  included in the interval [START, END), that is when the time is
       greater or equal to START and is lesser than END.

       COMMANDS consists of a sequence of one or more command  specifications,
       separated  by  ",", relating to that interval.  The syntax of a command
       specification is given by:

	       [<FLAGS>] <TARGET> <COMMAND> <ARG>

       FLAGS is optional and specifies the type of events relating to the time
       interval which enable sending the specified command, and must be a non-
       null sequence of identifier flags separated by "+" or "|" and  enclosed
       between "[" and "]".

       The following flags are recognized:

       enter
	   The	command	 is  sent  when the current frame timestamp enters the
	   specified interval. In other words, the command is  sent  when  the
	   previous  frame  timestamp  was  not in the given interval, and the
	   current is.

       leave
	   The command is sent when the current	 frame	timestamp  leaves  the
	   specified  interval.	 In  other words, the command is sent when the
	   previous frame timestamp was in the given interval, and the current
	   is not.

       expr
	   The	command	 ARG  is  interpreted  as  expression  and  result  of
	   expression is passed as ARG.

	   The	expression  is	evaluated through the eval API and can contain
	   the following constants:

	   POS Original position in the file of the  frame,  or	 undefined  if
	       undefined for the current frame. Deprecated, do not use.

	   PTS The presentation timestamp in input.

	   N   The  count of the input frame for video or audio, starting from
	       0.

	   T   The time in seconds of the current frame.

	   TS  The start time in seconds of the current command interval.

	   TE  The end time in seconds of the current command interval.

	   TI  The interpolated time of the current command interval, TI =  (T
	       - TS) / (TE - TS).

	   W   The video frame width.

	   H   The video frame height.

       If FLAGS is not specified, a default value of "[enter]" is assumed.

       TARGET  specifies  the  target  of the command, usually the name of the
       filter class or a specific filter instance name.

       COMMAND specifies the name of the command for the target filter.

       ARG is optional and specifies the optional list	of  argument  for  the
       given COMMAND.

       Between	 one  interval	specification  and  another,  whitespaces,  or
       sequences of characters starting with "#" until the end	of  line,  are
       ignored and can be used to annotate comments.

       A  simplified  BNF  description	of  the	 commands specification syntax
       follows:

	       <COMMAND_FLAG>  ::= "enter" | "leave"
	       <COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
	       <COMMAND>       ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
	       <COMMANDS>      ::= <COMMAND> [,<COMMANDS>]
	       <INTERVAL>      ::= <START>[-<END>] <COMMANDS>
	       <INTERVALS>     ::= <INTERVAL>[;<INTERVALS>]

       Examples

       •   Specify audio tempo change at second 4:

		   asendcmd=c='4.0 atempo tempo 1.5',atempo

       •   Target a specific filter instance:

		   asendcmd=c='4.0 atempo@my tempo 1.5',atempo@my

       •   Specify a list of drawtext and hue commands in a file.

		   # show text in the interval 5-10
		   5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
			    [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';

		   # desaturate the image in the interval 15-20
		   15.0-20.0 [enter] hue s 0,
			     [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
			     [leave] hue s 1,
			     [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';

		   # apply an exponential saturation fade-out effect, starting from time 25
		   25 [enter] hue s exp(25-t)

	   A filtergraph allowing to read and process the above	 command  list
	   stored in a file test.cmd, can be specified with:

		   sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

   setpts, asetpts
       Change the PTS (presentation timestamp) of the input frames.

       "setpts" works on video frames, "asetpts" on audio frames.

       This filter accepts the following options:

       expr
	   The	expression  which is evaluated for each frame to construct its
	   timestamp.

       The expression is evaluated through the eval API and  can  contain  the
       following constants:

       FRAME_RATE, FR
	   frame rate, only defined for constant frame-rate video

       PTS The presentation timestamp in input

       N   The	count  of  the input frame for video or the number of consumed
	   samples, not including the current frame for audio,	starting  from
	   0.

       NB_CONSUMED_SAMPLES
	   The	number	of  consumed  samples, not including the current frame
	   (only audio)

       NB_SAMPLES, S
	   The number of samples in the current frame (only audio)

       SAMPLE_RATE, SR
	   The audio sample rate.

       STARTPTS
	   The PTS of the first frame.

       STARTT
	   the time in seconds of the first frame

       INTERLACED
	   State whether the current frame is interlaced.

       T   the time in seconds of the current frame

       POS original position in	 the  file  of	the  frame,  or	 undefined  if
	   undefined for the current frame; deprecated, do not use

       PREV_INPTS
	   The previous input PTS.

       PREV_INT
	   previous input time in seconds

       PREV_OUTPTS
	   The previous output PTS.

       PREV_OUTT
	   previous output time in seconds

       RTCTIME
	   The	wallclock  (RTC) time in microseconds. This is deprecated, use
	   time(0) instead.

       RTCSTART
	   The wallclock (RTC) time at the start of the movie in microseconds.

       TB  The timebase of the input timestamps.

       T_CHANGE
	   Time of the first frame after command was applied or	 time  of  the
	   first frame if no commands.

       Examples

       •   Start counting PTS from zero

		   setpts=PTS-STARTPTS

       •   Apply fast motion effect:

		   setpts=0.5*PTS

       •   Apply slow motion effect:

		   setpts=2.0*PTS

       •   Set fixed rate of 25 frames per second:

		   setpts=N/(25*TB)

       •   Set fixed rate 25 fps with some jitter:

		   setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

       •   Apply an offset of 10 seconds to the input PTS:

		   setpts=PTS+10/TB

       •   Generate  timestamps	 from  a  "live	 source"  and  rebase onto the
	   current timebase:

		   setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

       •   Generate timestamps by counting samples:

		   asetpts=N/SR/TB

       Commands

       Both filters support all above options as commands.

   setrange
       Force color range for the output video frame.

       The "setrange" filter marks the color range  property  for  the	output
       frames.	It  does  not  change  the  input  frame,  but	only  sets the
       corresponding property, which affects  how  the	frame  is  treated  by
       following filters.

       The filter accepts the following options:

       range
	   Available values are:

	   auto
	       Keep the same color range property.

	   unspecified, unknown
	       Set the color range as unspecified.

	   limited, tv, mpeg
	       Set the color range as limited.

	   full, pc, jpeg
	       Set the color range as full.

   settb, asettb
       Set the timebase to use for the output frames timestamps.  It is mainly
       useful for testing timebase configuration.

       It accepts the following parameters:

       expr, tb
	   The expression which is evaluated into the output timebase.

       The  value  for tb is an arithmetic expression representing a rational.
       The expression can contain the constants "AVTB" (the default timebase),
       "intb" (the input timebase) and "sr" (the  sample  rate,	 audio	only).
       Default value is "intb".

       Examples

       •   Set the timebase to 1/25:

		   settb=expr=1/25

       •   Set the timebase to 1/10:

		   settb=expr=0.1

       •   Set the timebase to 1001/1000:

		   settb=1+0.001

       •   Set the timebase to 2*intb:

		   settb=2*intb

       •   Set the default timebase value:

		   settb=AVTB

   showcqt
       Convert	input  audio to a video output representing frequency spectrum
       logarithmically using Brown-Puckette  constant  Q  transform  algorithm
       with direct frequency domain coefficient calculation (but the transform
       itself  is  not	really	constant  Q,  instead the Q factor is actually
       variable/clamped), with musical tone scale, from E0 to D#10.

       The filter accepts the following options:

       size, s
	   Specify the video size for the output. It must  be  even.  For  the
	   syntax  of  this  option,  check  the  "Video  size" section in the
	   ffmpeg-utils manual.	 Default value is "1920x1080".

       fps, rate, r
	   Set the output frame rate. Default value is 25.

       bar_h
	   Set the bargraph height. It must be even. Default value is -1 which
	   computes the bargraph height automatically.

       axis_h
	   Set the axis height. It must be even. Default  value	 is  -1	 which
	   computes the axis height automatically.

       sono_h
	   Set the sonogram height. It must be even. Default value is -1 which
	   computes the sonogram height automatically.

       fullhd
	   Set	the  fullhd resolution. This option is deprecated, use size, s
	   instead. Default value is 1.

       sono_v, volume
	   Specify the sonogram volume expression. It can contain variables:

	   bar_v
	       the bar_v evaluated expression

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   a_weighting(f)
	       A-weighting of equal loudness

	   b_weighting(f)
	       B-weighting of equal loudness

	   c_weighting(f)
	       C-weighting of equal loudness.

	   Default value is 16.

       bar_v, volume2
	   Specify the bargraph volume expression. It can contain variables:

	   sono_v
	       the sono_v evaluated expression

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   a_weighting(f)
	       A-weighting of equal loudness

	   b_weighting(f)
	       B-weighting of equal loudness

	   c_weighting(f)
	       C-weighting of equal loudness.

	   Default value is "sono_v".

       sono_g, gamma
	   Specify the sonogram gamma. Lower gamma  makes  the	spectrum  more
	   contrast,  higher  gamma  makes  the	 spectrum  having  more range.
	   Default value is 3.	Acceptable range is "[1, 7]".

       bar_g, gamma2
	   Specify the bargraph gamma. Default value is 1. Acceptable range is
	   "[1, 7]".

       bar_t
	   Specify the bargraph transparency  level.  Lower  value  makes  the
	   bargraph  sharper.	Default	 value	is 1. Acceptable range is "[0,
	   1]".

       timeclamp, tc
	   Specify the transform timeclamp. At low frequency, there is	trade-
	   off	between	 accuracy  in  time  domain  and  frequency domain. If
	   timeclamp is lower,	event  in  time	 domain	 is  represented  more
	   accurately  (such  as fast bass drum), otherwise event in frequency
	   domain is  represented  more	 accurately  (such  as	bass  guitar).
	   Acceptable range is "[0.002, 1]". Default value is 0.17.

       attack
	   Set attack time in seconds. The default is 0 (disabled). Otherwise,
	   it  limits  future samples by applying asymmetric windowing in time
	   domain, useful when low latency is required. Accepted range is "[0,
	   1]".

       basefreq
	   Specify  the	  transform   base   frequency.	  Default   value   is
	   20.01523126408007475,   which  is  frequency	 50  cents  below  E0.
	   Acceptable range is "[10, 100000]".

       endfreq
	   Specify   the   transform   end   frequency.	  Default   value   is
	   20495.59681441799654,  which	 is  frequency	50  cents  above D#10.
	   Acceptable range is "[10, 100000]".

       coeffclamp
	   This option is deprecated and ignored.

       tlength
	   Specify the transform length in time domain.	 Use  this  option  to
	   control accuracy trade-off between time domain and frequency domain
	   at every frequency sample.  It can contain variables:

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option.

	   Default value is "384*tc/(384+tc*f)".

       count
	   Specify the transform count for every video frame. Default value is
	   6.  Acceptable range is "[1, 30]".

       fcount
	   Specify  the	 transform count for every single pixel. Default value
	   is 0, which makes it computed automatically.	 Acceptable  range  is
	   "[0, 10]".

       fontfile
	   Specify  font  file	for use with freetype to draw the axis. If not
	   specified, use embedded font. Note that drawing with font  file  or
	   embedded  font is not implemented with custom basefreq and endfreq,
	   use axisfile option instead.

       font
	   Specify fontconfig pattern. This has lower priority than  fontfile.
	   The	":" in the pattern may be replaced by "|" to avoid unnecessary
	   escaping.

       fontcolor
	   Specify font color expression. This is arithmetic  expression  that
	   should return integer value 0xRRGGBB. It can contain variables:

	   frequency, freq, f
	       the frequency where it is evaluated

	   timeclamp, tc
	       the value of timeclamp option

	   and functions:

	   midi(f)
	       midi  number of frequency f, some midi numbers: E0(16), C1(24),
	       C2(36), A4(69)

	   r(x), g(x), b(x)
	       red, green, and blue value of intensity x.

	   Default    value    is     "st(0,	 (midi(f)-59.5)/12);	 st(1,
	   if(between(ld(0),0,1),  0.5-0.5*cos(2*PI*ld(0)),  0)); r(1-ld(1)) +
	   b(ld(1))".

       axisfile
	   Specify image file to draw the axis. This option override  fontfile
	   and fontcolor option.

       axis, text
	   Enable/disable drawing text to the axis. If it is set to 0, drawing
	   to  the  axis  is  disabled, ignoring fontfile and axisfile option.
	   Default value is 1.

       csp Set colorspace. The accepted values are:

	   unspecified
	       Unspecified (default)

	   bt709
	       BT.709

	   fcc FCC

	   bt470bg
	       BT.470BG or BT.601-6 625

	   smpte170m
	       SMPTE-170M or BT.601-6 525

	   smpte240m
	       SMPTE-240M

	   bt2020ncl
	       BT.2020 with non-constant luminance

       cscheme
	   Set spectrogram color scheme. This is list of floating point values
	   with	 format	 "left_r|left_g|left_b|right_r|right_g|right_b".   The
	   default is "1|0.5|0|0|0.5|1".

       Examples

       •   Playing audio while showing the spectrum:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'

       •   Same as above, but with frame rate 30 fps:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'

       •   Playing at 1280x720:

		   ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=s=1280x720:count=4 [out0]'

       •   Disable sonogram display:

		   sono_h=0

       •   A1 and its harmonics: A1, A2, (near)E3, A3:

		   ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
				    asplit[a][out1]; [a] showcqt [out0]'

       •   Same as above, but with more accuracy in frequency domain:

		   ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
				    asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'

       •   Custom volume:

		   bar_v=10:sono_v=bar_v*a_weighting(f)

       •   Custom gamma, now spectrum is linear to the amplitude.

		   bar_g=2:sono_g=2

       •   Custom tlength equation:

		   tc=0.33:tlength='st(0,0.17); 384*tc / (384 / ld(0) + tc*f /(1-ld(0))) + 384*tc / (tc*f / ld(0) + 384 /(1-ld(0)))'

       •   Custom  fontcolor and fontfile, C-note is colored green, others are
	   colored blue:

		   fontcolor='if(mod(floor(midi(f)+0.5),12), 0x0000FF, g(1))':fontfile=myfont.ttf

       •   Custom font using fontconfig:

		   font='Courier New,Monospace,mono|bold'

       •   Custom frequency range with custom axis using image file:

		   axisfile=myaxis.png:basefreq=40:endfreq=10000

   showcwt
       Convert input audio to video  output  representing  frequency  spectrum
       using Continuous Wavelet Transform and Morlet wavelet.

       The filter accepts the following options:

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "640x512".

       rate, r
	   Set the output frame rate. Default value is 25.

       scale
	   Set the frequency scale used. Allowed values are:

	   linear
	   log
	   bark
	   mel
	   erbs
	   sqrt
	   cbrt
	   qdrt

	   Default value is "linear".

       iscale
	   Set the intensity scale used. Allowed values are:

	   linear
	   log
	   sqrt
	   cbrt
	   qdrt

	   Default value is "log".

       min Set	the minimum frequency that will be used in output.  Default is
	   20 Hz.

       max Set the maximum frequency that will be used in output.  Default  is
	   20000  Hz.  The real frequency upper limit depends on input audio's
	   sample rate and such will be enforced on this value when it is  set
	   to value greater than Nyquist frequency.

       imin
	   Set the minimum intensity that will be used in output.

       imax
	   Set the maximum intensity that will be used in output.

       logb
	   Set	the  logarithmic  basis	 for  brightness strength when mapping
	   calculated magnitude values to pixel values.	 Allowed range is from
	   0 to 1.  Default value is 0.0001.

       deviation
	   Set	the  frequency	deviation.   Lower  values  than  1  are  more
	   frequency  oriented,	 while	higher	values	than  1	 are more time
	   oriented.  Allowed range is from 0 to 10.  Default value is 1.

       pps Set the number of pixel output per each second in one row.  Allowed
	   range is from 1 to 1024.  Default value is 64.

       mode
	   Set the output visual mode. Allowed values are:

	   magnitude
	       Show magnitude.

	   phase
	       Show only phase.

	   magphase
	       Show combination of magnitude and phase.	 Magnitude  is	mapped
	       to brightness and phase to color.

	   channel
	       Show unique color per channel magnitude.

	   stereo
	       Show unique color per stereo difference.

	   Default value is "magnitude".

       slide
	   Set the output slide method. Allowed values are:

	   replace
	   scroll
	   frame
       direction
	   Set	the  direction	method for output slide method. Allowed values
	   are:

	   lr  Direction from left to right.

	   rl  Direction from right to left.

	   ud  Direction from up to down.

	   du  Direction from down to up.

       bar Set the ratio of bargraph display to display size. Default is 0.

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

   showfreqs
       Convert input audio  to	video  output  representing  the  audio	 power
       spectrum.  Audio amplitude is on Y-axis while frequency is on X-axis.

       The filter accepts the following options:

       size, s
	   Specify  size  of  video.  For the syntax of this option, check the
	   "Video size"	 section  in  the  ffmpeg-utils	 manual.   Default  is
	   "1024x512".

       rate, r
	   Set video rate. Default is 25.

       mode
	   Set	display	 mode.	 This  set  how	 each  frequency  bin  will be
	   represented.

	   It accepts the following values:

	   line
	   bar
	   dot

	   Default is "bar".

       ascale
	   Set amplitude scale.

	   It accepts the following values:

	   lin Linear scale.

	   sqrt
	       Square root scale.

	   cbrt
	       Cubic root scale.

	   log Logarithmic scale.

	   Default is "log".

       fscale
	   Set frequency scale.

	   It accepts the following values:

	   lin Linear scale.

	   log Logarithmic scale.

	   rlog
	       Reverse logarithmic scale.

	   Default is "lin".

       win_size
	   Set window size. Allowed range is from 16 to 65536.

	   Default is 2048

       win_func
	   Set windowing function.

	   It accepts the following values:

	   rect
	   bartlett
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default is "hanning".

       overlap
	   Set window overlap. In range "[0, 1]". Default is  1,  which	 means
	   optimal overlap for selected window function will be picked.

       averaging
	   Set	time averaging. Setting this to 0 will display current maximal
	   peaks.  Default is 1, which means time averaging is disabled.

       colors
	   Specify list of colors separated by space or by '|' which  will  be
	   used	 to  draw  channel frequencies. Unrecognized or missing colors
	   will be replaced by white color.

       cmode
	   Set channel display mode.

	   It accepts the following values:

	   combined
	   separate

	   Default is "combined".

       minamp
	   Set minimum amplitude used in "log" amplitude scaler.

       data
	   Set data display mode.

	   It accepts the following values:

	   magnitude
	   phase
	   delay

	   Default is "magnitude".

       channels
	   Set channels to use when  processing	 audio.	 By  default  all  are
	   processed.

   showspatial
       Convert	stereo input audio to a video output, representing the spatial
       relationship between two channels.

       The filter accepts the following options:

       size, s
	   Specify the video size for the  output.  For	 the  syntax  of  this
	   option,  check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "512x512".

       win_size
	   Set window size. Allowed range is from 1024 to 65536. Default  size
	   is 4096.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default value is "hann".

       rate, r
	   Set output framerate.

   showspectrum
       Convert input audio to a video output, representing the audio frequency
       spectrum.

       The filter accepts the following options:

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "640x512".

       slide
	   Specify how the spectrum should slide along the window.

	   It accepts the following values:

	   replace
	       the samples start again on the left when they reach the right

	   scroll
	       the samples scroll from right to left

	   fullframe
	       frames are only produced when the samples reach the right

	   rscroll
	       the samples scroll from left to right

	   lreplace
	       the samples start again on the right when they reach the left

	   Default value is "replace".

       mode
	   Specify display mode.

	   It accepts the following values:

	   combined
	       all channels are displayed in the same row

	   separate
	       all channels are displayed in separate rows

	   Default value is combined.

       color
	   Specify display color mode.

	   It accepts the following values:

	   channel
	       each channel is displayed in a separate color

	   intensity
	       each channel is displayed using the same color scheme

	   rainbow
	       each channel is displayed using the rainbow color scheme

	   moreland
	       each channel is displayed using the moreland color scheme

	   nebulae
	       each channel is displayed using the nebulae color scheme

	   fire
	       each channel is displayed using the fire color scheme

	   fiery
	       each channel is displayed using the fiery color scheme

	   fruit
	       each channel is displayed using the fruit color scheme

	   cool
	       each channel is displayed using the cool color scheme

	   magma
	       each channel is displayed using the magma color scheme

	   green
	       each channel is displayed using the green color scheme

	   viridis
	       each channel is displayed using the viridis color scheme

	   plasma
	       each channel is displayed using the plasma color scheme

	   cividis
	       each channel is displayed using the cividis color scheme

	   terrain
	       each channel is displayed using the terrain color scheme

	   Default value is channel.

       scale
	   Specify scale used for calculating intensity color values.

	   It accepts the following values:

	   lin linear

	   sqrt
	       square root, default

	   cbrt
	       cubic root

	   log logarithmic

	   4thrt
	       4th root

	   5thrt
	       5th root

	   Default value is sqrt.

       fscale
	   Specify frequency scale.

	   It accepts the following values:

	   lin linear

	   log logarithmic

	   Default value is lin.

       saturation
	   Set	saturation  modifier  for  displayed  colors.  Negative values
	   provide alternative color  scheme.  0  is  no  saturation  at  all.
	   Saturation must be in [-10.0, 10.0] range.  Default value is 1.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default value is "hann".

       orientation
	   Set	orientation  of	 time  vs frequency axis. Can be "vertical" or
	   "horizontal". Default is "vertical".

       overlap
	   Set ratio of overlap window. Default value is 0.  When value	 is  1
	   overlap  is	set  to	 recommended size for specific window function
	   currently used.

       gain
	   Set scale gain for calculating  intensity  color  values.   Default
	   value is 1.

       data
	   Set	which data to display. Can be "magnitude", default or "phase",
	   or unwrapped phase: "uphase".

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

       start
	   Set start frequency from which to display spectrogram.  Default  is
	   0.

       stop
	   Set stop frequency to which to display spectrogram. Default is 0.

       fps Set upper frame rate limit. Default is "auto", unlimited.

       legend
	   Draw time and frequency axes and legends. Default is disabled.

       drange
	   Set dynamic range used to calculate intensity color values. Default
	   is 120 dBFS.	 Allowed range is from 10 to 200.

       limit
	   Set upper limit of input audio samples volume in dBFS. Default is 0
	   dBFS.  Allowed range is from -100 to 100.

       opacity
	   Set	opacity	 strength  when	 using	pixel format output with alpha
	   component.

       The usage is very similar to the showwaves filter; see the examples  in
       that section.

       Examples

       •   Large window with logarithmic color scaling:

		   showspectrum=s=1280x480:scale=log

       •   Complete  example  for  a  colored and sliding spectrum per channel
	   using ffplay:

		   ffplay -f lavfi 'amovie=input.mp3, asplit [a][out1];
				[a] showspectrum=mode=separate:color=intensity:slide=1:scale=cbrt [out0]'

   showspectrumpic
       Convert input audio to a single video  frame,  representing  the	 audio
       frequency spectrum.

       The filter accepts the following options:

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "4096x2048".

       mode
	   Specify display mode.

	   It accepts the following values:

	   combined
	       all channels are displayed in the same row

	   separate
	       all channels are displayed in separate rows

	   Default value is combined.

       color
	   Specify display color mode.

	   It accepts the following values:

	   channel
	       each channel is displayed in a separate color

	   intensity
	       each channel is displayed using the same color scheme

	   rainbow
	       each channel is displayed using the rainbow color scheme

	   moreland
	       each channel is displayed using the moreland color scheme

	   nebulae
	       each channel is displayed using the nebulae color scheme

	   fire
	       each channel is displayed using the fire color scheme

	   fiery
	       each channel is displayed using the fiery color scheme

	   fruit
	       each channel is displayed using the fruit color scheme

	   cool
	       each channel is displayed using the cool color scheme

	   magma
	       each channel is displayed using the magma color scheme

	   green
	       each channel is displayed using the green color scheme

	   viridis
	       each channel is displayed using the viridis color scheme

	   plasma
	       each channel is displayed using the plasma color scheme

	   cividis
	       each channel is displayed using the cividis color scheme

	   terrain
	       each channel is displayed using the terrain color scheme

	   Default value is intensity.

       scale
	   Specify scale used for calculating intensity color values.

	   It accepts the following values:

	   lin linear

	   sqrt
	       square root, default

	   cbrt
	       cubic root

	   log logarithmic

	   4thrt
	       4th root

	   5thrt
	       5th root

	   Default value is log.

       fscale
	   Specify frequency scale.

	   It accepts the following values:

	   lin linear

	   log logarithmic

	   Default value is lin.

       saturation
	   Set	saturation  modifier  for  displayed  colors.  Negative values
	   provide alternative color  scheme.  0  is  no  saturation  at  all.
	   Saturation must be in [-10.0, 10.0] range.  Default value is 1.

       win_func
	   Set window function.

	   It accepts the following values:

	   rect
	   bartlett
	   hann
	   hanning
	   hamming
	   blackman
	   welch
	   flattop
	   bharris
	   bnuttall
	   bhann
	   sine
	   nuttall
	   lanczos
	   gauss
	   tukey
	   dolph
	   cauchy
	   parzen
	   poisson
	   bohman
	   kaiser

	   Default value is "hann".

       orientation
	   Set	orientation  of	 time  vs frequency axis. Can be "vertical" or
	   "horizontal". Default is "vertical".

       gain
	   Set scale gain for calculating  intensity  color  values.   Default
	   value is 1.

       legend
	   Draw time and frequency axes and legends. Default is enabled.

       rotation
	   Set color rotation, must be in [-1.0, 1.0] range.  Default value is
	   0.

       start
	   Set	start  frequency from which to display spectrogram. Default is
	   0.

       stop
	   Set stop frequency to which to display spectrogram. Default is 0.

       drange
	   Set dynamic range used to calculate intensity color values. Default
	   is 120 dBFS.	 Allowed range is from 10 to 200.

       limit
	   Set upper limit of input audio samples volume in dBFS. Default is 0
	   dBFS.  Allowed range is from -100 to 100.

       opacity
	   Set opacity strength when using  pixel  format  output  with	 alpha
	   component.

       Examples

       •   Extract  an audio spectrogram of a whole audio track in a 1024x1024
	   picture using ffmpeg:

		   ffmpeg -i audio.flac -lavfi showspectrumpic=s=1024x1024 spectrogram.png

   showvolume
       Convert input audio volume to a video output.

       The filter accepts the following options:

       rate, r
	   Set video rate.

       b   Set border width, allowed range is [0, 5]. Default is 1.

       w   Set channel width, allowed range is [80, 8192]. Default is 400.

       h   Set channel height, allowed range is [1, 900]. Default is 20.

       f   Set fade, allowed range is [0, 1]. Default is 0.95.

       c   Set volume color expression.

	   The expression can use the following variables:

	   VOLUME
	       Current max volume of channel in dB.

	   PEAK
	       Current peak.

	   CHANNEL
	       Current channel number, starting from 0.

       t   If set, displays channel names. Default is enabled.

       v   If set, displays volume values. Default is enabled.

       o   Set orientation, can be horizontal: "h" or vertical:	 "v",  default
	   is "h".

       s   Set	step  size, allowed range is [0, 5]. Default is 0, which means
	   step is disabled.

       p   Set background opacity, allowed range is [0, 1]. Default is 0.

       m   Set metering mode, can be peak: "p" or rms: "r", default is "p".

       ds  Set display scale, can be linear: "lin" or log: "log",  default  is
	   "lin".

       dm  In second.  If set to > 0., display a line for the max level in the
	   previous seconds.  default is disabled: 0.

       dmc The	color  of  the	max  line. Use when "dm" option is set to > 0.
	   default is: "orange"

   showwaves
       Convert input audio to a video output, representing the samples waves.

       The filter accepts the following options:

       size, s
	   Specify the video size for the  output.  For	 the  syntax  of  this
	   option,  check the "Video size" section in the ffmpeg-utils manual.
	   Default value is "600x240".

       mode
	   Set display mode.

	   Available values are:

	   point
	       Draw a point for each sample.

	   line
	       Draw a vertical line for each sample.

	   p2p Draw a point for each sample and a line between them.

	   cline
	       Draw a centered vertical line for each sample.

	   Default value is "point".

       n   Set the number of samples which are printed on the same  column.  A
	   larger  value  will	decrease  the  frame  rate. Must be a positive
	   integer. This option can be set only if the value for rate  is  not
	   explicitly specified.

       rate, r
	   Set	the  (approximate)  output frame rate. This is done by setting
	   the option n. Default value is "25".

       split_channels
	   Set if channels should be  drawn  separately	 or  overlap.  Default
	   value is 0.

       colors
	   Set	colors separated by '|' which are going to be used for drawing
	   of each channel.

       scale
	   Set amplitude scale.

	   Available values are:

	   lin Linear.

	   log Logarithmic.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   Default is linear.

       draw
	   Set the draw mode. This is mostly useful to set for high n.

	   Available values are:

	   scale
	       Scale pixel values for each drawn sample.

	   full
	       Draw every sample directly.

	   Default value is "scale".

       Examples

       •   Output  the	input  file  audio   and   the	 corresponding	 video
	   representation at the same time:

		   amovie=a.mp3,asplit[out0],showwaves[out1]

       •   Create  a  synthetic	 signal	 and show it with showwaves, forcing a
	   frame rate of 30 frames per second:

		   aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

   showwavespic
       Convert input audio to a single video frame, representing  the  samples
       waves.

       The filter accepts the following options:

       size, s
	   Specify  the	 video	size  for  the	output. For the syntax of this
	   option, check the "Video size" section in the ffmpeg-utils  manual.
	   Default value is "600x240".

       split_channels
	   Set	if  channels  should  be  drawn separately or overlap. Default
	   value is 0.

       colors
	   Set colors separated by '|' which are going to be used for  drawing
	   of each channel.

       scale
	   Set amplitude scale.

	   Available values are:

	   lin Linear.

	   log Logarithmic.

	   sqrt
	       Square root.

	   cbrt
	       Cubic root.

	   Default is linear.

       draw
	   Set the draw mode.

	   Available values are:

	   scale
	       Scale pixel values for each drawn sample.

	   full
	       Draw every sample directly.

	   Default value is "scale".

       filter
	   Set the filter mode.

	   Available values are:

	   average
	       Use average samples values for each drawn sample.

	   peak
	       Use peak samples values for each drawn sample.

	   Default value is "average".

       Examples

       •   Extract  a channel split representation of the wave form of a whole
	   audio track in a 1024x800 picture using ffmpeg:

		   ffmpeg -i audio.flac -lavfi showwavespic=split_channels=1:s=1024x800 waveform.png

   sidedata, asidedata
       Delete frame side data, or select frames based on it.

       This filter accepts the following options:

       mode
	   Set mode of operation of the filter.

	   Can be one of the following:

	   select
	       Select every frame with side data of "type".

	   delete
	       Delete side data of "type". If "type" is not  set,  delete  all
	       side data in the frame.

       type
	   Set	side  data  type used with all modes. Must be set for "select"
	   mode. For  the  list	 of  frame  side  data	types,	refer  to  the
	   "AVFrameSideDataType"  enum	in  libavutil/frame.h. For example, to
	   choose  "AV_FRAME_DATA_PANSCAN"  side  data,	  you	must   specify
	   "PANSCAN".

   spectrumsynth
       Synthesize  audio  from	2  input  video	 spectrums, first input stream
       represents magnitude across time and  second  represents	 phase	across
       time.   The filter will transform from frequency domain as displayed in
       videos back to time domain as presented in audio output.

       This filter is primarily created for reversing  processed  showspectrum
       filter  outputs,	 but can synthesize sound from other spectrograms too.
       But in such case results are going to be poor if the phase data is  not
       available,  because  in	such  cases  phase  data need to be recreated,
       usually it's just recreated from random noise.  For  best  results  use
       gray  only  output  ("channel"  color  mode in showspectrum filter) and
       "log" scale for magnitude video and "lin" scale	for  phase  video.  To
       produce	phase,	for 2nd video, use "data" option. Inputs videos should
       generally use "fullframe" slide mode as that saves resources needed for
       decoding video.

       The filter accepts the following options:

       sample_rate
	   Specify sample rate of output audio, the sample rate of audio  from
	   which spectrum was generated may differ.

       channels
	   Set number of channels represented in input video spectrums.

       scale
	   Set	scale which was used when generating magnitude input spectrum.
	   Can be "lin" or "log". Default is "log".

       slide
	   Set slide which was used when generating inputs spectrums.  Can  be
	   "replace",	"scroll",   "fullframe"	  or  "rscroll".   Default  is
	   "fullframe".

       win_func
	   Set window function used for resynthesis.

       overlap
	   Set window overlap. In range "[0, 1]". Default is  1,  which	 means
	   optimal overlap for selected window function will be picked.

       orientation
	   Set orientation of input videos. Can be "vertical" or "horizontal".
	   Default is "vertical".

       Examples

       •   First  create magnitude and phase videos from audio, assuming audio
	   is stereo with 44100 sample rate, then resynthesize videos back  to
	   audio with spectrumsynth:

		   ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=log:overlap=0.875:color=channel:slide=fullframe:data=magnitude -an -c:v rawvideo magnitude.nut
		   ffmpeg -i input.flac -lavfi showspectrum=mode=separate:scale=lin:overlap=0.875:color=channel:slide=fullframe:data=phase -an -c:v rawvideo phase.nut
		   ffmpeg -i magnitude.nut -i phase.nut -lavfi spectrumsynth=channels=2:sample_rate=44100:win_func=hann:overlap=0.875:slide=fullframe output.flac

   split, asplit
       Split input into several identical outputs.

       "asplit" works with audio input, "split" with video.

       The  filter  accepts  a	single parameter which specifies the number of
       outputs. If unspecified, it defaults to 2.

       Examples

       •   Create two separate outputs from the same input:

		   [in] split [out0][out1]

       •   To create 3 or more outputs, you need  to  specify  the  number  of
	   outputs, like in:

		   [in] asplit=3 [out0][out1][out2]

       •   Create  two	separate  outputs from the same input, one cropped and
	   one padded:

		   [in] split [splitout1][splitout2];
		   [splitout1] crop=100:100:0:0	   [cropout];
		   [splitout2] pad=200:200:100:100 [padout];

       •   Create 5 copies of the input audio with ffmpeg:

		   ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

   zmq, azmq
       Receive commands sent through a libzmq  client,	and  forward  them  to
       filters in the filtergraph.

       "zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted
       between	two  video filters, "azmq" between two audio filters. Both are
       capable to send messages to any filter type.

       To enable these filters you need to  install  the  libzmq  library  and
       headers and configure FFmpeg with "--enable-libzmq".

       For more information about libzmq see: <http://www.zeromq.org/>

       The  "zmq"  and	"azmq" filters work as a libzmq server, which receives
       messages sent through a network interface defined by  the  bind_address
       (or  the	 abbreviation  "b")  option.   Default value of this option is
       tcp://localhost:5555. You may want to alter this value to  your	needs,
       but do not forget to escape any ':' signs (see filtergraph escaping).

       The received message must be in the form:

	       <TARGET> <COMMAND> [<ARG>]

       TARGET  specifies  the  target  of the command, usually the name of the
       filter class or a specific filter instance  name.  The  default	filter
       instance	 name  uses  the pattern Parsed_<filter_name>_<index>, but you
       can override this by using the filter_name@id syntax  (see  Filtergraph
       syntax).

       COMMAND specifies the name of the command for the target filter.

       ARG  is optional and specifies the optional argument list for the given
       COMMAND.

       Upon reception, the message is processed and the corresponding  command
       is  injected  into the filtergraph. Depending on the result, the filter
       will send a reply to the client, adopting the format:

	       <ERROR_CODE> <ERROR_REASON>
	       <MESSAGE>

       MESSAGE is optional.

       Examples

       Look at tools/zmqsend for an example of a zmq client which can be  used
       to send commands processed by these filters.

       Consider	 the  following	 filtergraph  generated	 by  ffplay.   In this
       example the last overlay filter has an instance name. All other filters
       will have default instance names.

	       ffplay -dumpgraph 1 -f lavfi "
	       color=s=100x100:c=red  [l];
	       color=s=100x100:c=blue [r];
	       nullsrc=s=200x100, zmq [bg];
	       [bg][l]	 overlay     [bg+l];
	       [bg+l][r] overlay@my=x=100 "

       To change the color of the  left	 side  of  the	video,	the  following
       command can be used:

	       echo Parsed_color_0 c yellow | tools/zmqsend

       To change the right side:

	       echo Parsed_color_1 c pink | tools/zmqsend

       To change the position of the right side:

	       echo overlay@my x 150 | tools/zmqsend

MULTIMEDIA SOURCES
       Below is a description of the currently available multimedia sources.

   amovie
       This  is the same as movie source, except it selects an audio stream by
       default.

   avsynctest
       Generate an Audio/Video Sync Test.

       Generated stream periodically shows flash video frame and emits beep in
       audio.  Useful to inspect A/V sync issues.

       It accepts the following options:

       size, s
	   Set output video size. Default value is "hd720".

       framerate, fr
	   Set output video frame rate. Default value is 30.

       samplerate, sr
	   Set output audio sample rate. Default value is 44100.

       amplitude, a
	   Set output audio beep amplitude. Default value is 0.7.

       period, p
	   Set output audio beep period in seconds. Default value is 3.

       delay, dl
	   Set output video flash delay in number of frames. Default value  is
	   0.

       cycle, c
	   Enable cycling of video delays, by default is disabled.

       duration, d
	   Set stream output duration. By default duration is unlimited.

       fg, bg, ag
	   Set foreground/background/additional color.

       Commands

       This source supports the some above options as commands.

   movie
       Read audio and/or video stream(s) from a movie container.

       It accepts the following parameters:

       filename
	   The	name  of  the resource to read (not necessarily a file; it can
	   also be a device or a stream accessed through some protocol).

       format_name, f
	   Specifies the format assumed for the movie  to  read,  and  can  be
	   either  the	name  of  a  container	or  an	input  device.	If not
	   specified, the format is guessed from movie_name or by probing.

       seek_point, sp
	   Specifies the seek point in seconds.	 The  frames  will  be	output
	   starting  from  this	 seek  point.  The parameter is evaluated with
	   "av_strtod", so the numerical  value	 may  be  suffixed  by	an  IS
	   postfix. The default value is "0".

       streams, s
	   Specifies  the  streams  to read. Several streams can be specified,
	   separated by "+". The source will then have as many outputs, in the
	   same order. The syntax is  explained	 in  the  "Stream  specifiers"
	   section  in	the  ffmpeg  manual.  Two special names, "dv" and "da"
	   specify respectively the default  (best  suited)  video  and	 audio
	   stream.  Default  is	 "dv",	or  "da"  if  the  filter is called as
	   "amovie".

       stream_index, si
	   Specifies the index of the video stream to read. If	the  value  is
	   -1,	the most suitable video stream will be automatically selected.
	   The default value is "-1". Deprecated.  If  the  filter  is	called
	   "amovie", it will select audio instead of video.

       loop
	   Specifies  how  many	 times to read the stream in sequence.	If the
	   value is 0, the stream will be looped infinitely.  Default value is
	   "1".

	   Note that when the movie is looped the source  timestamps  are  not
	   changed,   so   it	will  generate	non  monotonically  increasing
	   timestamps.

       discontinuity
	   Specifies the time difference between frames above which the	 point
	   is  considered  a  timestamp	 discontinuity	which  is  removed  by
	   adjusting the later timestamps.

       dec_threads
	   Specifies the number of threads for decoding

       format_opts
	   Specify format options for the opened file. Format options  can  be
	   specified  as  a  list  of  key=value  pairs	 separated by ':'. The
	   following  example  shows  how  to	add   protocol_whitelist   and
	   protocol_blacklist options:

		   ffplay -f lavfi
		   "movie=filename='1.sdp':format_opts='protocol_whitelist=file,rtp,udp\:protocol_blacklist=http'"

       It  allows  overlaying  a  second  video	 on top of the main input of a
       filtergraph, as shown in this graph:

	       input -----------> deltapts0 --> overlay --> output
						   ^
						   |
	       movie --> scale--> deltapts1 -------+

       Examples

       •   Skip 3.2 seconds from the start of the AVI file in.avi, and overlay
	   it on top of the input labelled "in":

		   movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [over];
		   [in] setpts=PTS-STARTPTS [main];
		   [main][over] overlay=16:16 [out]

       •   Read from a video4linux2 device, and overlay it on top of the input
	   labelled "in":

		   movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [over];
		   [in] setpts=PTS-STARTPTS [main];
		   [main][over] overlay=16:16 [out]

       •   Read the first video stream and the audio stream with id 0x81  from
	   dvd.vob;  the  video	 is connected to the pad named "video" and the
	   audio is connected to the pad named "audio":

		   movie=dvd.vob:s=v:0+#0x81 [video] [audio]

       Commands

       Both movie and amovie support the following commands:

       seek
	   Perform  seek  using	 "av_seek_frame".    The   syntax   is:	  seek
	   stream_index|timestamp|flags

	   •   stream_index:  If  stream_index	is  -1,	 a  default  stream is
	       selected,  and  timestamp  is  automatically   converted	  from
	       AV_TIME_BASE units to the stream specific time_base.

	   •   timestamp:  Timestamp  in  AVStream.time_base  units  or, if no
	       stream is specified, in AV_TIME_BASE units.

	   •   flags: Flags which select direction and seeking mode.

       get_duration
	   Get movie duration in AV_TIME_BASE units.

EXTERNAL LIBRARIES
       FFmpeg can be hooked up with a number  of  external  libraries  to  add
       support	for  more formats. None of them are used by default, their use
       has to be explicitly requested by  passing  the	appropriate  flags  to
       ./configure.

   Alliance for Open Media (AOM)
       FFmpeg can make use of the AOM library for AV1 decoding and encoding.

       Go  to <http://aomedia.org/> and follow the instructions for installing
       the library. Then pass "--enable-libaom" to configure to enable it.

   AMD AMF/VCE
       FFmpeg can use the AMD Advanced Media Framework library for accelerated
       H.264 and HEVC(only windows) encoding on	 hardware  with	 Video	Coding
       Engine (VCE).

       To   enable   support   you   must  obtain  the	AMF  framework	header
       files(version			   1.4.9+)			  from
       <https://github.com/GPUOpen-LibrariesAndSDKs/AMF.git>.

       Create  an  "AMF/"  directory  in  the  system  include path.  Copy the
       contents	 of  "AMF/amf/public/include/"	into  that  directory.	  Then
       configure FFmpeg with "--enable-amf".

       Initialization  of  amf	encoder	 occurs	 in  this  order: 1) trying to
       initialize through dx11(only windows) 2) trying to  initialize  through
       dx9(only windows) 3) trying to initialize through vulkan

       To  use	h.264(AMD  VCE) encoder on linux amdgru-pro version 19.20+ and
       amf-amdgpu-pro  package(amdgru-pro  contains,  but  does	 not   install
       automatically) are required.

       This  driver  can  be  installed	 using	amdgpu-pro-install  script  in
       official amd driver archive.

   AviSynth
       FFmpeg can read AviSynth scripts as  input.  To	enable	support,  pass
       "--enable-avisynth"  to configure after installing the headers provided
       by  <https://github.com/AviSynth/AviSynthPlus>.	  AviSynth+   can   be
       configured   to	 install   only	  the	headers	  by   either  passing
       "-DHEADERS_ONLY:bool=on" to the normal CMake-based build system, or  by
       using the supplied "GNUmakefile".

       For  Windows,  supported AviSynth variants are <http://avisynth.nl> for
       32-bit builds and <http://avisynth.nl/index.php/AviSynth+>  for	32-bit
       and 64-bit builds.

       For  Linux,  macOS,  and	 BSD,  the  only supported AviSynth variant is
       <https://github.com/AviSynth/AviSynthPlus>, starting with version 3.5.

	   In 2016, AviSynth+ added support for building  with	GCC.  However,
	   due	to  the eccentricities of Windows' calling conventions, 32-bit
	   GCC builds of AviSynth+ are	not  compatible	 with  typical	32-bit
	   builds of FFmpeg.

	   By default, FFmpeg assumes compatibility with 32-bit MSVC builds of
	   AviSynth+  since  that is the most widely-used and entrenched build
	   configuration.  Users can override  this  and  enable  support  for
	   32-bit  GCC	builds of AviSynth+ by passing "-DAVSC_WIN32_GCC32" to
	   "--extra-cflags" when configuring FFmpeg.

	   64-bit builds of FFmpeg are not affected, and can use  either  MSVC
	   or GCC builds of AviSynth+ without any special flags.

	   AviSynth(+)	is  loaded dynamically.	 Distributors can build FFmpeg
	   with "--enable-avisynth", and the binaries will work regardless  of
	   the	end user having AviSynth installed.  If/when an end user would
	   like to use AviSynth scripts, then they can install AviSynth(+) and
	   FFmpeg will be able to find and use it to open scripts.

   Chromaprint
       FFmpeg can make use of the Chromaprint  library	for  generating	 audio
       fingerprints.   Pass  "--enable-chromaprint" to configure to enable it.
       See <https://acoustid.org/chromaprint>.

   codec2
       FFmpeg can make use of the  codec2  library  for	 codec2	 decoding  and
       encoding.   There  is currently no native decoder, so libcodec2 must be
       used for decoding.

       Go to <http://freedv.org/>, download "Codec 2 source  archive".	 Build
       and  install  using  CMake.  Debian users can install the libcodec2-dev
       package	instead.   Once	 libcodec2   is	  installed   you   can	  pass
       "--enable-libcodec2" to configure to enable it.

       The easiest way to use codec2 is with .c2 files, since they contain the
       mode  information  required for decoding.  To encode such a file, use a
       .c2 file extension and give the libcodec2  encoder  the	-mode  option:
       "ffmpeg	-i  input.wav -mode 700C output.c2".  Playback is as simple as
       "ffplay output.c2".  For a list of  supported  modes,  run  "ffmpeg  -h
       encoder=libcodec2".   Raw  codec2  files	 are  also supported.  To make
       sense of them the mode in use needs to be specified as a format option:
       "ffmpeg -f codec2raw -mode 1300 -i input.raw output.wav".

   dav1d
       FFmpeg can make use of the dav1d library for AV1 video decoding.

       Go  to  <https://code.videolan.org/videolan/dav1d>   and	  follow   the
       instructions  for installing the library. Then pass "--enable-libdav1d"
       to configure to enable it.

   davs2
       FFmpeg can make use of the davs2 library for  AVS2-P2/IEEE1857.4	 video
       decoding.

       Go to <https://github.com/pkuvcl/davs2> and follow the instructions for
       installing  the	library. Then pass "--enable-libdavs2" to configure to
       enable it.

	   libdavs2 is under the GNU Public License Version 2  or  later  (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html>	   for
	   details), you must upgrade FFmpeg's license to GPL in order to  use
	   it.

   uavs3d
       FFmpeg can make use of the uavs3d library for AVS3-P2/IEEE1857.10 video
       decoding.

       Go to <https://github.com/uavs3/uavs3d> and follow the instructions for
       installing  the library. Then pass "--enable-libuavs3d" to configure to
       enable it.

   Game Music Emu
       FFmpeg can make use of the Game Music Emu library to  read  audio  from
       supported  video	 game  music  file  formats. Pass "--enable-libgme" to
       configure	    to		  enable	    it.		   See
       <https://bitbucket.org/mpyne/game-music-emu/overview>.

   Intel QuickSync Video
       FFmpeg can use Intel QuickSync Video (QSV) for accelerated decoding and
       encoding	 of multiple codecs. To use QSV, FFmpeg must be linked against
       the "libmfx" dispatcher, which loads the actual decoding libraries.

       The  dispatcher	is  open   source   and	  can	be   downloaded	  from
       <https://github.com/lu-zero/mfx_dispatch.git>.	FFmpeg	 needs	to  be
       configured with the "--enable-libmfx" option and "pkg-config" needs  to
       be able to locate the dispatcher's ".pc" files.

   Kvazaar
       FFmpeg can make use of the Kvazaar library for HEVC encoding.

       Go    to	  <https://github.com/ultravideo/kvazaar>   and	  follow   the
       instructions    for    installing    the	    library.	 Then	  pass
       "--enable-libkvazaar" to configure to enable it.

   LAME
       FFmpeg can make use of the LAME library for MP3 encoding.

       Go  to  <http://lame.sourceforge.net/>  and follow the instructions for
       installing the library.	Then pass "--enable-libmp3lame"	 to  configure
       to enable it.

   libilbc
       iLBC  is	 a narrowband speech codec that has been made freely available
       by Google as part  of  the  WebRTC  project.  libilbc  is  a  packaging
       friendly	 copy  of  the	iLBC codec. FFmpeg can make use of the libilbc
       library for iLBC decoding and encoding.

       Go   to	 <https://github.com/TimothyGu/libilbc>	  and	 follow	   the
       instructions  for  installing the library. Then pass "--enable-libilbc"
       to configure to enable it.

   libjxl
       JPEG XL is an image format intended to fully replace legacy JPEG for an
       extended	 period	 of  life.   See   <https://jpegxl.info/>   for	  more
       information, and see <https://github.com/libjxl/libjxl> for the library
       source. You can pass "--enable-libjxl" to configure in order enable the
       libjxl wrapper.

   libvpx
       FFmpeg  can  make  use  of  the libvpx library for VP8/VP9 decoding and
       encoding.

       Go to <http://www.webmproject.org/> and	follow	the  instructions  for
       installing  the	library.  Then	pass "--enable-libvpx" to configure to
       enable it.

   ModPlug
       FFmpeg can make use of this library, originating	 in  Modplug-XMMS,  to
       read	   from	       MOD-like	       music	    files.	   See
       <https://github.com/Konstanty/libmodplug>.  Pass	 "--enable-libmodplug"
       to configure to enable it.

   OpenCORE, VisualOn, and Fraunhofer libraries
       Spun  off  Google  Android  sources,  OpenCore, VisualOn and Fraunhofer
       libraries provide encoders for a number of audio codecs.

	   OpenCORE and VisualOn libraries are under the  Apache  License  2.0
	   (see	  <http://www.apache.org/licenses/LICENSE-2.0>	for  details),
	   which is incompatible to the LGPL version 2.1 and  GPL  version  2.
	   You	have  to upgrade FFmpeg's license to LGPL version 3 (or if you
	   have	 enabled  GPL  components,   GPL   version   3)	  by   passing
	   "--enable-version3" to configure in order to use it.

	   The	license of the Fraunhofer AAC library is incompatible with the
	   GPL.	   Therefore,	for   GPL   builds,   you   have    to	  pass
	   "--enable-nonfree"  to configure in order to use it. To the best of
	   our knowledge, it is compatible with the LGPL.

       OpenCORE AMR

       FFmpeg  can  make  use	of   the   OpenCORE   libraries	  for	AMR-NB
       decoding/encoding and AMR-WB decoding.

       Go  to  <http://sourceforge.net/projects/opencore-amr/>	and follow the
       instructions    for    installing    the	   libraries.	  Then	  pass
       "--enable-libopencore-amrnb"   and/or  "--enable-libopencore-amrwb"  to
       configure to enable them.

       VisualOn AMR-WB encoder library

       FFmpeg can make use  of	the  VisualOn  AMR-WBenc  library  for	AMR-WB
       encoding.

       Go  to  <http://sourceforge.net/projects/opencore-amr/>	and follow the
       instructions    for    installing    the	   library.	 Then	  pass
       "--enable-libvo-amrwbenc" to configure to enable it.

       Fraunhofer AAC library

       FFmpeg  can  make  use of the Fraunhofer AAC library for AAC decoding &
       encoding.

       Go to <http://sourceforge.net/projects/opencore-amr/>  and  follow  the
       instructions	for	installing    the    library.	  Then	  pass
       "--enable-libfdk-aac" to configure to enable it.

   OpenH264
       FFmpeg can make use of the OpenH264  library  for  H.264	 decoding  and
       encoding.

       Go  to  <http://www.openh264.org/>  and	follow	the  instructions  for
       installing the library. Then pass "--enable-libopenh264"	 to  configure
       to enable it.

       For  decoding,  this  library  is  much	more limited than the built-in
       decoder in  libavcodec;	currently,  this  library  lacks  support  for
       decoding	 B-frames  and	some  other  main/high	profile	 features. (It
       currently only supports constrained baseline profile and CABAC.)	 Using
       it  is  mostly  useful  for testing and for taking advantage of Cisco's
       patent			     portfolio			       license
       (<http://www.openh264.org/BINARY_LICENSE.txt>).

   OpenJPEG
       FFmpeg can use the OpenJPEG libraries for decoding/encoding J2K videos.
       Go  to  <http://www.openjpeg.org/>  to get the libraries and follow the
       installation instructions.  To enable using OpenJPEG  in	 FFmpeg,  pass
       "--enable-libopenjpeg" to ./configure.

   rav1e
       FFmpeg  can  make use of rav1e (Rust AV1 Encoder) via its C bindings to
       encode videos.  Go to <https://github.com/xiph/rav1e/> and  follow  the
       instructions  to	 build the C library. To enable using rav1e in FFmpeg,
       pass "--enable-librav1e" to ./configure.

   SVT-AV1
       FFmpeg can make use of the Scalable Video Technology  for  AV1  library
       for AV1 encoding.

       Go   to	 <https://gitlab.com/AOMediaCodec/SVT-AV1/>   and  follow  the
       instructions for installing the library. Then pass "--enable-libsvtav1"
       to configure to enable it.

   TwoLAME
       FFmpeg can make use of the TwoLAME library for MP2 encoding.

       Go  to  <http://www.twolame.org/>  and  follow  the  instructions   for
       installing  the	library.  Then pass "--enable-libtwolame" to configure
       to enable it.

   VapourSynth
       FFmpeg can read VapourSynth scripts as input. To enable	support,  pass
       "--enable-vapoursynth"	to  configure.	Vapoursynth  is	 detected  via
       "pkg-config".	Versions    42	  or	greater	   supported.	   See
       <http://www.vapoursynth.com/>.

       Due  to security concerns, Vapoursynth scripts will not be autodetected
       so the input format has to be  forced.  For  ff*	 CLI  tools,  add  "-f
       vapoursynth" before the input "-i yourscript.vpy".

   x264
       FFmpeg can make use of the x264 library for H.264 encoding.

       Go  to  <http://www.videolan.org/developers/x264.html>  and  follow the
       instructions for installing the library. Then  pass  "--enable-libx264"
       to configure to enable it.

	   x264	 is  under  the	 GNU  Public  License  Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html>	   for
	   details),  you must upgrade FFmpeg's license to GPL in order to use
	   it.

   x265
       FFmpeg can make use of the x265 library for HEVC encoding.

       Go to <http://x265.org/developers.html> and follow the instructions for
       installing the library. Then pass "--enable-libx265"  to	 configure  to
       enable it.

	   x265	 is  under  the	 GNU  Public  License  Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html>	   for
	   details),  you must upgrade FFmpeg's license to GPL in order to use
	   it.

   xavs
       FFmpeg can make use of the xavs library for AVS encoding.

       Go to <http://xavs.sf.net/> and follow the instructions for  installing
       the library. Then pass "--enable-libxavs" to configure to enable it.

   xavs2
       FFmpeg  can  make use of the xavs2 library for AVS2-P2/IEEE1857.4 video
       encoding.

       Go to <https://github.com/pkuvcl/xavs2> and follow the instructions for
       installing the library. Then pass "--enable-libxavs2" to	 configure  to
       enable it.

	   libxavs2  is	 under	the GNU Public License Version 2 or later (see
	   <http://www.gnu.org/licenses/old-licenses/gpl-2.0.html>	   for
	   details),  you must upgrade FFmpeg's license to GPL in order to use
	   it.

   ZVBI
       ZVBI is a VBI decoding library which can be used by  FFmpeg  to	decode
       DVB teletext pages and DVB teletext subtitles.

       Go   to	 <http://sourceforge.net/projects/zapping/>   and  follow  the
       instructions for installing the library. Then  pass  "--enable-libzvbi"
       to configure to enable it.

SUPPORTED FILE FORMATS
       You  can use the "-formats" and "-codecs" options to have an exhaustive
       list.

   File Formats
       FFmpeg supports the following file formats  through  the	 "libavformat"
       library:

       Name  :	Encoding @tab Decoding @tab Comments
       3dostr			  :    @tab X
       4xm			  :    @tab X
	       @tab 4X Technologies format, used in some games.

       8088flex TMV		  :    @tab X
       AAX			  :    @tab X
	       @tab Audible Enhanced Audio format, used in audiobooks.

       AA			  :    @tab X
	       @tab Audible Format 2, 3, and 4, used in audiobooks.

       ACT Voice		  :    @tab X
	       @tab contains G.729 audio

       Adobe Filmstrip		  :  X @tab X
       Audio IFF (AIFF)		  :  X @tab X
       American Laser Games MM	  :    @tab X
	       @tab Multimedia format used in games like Mad Dog McCree.

       3GPP AMR			  :  X @tab X
       Amazing Studio Packed Animation File   :	   @tab X
	       @tab Multimedia format used in game Heart Of Darkness.

       Apple HTTP Live Streaming  :    @tab X
       Artworx Data Format	  :    @tab X
       Interplay ACM		  :    @tab X
	       @tab Audio only format used in some Interplay games.

       ADP			  :    @tab X
	       @tab Audio format used on the Nintendo Gamecube.

       AFC			  :    @tab X
	       @tab Audio format used on the Nintendo Gamecube.

       ADS/SS2			  :    @tab X
	       @tab Audio format used on the PS2.

       APNG			  :  X @tab X
       ASF			  :  X @tab X
	       @tab Advanced / Active Streaming Format.

       AST			  :  X @tab X
	       @tab Audio format used on the Nintendo Wii.

       AVI			  :  X @tab X
       AviSynth			  :    @tab X
       AVR			  :    @tab X
	       @tab Audio format used on Mac.

       AVS			  :    @tab X
	       @tab Multimedia format used by the Creature Shock game.

       Beam Software SIFF	  :    @tab X
	       @tab Audio and video format used in some games by Beam Software.

       Bethesda Softworks VID	  :    @tab X
	       @tab Used in some games from Bethesda Softworks.

       Binary text		  :    @tab X
       Bink			  :    @tab X
	       @tab Multimedia format used by many games.

       Bink Audio		  :    @tab X
	       @tab Audio only multimedia format used by some games.

       Bitmap Brothers JV	  :    @tab X
	       @tab Used in Z and Z95 games.

       BRP			  :    @tab X
	       @tab Argonaut Games format.

       Brute Force & Ignorance	  :    @tab X
	       @tab Used in the game Flash Traffic: City of Angels.

       BFSTM			  :    @tab X
	       @tab Audio format used on the Nintendo WiiU (based on BRSTM).

       BRSTM			  :    @tab X
	       @tab Audio format used on the Nintendo Wii.

       BW64			  :    @tab X
	       @tab Broadcast Wave 64bit.

       BWF			  :  X @tab X
       codec2 (raw)		  :  X @tab X
	       @tab Must be given -mode format option to decode correctly.

       codec2 (.c2 files)	  :  X @tab X
	       @tab Contains header with version and mode info, simplifying playback.

       CRI ADX			  :  X @tab X
	       @tab Audio-only format used in console video games.

       CRI AIX			  :    @tab X
       CRI HCA			  :    @tab X
	       @tab Audio-only format used in console video games.

       Discworld II BMV		  :    @tab X
       Interplay C93		  :    @tab X
	       @tab Used in the game Cyberia from Interplay.

       Delphine Software International CIN  :	 @tab X
	       @tab Multimedia format used by Delphine Software games.

       Digital Speech Standard (DSS)  :	   @tab X
       CD+G			  :    @tab X
	       @tab Video format used by CD+G karaoke disks

       Phantom Cine		  :    @tab X
       Commodore CDXL		  :    @tab X
	       @tab Amiga CD video format

       Core Audio Format	  :  X @tab X
	       @tab Apple Core Audio Format

       CRC testing format	  :  X @tab
       Creative Voice		  :  X @tab X
	       @tab Created for the Sound Blaster Pro.

       CRYO APC			  :    @tab X
	       @tab Audio format used in some games by CRYO Interactive Entertainment.

       D-Cinema audio		  :  X @tab X
       Deluxe Paint Animation	  :    @tab X
       DCSTR			  :    @tab X
       DFA			  :    @tab X
	       @tab This format is used in Chronomaster game

       DirectDraw Surface	  :    @tab X
       DSD Stream File (DSF)	  :    @tab X
       DV video			  :  X @tab X
       DXA			  :    @tab X
	       @tab This format is used in the non-Windows version of the Feeble Files
		    game and different game cutscenes repacked for use with ScummVM.

       Electronic Arts cdata   :     @tab X
       Electronic Arts Multimedia   :	  @tab X
	       @tab Used in various EA games; files have extensions like WVE and UV2.

       Ensoniq Paris Audio File	  :    @tab X
       FFM (FFserver live feed)	  :  X @tab X
       Flash (SWF)		  :  X @tab X
       Flash 9 (AVM2)		  :  X @tab X
	       @tab Only embedded audio is decoded.

       FLI/FLC/FLX animation	  :    @tab X
	       @tab .fli/.flc files

       Flash Video (FLV)	  :  X @tab X
	       @tab Macromedia Flash video files

       framecrc testing format	  :  X @tab
       FunCom ISS		  :    @tab X
	       @tab Audio format used in various games from FunCom like The Longest Journey.

       G.723.1			  :  X @tab X
       G.726			  :    @tab X @tab Both left- and right-
       justified.
       G.729 BIT		  :  X @tab X
       G.729 raw		  :    @tab X
       GENH			  :    @tab X
	       @tab Audio format for various games.

       GIF Animation		  :  X @tab X
       GXF			  :  X @tab X
	       @tab General eXchange Format SMPTE 360M, used by Thomson Grass Valley
		    playout servers.

       HNM  :	 @tab X
	       @tab Only version 4 supported, used in some games from Cryo Interactive

       iCEDraw File		  :    @tab X
       ICO			  :  X @tab X
	       @tab Microsoft Windows ICO

       id Quake II CIN video	  :    @tab X
       id RoQ			  :  X @tab X
	       @tab Used in Quake III, Jedi Knight 2 and other computer games.

       IEC61937 encapsulation  :  X @tab X
       IFF			  :    @tab X
	       @tab Interchange File Format

       IFV			  :    @tab X
	       @tab A format used by some old CCTV DVRs.

       iLBC			  :  X @tab X
       Interplay MVE		  :    @tab X
	       @tab Format used in various Interplay computer games.

       Iterated Systems ClearVideo  :	   @tab	 X
	       @tab I-frames only

       IV8			  :    @tab X
	       @tab A format generated by IndigoVision 8000 video server.

       IVF (On2)		  :  X @tab X
	       @tab A format used by libvpx

       Internet Video Recording	  :    @tab X
       IRCAM			  :  X @tab X
       LAF			  :    @tab X
	       @tab Limitless Audio Format

       LATM			  :  X @tab X
       LMLM4			  :    @tab X
	       @tab Used by Linux Media Labs MPEG-4 PCI boards

       LOAS			  :    @tab X
	       @tab contains LATM multiplexed AAC audio

       LRC			  :  X @tab X
       LVF			  :    @tab X
       LXF			  :    @tab X
	       @tab VR native stream format, used by Leitch/Harris' video servers.

       Magic Lantern Video (MLV)  :    @tab X
       Matroska			  :  X @tab X
       Matroska audio		  :  X @tab
       FFmpeg metadata		  :  X @tab X
	       @tab Metadata in text format.

       MAXIS XA			  :    @tab X
	       @tab Used in Sim City 3000; file extension .xa.

       MCA			  :    @tab X
	       @tab Used in some games from Capcom; file extension .mca.

       MD Studio		  :    @tab X
       Metal Gear Solid: The Twin Snakes  :  @tab X
       Megalux Frame		  :    @tab X
	       @tab Used by Megalux Ultimate Paint

       MobiClip MODS		  :    @tab X
       MobiClip MOFLEX		  :    @tab X
       Mobotix .mxg		  :    @tab X
       Monkey's Audio		  :    @tab X
       Motion Pixels MVI	  :    @tab X
       MOV/QuickTime/MP4	  :  X @tab X
	       @tab 3GP, 3GP2, PSP, iPod variants supported

       MP2			  :  X @tab X
       MP3			  :  X @tab X
       MPEG-1 System		  :  X @tab X
	       @tab muxed audio and video, VCD format supported

       MPEG-PS (program stream)	  :  X @tab X
	       @tab also known as C<VOB> file, SVCD and DVD format supported

       MPEG-TS (transport stream)  :  X @tab X
	       @tab also known as DVB Transport Stream

       MPEG-4			  :  X @tab X
	       @tab MPEG-4 is a variant of QuickTime.

       MSF			  :    @tab X
	       @tab Audio format used on the PS3.

       Mirillis FIC video	  :    @tab X
	       @tab No cursor rendering.

       MIDI Sample Dump Standard  :    @tab X
       MIME multipart JPEG	  :  X @tab
       MSN TCP webcam		  :    @tab X
	       @tab Used by MSN Messenger webcam streams.

       MTV			  :    @tab X
       Musepack			  :    @tab X
       Musepack SV8		  :    @tab X
       Material eXchange Format (MXF)  :  X @tab X
	       @tab SMPTE 377M, used by D-Cinema, broadcast industry.

       Material eXchange Format (MXF), D-10 Mapping  :	X @tab X
	       @tab SMPTE 386M, D-10/IMX Mapping.

       NC camera feed		  :    @tab X
	       @tab NC (AVIP NC4600) camera streams

       NIST SPeech HEader REsources  :	  @tab X
       Computerized Speech Lab NSP  :	 @tab X
       NTT TwinVQ (VQF)		  :    @tab X
	       @tab Nippon Telegraph and Telephone Corporation TwinVQ.

       Nullsoft Streaming Video	  :    @tab X
       NuppelVideo		  :    @tab X
       NUT			  :  X @tab X
	       @tab NUT Open Container Format

       Ogg			  :  X @tab X
       Playstation Portable PMP	  :    @tab X
       Portable Voice Format	  :    @tab X
       RK Audio (RKA)		  :    @tab X
       TechnoTrend PVA		  :    @tab X
	       @tab Used by TechnoTrend DVB PCI boards.

       QCP			  :    @tab X
       raw ADTS (AAC)		  :  X @tab X
       raw AC-3			  :  X @tab X
       raw AMR-NB		  :    @tab X
       raw AMR-WB		  :    @tab X
       raw APAC			  :    @tab X
       raw aptX			  :  X @tab X
       raw aptX HD		  :  X @tab X
       raw Bonk			  :    @tab X
       raw Chinese AVS video	  :  X @tab X
       raw DFPWM		  :  X @tab X
       raw Dirac		  :  X @tab X
       raw DNxHD		  :  X @tab X
       raw DTS			  :  X @tab X
       raw DTS-HD		  :    @tab X
       raw E-AC-3		  :  X @tab X
       raw FLAC			  :  X @tab X
       raw GSM			  :    @tab X
       raw H.261		  :  X @tab X
       raw H.263		  :  X @tab X
       raw H.264		  :  X @tab X
       raw HEVC			  :  X @tab X
       raw Ingenient MJPEG	  :    @tab X
       raw MJPEG		  :  X @tab X
       raw MLP			  :    @tab X
       raw MPEG			  :    @tab X
       raw MPEG-1		  :    @tab X
       raw MPEG-2		  :    @tab X
       raw MPEG-4		  :  X @tab X
       raw NULL			  :  X @tab
       raw video		  :  X @tab X
       raw id RoQ		  :  X @tab
       raw OBU			  :  X @tab X
       raw OSQ			  :    @tab X
       raw SBC			  :  X @tab X
       raw Shorten		  :    @tab X
       raw TAK			  :    @tab X
       raw TrueHD		  :  X @tab X
       raw VC-1			  :  X @tab X
       raw PCM A-law		  :  X @tab X
       raw PCM mu-law		  :  X @tab X
       raw PCM Archimedes VIDC	  :  X @tab X
       raw PCM signed 8 bit	  :  X @tab X
       raw PCM signed 16 bit big-endian	  :  X @tab X
       raw PCM signed 16 bit little-endian   :	X @tab X
       raw PCM signed 24 bit big-endian	  :  X @tab X
       raw PCM signed 24 bit little-endian   :	X @tab X
       raw PCM signed 32 bit big-endian	  :  X @tab X
       raw PCM signed 32 bit little-endian   :	X @tab X
       raw PCM signed 64 bit big-endian	  :  X @tab X
       raw PCM signed 64 bit little-endian   :	X @tab X
       raw PCM unsigned 8 bit	  :  X @tab X
       raw PCM unsigned 16 bit big-endian   :  X @tab X
       raw PCM unsigned 16 bit little-endian   :  X @tab X
       raw PCM unsigned 24 bit big-endian   :  X @tab X
       raw PCM unsigned 24 bit little-endian   :  X @tab X
       raw PCM unsigned 32 bit big-endian   :  X @tab X
       raw PCM unsigned 32 bit little-endian   :  X @tab X
       raw PCM 16.8 floating point little-endian  :    @tab X
       raw PCM 24.0 floating point little-endian  :    @tab X
       raw PCM floating-point 32 bit big-endian	  :  X @tab X
       raw PCM floating-point 32 bit little-endian   :	X @tab X
       raw PCM floating-point 64 bit big-endian	  :  X @tab X
       raw PCM floating-point 64 bit little-endian   :	X @tab X
       RDT			  :    @tab X
       REDCODE R3D		  :    @tab X
	       @tab File format used by RED Digital cameras, contains JPEG 2000 frames and PCM audio.

       RealMedia		  :  X @tab X
       Redirector		  :    @tab X
       RedSpark			  :    @tab X
       Renderware TeXture Dictionary  :	   @tab X
       Resolume DXV		  :    @tab X
       RF64			  :    @tab X
       RL2			  :    @tab X
	       @tab Audio and video format used in some games by Entertainment Software Partners.

       RPL/ARMovie		  :    @tab X
       Lego Mindstorms RSO	  :  X @tab X
       RSD			  :    @tab X
       RTMP			  :  X @tab X
	       @tab Output is performed by publishing stream to RTMP server

       RTP			  :  X @tab X
       RTSP			  :  X @tab X
       Sample Dump eXchange	  :    @tab X
       SAP			  :  X @tab X
       SBG			  :    @tab X
       SDNS			  :    @tab X
       SDP			  :    @tab X
       SER			  :    @tab X
       Digital Pictures SGA	  :    @tab X
       Sega FILM/CPK		  :  X @tab X
	       @tab Used in many Sega Saturn console games.

       Silicon Graphics Movie	  :    @tab X
       Sierra SOL		  :    @tab X
	       @tab .sol files used in Sierra Online games.

       Sierra VMD		  :    @tab X
	       @tab Used in Sierra CD-ROM games.

       Smacker			  :    @tab X
	       @tab Multimedia format used by many games.

       SMJPEG			  :  X @tab X
	       @tab Used in certain Loki game ports.

       SMPTE 337M encapsulation	  :    @tab X
       Smush			  :    @tab X
	       @tab Multimedia format used in some LucasArts games.

       Sony OpenMG (OMA)	  :  X @tab X
	       @tab Audio format used in Sony Sonic Stage and Sony Vegas.

       Sony PlayStation STR	  :    @tab X
       Sony Wave64 (W64)	  :  X @tab X
       SoX native format	  :  X @tab X
       SUN AU format		  :  X @tab X
       SUP raw PGS subtitles	  :  X @tab X
       SVAG			  :    @tab X
	       @tab Audio format used in Konami PS2 games.

       TDSC			  :    @tab X
       Text files		  :    @tab X
       THP			  :    @tab X
	       @tab Used on the Nintendo GameCube.

       Tiertex Limited SEQ	  :    @tab X
	       @tab Tiertex .seq files used in the DOS CD-ROM version of the game Flashback.

       True Audio		  :  X @tab X
       VAG			  :    @tab X
	       @tab Audio format used in many Sony PS2 games.

       VC-1 test bitstream	  :  X @tab X
       Vidvox Hap		  :  X @tab X
       Vivo			  :    @tab X
       VPK			  :    @tab X
	       @tab Audio format used in Sony PS games.

       Marble WADY		  :    @tab X
       WAV			  :  X @tab X
       Waveform Archiver	  :    @tab X
       WavPack			  :  X @tab X
       WebM			  :  X @tab X
       Windows Televison (WTV)	  :  X @tab X
       Wing Commander III movie	  :    @tab X
	       @tab Multimedia format used in Origin's Wing Commander III computer game.

       Westwood Studios audio	  :  X @tab X
	       @tab Multimedia format used in Westwood Studios games.

       Westwood Studios VQA	  :    @tab X
	       @tab Multimedia format used in Westwood Studios games.

       Wideband Single-bit Data (WSD)  :    @tab X
       WVE			  :    @tab X
       Konami XMD		  :    @tab X
       XMV			  :    @tab X
	       @tab Microsoft video container used in Xbox games.

       XVAG			  :    @tab X
	       @tab Audio format used on the PS3.

       xWMA			  :    @tab X
	       @tab Microsoft audio container used by XAudio 2.

       eXtended BINary text (XBIN)  :  @tab X
       YUV4MPEG pipe		  :  X @tab X
       Psygnosis YOP		  :    @tab X

       "X"  means  that	 the  feature  in that column (encoding / decoding) is
       supported.

   Image Formats
       FFmpeg can read and write images for each frame of  a  video  sequence.
       The following image formats are supported:

       Name  :	Encoding @tab Decoding @tab Comments
       .Y.U.V	     :	X @tab X
	       @tab one raw file per component

       Alias PIX     :	X @tab X
	       @tab Alias/Wavefront PIX image format

       animated GIF  :	X @tab X
       APNG	     :	X @tab X
	       @tab Animated Portable Network Graphics

       BMP	     :	X @tab X
	       @tab Microsoft BMP image

       BRender PIX   :	  @tab X
	       @tab Argonaut BRender 3D engine image format.

       CRI	     :	  @tab X
	       @tab Cintel RAW

       DPX	     :	X @tab X
	       @tab Digital Picture Exchange

       EXR	     :	  @tab X
	       @tab OpenEXR

       FITS	     :	X @tab X
	       @tab Flexible Image Transport System

       HDR	     :	X @tab X
	       @tab Radiance HDR RGBE Image format

       IMG	     :	  @tab X
	       @tab GEM Raster image

       JPEG	     :	X @tab X
	       @tab Progressive JPEG is not supported.

       JPEG 2000     :	X @tab X
       JPEG-LS	     :	X @tab X
       LJPEG	     :	X @tab
	       @tab Lossless JPEG

       Media 100     :	  @tab X
       MSP	     :	  @tab X
	       @tab Microsoft Paint image

       PAM	     :	X @tab X
	       @tab PAM is a PNM extension with alpha support.

       PBM	     :	X @tab X
	       @tab Portable BitMap image

       PCD	     :	  @tab X
	       @tab PhotoCD

       PCX	     :	X @tab X
	       @tab PC Paintbrush

       PFM	     :	X @tab X
	       @tab Portable FloatMap image

       PGM	     :	X @tab X
	       @tab Portable GrayMap image

       PGMYUV	     :	X @tab X
	       @tab PGM with U and V components in YUV 4:2:0

       PGX	     :	  @tab X
	       @tab PGX file decoder

       PHM	     :	X @tab X
	       @tab Portable HalfFloatMap image

       PIC	     :	@tab X
	       @tab Pictor/PC Paint

       PNG	     :	X @tab X
	       @tab Portable Network Graphics image

       PPM	     :	X @tab X
	       @tab Portable PixelMap image

       PSD	     :	  @tab X
	       @tab Photoshop

       PTX	     :	  @tab X
	       @tab V.Flash PTX format

       QOI	     :	X @tab X
	       @tab Quite OK Image format

       SGI	     :	X @tab X
	       @tab SGI RGB image format

       Sun Rasterfile	:  X @tab X
	       @tab Sun RAS image format

       TIFF	     :	X @tab X
	       @tab YUV, JPEG and some extension is not supported yet.

       Truevision Targa	  :  X @tab X
	       @tab Targa (.TGA) image format

       VBN   :	X @tab X
	       @tab Vizrt Binary Image format

       WBMP	     :	X @tab X
	       @tab Wireless Application Protocol Bitmap image format

       WebP	     :	E @tab X
	       @tab WebP image format, encoding supported through external library libwebp

       XBM   :	X @tab X
	       @tab X BitMap image format

       XFace  :	 X @tab X
	       @tab X-Face image format

       XPM   :	  @tab X
	       @tab X PixMap image format

       XWD   :	X @tab X
	       @tab X Window Dump image format

       "X"  means  that	 the  feature  in that column (encoding / decoding) is
       supported.

       "E" means that support is provided through an external library.

   Video Codecs
       Name  :	Encoding @tab Decoding @tab Comments
       4X Movie		       :      @tab  X
	       @tab Used in certain computer games.

       8088flex TMV	       :      @tab  X
       A64 multicolor	       :   X  @tab
	       @tab Creates video suitable to be played on a commodore 64 (multicolor mode).

       Amazing Studio PAF Video	 :	@tab  X
       American Laser Games MM	 :     @tab X
	       @tab Used in games like Mad Dog McCree.

       Amuse Graphics Movie    :      @tab  X
       AMV Video	       :   X  @tab  X
	       @tab Used in Chinese MP3 players.

       ANSI/ASCII art	       :      @tab  X
       Apple Intermediate Codec	 :	@tab  X
       Apple MJPEG-B	       :      @tab  X
       Apple Pixlet	       :      @tab  X
       Apple ProRes	       :   X  @tab  X
	       @tab fourcc: apch,apcn,apcs,apco,ap4h,ap4x

       Apple QuickDraw	       :      @tab  X
	       @tab fourcc: qdrw

       Argonaut Video	       :      @tab  X
	       @tab Used in some Argonaut games.

       Asus v1		       :   X  @tab  X
	       @tab fourcc: ASV1

       Asus v2		       :   X  @tab  X
	       @tab fourcc: ASV2

       ATI VCR1		       :      @tab  X
	       @tab fourcc: VCR1

       ATI VCR2		       :      @tab  X
	       @tab fourcc: VCR2

       Auravision Aura	       :      @tab  X
       Auravision Aura 2       :      @tab  X
       Autodesk Animator Flic video   :	     @tab  X
       Autodesk RLE	       :      @tab  X
	       @tab fourcc: AASC

       AV1		       :   E  @tab  E
	       @tab Supported through external libraries libaom, libdav1d, librav1e and libsvtav1

       Avid 1:1 10-bit RGB Packer   :	X  @tab	 X
	       @tab fourcc: AVrp

       AVS (Audio Video Standard) video	  :	 @tab  X
	       @tab Video encoding used by the Creature Shock game.

       AVS2-P2/IEEE1857.4      :   E  @tab  E
	       @tab Supported through external libraries libxavs2 and libdavs2

       AVS3-P2/IEEE1857.10     :      @tab  E
	       @tab Supported through external library libuavs3d

       AYUV		       :   X  @tab  X
	       @tab Microsoft uncompressed packed 4:4:4:4

       Beam Software VB	       :      @tab  X
       Bethesda VID video      :      @tab  X
	       @tab Used in some games from Bethesda Softworks.

       Bink Video	       :      @tab  X
       BitJazz SheerVideo      :      @tab  X
       Bitmap Brothers JV video	  :    @tab X
       y41p Brooktree uncompressed 4:1:1 12-bit	     :	 X  @tab  X
       Brooktree ProSumer Video	  :	 @tab  X
	       @tab fourcc: BT20

       Brute Force & Ignorance	  :    @tab X
	       @tab Used in the game Flash Traffic: City of Angels.

       C93 video	       :      @tab  X
	       @tab Codec used in Cyberia game.

       CamStudio	       :      @tab  X
	       @tab fourcc: CSCD

       CD+G		       :      @tab  X
	       @tab Video codec for CD+G karaoke disks

       CDXL		       :      @tab  X
	       @tab Amiga CD video codec

       Chinese AVS video       :   E  @tab  X
	       @tab AVS1-P2, JiZhun profile, encoding through external library libxavs

       Delphine Software International CIN video   :	  @tab	X
	       @tab Codec used in Delphine Software International games.

       Discworld II BMV Video  :      @tab  X
       CineForm HD	       :   X  @tab  X
       Canopus HQ	       :      @tab  X
       Canopus HQA	       :      @tab  X
       Canopus HQX	       :      @tab  X
       Canopus Lossless Codec  :      @tab  X
       CDToons		       :      @tab  X
	       @tab Codec used in various Broderbund games.

       Cinepak		       :      @tab  X
       Cirrus Logic AccuPak    :   X  @tab  X
	       @tab fourcc: CLJR

       CPiA Video Format       :      @tab  X
       Creative YUV (CYUV)     :      @tab  X
       DFA		       :      @tab  X
	       @tab Codec used in Chronomaster game.

       Dirac		       :   E  @tab  X
	       @tab supported though the native vc2 (Dirac Pro) encoder

       Deluxe Paint Animation  :      @tab  X
       DNxHD		       :    X @tab  X
	       @tab aka SMPTE VC3

       Duck TrueMotion 1.0    :	     @tab  X
	       @tab fourcc: DUCK

       Duck TrueMotion 2.0     :      @tab  X
	       @tab fourcc: TM20

       Duck TrueMotion 2.0 RT  :      @tab  X
	       @tab fourcc: TR20

       DV (Digital Video)      :   X  @tab  X
       Dxtory capture format   :      @tab  X
       Feeble Files/ScummVM DXA	  :	 @tab  X
	       @tab Codec originally used in Feeble Files game.

       Electronic Arts CMV video   :	  @tab	X
	       @tab Used in NHL 95 game.

       Electronic Arts Madcow video   :	     @tab  X
       Electronic Arts TGV video   :	  @tab	X
       Electronic Arts TGQ video   :	  @tab	X
       Electronic Arts TQI video   :	  @tab	X
       Escape 124	       :      @tab  X
       Escape 130	       :      @tab  X
       FFmpeg video codec #1   :   X  @tab  X
	       @tab lossless codec (fourcc: FFV1)

       Flash Screen Video v1   :   X  @tab  X
	       @tab fourcc: FSV1

       Flash Screen Video v2   :   X  @tab  X
       Flash Video (FLV)       :   X  @tab  X
	       @tab Sorenson H.263 used in Flash

       FM Screen Capture Codec	 :	@tab  X
       Forward Uncompressed    :      @tab  X
       Fraps		       :      @tab  X
       Go2Meeting	       :      @tab  X
	       @tab fourcc: G2M2, G2M3

       Go2Webinar	       :      @tab  X
	       @tab fourcc: G2M4

       Gremlin Digital Video   :      @tab  X
       H.261		       :   X  @tab  X
       H.263 / H.263-1996      :   X  @tab  X
       H.263+ / H.263-1998 / H.263 version 2   :   X  @tab  X
       H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10   :   E  @tab	X
	       @tab encoding supported through external library libx264 and OpenH264

       HEVC		       :   X  @tab  X
	       @tab encoding supported through external library libx265 and libkvazaar

       HNM version 4	       :      @tab  X
       HuffYUV		       :   X  @tab  X
       HuffYUV FFmpeg variant  :   X  @tab  X
       IBM Ultimotion	       :      @tab  X
	       @tab fourcc: ULTI

       id Cinematic video      :      @tab  X
	       @tab Used in Quake II.

       id RoQ video	       :   X  @tab  X
	       @tab Used in Quake III, Jedi Knight 2, other computer games.

       IFF ILBM		       :      @tab  X
	       @tab IFF interleaved bitmap

       IFF ByteRun1	       :      @tab  X
	       @tab IFF run length encoded bitmap

       Infinity IMM4	       :      @tab  X
       Intel H.263	       :      @tab  X
       Intel Indeo 2	       :      @tab  X
       Intel Indeo 3	       :      @tab  X
       Intel Indeo 4	       :      @tab  X
       Intel Indeo 5	       :      @tab  X
       Interplay C93	       :      @tab  X
	       @tab Used in the game Cyberia from Interplay.

       Interplay MVE video     :      @tab  X
	       @tab Used in Interplay .MVE files.

       J2K  :	X  @tab	 X
       Karl Morton's video codec   :	  @tab	X
	       @tab Codec used in Worms games.

       Kega Game Video (KGV1)  :       @tab  X
	       @tab Kega emulator screen capture codec.

       Lagarith		       :      @tab  X
       LCL (LossLess Codec Library) MSZH   :	  @tab	X
       LCL (LossLess Codec Library) ZLIB   :   E  @tab	E
       LOCO		       :      @tab  X
       LucasArts SANM/Smush    :      @tab  X
	       @tab Used in LucasArts games / SMUSH animations.

       lossless MJPEG	       :   X  @tab  X
       MagicYUV Video	       :   X  @tab  X
       Mandsoft Screen Capture Codec   :      @tab  X
       Microsoft ATC Screen    :      @tab  X
	       @tab Also known as Microsoft Screen 3.

       Microsoft Expression Encoder Screen   :	    @tab  X
	       @tab Also known as Microsoft Titanium Screen 2.

       Microsoft RLE	       :   X  @tab  X
       Microsoft Screen 1      :      @tab  X
	       @tab Also known as Windows Media Video V7 Screen.

       Microsoft Screen 2      :      @tab  X
	       @tab Also known as Windows Media Video V9 Screen.

       Microsoft Video 1       :      @tab  X
       Mimic		       :      @tab  X
	       @tab Used in MSN Messenger Webcam streams.

       Miro VideoXL	       :      @tab  X
	       @tab fourcc: VIXL

       MJPEG (Motion JPEG)     :   X  @tab  X
       Mobotix MxPEG video     :      @tab  X
       Motion Pixels video     :      @tab  X
       MPEG-1 video	       :   X  @tab  X
       MPEG-2 video	       :   X  @tab  X
       MPEG-4 part 2	       :   X  @tab  X
	       @tab libxvidcore can be used alternatively for encoding.

       MPEG-4 part 2 Microsoft variant version 1   :	  @tab	X
       MPEG-4 part 2 Microsoft variant version 2   :   X  @tab	X
       MPEG-4 part 2 Microsoft variant version 3   :   X  @tab	X
       Newtek SpeedHQ		     :	 X  @tab  X
       Nintendo Gamecube THP video   :	    @tab  X
       NotchLC		       :      @tab  X
       NuppelVideo/RTjpeg      :      @tab  X
	       @tab Video encoding used in NuppelVideo files.

       On2 VP3		       :      @tab  X
	       @tab still experimental

       On2 VP4		       :      @tab  X
	       @tab fourcc: VP40

       On2 VP5		       :      @tab  X
	       @tab fourcc: VP50

       On2 VP6		       :      @tab  X
	       @tab fourcc: VP60,VP61,VP62

       On2 VP7		       :      @tab  X
	       @tab fourcc: VP70,VP71

       VP8		       :   E  @tab  X
	       @tab fourcc: VP80, encoding supported through external library libvpx

       VP9		       :   E  @tab  X
	       @tab encoding supported through external library libvpx

       Pinnacle TARGA CineWave YUV16  :	     @tab  X
	       @tab fourcc: Y216

       Q-team QPEG	       :      @tab  X
	       @tab fourccs: QPEG, Q1.0, Q1.1

       QuickTime 8BPS video    :      @tab  X
       QuickTime Animation (RLE) video	 :   X	@tab  X
	       @tab fourcc: 'rle '

       QuickTime Graphics (SMC)	  :   X	 @tab  X
	       @tab fourcc: 'smc '

       QuickTime video (RPZA)  :   X  @tab  X
	       @tab fourcc: rpza

       R10K AJA Kona 10-bit RGB Codec	   :   X  @tab	X
       R210 Quicktime Uncompressed RGB 10-bit	   :   X  @tab	X
       Raw Video	       :   X  @tab  X
       RealVideo 1.0	       :   X  @tab  X
       RealVideo 2.0	       :   X  @tab  X
       RealVideo 3.0	       :      @tab  X
	       @tab still far from ideal

       RealVideo 4.0	       :      @tab  X
       Renderware TXD (TeXture Dictionary)   :	    @tab  X
	       @tab Texture dictionaries used by the Renderware Engine.

       RivaTuner Video	       :      @tab  X
	       @tab fourcc: 'RTV1'

       RL2 video	       :      @tab  X
	       @tab used in some games by Entertainment Software Partners

       ScreenPressor	       :      @tab  X
       Screenpresso	       :      @tab  X
       Screen Recorder Gold Codec   :	   @tab	 X
       Sierra VMD video	       :      @tab  X
	       @tab Used in Sierra VMD files.

       Silicon Graphics Motion Video Compressor 1 (MVC1)   :	  @tab	X
       Silicon Graphics Motion Video Compressor 2 (MVC2)   :	  @tab	X
       Silicon Graphics RLE 8-bit video	  :	 @tab  X
       Smacker video	       :      @tab  X
	       @tab Video encoding used in Smacker.

       SMPTE VC-1	       :      @tab  X
       Snow		       :   X  @tab  X
	       @tab experimental wavelet codec (fourcc: SNOW)

       Sony PlayStation MDEC (Motion DECoder)	:      @tab  X
       Sorenson Vector Quantizer 1   :	 X  @tab  X
	       @tab fourcc: SVQ1

       Sorenson Vector Quantizer 3   :	    @tab  X
	       @tab fourcc: SVQ3

       Sunplus JPEG (SP5X)     :      @tab  X
	       @tab fourcc: SP5X

       TechSmith Screen Capture Codec	:      @tab  X
	       @tab fourcc: TSCC

       TechSmith Screen Capture Codec 2	  :	 @tab  X
	       @tab fourcc: TSC2

       Theora		       :   E  @tab  X
	       @tab encoding supported through external library libtheora

       Tiertex Limited SEQ video   :	  @tab	X
	       @tab Codec used in DOS CD-ROM FlashBack game.

       Ut Video		       :   X  @tab  X
       v210 QuickTime uncompressed 4:2:2 10-bit	     :	 X  @tab  X
       v308 QuickTime uncompressed 4:4:4	     :	 X  @tab  X
       v408 QuickTime uncompressed 4:4:4:4	     :	 X  @tab  X
       v410 QuickTime uncompressed 4:4:4 10-bit	     :	 X  @tab  X
       VBLE Lossless Codec     :      @tab  X
       vMix Video	       :      @tab  X
	       @tab fourcc: 'VMX1'

       VMware Screen Codec / VMware Video   :	   @tab	 X
	       @tab Codec used in videos captured by VMware.

       Westwood Studios VQA (Vector Quantized Animation) video	 :	@tab
       X
       Windows Media Image     :      @tab  X
       Windows Media Video 7   :   X  @tab  X
       Windows Media Video 8   :   X  @tab  X
       Windows Media Video 9   :      @tab  X
	       @tab not completely working

       Wing Commander III / Xan	  :	 @tab  X
	       @tab Used in Wing Commander III .MVE files.

       Wing Commander IV / Xan	 :	@tab  X
	       @tab Used in Wing Commander IV.

       Winnov WNV1	       :      @tab  X
       WMV7		       :   X  @tab  X
       YAMAHA SMAF	       :   X  @tab  X
       Psygnosis YOP Video     :      @tab  X
       yuv4		       :   X  @tab  X
	       @tab libquicktime uncompressed packed 4:2:0

       ZeroCodec Lossless Video	 :	@tab  X
       ZLIB		       :   X  @tab  X
	       @tab part of LCL, encoder experimental

       Zip Motion Blocks Video	 :    X @tab  X
	       @tab Encoder works only in PAL8.

       "X" means that the feature in that  column  (encoding  /	 decoding)  is
       supported.

       "E" means that support is provided through an external library.

   Audio Codecs
       Name  :	Encoding @tab Decoding @tab Comments
       8SVX exponential	       :      @tab  X
       8SVX fibonacci	       :      @tab  X
       AAC		       :  EX  @tab  X
	       @tab encoding supported through internal encoder and external library libfdk-aac

       AAC+		       :   E  @tab  IX
	       @tab encoding supported through external library libfdk-aac

       AC-3		       :  IX  @tab  IX
       ACELP.KELVIN	       :      @tab  X
       ADPCM 4X Movie	       :      @tab  X
       ADPCM Yamaha AICA       :      @tab  X
       ADPCM AmuseGraphics Movie  :	@tab  X
       ADPCM Argonaut Games    :  X   @tab  X
       ADPCM CDROM XA	       :      @tab  X
       ADPCM Creative Technology  :	 @tab  X
	       @tab 16 -E<gt> 4, 8 -E<gt> 4, 8 -E<gt> 3, 8 -E<gt> 2

       ADPCM Electronic Arts   :      @tab  X
	       @tab Used in various EA titles.

       ADPCM Electronic Arts Maxis CDROM XS   :	     @tab  X
	       @tab Used in Sim City 3000.

       ADPCM Electronic Arts R1	  :	 @tab  X
       ADPCM Electronic Arts R2	  :	 @tab  X
       ADPCM Electronic Arts R3	  :	 @tab  X
       ADPCM Electronic Arts XAS  :	 @tab  X
       ADPCM G.722	       :   X  @tab  X
       ADPCM G.726	       :   X  @tab  X
       ADPCM IMA Acorn Replay  :      @tab  X
       ADPCM IMA AMV	       :   X  @tab  X
	       @tab Used in AMV files

       ADPCM IMA Cunning Developments	:      @tab  X
       ADPCM IMA Electronic Arts EACS	:      @tab  X
       ADPCM IMA Electronic Arts SEAD	:      @tab  X
       ADPCM IMA Funcom	       :      @tab  X
       ADPCM IMA High Voltage Software ALP	 :   X	@tab  X
       ADPCM IMA Mobiclip MOFLEX   :	  @tab	X
       ADPCM IMA QuickTime     :   X  @tab  X
       ADPCM IMA Simon & Schuster Interactive	 :   X	@tab  X
       ADPCM IMA Ubisoft APM   :   X  @tab  X
       ADPCM IMA Loki SDL MJPEG	  :	 @tab  X
       ADPCM IMA WAV	       :   X  @tab  X
       ADPCM IMA Westwood      :      @tab  X
       ADPCM ISS IMA	       :      @tab  X
	       @tab Used in FunCom games.

       ADPCM IMA Dialogic      :      @tab  X
       ADPCM IMA Duck DK3      :      @tab  X
	       @tab Used in some Sega Saturn console games.

       ADPCM IMA Duck DK4      :      @tab  X
	       @tab Used in some Sega Saturn console games.

       ADPCM IMA Radical       :      @tab  X
       ADPCM Microsoft	       :   X  @tab  X
       ADPCM MS IMA	       :   X  @tab  X
       ADPCM Nintendo Gamecube AFC   :	    @tab  X
       ADPCM Nintendo Gamecube DTK   :	    @tab  X
       ADPCM Nintendo THP   :	   @tab	 X
       ADPCM Playstation       :      @tab  X
       ADPCM QT IMA	       :   X  @tab  X
       ADPCM SEGA CRI ADX      :   X  @tab  X
	       @tab Used in Sega Dreamcast games.

       ADPCM Shockwave Flash   :   X  @tab  X
       ADPCM Sound Blaster Pro 2-bit   :      @tab  X
       ADPCM Sound Blaster Pro 2.6-bit	 :	@tab  X
       ADPCM Sound Blaster Pro 4-bit   :      @tab  X
       ADPCM VIMA	       :      @tab  X
	       @tab Used in LucasArts SMUSH animations.

       ADPCM Konami XMD	       :      @tab  X
       ADPCM Westwood Studios IMA	:   X @tab  X
	       @tab Used in Westwood Studios games like Command and Conquer.

       ADPCM Yamaha	       :   X  @tab  X
       ADPCM Zork	       :      @tab  X
       AMR-NB		       :   E  @tab  X
	       @tab encoding supported through external library libopencore-amrnb

       AMR-WB		       :   E  @tab  X
	       @tab encoding supported through external library libvo-amrwbenc

       Amazing Studio PAF Audio	 :	@tab  X
       Apple lossless audio    :   X  @tab  X
	       @tab QuickTime fourcc 'alac'

       aptX		       :   X  @tab  X
	       @tab Used in Bluetooth A2DP

       aptX HD		       :   X  @tab  X
	       @tab Used in Bluetooth A2DP

       ATRAC1		       :      @tab  X
       ATRAC3		       :      @tab  X
       ATRAC3+		       :      @tab  X
       ATRAC9		       :      @tab  X
       Bink Audio	       :      @tab  X
	       @tab Used in Bink and Smacker files in many games.

       Bonk audio	       :      @tab  X
       CELT		       :      @tab  E
	       @tab decoding supported through external library libcelt

       codec2		       :   E  @tab  E
	       @tab en/decoding supported through external library libcodec2

       CRI HCA		       :      @tab X
       Delphine Software International CIN audio   :	  @tab	X
	       @tab Codec used in Delphine Software International games.

       DFPWM		       :   X  @tab  X
       Digital Speech Standard - Standard Play mode (DSS SP)  :	     @tab  X
       Discworld II BMV Audio  :      @tab  X
       COOK		       :      @tab  X
	       @tab All versions except 5.1 are supported.

       DCA (DTS Coherent Acoustics)   :	  X  @tab  X
	       @tab supported extensions: XCh, XXCH, X96, XBR, XLL, LBR (partially)

       Dolby E	 :	@tab  X
       DPCM Cuberoot-Delta-Exact  :   @tab  X
	       @tab Used in few games.

       DPCM Gremlin	       :      @tab  X
       DPCM id RoQ	       :   X  @tab  X
	       @tab Used in Quake III, Jedi Knight 2 and other computer games.

       DPCM Marble WADY	       :      @tab  X
       DPCM Interplay	       :      @tab  X
	       @tab Used in various Interplay computer games.

       DPCM Squareroot-Delta-Exact   :	 @tab  X
	       @tab Used in various games.

       DPCM Sierra Online      :      @tab  X
	       @tab Used in Sierra Online game audio files.

       DPCM Sol		       :      @tab  X
       DPCM Xan		       :      @tab  X
	       @tab Used in Origin's Wing Commander IV AVI files.

       DPCM Xilam DERF	       :      @tab  X
       DSD (Direct Stream Digital), least significant bit first	  :   @tab  X
       DSD (Direct Stream Digital), most significant bit first	  :   @tab  X
       DSD (Direct Stream Digital), least significant bit first, planar	  :
       @tab  X
       DSD (Direct Stream Digital), most significant bit first, planar	  :
       @tab  X
       DSP Group TrueSpeech    :      @tab  X
       DST (Direct Stream Transfer)  :	 @tab  X
       DV audio		       :      @tab  X
       Enhanced AC-3	       :   X  @tab  X
       EVRC (Enhanced Variable Rate Codec)  :	   @tab	 X
       FLAC (Free Lossless Audio Codec)	  :   X	 @tab  IX
       FTR Voice	       :      @tab  X
       G.723.1		       :  X   @tab  X
       G.729		       :      @tab  X
       GSM		       :   E  @tab  X
	       @tab encoding supported through external library libgsm

       GSM Microsoft variant   :   E  @tab  X
	       @tab encoding supported through external library libgsm

       IAC (Indeo Audio Coder)	 :	@tab  X
       iLBC (Internet Low Bitrate Codec)  :   E	 @tab  EX
	       @tab encoding and decoding supported through external library libilbc

       IMC (Intel Music Coder)	 :	@tab  X
       Interplay ACM		 :	@tab  X
       MACE (Macintosh Audio Compression/Expansion) 3:1	  :	 @tab  X
       MACE (Macintosh Audio Compression/Expansion) 6:1	  :	 @tab  X
       Marian's A-pac audio	 :	@tab  X
       MI-SC4 (Micronas SC-4 Audio)   :	     @tab  X
       MLP (Meridian Lossless Packing)	 :   X	@tab  X
	       @tab Used in DVD-Audio discs.

       Monkey's Audio	       :      @tab  X
       MP1 (MPEG audio layer 1)	  :	 @tab IX
       MP2 (MPEG audio layer 2)	  :  IX	 @tab IX
	       @tab encoding supported also through external library TwoLAME

       MP3 (MPEG audio layer 3)	  :   E	 @tab IX
	       @tab encoding supported through external library LAME, ADU MP3 and MP3onMP4 also supported

       MPEG-4 Audio Lossless Coding (ALS)   :	   @tab	 X
       MobiClip FastAudio      :      @tab  X
       Musepack SV7	       :      @tab  X
       Musepack SV8	       :      @tab  X
       Nellymoser Asao	       :   X  @tab  X
       On2 AVC (Audio for Video Codec)	:      @tab  X
       Opus		       :   E  @tab  X
	       @tab encoding supported through external library libopus

       OSQ (Original Sound Quality)   :	     @tab  X
       PCM A-law	       :   X  @tab  X
       PCM mu-law	       :   X  @tab  X
       PCM Archimedes VIDC     :   X  @tab  X
       PCM signed 8-bit planar	 :   X	@tab  X
       PCM signed 16-bit big-endian planar   :	 X  @tab  X
       PCM signed 16-bit little-endian planar	:   X  @tab  X
       PCM signed 24-bit little-endian planar	:   X  @tab  X
       PCM signed 32-bit little-endian planar	:   X  @tab  X
       PCM 32-bit floating point big-endian   :	  X  @tab  X
       PCM 32-bit floating point little-endian	 :   X	@tab  X
       PCM 64-bit floating point big-endian   :	  X  @tab  X
       PCM 64-bit floating point little-endian	 :   X	@tab  X
       PCM D-Cinema audio signed 24-bit	   :   X  @tab	X
       PCM signed 8-bit	       :   X  @tab  X
       PCM signed 16-bit big-endian   :	  X  @tab  X
       PCM signed 16-bit little-endian	 :   X	@tab  X
       PCM signed 24-bit big-endian   :	  X  @tab  X
       PCM signed 24-bit little-endian	 :   X	@tab  X
       PCM signed 32-bit big-endian   :	  X  @tab  X
       PCM signed 32-bit little-endian	 :   X	@tab  X
       PCM signed 16/20/24-bit big-endian in MPEG-TS   :      @tab  X
       PCM unsigned 8-bit      :   X  @tab  X
       PCM unsigned 16-bit big-endian	:   X  @tab  X
       PCM unsigned 16-bit little-endian   :   X  @tab	X
       PCM unsigned 24-bit big-endian	:   X  @tab  X
       PCM unsigned 24-bit little-endian   :   X  @tab	X
       PCM unsigned 32-bit big-endian	:   X  @tab  X
       PCM unsigned 32-bit little-endian   :   X  @tab	X
       PCM SGA		       :      @tab  X
       QCELP / PureVoice       :      @tab  X
       QDesign Music Codec 1   :      @tab  X
       QDesign Music Codec 2   :      @tab  X
	       @tab There are still some distortions.

       RealAudio 1.0 (14.4K)   :   X  @tab  X
	       @tab Real 14400 bit/s codec

       RealAudio 2.0 (28.8K)   :      @tab  X
	       @tab Real 28800 bit/s codec

       RealAudio 3.0 (dnet)    :  IX  @tab  X
	       @tab Real low bitrate AC-3 codec

       RealAudio Lossless      :      @tab  X
       RealAudio SIPR / ACELP.NET  :	  @tab	X
       RK Audio (RKA)	       :      @tab  X
       SBC (low-complexity subband codec)  :   X  @tab	X
	       @tab Used in Bluetooth A2DP

       Shorten		       :      @tab  X
       Sierra VMD audio	       :      @tab  X
	       @tab Used in Sierra VMD files.

       Smacker audio	       :      @tab  X
       SMPTE 302M AES3 audio   :   X  @tab  X
       Sonic		       :   X  @tab  X
	       @tab experimental codec

       Sonic lossless	       :   X  @tab  X
	       @tab experimental codec

       Speex		       :   E  @tab  EX
	       @tab supported through external library libspeex

       TAK (Tom's lossless Audio Kompressor)   :      @tab  X
       True Audio (TTA)	       :   X  @tab  X
       TrueHD		       :   X  @tab  X
	       @tab Used in HD-DVD and Blu-Ray discs.

       TwinVQ (VQF flavor)     :      @tab  X
       VIMA		       :      @tab  X
	       @tab Used in LucasArts SMUSH animations.

       ViewQuest VQC	       :      @tab  X
       Vorbis		       :   E  @tab  X
	       @tab A native but very primitive encoder exists.

       Voxware MetaSound       :      @tab  X
       Waveform Archiver       :      @tab  X
       WavPack		       :   X  @tab  X
       Westwood Audio (SND1)   :      @tab  X
       Windows Media Audio 1   :   X  @tab  X
       Windows Media Audio 2   :   X  @tab  X
       Windows Media Audio Lossless  :	 @tab  X
       Windows Media Audio Pro	:     @tab  X
       Windows Media Audio Voice  :   @tab  X
       Xbox Media Audio 1      :      @tab  X
       Xbox Media Audio 2      :      @tab  X

       "X"  means  that	 the  feature  in that column (encoding / decoding) is
       supported.

       "E" means that support is provided through an external library.

       "I" means that an integer-only version is available, too (ensures  high
       performance on systems without hardware floating point support).

   Subtitle Formats
       Name  :	Muxing @tab Demuxing @tab Encoding @tab Decoding
       3GPP Timed Text	 :    @tab   @tab X @tab X
       AQTitle		 :    @tab X @tab   @tab X
       DVB		 :  X @tab X @tab X @tab X
       DVB teletext	 :    @tab X @tab   @tab E
       DVD		 :  X @tab X @tab X @tab X
       JACOsub		 :  X @tab X @tab   @tab X
       MicroDVD		 :  X @tab X @tab   @tab X
       MPL2		 :    @tab X @tab   @tab X
       MPsub (MPlayer)	 :    @tab X @tab   @tab X
       PGS		 :    @tab   @tab   @tab X
       PJS (Phoenix)	 :    @tab X @tab   @tab X
       RealText		 :    @tab X @tab   @tab X
       SAMI		 :    @tab X @tab   @tab X
       Spruce format (STL)  :	 @tab X @tab   @tab X
       SSA/ASS		 :  X @tab X @tab X @tab X
       SubRip (SRT)	 :  X @tab X @tab X @tab X
       SubViewer v1	 :    @tab X @tab   @tab X
       SubViewer	 :    @tab X @tab   @tab X
       TED Talks captions  :  @tab X @tab   @tab X
       TTML		 :  X @tab   @tab X @tab
       VobSub (IDX+SUB)	 :    @tab X @tab   @tab X
       VPlayer		 :    @tab X @tab   @tab X
       WebVTT		 :  X @tab X @tab X @tab X
       XSUB		 :    @tab   @tab X @tab X

       "X" means that the feature is supported.

       "E" means that support is provided through an external library.

   Network Protocols
       Name	     :	Support
       AMQP	     :	E
       file	     :	X
       FTP	     :	X
       Gopher	     :	X
       Gophers	     :	X
       HLS	     :	X
       HTTP	     :	X
       HTTPS	     :	X
       Icecast	     :	X
       MMSH	     :	X
       MMST	     :	X
       pipe	     :	X
       Pro-MPEG FEC  :	X
       RTMP	     :	X
       RTMPE	     :	X
       RTMPS	     :	X
       RTMPT	     :	X
       RTMPTE	     :	X
       RTMPTS	     :	X
       RTP	     :	X
       SAMBA	     :	E
       SCTP	     :	X
       SFTP	     :	E
       TCP	     :	X
       TLS	     :	X
       UDP	     :	X
       ZMQ	     :	E

       "X" means that the protocol is supported.

       "E" means that support is provided through an external library.

   Input/Output Devices
       Name		  :  Input  @tab Output
       ALSA		  :  X	    @tab X
       BKTR		  :  X	    @tab
       caca		  :	    @tab X
       DV1394		  :  X	    @tab
       Lavfi virtual device  :	X   @tab
       Linux framebuffer  :  X	    @tab X
       JACK		  :  X	    @tab
       LIBCDIO		  :  X
       LIBDC1394	  :  X	    @tab
       OpenAL		  :  X
       OpenGL		  :	    @tab X
       OSS		  :  X	    @tab X
       PulseAudio	  :  X	    @tab X
       SDL		  :	    @tab X
       Video4Linux2	  :  X	    @tab X
       VfW capture	  :  X	    @tab
       X11 grabbing	  :  X	    @tab
       Win32 grabbing	  :  X	    @tab

       "X" means that input/output is supported.

   Timecode
       Codec/format	  :  Read   @tab Write
       AVI		  :  X	    @tab X
       DV		  :  X	    @tab X
       GXF		  :  X	    @tab X
       MOV		  :  X	    @tab X
       MPEG1/2		  :  X	    @tab X
       MXF		  :  X	    @tab X

SEE ALSO
       ffmpeg(1),  ffplay(1),  ffprobe(1),  ffmpeg-utils(1), ffmpeg-scaler(1),
       ffmpeg-resampler(1),   ffmpeg-codecs(1),	  ffmpeg-bitstream-filters(1),
       ffmpeg-formats(1),	 ffmpeg-devices(1),	  ffmpeg-protocols(1),
       ffmpeg-filters(1)

AUTHORS
       The FFmpeg developers.

       For details about the authorship, see the Git history  of  the  project
       (https://git.ffmpeg.org/ffmpeg),	 e.g. by typing the command git log in
       the FFmpeg source directory,  or	 browsing  the	online	repository  at
       <https://git.ffmpeg.org/ffmpeg>.

       Maintainers  for	 the  specific	components  are	 listed	 in  the  file
       MAINTAINERS in the source code tree.

								 FFMPEG-ALL(1)
