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

NAME
       ffmpeg - ffmpeg video converter

SYNOPSIS
       ffmpeg [global_options] {[input_file_options] -i input_file} ...
       {[output_file_options] output_file} ...

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

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

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

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

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

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

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

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

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

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

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

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

       The format option may be needed for raw input files.

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

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

       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 |  simple filtergraph	| filtered |  encoder	| encoded data |
	       | frames	 | -------------------> | frames   | ---------> | packets      |
	       |_________|			|__________|		|______________|

       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 | ---> | fps | ---> | output |
	       |_______|      |_____________|	   |_______|	  |_____|      |________|

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

       Complex filtergraphs

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

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

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

       The -lavfi option is equivalent to -filter_complex.

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

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

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

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

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

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

OPTIONS
       All the numerical options, if not specified otherwise, accept a string
       representing a number as input, which may be followed by one of the SI
       unit prefixes, for example: 'K', 'M', or 'G'.

       If 'i' is appended to the SI unit prefix, the complete prefix will be
       interpreted as a unit prefix for binary multiplies, which are based on
       powers of 1024 instead of powers of 1000. Appending 'B' to the SI unit
       prefix multiplies the value by 8. This allows using, for example: 'KB',
       'MiB', 'G' and 'B' as number suffixes.

       Options which do not take arguments are boolean options, and set the
       corresponding value to true. They can be set to false by prefixing the
       option name with "no". For example using "-nofoo" will set the boolean
       option with name "foo" to false.

   Stream specifiers
       Some options are applied per-stream, e.g. bitrate or codec. Stream
       specifiers are used to precisely specify which stream(s) a given option
       belongs to.

       A stream specifier is a string generally appended to the option name
       and separated from it by a colon. E.g. "-codec:a:1 ac3" contains the
       "a:1" stream specifier, which matches the second audio stream.
       Therefore, it would select the ac3 codec for the second audio stream.

       A stream specifier can match several streams, so that the option is
       applied to all of them. E.g. the stream specifier in "-b:a 128k"
       matches all audio streams.

       An empty stream specifier matches all streams. For example, "-codec
       copy" or "-codec: copy" would copy all the streams without reencoding.

       Possible forms of stream specifiers are:

       stream_index
	   Matches the stream with this index. E.g. "-threads:1 4" would set
	   the thread count for the second stream to 4.

       stream_type[:stream_index]
	   stream_type is one of following: 'v' for video, 'a' for audio, 's'
	   for subtitle, 'd' for data, and 't' for attachments. If
	   stream_index is given, then it matches stream number stream_index
	   of this type. Otherwise, it matches all streams of this type.

       p:program_id[:stream_index]
	   If stream_index is given, then it matches the stream with number
	   stream_index in the program with the id program_id. Otherwise, it
	   matches all streams in the program.

       #stream_id
	   Matches the stream by a format-specific ID.

   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.

	   Possible values of arg are:

	   decoder=decoder_name
	       Print detailed information about the decoder named
	       decoder_name. Use the -decoders option to get a list of all
	       decoders.

	   encoder=encoder_name
	       Print detailed information about the encoder named
	       encoder_name. Use the -encoders option to get a list of all
	       encoders.

	   demuxer=demuxer_name
	       Print detailed information about the demuxer named
	       demuxer_name. Use the -formats option to get a list of all
	       demuxers and muxers.

	   muxer=muxer_name
	       Print detailed information about the muxer named muxer_name.
	       Use the -formats option to get a list of all muxers and
	       demuxers.

	   filter=filter_name
	       Print detailed information about the filter name filter_name.
	       Use the -filters option to get a list of all filters.

       -version
	   Show version.

       -formats
	   Show available formats.

       -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.

       -loglevel [repeat+]loglevel | -v [repeat+]loglevel
	   Set the logging level used by the library.  Adding "repeat+"
	   indicates that repeated log output should not be compressed to the
	   first line and the "Last message repeated n times" line will be
	   omitted. "repeat" can also be used alone.  If "repeat" is used
	   alone, and with no prior loglevel set, the default loglevel will be
	   used. If multiple loglevel parameters are given, using 'repeat'
	   will not change the loglevel.  loglevel is a number or a string
	   containing one of the following values:

	   quiet
	       Show nothing at all; be silent.

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

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

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

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

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

	   verbose
	       Same as "info", except more verbose.

	   debug
	       Show everything, including debugging information.

	   By default the program logs to stderr, if coloring is supported by
	   the terminal, colors are used to mark errors and warnings. Log
	   coloring can be disabled setting the environment variable
	   AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting the
	   environment variable AV_LOG_FORCE_COLOR.  The use of the
	   environment variable NO_COLOR is deprecated and will be dropped in
	   a following FFmpeg version.

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

	   Setting the environment variable "FFREPORT" to any value has the
	   same effect. If the value is a ':'-separated key=value sequence,
	   these options will affect the report; options 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 option is 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 "%"

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

       -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
	       xop
	       fma4
	       3dnow
	       3dnowext
	       cmov
	   ARM
	       armv5te
	       armv6
	       armv6t2
	       vfp
	       vfpv3
	       neon
	   PowerPC
	       altivec
	   Specific Processors
	       pentium2
	       pentium3
	       pentium4
	       k6
	       k62
	       athlon
	       athlonxp
	       k8
       -opencl_options options (global)
	   Set OpenCL environment options. This option is only available when
	   FFmpeg has been compiled with "--enable-opencl".

	   options must be a list of key=value option pairs separated by ':'.
	   See the ``OpenCL Options'' section in the ffmpeg-utils manual for
	   the list of supported options.

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

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

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

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

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

       All codec AVOptions are obviously per-stream, so the chapter on stream
       specifiers applies to them

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

       Note2 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 filename (input)
	   input file name

       -y (global)
	   Overwrite output files without asking.

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

       -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 (output)
	   Stop writing the output after its duration reaches duration.
	   duration may be a number in seconds, or in "hh:mm:ss[.xxx]" form.

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

       -to position (output)
	   Stop writing the output at position.	 position may be a number in
	   seconds, or in "hh:mm:ss[.xxx]" form.

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

       -fs limit_size (output)
	   Set the file size limit, expressed in bytes.

       -ss position (input/output)
	   When used as an input option (before "-i"), seeks in this input
	   file to position. When used as an output option (before an output
	   filename), decodes but discards input until the timestamps reach
	   position. This is slower, but more accurate.

	   position may be either in seconds or in "hh:mm:ss[.xxx]" form.

       -itsoffset offset (input)
	   Set the input time offset in seconds.  "[-]hh:mm:ss[.xxx]" syntax
	   is also supported.  The offset is added to the timestamps of the
	   input files.	 Specifying a positive offset means that the
	   corresponding streams are delayed by offset seconds.

       -timestamp time (output)
	   Set the recording timestamp in the container.  The syntax for time
	   is:

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

	   If the value is "now" it takes the current time.  Time is local
	   time unless 'Z' or '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.

       -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 or chapters. 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:1 language=eng OUTPUT

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

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

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

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

       -dframes number (output)
	   Set the number of data frames to record. This is an alias for
	   "-frames:d".

       -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 is codec-dependent.

       -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.

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

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

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

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

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

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

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

	   See also the option "-fdebug ts".

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

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

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

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

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

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

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

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

		   ffmpeg -dump_attachment:t "" -i INPUT

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

   Video Options
       -vframes number (output)
	   Set the number of video frames to record. This is an alias for
	   "-frames:v".

       -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.

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

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

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

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

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

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

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

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

       -vn (output)
	   Disable video recording.

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

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

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

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

       -vlang code
	   Set the ISO 639 language code (3 letters) of the current video
	   stream.

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

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

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

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

       -vdt n
	   Discard threshold.

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

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

       -psnr
	   Calculate PSNR of compressed frames.

       -vstats
	   Dump video coding statistics to vstats_HHMMSS.log.

       -vstats_file file
	   Dump video coding statistics to file.

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

       -dc precision
	   Intra_dc_precision.

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

       -qphist (global)
	   Show QP histogram

       -vbsf bitstream_filter
	   Deprecated see -bsf

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

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

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

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

		   -force_key_frames 0:05:00,chapters-0.1

	   The expression in expr can contain the following constants:

	   n   the number of current processed frame, starting from 0

	   n_forced
	       the number of forced frames

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

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

	   t   the time of the current processed frame

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

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

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

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

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

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

   Audio Options
       -aframes number (output)
	   Set the number of audio frames to record. This is an alias for
	   "-frames:a".

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

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

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

       -an (output)
	   Disable audio recording.

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

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

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

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

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

       -absf bitstream_filter
	   Deprecated, see -bsf

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

   Subtitle options:
       -slang code
	   Set the ISO 639 language code (3 letters) of the current subtitle
	   stream.

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

       -sn (output)
	   Disable subtitle recording.

       -sbsf bitstream_filter
	   Deprecated, see -bsf

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

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

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

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

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

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

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

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

		   ffmpeg -i INPUT -map 0 output

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

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

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

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

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

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

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

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

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

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

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

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

	   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

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

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

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

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

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

	   If metadata specifier is omitted, it defaults to global.

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

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

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

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

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

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

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

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

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

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

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

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

       -re (input)
	   Read input at native frame rate. Mainly used to simulate a grab
	   device.  By default ffmpeg attempts to read the input(s) as fast as
	   possible.  This option will slow down the reading of the input(s)
	   to the native frame rate of the input(s). It is useful for real-
	   time output (e.g. live streaming). If your input(s) is coming from
	   some other live streaming source (through HTTP or UDP for example)
	   the server might already be in real-time, thus the option will
	   likely not be required. On the other hand, this is meaningful if
	   your input(s) is a file you are trying to push in real-time.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       -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.

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

       -dts_delta_threshold
	   Timestamp discontinuity delta threshold.

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

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

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

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

		   ffmpeg -i infile -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 (per-stream)
	   Force a tag/fourcc for matching streams.

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

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

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

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

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

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

	   For example, to overlay an image over video

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

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

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

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

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

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

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

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

       -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.

       -override_ffserver (global)
	   Overrides the input specifications from ffserver. Using this option
	   you can map any input stream to ffserver and control many aspects
	   of the encoding from ffmpeg. Without this option ffmpeg will
	   transmit to ffserver what is requested by ffserver.	The option is
	   intended for cases where features are needed that cannot be
	   specified to ffserver but can be to ffmpeg.

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

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

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

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

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

       Preset 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.

TIPS
       ·   For streaming at very low bitrate application, use a low frame rate
	   and a small GOP size. This is especially true for RealVideo where
	   the Linux player does not seem to be very fast, so it can miss
	   frames. An example is:

		   ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm

       ·   The parameter 'q' which is displayed while encoding is the current
	   quantizer. The value 1 indicates that a very good quality could be
	   achieved. The value 31 indicates the worst quality. If q=31 appears
	   too often, it means that the encoder cannot compress enough to meet
	   your bitrate. You must either increase the bitrate, decrease the
	   frame rate or decrease the frame size.

       ·   If your computer is not fast enough, you can speed up the
	   compression at the expense of the compression ratio. You can use
	   '-me zero' to speed up motion estimation, and '-g 0' to disable
	   motion estimation completely (you have only I-frames, which means
	   it is about as good as JPEG compression).

       ·   To have very low audio bitrates, reduce the sampling frequency
	   (down to 22050 Hz for MPEG audio, 22050 or 11025 for AC-3).

       ·   To have a constant quality (but a variable bitrate), use the option
	   '-qscale n' when 'n' is between 1 (excellent quality) and 31 (worst
	   quality).

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

       Preset files are specified with the "pre" option, this option takes a
       preset name as input.  FFmpeg searches for a file named
       preset_name.avpreset in the directories $AVCONV_DATADIR (if set), and
       $HOME/.ffmpeg, and in the data directory defined at configuration time
       (usually $PREFIX/share/ffmpeg) in that order.  For example, if the
       argument is "libx264-max", it will search for the file
       libx264-max.avpreset.

   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 -s cif -r 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 -s cif -r 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 -formats".

       ·   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 -vframes 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 -i foo-%03d.jpeg -r 12 -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 -i 'foo-*.jpeg' -r 12 -s WxH foo.avi

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

		   ffmpeg -i test1.avi -i test2.avi -map 0:3 -map 0:2 -map 0:1 -map 0:0 -c copy test12.nut

	   The resulting output file test12.avi will contain first four
	   streams from the input file 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
       The accepted syntax is:

	       [-][HH:]MM:SS[.m...]
	       [-]S+[.m...]

       HH expresses the number of hours, MM the number a of minutes and SS the
       number of 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

   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 (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 (0x00/0.0 means completely transparent,
       0xff/1.0 completely opaque).  If the alpha component is not specified
       then 0xff is assumed.

       The string "random" will result in a random color.

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.

       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".

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

       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 maximum 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.

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

       tan(x)
	   Compute tangent of x.

       tanh(x)
	   Compute hyperbolic tangent of x.

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

	   When the series does not converge the result is undefined.

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

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

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

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

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

       The following constants are available:

       PI  area of the unit disc, approximately 3.14

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

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

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

       "*" works like AND

       "+" works like OR

       For example the construct:

	       if (A AND B) then C

       is equivalent to:

	       if(A*B, C)

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

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

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

       y   10^-24 / 2^-80

       z   10^-21 / 2^-70

       a   10^-18 / 2^-60

       f   10^-15 / 2^-50

       p   10^-12 / 2^-40

       n   10^-9 / 2^-30

       u   10^-6 / 2^-20

       m   10^-3 / 2^-10

       c   10^-2

       d   10^-1

       h   10^2

       k   10^3 / 2^10

       K   10^3 / 2^10

       M   10^6 / 2^20

       G   10^9 / 2^30

       T   10^12 / 2^40

       P   10^15 / 2^40

       E   10^18 / 2^50

       Z   10^21 / 2^60

       Y   10^24 / 2^70

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

       The list of supported options follows:

       build_options
	   Set build options used to compile the registered kernels.

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

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

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

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

	   The specifed index must be one of the indexes in the device list
	   which can be obtained with "av_opencl_get_device_list()".

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

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

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

       The list of supported options follow:

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

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

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

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

	   Possible values:

	   mv4 Use four motion vector by macroblock (mpeg4).

	   qpel
	       Use 1/4 pel motion compensation.

	   loop
	       Use loop filter.

	   qscale
	       Use fixed qscale.

	   gmc Use gmc.

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

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

	   pass2
	       Use internal 2pass ratecontrol in second pass mode.

	   gray
	       Only decode/encode grayscale.

	   emu_edge
	       Do not draw edges.

	   psnr
	       Set error[?] variables during encoding.

	   truncated
	   naq Normalize adaptive quantization.

	   ildct
	       Use interlaced DCT.

	   low_delay
	       Force low delay.

	   global_header
	       Place global headers in extradata instead of every keyframe.

	   bitexact
	       Use only bitexact stuff (except (I)DCT).

	   aic Apply H263 advanced intra coding / mpeg4 ac prediction.

	   cbp Deprecated, use mpegvideo private options instead.

	   qprd
	       Deprecated, use mpegvideo private options instead.

	   ilme
	       Apply interlaced motion estimation.

	   cgop
	       Use closed gop.

       sub_id integer
	   Deprecated, currently unused.

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

	   Possible values:

	   zero
	       zero motion estimation (fastest)

	   full
	       full motion estimation (slowest)

	   epzs
	       EPZS motion estimation (default)

	   esa esa motion estimation (alias for full)

	   tesa
	       tesa motion estimation

	   dia dia motion estimation (alias for epzs)

	   log log motion estimation

	   phods
	       phods motion estimation

	   x1  X1 motion estimation

	   hex hex motion estimation

	   umh umh motion estimation

	   iter
	       iter motion estimation

       extradata_size integer
	   Set extradata size.

       time_base rational number
	   Set codec time base.

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

       g integer (encoding,video)
	   Set the group of picture 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.

       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.

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

       rc_strategy integer (encoding,video)
	   Set ratecontrol method.

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

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

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

	   Possible values:

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

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

	   ump4
	       (autodetected if fourcc==UMP4)

	   no_padding
	       padding bug (autodetected)

	   amv
	   ac_vlc
	       illegal vlc bug (autodetected per fourcc)

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

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

	   edge
	       edge padding bug (autodetected per fourcc/version)

	   hpel_chroma
	   dc_clip
	   ms  Workaround various bugs in microsoft broken decoders.

	   trunc
	       trancated frames

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

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

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

	   Possible values:

	   very
	       strictly conform to a 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

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

	   compliant
	       consider all spec non compliancies as errors

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

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

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

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

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

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

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

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

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

       rc_buf_aggressivity float (encoding,video)
	   Currently useless.

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

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

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

       dct integer (encoding,video)
	   Set DCT algorithm.

	   Possible values:

	   auto
	       autoselect a good one (default)

	   fastint
	       fast integer

	   int accurate integer

	   mmx
	   altivec
	   faan
	       floating point AAN DCT

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

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

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

       p_mask float (encoding,video)
	   Set inter masking.

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

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

	   Possible values:

	   auto
	   int
	   simple
	   simplemmx
	   libmpeg2mmx
	   mmi
	   arm
	   altivec
	   sh4
	   simplearm
	   simplearmv5te
	   simplearmv6
	   simpleneon
	   simplealpha
	   h264
	   vp3
	   ipp
	   xvidmmx
	   faani
	       floating point AAN IDCT

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

	   Possible values:

	   guess_mvs
	       iterative motion vector (MV) search (slow)

	   deblock
	       use strong deblock filter for damaged MBs

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

	   Possible values:

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

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

	   Possible values:

	   pict
	       picture info

	   rc  rate control

	   bitstream
	   mb_type
	       macroblock (MB) type

	   qp  per-block quantization parameter (QP)

	   mv  motion vector

	   dct_coeff
	   skip
	   startcode
	   pts
	   er  error recognition

	   mmco
	       memory management control operations (H.264)

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

	   vis_mb_type
	       visualize block types

	   buffers
	       picture buffer allocations

	   thread_ops
	       threading operations

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

	   Possible values:

	   pf  forward predicted MVs of P-frames

	   bf  forward predicted MVs of B-frames

	   bb  backward predicted MVs of B-frames

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

	   vlc variable length coder / huffman coder

	   ac  arithmetic coder

	   raw raw (no encoding)

	   rle run-length coder

	   deflate
	       deflate-based coder

       context integer (encoding,video)
	   Set context model.

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

	   Possible values:

	   simple
	       use mbcmp (default)

	   bits
	       use fewest bits

	   rd  use best rate distortion

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

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

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

       nr integer (encoding,video)
	   Set noise reduction.

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

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

	   fast
	       allow non spec compliant speedup tricks

	   sgop
	       Deprecated, use mpegvideo private options instead

	   noout
	       skip bitstream encoding

	   local_header
	       place global headers at every keyframe instead of in extradata

	   chunks
	       Frame data might be split into multiple chunks

	   showall
	       Show all frames before the first keyframe

	   skiprd
	       Deprecated, use mpegvideo private options instead

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

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

	   auto
	       detect a good number of threads

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

       mb_threshold integer (encoding,video)
	   Set macroblock threshold.

       dc integer (encoding,video)
	   Set intra_dc_precision.

       nssew integer (encoding,video)
	   Set nsse weight.

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

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

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

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

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

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

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

       skip_exp integer (encoding,video)
	   Set frame skip exponent.

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

	   Possible values:

	   sad sum of absolute differences, fast (default)

	   sse sum of squared errors

	   satd
	       sum of absolute Hadamard transformed differences

	   dct sum of absolute DCT transformed differences

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

	   bit number of bits needed for the block

	   rd  rate distortion optimal, slow

	   zero
	       0

	   vsad
	       sum of absolute vertical differences

	   vsse
	       sum of squared vertical differences

	   nsse
	       noise preserving sum of squared differences

	   w53 5/3 wavelet, only used in snow

	   w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

	   Possible values:

	   none
	       Discard no frame.

	   default
	       Discard useless frames like 0-sized frames.

	   noref
	       Discard all non-reference frames.

	   bidir
	       Discard all bidirectional frames.

	   nokey
	       Discard all frames excepts keyframes.

	   all Discard all frames.

	   Default value is default.

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

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

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

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

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

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

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

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

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

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

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

       request_channel_layout integer (decoding,audio)
	   Possible values:

       rc_max_vbv_use float (encoding,video)
       rc_min_vbv_use float (encoding,video)
       ticks_per_frame integer (decoding/encoding,audio,video)
       color_primaries integer (decoding/encoding,video)
       color_trc integer (decoding/encoding,video)
       colorspace integer (decoding/encoding,video)
       color_range integer (decoding/encoding,video)
       chroma_sample_location integer (decoding/encoding,video)
       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 multithreading type.

	   Possible values:

	   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.

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

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

   rawvideo
       Raw video decoder.

       This decoder decodes rawvideo streams.

       Options

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

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

	   0   bottom-field-first is assumed

	   1   top-field-first is assumed

AUDIO DECODERS
   ffwavesynth
       Internal wave synthetizer.

       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".

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

       Options

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

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

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 "-codecs" 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 an experimental FFmpeg-native AAC encoder. Currently
       only the low complexity (AAC-LC) profile is supported. To use this
       encoder, you must set strict option to experimental or lower.

       As this encoder is experimental, unexpected behavior may exist from
       time to time. For a more stable AAC encoder, see libvo-aacenc. However,
       be warned that it has a worse quality reported by some users.

       Options

       b   Set bit rate in bits/s. Setting this automatically activates
	   constant bit rate (CBR) mode.

       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.

       stereo_mode
	   Set stereo encoding mode. Possible values:

	   auto
	       Automatically selected by the encoder.

	   ms_off
	       Disable middle/side encoding. This is the default.

	   ms_force
	       Force middle/side encoding.

       aac_coder
	   Set AAC encoder coding method. Possible values:

	   0   FAAC-inspired method.

	       This method is a simplified reimplementation of the method used
	       in FAAC, which sets thresholds proportional to the band
	       energies, and then decreases all the thresholds with quantizer
	       steps to find the appropriate quantization with distortion
	       below threshold band by band.

	       The quality of this method is comparable to the two loop
	       searching method descibed below, but somewhat a little better
	       and slower.

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

	       This has a theoretic best quality out of all the coding
	       methods, but at the cost of the slowest speed.

	   2   Two loop searching (TLS) method.

	       This method first sets quantizers depending on band thresholds
	       and then tries to find an optimal combination by adding or
	       subtracting a specific value from all quantizers and adjusting
	       some individual quantizer a little.

	       This method produces similar quality with the FAAC method and
	       is the default.

	   3   Constant quantizer method.

	       This method sets a constant quantizer for all bands. This is
	       the fastest of all the methods, yet produces the worst quality.

       Tips and Tricks

       According to some reports (e.g.
       <http://d.hatena.ne.jp/kamedo2/20120729/1343545890>), setting the
       cutoff option to 15000 Hz greatly improves the quality of the output
       quality. As a result, we encourage you to do the same.

   ac3 and ac3_fixed
       AC-3 audio encoders.

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

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

       AC-3 Metadata

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

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

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

       Metadata Control Options

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

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

	   1   Metadata values can be changed before encoding each frame.

       Downmix Levels

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

	   0.707
	       Apply -3dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6dB gain

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

	   0.707
	       Apply -3dB gain

	   0.500
	       Apply -6dB gain (default)

	   0.000
	       Silence Surround Channel(s)

       Audio Production Information

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

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

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   large
	       Large Room

	   2
	   small
	       Small Room

       Other Metadata Options

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

	   0
	   off No Copyright Exists (default)

	   1
	   on  Copyright Exists

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

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   off Not Dolby Surround Encoded

	   2
	   on  Dolby Surround Encoded

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

	   0
	   off Not Original Source

	   1
	   on  Original Source (default)

       Extended Bitstream Information

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

       Extended Bitstream Information - Part 1

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   ltrt
	       Lt/Rt Downmix Preferred

	   2
	   loro
	       Lo/Ro Downmix Preferred

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

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

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

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround Channel(s)

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

	   1.414
	       Apply +3dB gain

	   1.189
	       Apply +1.5dB gain

	   1.000
	       Apply 0dB gain

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain (default)

	   0.500
	       Apply -6.0dB gain

	   0.000
	       Silence Center Channel

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

	   0.841
	       Apply -1.5dB gain

	   0.707
	       Apply -3.0dB gain

	   0.595
	       Apply -4.5dB gain

	   0.500
	       Apply -6.0dB gain (default)

	   0.000
	       Silence Surround Channel(s)

       Extended Bitstream Information - Part 2

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Surround EX Off

	   2
	   off Dolby Surround EX On

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

	   0
	   notindicated
	       Not Indicated (default)

	   1
	   on  Dolby Headphone Off

	   2
	   off Dolby Headphone On

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

	   0
	   standard
	       Standard A/D Converter (default)

	   1
	   hdcd
	       HDCD A/D Converter

       Other AC-3 Encoding Options

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

       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)

   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".

       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. 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.

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

   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).

   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-aacenc
       VisualOn AAC encoder.

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

       This encoder is considered to be worse than the native experimental
       FFmpeg AAC encoder, according to multiple sources.

       Options

       The VisualOn AAC encoder only support encoding AAC-LC and up to 2
       channels. It is also CBR-only.

       b   Set bit rate in bits/s.

   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).

   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 modeled 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 their 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 of FFmpeg is 10ms, but is 20ms in
	   opusenc.

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

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

	   voip
	       Favor improved speech intelligibility.

	   audio
	       Favor faithfulness to the input (the default).

	   lowdelay
	       Restrict to only the lowest delay modes.

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

   libwavpack
       A wrapper providing WavPack encoding through libwavpack.

       Only lossless mode using 32-bit integer samples is supported currently.
       The compression_level option can be used to control speed vs.
       compression tradeoff, with the values mapped to libwavpack as follows:

       0   Fast mode - corresponding to the wavpack -f option.

       1   Normal (default) settings.

       2   High quality - corresponding to the wavpack -h option.

       3   Very high quality - corresponding to the wavpack -hh option.

       4-8 Same as 3, but with extra processing enabled - corresponding to the
	   wavpack -x option. I.e. 4 is the same as -x2 and 8 is the same as
	   -x6.

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

   libtheora
       Theora format supported through libtheora.

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

       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, only works if the "qscale" flag in flags is
	   not enabled.

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

       g   Set the GOP size.

       global_quality
	   Set the global quality in lambda units, only works if the "qscale"
	   flag in flags is enabled. The value is clipped in the [0 -
	   10*"FF_QP2LAMBDA"] range, and then multiplied for 6.3 to get a
	   value in the native libtheora range [0-63]. A higher value
	   corresponds to a higher quality.

	   For example, to set maximum constant quality encoding with ffmpeg:

		   ffmpeg -i INPUT -flags:v qscale -global_quality:v "10*QP2LAMBDA" -codec:v libtheora OUTPUT.ogg

   libvpx
       VP8 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

       Mapping from FFmpeg to libvpx options with conversion notes in
       parentheses.

       threads
	   g_threads

       profile
	   g_profile

       vb  rc_target_bitrate

       g   kf_max_dist

       keyint_min
	   kf_min_dist

       qmin
	   rc_min_quantizer

       qmax
	   rc_max_quantizer

       bufsize, vb
	   rc_buf_sz "(bufsize * 1000 / vb)"

	   rc_buf_optimal_sz "(bufsize * 1000 / vb * 5 / 6)"

       rc_init_occupancy, vb
	   rc_buf_initial_sz "(rc_init_occupancy * 1000 / vb)"

       rc_buffer_aggressivity
	   rc_undershoot_pct

       skip_threshold
	   rc_dropframe_thresh

       qcomp
	   rc_2pass_vbr_bias_pct

       maxrate, vb
	   rc_2pass_vbr_maxsection_pct "(maxrate * 100 / vb)"

       minrate, vb
	   rc_2pass_vbr_minsection_pct "(minrate * 100 / vb)"

       minrate, maxrate, vb
	   "VPX_CBR" "(minrate == maxrate == vb)"

       crf "VPX_CQ", "VP8E_SET_CQ_LEVEL"

       quality
	   best
	       "VPX_DL_BEST_QUALITY"

	   good
	       "VPX_DL_GOOD_QUALITY"

	   realtime
	       "VPX_DL_REALTIME"

       speed
	   "VP8E_SET_CPUUSED"

       nr  "VP8E_SET_NOISE_SENSITIVITY"

       mb_threshold
	   "VP8E_SET_STATIC_THRESHOLD"

       slices
	   "VP8E_SET_TOKEN_PARTITIONS"

       max-intra-rate
	   "VP8E_SET_MAX_INTRA_BITRATE_PCT"

       force_key_frames
	   "VPX_EFLAG_FORCE_KF"

       Alternate reference frame related
	   vp8flags altref
	       "VP8E_SET_ENABLEAUTOALTREF"

	   arnr_max_frames
	       "VP8E_SET_ARNR_MAXFRAMES"

	   arnr_type
	       "VP8E_SET_ARNR_TYPE"

	   arnr_strength
	       "VP8E_SET_ARNR_STRENGTH"

	   rc_lookahead
	       g_lag_in_frames

       vp8flags error_resilient
	   g_error_resilient

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

   libx264
       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 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>.

       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 --full-help 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)
       qmax (qpmax)
       qmin (qpmin)
       qdiff (qpstep)
       qblur (qblur)
       qcomp (qcomp)
       refs (ref)
       sc_threshold (scenecut)
       trellis (trellis)
       nr  (nr)
       me_range (merange)
       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).

       subq (subme)
       b_strategy (b-adapt)
       keyint_min (min-keyint)
       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 comparation 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)
       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.

       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 -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

       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 compability 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).

   libxvid
       Xvid MPEG-4 Part 2 encoder wrapper.

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

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

       Options

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

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

	   mv4 Use four motion vector by macroblock.

	   aic Enable high quality AC prediction.

	   gray
	       Only encode grayscale.

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

	   qpel
	       Enable quarter-pixel motion compensation.

	   cgop
	       Enable closed GOP.

	   global_header
	       Place global headers in extradata instead of every keyframe.

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

	   zero
	       Use no motion estimation (default).

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

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

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

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

	   simple
	       Use macroblock comparing function algorithm (default).

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

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

   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 choosen with the "-vcodec" option.

       Private Options for prores-ks

       profile integer
	   Select the ProRes profile to encode

	   proxy
	   lt
	   standard
	   hq
	   4444
       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 produc 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.

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.

       Below is a description of the currently available bitstream filters.

   aac_adtstoasc
       Convert MPEG-2/4 AAC ADTS to MPEG-4 Audio Specific Configuration
       bitstream filter.

       This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS
       header and removes the ADTS header.

       This is required for example when copying an AAC stream from a raw ADTS
       AAC container to a FLV or a MOV/MP4 file.

   chomp
       Remove zero padding at the end of a packet.

   dump_extradata
   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 ("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

   imx_dump_header
   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

   mjpega_dump_header
   movsub
   mp3_header_compress
   mp3_header_decompress
   noise
   remove_extradata
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 allow to detect more
	   information in case it is dispersed into the stream, but will
	   increase latency. Must be an integer not lesser than 32. It is
	   5000000 by default.

       packetsize integer (output)
	   Set packet size.

       fflags flags (input/output)
	   Set format flags.

	   Possible values:

	   ignidx
	       Ignore index.

	   genpts
	       Generate PTS.

	   nofillin
	       Do not fill in missing values that can be exactly calculated.

	   noparse
	       Disable AVParsers, this needs "+nofillin" too.

	   igndts
	       Ignore DTS.

	   discardcorrupt
	       Discard corrupted frames.

	   sortdts
	       Try to interleave output packets by DTS.

	   keepside
	       Do not merge side data.

	   latm
	       Enable RTP MP4A-LATM payload.

	   nobuffer
	       Reduce the latency introduced by optional buffering

       seek2any integer (input)
	   Forces seeking to enable seek to any mode if set to 1. Default is
	   0.

       analyzeduration integer (input)
	   Specify how many microseconds are analyzed to probe the input. A
	   higher value will allow to detect 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.

       use_wallclock_as_timestamps integer (input)
	   Use wallclock as timestamps.

       avoid_negative_ts integer (output)
	   Shift timestamps to make them non-negative. A value of 1 enables
	   shifting, a value of 0 disables it, the default value of -1 enables
	   shifting when required by the target format.

	   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.

	   Also note that this affects only leading negative timestamps, and
	   not non-monotonic negative timestamps.

       skip_initial_bytes integer (input)
	   Set number initial bytes to skip. 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 1
	   enables it, and has the effect of reducing the latency; 0 disables
	   it and may slightly increase performance in some cases.

DEMUXERS
       Demuxers are configured elements in FFmpeg which allow to 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 "-formats" of the ff* tools will display the list of enabled
       demuxers.

       The description of some of the currently available demuxers follows.

   applehttp
       Apple HTTP Live Streaming demuxer.

       This demuxer presents all AVStreams from all variant streams.  The id
       field is set to the bitrate variant index number. By setting the
       discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the
       caller can decide which variant streams to actually receive.  The total
       bitrate of the variant that the stream belongs to is available in a
       metadata key named "variant_bitrate".

   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 packet 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 directives apply to that file.

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

	   To make FFmpeg recognize the format automatically, this directive
	   must appears 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.

       Options

       This demuxer accepts the following option:

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

	   If set to 0, any file name is accepted.

	   The default is -1, it is equivalent to 1 if the format was
	   automatically probed and 0 otherwise.

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

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

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

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

   libquvi
       Play media from Internet services using the quvi project.

       The demuxer accepts a format option to request a specific quality. It
       is by default set to best.

       See <http://quvi.sourceforge.net/> for more information.

       FFmpeg needs to be built with "--enable-libquvi" for this demuxer to be
       enabled.

   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.

	   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.

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

       Examples

       ·   Use ffmpeg for creating a video from the images in the file
	   sequence img-001.jpeg, img-002.jpeg, ..., assuming an input frame
	   rate of 10 frames per second:

		   ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv

       ·   As above, but start by reading from a file with index 100 in the
	   sequence:

		   ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv

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

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

   rawvideo
       Raw video demuxer.

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

       This demuxer accepts the following options:

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

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

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

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

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

   sbg
       SBaGen script demuxer.

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

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

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

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

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

       This demuxer accepts the following option:

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

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

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

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

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

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

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

       A description of some of the currently available muxers follows.

   aiff
       Audio Interchange File Format muxer.

       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.

   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.

       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 -

       See also the framecrc muxer.

   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.

       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.

   framemd5
       Per-packet MD5 testing format.

       This muxer computes and prints the MD5 hash 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 hash.

       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>, <MD5>

       MD5 is a hexadecimal number representing the computed MD5 hash for the
       packet.

       For example to compute the MD5 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 md5 muxer.

   hls
       Apple HTTP Live Streaming muxer that segments MPEG-TS according to the
       HTTP Live Streaming specification.

       It creates a playlist file and numbered segment files. The output
       filename specifies the playlist filename; the segment filenames receive
       the same basename as the playlist, a sequential number and a .ts
       extension.

	       ffmpeg -i in.nut out.m3u8

       -hls_time seconds
	   Set the segment length in seconds.

       -hls_list_size size
	   Set the maximum number of playlist entries.

       -hls_wrap wrap
	   Set the number after which index wraps.

       -start_number number
	   Start the sequence from number.

   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 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 1 -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 1 -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 input video
       you can employ the command:

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

       start_number number
	   Start the sequence from number. Default value is 1. Must be a
	   positive number.

       -update number
	   If number is nonzero, the filename will always be interpreted as
	   just a filename, not a pattern, and this file will be continuously
	   overwritten with new images.

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

   matroska
       Matroska container muxer.

       This muxer implements the matroska and webm container specs.

       The recognized metadata settings in this muxer are:

       title=title name
	   Name provided to a single track

       language=language name
	   Specifies the language of the track in the Matroska languages form

       stereo_mode=mode
	   Stereo 3D video layout of two views in a single video track

	   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

       This muxer supports the following options:

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

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

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

   md5
       MD5 testing format.

       This muxer computes and prints the MD5 hash 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 hash.

       The output of the muxer consists of a single line of the form: MD5=MD5,
       where MD5 is a hexadecimal number representing the computed MD5 hash.

       For example 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 framemd5 muxer.

   MOV/MP4/ISMV
       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 AVOptions that define
       how to cut the file into fragments:

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

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

       -frag_duration duration
	   Create fragments that are duration microseconds long.

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

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

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

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

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

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

	   Files written with this option set do not work in QuickTime.	 This
	   option is implicitly set when writing ismv (Smooth Streaming)
	   files.

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

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

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

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

       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 an ID3v2 header at the
       beginning and optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4
       are supported, the "id3v2_version" option controls which one is used.
       The legacy ID3v1 tag is not written by default, but may be enabled with
       the "write_id3v1" option.

       For seekable output the muxer also writes a Xing frame at the
       beginning, which contains the number of frames in the file. It is
       useful for computing duration of VBR files.

       The muxer supports writing ID3v2 attached pictures (APIC frames). 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.

       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

   mpegts
       MPEG transport stream muxer.

       This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

       The muxer options are:

       -mpegts_original_network_id number
	   Set the original_network_id (default 0x0001). 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.

       -mpegts_transport_stream_id number
	   Set the transport_stream_id (default 0x0001). This identifies a
	   transponder in DVB.

       -mpegts_service_id number
	   Set the service_id (default 0x0001) also known as program in DVB.

       -mpegts_pmt_start_pid number
	   Set the first PID for PMT (default 0x1000, max 0x1f00).

       -mpegts_start_pid number
	   Set the first PID for data packets (default 0x0100, max 0x0f00).

       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".

	       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" \
		    -y out.ts

   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 -

   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.

   segment, stream_segment, ssegment
       Basic stream segmenter.

       The segmenter 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.

       "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 segment muxer supports the following options:

       reference_stream specifier
	   Set the reference stream, as specified by the string specifier.  If
	   specifier is set to "auto", the reference is choosen 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_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.

	   Default value is "samp".

       segment_list_size size
	   Update the list file so that it contains at most the last size
	   segments. If 0 the list file will contain all the segments. Default
	   value is 0.

       segment_list_type type
	   Specify the format for the segment list file.

	   The following values are recognized:

	   flat
	       Generate a flat list for the created segments, one segment per
	       line.

	   csv, ext
	       Generate a list for the created segments, one segment per line,
	       each line matching the format (comma-separated values):

		       <segment_filename>,<segment_start_time>,<segment_end_time>

	       segment_filename is the name of the output file generated by
	       the muxer according to the provided pattern. CSV escaping
	       (according to RFC4180) is applied if required.

	       segment_start_time and segment_end_time specify the segment
	       start and end time expressed in seconds.

	       A list file with the suffix ".csv" or ".ext" will auto-select
	       this format.

	       ext is deprecated in favor or csv.

	   ffconcat
	       Generate an ffconcat file for the created segments. The
	       resulting file can be read using the FFmpeg concat demuxer.

	       A list file with the suffix ".ffcat" or ".ffconcat" will auto-
	       select this format.

	   m3u8
	       Generate an extended M3U8 file, version 3, compliant with
	       <http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

	       A list file with the suffix ".m3u8" will auto-select this
	       format.

	   If not specified the type is guessed from the list file name
	   suffix.

       segment_time time
	   Set segment duration to time, the value must be a duration
	   specification. Default value is "2". See also the segment_times
	   option.

	   Note that splitting may not be accurate, unless you force the
	   reference stream key-frames at the given time. See the introductory
	   notice and the examples below.

       segment_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.

       reset_timestamps 1|0
	   Reset timestamps at the begin 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.

       Examples

       ·   To remux the content of file in.mkv to a list of segments
	   out-000.nut, out-001.nut, etc., and write the list of generated
	   segments to out.list:

		   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut

       ·   As the example above, but 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

       ·   As the example above, but 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

       ·   To convert the in.mkv to TS segments using the "libx264" and
	   "libfaac" encoders:

		   ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -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

   tee
       The tee muxer can be used to write the same data to several files or
       any other kind of muxer. It can be used, for example, to both stream a
       video to the network and save it to disk at the same time.

       It is different from specifying several outputs to the ffmpeg command-
       line tool because the audio and video data will be encoded only once
       with the tee muxer; encoding can be a very expensive process. It is not
       useful when using the libavformat API directly because it is then
       possible to feed the same packets to several muxers directly.

       The slave outputs are specified in the file name given to the muxer,
       separated by '|'. If any of the slave name contains the '|' separator,
       leading or trailing spaces or any special character, it must be escaped
       (see the ``Quoting and escaping'' section in the ffmpeg-utils manual).

       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.

       Example: encode something and both archive it in a WebM file and stream
       it as MPEG-TS over UDP (the streams need to be explicitly mapped):

	       ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
		 "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       Note: some codecs may need different options depending on the output
       format; the auto-detection of this can not work with the tee muxer. The
       main example is the global_header flag.

METADATA
       FFmpeg is able to dump metadata from media files into a simple
       UTF-8-encoded INI-like text file and then load it back using the
       metadata muxer/demuxer.

       The file format is as follows:

       1.  A file consists of a header and a number of metadata tags divided
	   into sections, each on its own line.

       2.  The header is a ';FFMETADATA' string, followed by a version number
	   (now 1).

       3.  Metadata tags are of the form 'key=value'

       4.  Immediately after header follows global metadata

       5.  After global metadata there may be sections with
	   per-stream/per-chapter metadata.

       6.  A section starts with the section name in uppercase (i.e. STREAM or
	   CHAPTER) in brackets ('[', ']') and ends with next section or end
	   of file.

       7.  At the beginning of a chapter section there may be an optional
	   timebase to be used for start/end values. It must be in form
	   'TIMEBASE=num/den', where num and den are integers. If the timebase
	   is missing then start/end times are assumed to be in milliseconds.
	   Next a chapter section must contain chapter start and end times in
	   form 'START=num', 'END=num', where num is a positive integer.

       8.  Empty lines and lines starting with ';' or '#' are ignored.

       9.  Metadata keys or values containing special characters ('=', ';',
	   '#', '\' and a newline) must be escaped with a backslash '\'.

       10. Note that whitespace in metadata (e.g. foo = bar) is considered to
	   be a part of the tag (in the example above key is 'foo ', value is
	   ' bar').

       A ffmetadata file might look like this:

	       ;FFMETADATA1
	       title=bike\\shed
	       ;this is a comment
	       artist=FFmpeg troll team

	       [CHAPTER]
	       TIMEBASE=1/1000
	       START=0
	       #chapter ends at 0:01:00
	       END=60000
	       title=chapter \#1
	       [STREAM]
	       title=multi\
	       line

       By using the ffmetadata muxer and demuxer it is possible to extract
       metadata from an input file to an ffmetadata file, and then transcode
       the file into an output file with the edited ffmetadata file.

       Extracting an ffmetadata file with ffmpeg goes as follows:

	       ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

       Reinserting edited metadata information from the FFMETADATAFILE file
       can be done as:

	       ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

PROTOCOLS
       Protocols are configured elements in FFmpeg which allow to access
       resources which require the use of a particular protocol.

       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.

       A description of the currently available protocols follows.

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
	   BluRay angle

       chapter
	   Start chapter (1...N)

       playlist
	   Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

       Read longest playlist from BluRay mounted to /mnt/bluray:

	       bluray:/mnt/bluray

       Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start
       from chapter 2:

	       -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

   cache
       Caching wrapper for input stream.

       Cache the input stream to temporary file. It brings seeking capability
       to live streams.

	       cache:<URL>

   concat
       Physical concatenation protocol.

       Allow to read and seek from many resource in sequence as if they were a
       unique resource.

       A URL accepted by this protocol has the syntax:

	       concat:<URL1>|<URL2>|...|<URLN>

       where URL1, URL2, ..., URLN are the urls of the resource to be
       concatenated, each one possibly specifying a distinct protocol.

       For example to read a sequence of files split1.mpeg, split2.mpeg,
       split3.mpeg with ffplay use the command:

	       ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

       Note that you may need to escape the character "|" which is special for
       many shells.

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set the AES decryption key binary block from given hexadecimal
	   representation.

       iv  Set the AES decryption initialization vector binary block from
	   given hexadecimal representation.

       Accepted URL formats:

	       crypto:<URL>
	       crypto+<URL>

   data
       Data in-line in the URI. See
       <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given inline with ffmpeg:

	       ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   file
       File access protocol.

       Allow to read from or read to a file.

       For example to read from a file input.mpeg with ffmpeg use the command:

	       ffmpeg -i file:input.mpeg output.mpeg

       The ff* tools default to the file protocol, that is a resource
       specified with the name "FILE.mpeg" is interpreted as the URL
       "file:FILE.mpeg".

   ftp
       FTP (File Transfer Protocol).

       Allow to 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 of socket I/O operations used by the underlying low
	   level operation. By default it is set to -1, which means that the
	   timeout is not specified.

       ftp-anonymous-password
	   Password used when login as anonymous user. Typically an e-mail
	   address should be used.

       ftp-write-seekable
	   Control seekability of connection during encoding. If set to 1 the
	   resource is supposed to be seekable, if set to 0 it is assumed not
	   to be seekable. Default value is 0.

       NOTE: Protocol can be used as output, but it is recommended to not do
       it, unless special care is taken (tests, customized server
       configuration etc.). Different FTP servers behave in different way
       during seek operation. ff* tools may produce incomplete content due to
       server limitations.

   gopher
       Gopher protocol.

   hls
       Read Apple HTTP Live Streaming compliant segmented stream as a uniform
       one. The M3U8 playlists describing the segments can be remote HTTP
       resources or local files, accessed using the standard file protocol.
       The nested protocol is declared by specifying "+proto" after the hls
       URI scheme name, where proto is either "file" or "http".

	       hls+http://host/path/to/remote/resource.m3u8
	       hls+file://path/to/local/resource.m3u8

       Using this protocol is discouraged - the hls demuxer should work just
       as well (if not, please report the issues) and is more complete.	 To
       use the hls demuxer instead, simply use the direct URLs to the m3u8
       files.

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options.

       seekable
	   Control seekability of connection. If set to 1 the resource is
	   supposed to be seekable, if set to 0 it is assumed not to be
	   seekable, if set to -1 it will try to autodetect if it is seekable.
	   Default value is -1.

       chunked_post
	   If set to 1 use chunked transfer-encoding for posts, default is 1.

       headers
	   Set custom HTTP headers, can override built in default headers. The
	   value must be a string encoding the headers.

       content_type
	   Force a content type.

       user-agent
	   Override User-Agent header. If not specified the protocol will use
	   a string describing the libavformat build.

       multiple_requests
	   Use persistent connections if set to 1. By default it is 0.

       post_data
	   Set custom HTTP post data.

       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.

       mime_type
	   Set MIME type.

       icy If set to 1 request ICY (SHOUTcast) metadata from the server. If
	   the server supports this, the metadata has to be retrieved by the
	   application by reading the icy_metadata_headers and
	   icy_metadata_packet options.	 The default is 0.

       icy_metadata_headers
	   If the server supports ICY metadata, this contains the ICY specific
	   HTTP reply headers, separated with 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.

       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.

       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

   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.

       Allow to read and write from UNIX pipes.

       The accepted syntax is:

	       pipe:[<number>]

       number is the number corresponding to the file descriptor of the pipe
       (e.g. 0 for stdin, 1 for stdout, 2 for stderr).	If number is not
       specified, by default the stdout file descriptor will be used for
       writing, stdin for reading.

       For example to read from stdin with ffmpeg:

	       cat test.wav | ffmpeg -i pipe:0
	       # ...this is the same as...
	       cat test.wav | ffmpeg -i pipe:

       For writing to stdout with ffmpeg:

	       ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
	       # ...this is the same as...
	       ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       Note that some formats (typically MOV), require the output protocol to
       be seekable, so they will fail with the pipe output protocol.

   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://<server>[:<port>][/<app>][/<instance>][/<playpath>]

       The accepted parameters are:

       server
	   The address of the RTMP server.

       port
	   The number of the TCP port to use (by default is 1935).

       app It is the name of the application to access. It usually corresponds
	   to the path where the application is installed on the RTMP server
	   (e.g. /ondemand/, /flash/live/, etc.). You can override the value
	   parsed from the URI through the "rtmp_app" option, too.

       playpath
	   It is the path or name of the resource to play with reference to
	   the application specified in app, may be prefixed by "mp4:". You
	   can override the value parsed from the URI through the
	   "rtmp_playpath" option, too.

       listen
	   Act as a server, listening for an incoming connection.

       timeout
	   Maximum time to wait for the incoming connection. Implies listen.

       Additionally, the following parameters can be set via command line
       options (or in code via "AVOption"s):

       rtmp_app
	   Name of application to connect on the RTMP server. This option
	   overrides the parameter specified in the URI.

       rtmp_buffer
	   Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
	   Extra arbitrary AMF connection parameters, parsed from a string,
	   e.g. like "B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each
	   value is prefixed by a single character denoting the type, B for
	   Boolean, N for number, S for string, O for object, or Z for null,
	   followed by a colon. For Booleans the data must be either 0 or 1
	   for FALSE or TRUE, respectively.  Likewise for Objects the data
	   must be 0 or 1 to end or begin an object, respectively. Data items
	   in subobjects may be named, by prefixing the type with 'N' and
	   specifying the name before the value (i.e. "NB:myFlag:1"). This
	   option may be used multiple times to construct arbitrary AMF
	   sequences.

       rtmp_flashver
	   Version of the Flash plugin used to run the SWF player. The default
	   is LNX 9,0,124,2.

       rtmp_flush_interval
	   Number of packets flushed in the same request (RTMPT only). The
	   default is 10.

       rtmp_live
	   Specify that the media is a live stream. No resuming or seeking in
	   live streams is possible. The default value is "any", which means
	   the subscriber first tries to play the live stream specified in the
	   playpath. If a live stream of that name is not found, it plays the
	   recorded stream. The other possible values are "live" and
	   "recorded".

       rtmp_pageurl
	   URL of the web page in which the media was embedded. By default no
	   value will be sent.

       rtmp_playpath
	   Stream identifier to play or to publish. This option overrides the
	   parameter specified in the URI.

       rtmp_subscribe
	   Name of live stream to subscribe to. By default no value will be
	   sent.  It is only sent if the option is specified or if rtmp_live
	   is set to live.

       rtmp_swfhash
	   SHA256 hash of the decompressed SWF file (32 bytes).

       rtmp_swfsize
	   Size of the decompressed SWF file, required for SWFVerification.

       rtmp_swfurl
	   URL of the SWF player for the media. By default no value will be
	   sent.

       rtmp_swfverify
	   URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
	   URL of the target stream. Defaults to proto://host[:port]/app.

       For example to read with ffplay a multimedia resource named "sample"
       from the application "vod" from an RTMP server "myserver":

	       ffplay rtmp://myserver/vod/sample

   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.

   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 Protocol.

   rtsp
       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 <http://github.com/revmischa/rtsp-server>).

       The required syntax for a RTSP url is:

	       rtsp://<hostname>[:<port>]/<path>

       The following options (set on the ffmpeg/ffplay command line, or set in
       code via "AVOption"s or in "avformat_open_input"), are supported:

       Flags for "rtsp_transport":

       udp Use UDP as lower transport protocol.

       tcp Use TCP (interleaving within the RTSP control channel) as lower
	   transport protocol.

       udp_multicast
	   Use UDP multicast as lower transport protocol.

       http
	   Use HTTP tunneling as lower transport protocol, which is useful for
	   passing proxies.

       Multiple lower transport protocols may be specified, in that case they
       are tried one at a time (if the setup of one fails, the next one is
       tried).	For the muxer, only the "tcp" and "udp" options are supported.

       Flags for "rtsp_flags":

       filter_src
	   Accept packets only from negotiated peer address and port.

       listen
	   Act as a server, listening for an incoming connection.

       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".

       Example command lines:

       To 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

       To watch a stream tunneled over HTTP:

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

       To 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

       To receive a stream in realtime:

	       ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

       stimeout
	   Socket IO timeout in micro seconds.

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a
       protocol handler in libavformat, it is a muxer and demuxer.  It is used
       for signalling of RTP streams, by announcing the SDP for the streams
       regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

	       sap://<destination>[:<port>][?<options>]

       The RTP packets are sent to destination on port port, or to port 5004
       if no port is specified.	 options is a "&"-separated list. The
       following options are supported:

       announce_addr=address
	   Specify the destination IP address for sending the announcements
	   to.	If omitted, the announcements are sent to the commonly used
	   SAP announcement multicast address 224.2.127.254 (sap.mcast.net),
	   or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
	   Specify the port to send the announcements on, defaults to 9875 if
	   not specified.

       ttl=ttl
	   Specify the time to live value for the announcements and RTP
	   packets, defaults to 255.

       same_port=0|1
	   If set to 1, send all RTP streams on the same port pair. If zero
	   (the default), all streams are sent on unique ports, with each
	   stream on a port 2 numbers higher than the previous.	 VLC/Live555
	   requires this to be set to 1, to be able to receive the stream.
	   The RTP stack in libavformat for receiving requires all streams to
	   be sent on unique ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

	       ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

	       ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

	       sap://[<address>][:<port>]

       address is the multicast address to listen for announcements on, if
       omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the
       port that is listened on, 9875 if omitted.

       The demuxers listens for announcements on the given address and port.
       Once an announcement is received, it tries to receive that particular
       stream.

       Example command lines follow.

       To play back the first stream announced on the normal SAP multicast
       address:

	       ffplay sap://

       To play back the first stream announced on one the default IPv6 SAP
       multicast address:

	       ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

	       sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

       listen
	   If set to any value, listen for an incoming connection. Outgoing
	   connection is done by default.

       max_streams
	   Set the maximum number of streams. By default no limit is set.

   srtp
       Secure Real-time Transport Protocol.

       The accepted options are:

       srtp_in_suite
       srtp_out_suite
	   Select input and output encoding suites.

	   Supported values:

	   AES_CM_128_HMAC_SHA1_80
	   SRTP_AES128_CM_HMAC_SHA1_80
	   AES_CM_128_HMAC_SHA1_32
	   SRTP_AES128_CM_HMAC_SHA1_32
       srtp_in_params
       srtp_out_params
	   Set input and output encoding parameters, which are expressed by a
	   base64-encoded representation of a binary block. The first 16 bytes
	   of this binary block are used as master key, the following 14 bytes
	   are used as master salt.

   tcp
       Trasmission Control Protocol.

       The required syntax for a TCP url is:

	       tcp://<hostname>:<port>[?<options>]

       listen
	   Listen for an incoming connection

       timeout=microseconds
	   In read mode: if no data arrived in more than this time interval,
	   raise error.	 In write mode: if socket cannot be written in more
	   than this time interval, raise error.  This also sets timeout on
	   TCP connection establishing.

		   ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
		   ffplay tcp://<hostname>:<port>

   tls
       Transport Layer Security/Secure Sockets Layer

       The required syntax for a TLS/SSL url is:

	       tls://<hostname>:<port>[?<options>]

       listen
	   Act as a server, listening for an incoming connection.

       cafile=filename
	   Certificate authority file. The file must be in OpenSSL PEM format.

       cert=filename
	   Certificate file. The file must be in OpenSSL PEM format.

       key=filename
	   Private key file.

       verify=0|1
	   Verify the peer's certificate.

       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 a 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 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 socket buffer size in bytes. This is used both for the
	   receiving and the sending buffer size.

       localport=port
	   Override the local UDP port to bind with.

       localaddr=addr
	   Choose the local IP address. This is useful e.g. if sending
	   multicast and the host has multiple interfaces, where the user can
	   choose which interface to send on by specifying the IP address of
	   that interface.

       pkt_size=size
	   Set the size in bytes of UDP packets.

       reuse=1|0
	   Explicitly allow or disallow reusing UDP sockets.

       ttl=ttl
	   Set the time to live value (for multicast only).

       connect=1|0
	   Initialize the UDP socket with "connect()". In this case, the
	   destination address can't be changed with ff_udp_set_remote_url
	   later.  If the destination address isn't known at the start, this
	   option can be specified in ff_udp_set_remote_url, too.  This allows
	   finding out the source address for the packets with getsockname,
	   and makes writes return with AVERROR(ECONNREFUSED) if "destination
	   unreachable" is received.  For receiving, this gives the benefit of
	   only receiving packets from the specified peer address/port.

       sources=address[,address]
	   Only receive packets sent to the multicast group from one of the
	   specified sender IP addresses.

       block=address[,address]
	   Ignore packets sent to the multicast group from the specified
	   sender IP addresses.

       fifo_size=units
	   Set the UDP receiving circular buffer size, expressed as a number
	   of packets with size of 188 bytes. If not specified defaults to
	   7*4096.

       overrun_nonfatal=1|0
	   Survive in case of UDP receiving circular buffer overrun. Default
	   value is 0.

       timeout=microseconds
	   In read mode: if no data arrived in more than this time interval,
	   raise error.

       Some usage examples of the UDP protocol with ffmpeg follow.

       To stream over UDP to a remote endpoint:

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

       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

       To receive over UDP from a remote endpoint:

	       ffmpeg -i udp://[<multicast-address>]:<port>

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 allow to access
       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 "-formats" of the ff* tools will display the list of
       supported input devices (amongst the demuxers).

       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>

   bktr
       BSD video input device.

   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.

       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 same name (starts at 0,
	   defaults to 0).

       audio_device_number
	   Set audio device number for devices with 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>

       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"

   dv1394
       Linux DV 1394 input device.

   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.

       To record from the framebuffer device /dev/fb0 with ffmpeg:

	       ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi

       You can take a single screenshot image with the command:

	       ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg

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

   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 maxiumum 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 it's GUID. Capturing will
	   only be performed from the specified device and fails if no device
	   with the given GUID is found. This is useful to select the input if
	   multiple devices are connected at the same time.  Look at
	   /sys/bus/firewire/devices to find out the GUIDs.

       Examples

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

		   ffplay -f iec61883 -i auto

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

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

   jack
       JACK input device.

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

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

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

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

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

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

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

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

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

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

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

   lavfi
       Libavfilter input virtual device.

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

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

       Options

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

	   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.

       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]"

   libdc1394
       IIDC1394 input device, based on libdc1394 and libraw1394.

   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 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>

   pulse
       pulseaudio input device.

       To enable this input device during configuration you need libpulse-
       simple installed in your system.

       The filename to provide to the input device is a source device or the
       string "default"

       To list the pulse source devices and their properties you can invoke
       the command pactl list sources.

	       ffmpeg -f pulse -i default /tmp/pulse.wav

       server AVOption

       The syntax is:

	       -server <server name>

       Connects to a specific server.

       name AVOption

       The syntax is:

	       -name <application name>

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

       stream_name AVOption

       The syntax is:

	       -stream_name <stream name>

       Specify the stream name pulse will use when showing active streams, by
       default it is "record"

       sample_rate AVOption

       The syntax is:

	       -sample_rate <samplerate>

       Specify the samplerate in Hz, by default 48kHz is used.

       channels AVOption

       The syntax is:

	       -channels <N>

       Specify the channels in use, by default 2 (stereo) is set.

       frame_size AVOption

       The syntax is:

	       -frame_size <bytes>

       Specify the number of byte per frame, by default it is set to 1024.

       fragment_size AVOption

       The syntax is:

	       -fragment_size <bytes>

       Specify the minimal buffering fragment in pulseaudio, it will affect
       the audio latency. By default it is unset.

   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

   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:

       ·   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 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".

   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.

   x11grab
       X11 video input device.

       This device allows 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 dpyinfo 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 -r 25 -s cif -i :0.0 out.mpg

       Grab at position "10,20":

	       ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg

       Options

       draw_mouse
	   Specify whether to draw the mouse pointer. A value of 0 specify 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 -r 25 -s 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 -r 25 -s 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.

	   For example:

		   ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg

	   With follow_mouse:

		   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg

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

OUTPUT DEVICES
       Output devices are configured elements in FFmpeg which allow to 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 "-formats" of the ff* tools will display the list of enabled
       output devices (amongst the muxers).

       A description of the currently available output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) output device.

   caca
       CACA output device.

       This output device allows 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 -vcodec 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 -

   oss
       OSS (Open Sound System) output device.

   sdl
       SDL (Simple DirectMedia Layer) output device.

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

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

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

       Options

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

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

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

       Examples

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

	       ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

   sndio
       sndio audio output device.

   xv
       XV (XVideo) output device.

       This output device allows 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_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.

       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.

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

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

       Examples

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

		   ffmpeg -i INPUT OUTPUT -f xv display

       ·   Decode and display the input video to multiple X11 windows:

		   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS
       The audio resampler supports the following named options.

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

       ich, in_channel_count
	   Set the number of input channels. Default value is 0. Setting this
	   value is not mandatory if the corresponding channel layout
	   in_channel_layout is set.

       och, out_channel_count
	   Set the number of output channels. Default value is 0. Setting this
	   value is not mandatory if the corresponding channel layout
	   out_channel_layout is set.

       uch, used_channel_count
	   Set the number of used input channels. Default value is 0. This
	   option is only used for special remapping.

       isr, in_sample_rate
	   Set the input sample rate. Default value is 0.

       osr, out_sample_rate
	   Set the output sample rate. Default value is 0.

       isf, in_sample_fmt
	   Specify the input sample format. It is set by default to "none".

       osf, out_sample_fmt
	   Specify the output sample format. It is set by default to "none".

       tsf, internal_sample_fmt
	   Set the internal sample format. Default value is "none".  This will
	   automatically be chosen when it is not explicitly set.

       icl, in_channel_layout
	   Set the input channel layout.

       ocl, out_channel_layout
	   Set the output channel layout.

       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.

       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, 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 if set to 1, default value is 0.

       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 an integer in
	   the interval [2,16], default value is 9.

       output_sample_bits
	   For swr only, set number of used output sample bits for dithering.
	   Must be an integer in the interval [0,64], default value is 0,
	   which means it's not used.

SCALER OPTIONS
       The video scaler supports the following named options.

       Options may be set by specifying -option value in the FFmpeg tools. For
       programmatic use, they can be set explicitly in the "SwsContext"
       options or through the libavutil/opt.h API.

       sws_flags
	   Set the scaler flags. This is also used to set the scaling
	   algorithm. Only a single algorithm should be selected.

	   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.

	   bicubiclin
	       Select bicubic scaling algorithm for the luma component,
	       bilinear for chroma components.

	   gauss
	       Select Gaussian rescaling algorithm.

	   sinc
	       Select sinc rescaling algorithm.

	   lanczos
	       Select lanczos rescaling algorithm.

	   spline
	       Select natural bicubic spline rescaling algorithm.

	   print_info
	       Enable printing/debug logging.

	   accurate_rnd
	       Enable accurate rounding.

	   full_chroma_int
	       Enable full chroma interpolation.

	   full_chroma_inp
	       Select full chroma input.

	   bitexact
	       Enable bitexact output.

       srcw
	   Set source width.

       srch
	   Set source height.

       dstw
	   Set destination width.

       dsth
	   Set destination height.

       src_format
	   Set source pixel format (must be expressed as an integer).

       dst_format
	   Set destination pixel format (must be expressed as an integer).

       src_range
	   Select source range.

       dst_range
	   Select destination range.

       param0, param1
	   Set scaling algorithm parameters. The specified values are specific
	   of some scaling algorithms and ignored by others. The specified
	   values are floating point number values.

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.

	       input --> split ---------------------> overlay --> output
			   |				 ^
			   |				 |
			   +-----> crop --> vflip -------+

       This filtergraph splits the input stream in two streams, 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 in output the top half of the video is mirrored
       onto the bottom half.

       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 the 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", a filter with no
       output pads is called a "sink".

   Filtergraph syntax
       A filtergraph can be represented using a textual representation, which
       is recognized by the -filter/-vf and -filter_complex options in ffmpeg
       and -vf in ffplay, and by the
       "avfilter_graph_parse()"/"avfilter_graph_parse2()" 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=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.  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 the following forms:

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

       ·   A ':'-separated list of value. In this case, the keys are assumed
	   to be the option names in the order they are declared. E.g. the
	   "fade" filter declares three options in this order -- type,
	   start_frame and nb_frames. Then the parameter list in:0:30 means
	   that the value in is assigned to the option type, 0 to start_frame
	   and 30 to nb_frames.

       ·   A ':'-separated list of mixed direct value and long key=value
	   pairs. The direct value must precede the key=value pairs, and
	   follow the same constraints order of the previous point. The
	   following key=value pairs can be set in any preferred order.

       If the option value itself is a list of items (e.g. the "format" filter
       takes a list of pixel formats), the items in the list are usually
       separated by '|'.

       The list of arguments can be quoted using the character "'" as initial
       and ending mark, and the character '\' for escaping the characters
       within the quoted text; otherwise the argument string is considered
       terminated when the next special character (belonging to the set
       "[]=;,") is encountered.

       The name and arguments of the filter are optionally preceded and
       followed by a list of link labels.  A link label allows 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 complete filterchain all the unlabelled filter input and output
       pads must be connected. A filtergraph is considered valid if all the
       filter input and output pads of all the filterchains are connected.

       Libavfilter will automatically insert scale filters where format
       conversion is required. It is possible to specify swscale flags for
       those automatically inserted scalers by prepending "sws_flags=flags;"
       to the filtergraph description.

       Follows a BNF description for the filtergraph syntax:

	       <NAME>		  ::= sequence of alphanumeric characters and '_'
	       <LINKLABEL>	  ::= "[" <NAME> "]"
	       <LINKLABELS>	  ::= <LINKLABEL> [<LINKLABELS>]
	       <FILTER_ARGUMENTS> ::= sequence of chars (eventually quoted)
	       <FILTER>		  ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
	       <FILTERCHAIN>	  ::= <FILTER> [,<FILTERCHAIN>]
	       <FILTERGRAPH>	  ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

   Notes on filtergraph escaping
       Some filter arguments require the use of special characters, typically
       ":" to separate key=value pairs in a named options list. In this case
       the user should perform a first level escaping when specifying the
       filter arguments. For example, consider the following literal string to
       be embedded in the drawtext filter arguments:

	       this is a 'string': may contain one, or more, special characters

       Since ":" is special for the filter arguments syntax, it needs to be
       escaped, so you get:

	       text=this is a \'string\'\: may contain one, or more, special characters

       A second level of escaping is required when embedding the filter
       arguments 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

       Finally an additional level of escaping may be 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"

       Sometimes, it might be more convenient to employ quoting in place of
       escaping. For example the string:

	       Caesar: tu quoque, Brute, fili mi

       Can be quoted in the filter arguments as:

	       text='Caesar: tu quoque, Brute, fili mi'

       And finally inserted in a filtergraph like:

	       drawtext=text=\'Caesar: tu quoque\, Brute\, fili mi\'

       See the ``Quoting and escaping'' section in the ffmpeg-utils manual for
       more information about the escaping and quoting rules adopted by
       FFmpeg.

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

       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

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.

   aconvert
       Convert the input audio format to the specified formats.

       This filter is deprecated. Use aformat instead.

       The filter accepts a string of the form:
       "sample_format:channel_layout".

       sample_format specifies the sample format, and can be a string or the
       corresponding numeric value defined in libavutil/samplefmt.h. Use 'p'
       suffix for a planar sample format.

       channel_layout specifies the channel layout, and can be a string or the
       corresponding number value defined in libavutil/channel_layout.h.

       The special parameter "auto", signifies that the filter will
       automatically select the output format depending on the output filter.

       Examples

       ·   Convert input to float, planar, stereo:

		   aconvert=fltp:stereo

       ·   Convert input to unsigned 8-bit, automatically select out channel
	   layout:

		   aconvert=u8:auto

   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 time for starting to apply the fade effect. Default is 0.
	   The accepted syntax is:

		   [-]HH[:MM[:SS[.m...]]]
		   [-]S+[.m...]

	   See also the function "av_parse_time()".  If set this option is
	   used instead of start_sample one.

       duration, d
	   Specify the duration for which the fade effect has to last. Default
	   is 0.  The accepted syntax is:

		   [-]HH[:MM[:SS[.m...]]]
		   [-]S+[.m...]

	   See also the function "av_parse_time()".  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.	If set this option is used instead of nb_samples one.

       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

	   par select inverted parabola

	   qua select quadratic

	   cub select cubic

	   squ select square root

	   cbr select cubic root

       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

   aformat
       Set output format constraints for the input audio. The framework will
       negotiate the most appropriate format to minimize conversions.

       The filter accepts the following named parameters:

       sample_fmts
	   A '|'-separated list of requested sample formats.

       sample_rates
	   A '|'-separated list of requested sample rates.

       channel_layouts
	   A '|'-separated list of requested channel layouts.

       If a parameter is omitted, all values are allowed.

       For example to force the output to either unsigned 8-bit or signed
       16-bit stereo:

	       aformat=sample_fmts=u8|s16:channel_layouts=stereo

   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
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

   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.

       For example

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

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

       The filter accepts the following named parameters:

       inputs
	   Number of inputs. If unspecified, it defaults to 2.

       duration
	   How to determine the end-of-stream.

	   longest
	       Duration of longest input. (default)

	   shortest
	       Duration of shortest input.

	   first
	       Duration of first input.

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

   anull
       Pass the audio source unchanged to the output.

   apad
       Pad the end of a audio stream with silence, this can be used together
       with -shortest to extend audio streams to the same length as the video
       stream.

   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

   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 ffmpeg-resampler 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

   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
       signal 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.

       A description of each shown parameter follows:

       n   sequential number of the input frame, starting from 0

       pts 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
	   presentation timestamp of the input frame in seconds

       pos position of the frame in the input stream, -1 if this information
	   in unavailable and/or meaningless (for example in case of synthetic
	   audio)

       fmt sample format

       chlayout
	   channel layout

       rate
	   sample rate for the audio frame

       nb_samples
	   number of samples (per channel) in the frame

       checksum
	   Adler-32 checksum (printed in hexadecimal) of the audio data. For
	   planar audio the data is treated as if all the planes were
	   concatenated.

       plane_checksums
	   A list of Adler-32 checksums for each data plane.

   astats
       Display time domain statistical information about the audio channels.
       Statistics are calculated and displayed for each audio channel and,
       where applicable, an overall figure is also given.

       The filter accepts the following option:

       length
	   Short window length in seconds, used for peak and trough RMS
	   measurement.	 Default is 0.05 (50 miliseconds). Allowed range is
	   "[0.1 - 10]".

       A description of each shown parameter follows:

       DC offset
	   Mean amplitude displacement from zero.

       Min level
	   Minimal sample level.

       Max level
	   Maximal sample level.

       Peak level dB
       RMS level dB
	   Standard peak and RMS level measured in dBFS.

       RMS peak dB
       RMS trough dB
	   Peak and trough values for RMS level measured over a short window.

       Crest factor
	   Standard ratio of peak to RMS level (note: not in dB).

       Flat factor
	   Flatness (i.e. consecutive samples with the same value) of the
	   signal at its peak levels (i.e. either Min level or Max level).

       Peak count
	   Number of occasions (not the number of samples) that the signal
	   attained either Min level or Max level.

   astreamsync
       Forward two audio streams and control the order the buffers are
       forwarded.

       The filter accepts the following options:

       expr, e
	   Set the expression deciding which stream should be forwarded next:
	   if the result is negative, the first stream is forwarded; if the
	   result is positive or zero, the second stream is forwarded. It can
	   use the following variables:

	   b1 b2
	       number of buffers forwarded so far on each stream

	   s1 s2
	       number of samples forwarded so far on each stream

	   t1 t2
	       current timestamp of each stream

	   The default value is "t1-t2", which means to always forward the
	   stream that has a smaller timestamp.

       Examples

       Stress-test "amerge" by randomly sending buffers on the wrong input,
       while avoiding too much of a desynchronization:

	       amovie=file.ogg [a] ; amovie=file.mp3 [b] ;
	       [a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ;
	       [a2] [b2] amerge

   asyncts
       Synchronize audio data with timestamps by squeezing/stretching it
       and/or dropping samples/adding silence when needed.

       This filter is not built by default, please use aresample to do
       squeezing/stretching.

       The filter accepts the following named parameters:

       compensate
	   Enable stretching/squeezing the data to make it match the
	   timestamps. Disabled by default. When disabled, time gaps are
	   covered with silence.

       min_delta
	   Minimum difference between timestamps and audio data (in seconds)
	   to trigger adding/dropping samples. Default value is 0.1. If you
	   get non-perfect sync with this filter, try setting this parameter
	   to 0.

       max_comp
	   Maximum compensation in samples per second. Relevant only with
	   compensate=1.  Default value 500.

       first_pts
	   Assume the first pts should be this value. The time base 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.

   atempo
       Adjust audio tempo.

       The filter accepts exactly one parameter, the audio tempo. If not
       specified then the filter will assume nominal 1.0 tempo. Tempo must be
       in the [0.5, 2.0] range.

       Examples

       ·   Slow down audio to 80% tempo:

		   atempo=0.8

       ·   To speed up audio to 125% tempo:

		   atempo=1.25

   atrim
       Trim the input so that the output contains one continuous subpart of
       the input.

       This filter accepts the following options:

       start
	   Timestamp (in seconds) of the start of the kept section. I.e. the
	   audio sample with the timestamp start will be the first sample in
	   the output.

       end Timestamp (in seconds) 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
	   Maximum duration of the output in seconds.

       start_sample
	   Number of the first sample that should be passed to output.

       end_sample
	   Number of the first sample that should be dropped.

       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 that the output timestamps start
       at zero, insert the asetpts filter after the atrim filter.

       If multiple start or end options are set, this filter tries to be
       greedy and keep all samples that match at least one of the specified
       constraints. To keep only the part that matches all the constraints at
       once, chain multiple atrim filters.

       The defaults are such that all the input is kept. So it is possible to
       set e.g.	 just the end values to keep everything before the specified
       time.

       Examples:

       ·   drop everything except the second minute of input

		   ffmpeg -i INPUT -af atrim=60:120

       ·   keep only the first 1000 samples

		   ffmpeg -i INPUT -af atrim=end_sample=1000

   bandpass
       Apply a two-pole Butterworth band-pass filter with central frequency
       frequency, and (3dB-point) band-width width.  The csg option selects a
       constant skirt gain (peak gain = Q) instead of the default: constant
       0dB peak gain.  The filter roll off at 6dB per octave (20dB per
       decade).

       The filter accepts the following options:

       frequency, f
	   Set the filter's central frequency. Default is 3000.

       csg Constant skirt gain if set to 1. Defaults to 0.

       width_type
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

   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
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

   bass
       Boost or cut the bass (lower) frequencies of the audio using a two-pole
       shelving filter with a response similar to that of a standard hi-fi's
       tone-controls. This is also known as shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
	   Give the gain at 0 Hz. Its useful range is about -20 (for a large
	   cut) to +20 (for a large boost).  Beware of clipping when using a
	   positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 100 Hz.

       width_type
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Determine how steep is the filter's shelf transition.

   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.

   channelmap
       Remap input channels to new locations.

       This filter accepts the following named parameters:

       channel_layout
	   Channel layout of the output stream.

       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.

       If no mapping is present, the filter will implicitly map input channels
       to output channels preserving index.

       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:channel_layout=5.1' out.wav

   channelsplit
       Split each channel in input audio stream into a separate output stream.

       This filter accepts the following named parameters:

       channel_layout
	   Channel layout of the input stream. Default is "stereo".

       For example, assuming a stereo input MP3 file

	       ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

       will create an output Matroska file with two audio streams, one
       containing only the left channel and the other the right channel.

       To 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

   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
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.

       gain, g
	   Set the required gain or attenuation in dB.	Beware of clipping
	   when using a positive gain.

   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
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.  Applies
	   only to double-pole filter.	The default is 0.707q and gives a
	   Butterworth response.

   join
       Join multiple input streams into one multi-channel stream.

       The filter accepts the following named parameters:

       inputs
	   Number of input streams. Defaults to 2.

       channel_layout
	   Desired output channel layout. 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 those 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.

       E.g. to join 3 inputs (with properly set channel layouts)

	       ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

       To 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

   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
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Specify the band-width of a filter in width_type units.  Applies
	   only to double-pole filter.	The default is 0.707q and gives a
	   Butterworth response.

   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 remap efficiently 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=1: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"

   resample
       Convert the audio sample format, sample rate and channel layout. This
       filter is not meant to be used directly.

   silencedetect
       Detect silence in an audio stream.

       This filter logs a message when it detects that the input audio volume
       is less or equal to a noise tolerance value for a duration greater or
       equal to the minimum detected noise duration.

       The printed times and duration are expressed in seconds.

       The filter accepts the following options:

       duration, d
	   Set silence duration until notification (default is 2 seconds).

       noise, n
	   Set noise tolerance. Can be specified in dB (in case "dB" is
	   appended to the specified value) or amplitude ratio. Default is
	   -60dB, or 0.001.

       Examples

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

		   silencedetect=n=-50dB:d=5

       ·   Complete example with ffmpeg to detect silence with 0.0001 noise
	   tolerance in silence.mp3:

		   ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

   treble
       Boost or cut treble (upper) frequencies of the audio using a two-pole
       shelving filter with a response similar to that of a standard hi-fi's
       tone-controls. This is also known as shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
	   Give the gain at whichever is the lower of ~22 kHz and the Nyquist
	   frequency. Its useful range is about -20 (for a large cut) to +20
	   (for a large boost). Beware of clipping when using a positive gain.

       frequency, f
	   Set the filter's central frequency and so can be used to extend or
	   reduce the frequency range to be boosted or cut.  The default value
	   is 3000 Hz.

       width_type
	   Set method to specify band-width of filter.

	   h   Hz

	   q   Q-Factor

	   o   octave

	   s   slope

       width, w
	   Determine how steep is the filter's shelf transition.

   volume
       Adjust the input audio volume.

       The filter accepts the following options:

       volume
	   Expresses how the audio volume will be increased or decreased.

	   Output values are clipped to the maximum value.

	   The output audio volume is given by the relation:

		   <output_volume> = <volume> * <input_volume>

	   Default value for volume is 1.0.

       precision
	   Set the mathematical precision.

	   This determines which input sample formats will be allowed, which
	   affects the precision of the volume scaling.

	   fixed
	       8-bit fixed-point; limits input sample format to U8, S16, and
	       S32.

	   float
	       32-bit floating-point; limits input sample format to FLT.
	       (default)

	   double
	       64-bit floating-point; limits input sample format to DBL.

       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

   volumedetect
       Detect the volume of the input video.

       The filter has no parameters. The input is not modified. Statistics
       about the volume will be printed in the log when the input stream end
       is reached.

       In particular it will show the mean volume (root mean square), maximum
       volume (on a per-sample basis), and the beginning of an histogram of
       the registered volume values (from the maximum value to a cumulated
       1/1000 of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

	       [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
	       [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
	       [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
	       [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
	       [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
	       [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
	       [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
	       [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
	       [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means that:

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

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

       ·   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

       In other words, raising the volume by +4 dB does not cause any
       clipping, raising it by +5 dB causes clipping for 6 samples, etc.

AUDIO SOURCES
       Below is a description of the currently available audio sources.

   abuffer
       Buffer audio frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in libavfilter/asrc_abuffer.h.

       It accepts the following named parameters:

       time_base
	   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 corresponging 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.

       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 function
	   "av_parse_time()" for the accepted format.  Note that the resulting
	   duration may be greater than the specified duration, as the
	   generated audio is always cut at the end of a complete frame.

	   If not specified, or the expressed duration is negative, the audio
	   is supposed to be generated forever.

       nb_samples, n
	   Set the number of samples per channel per each output frame,
	   default to 1024.

       sample_rate, s
	   Specify the sample rate, default to 44100.

       Each expression in exprs can contain the following constants:

       n   number of the evaluated sample, starting from 0

       t   time of the evaluated sample expressed in seconds, starting from 0

       s   sample rate

       Examples

       ·   Generate silence:

		   aevalsrc=0

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

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

       ·   Generate a two channels signal, specify the channel layout (Front
	   Center + Back Center) explicitly:

		   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       ·   Generate white noise:

		   aevalsrc="-2+random(0)"

       ·   Generate an amplitude modulated signal:

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

       ·   Generate 2.5 Hz binaural beats on a 360 Hz carrier:

		   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

   anullsrc
       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
	   Specify 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
	   Specify the sample rate, and defaults to 44100.

       nb_samples, n
	   Set the number of samples per requested frames.

       Examples

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

		   anullsrc=r=48000:cl=4

       ·   Do the same operation with a more obvious syntax:

		   anullsrc=r=48000:cl=mono

       All the parameters need to be explicitly defined.

   flite
       Synthesize a voice utterance using the libflite library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libflite".

       Note that the flite library is not thread-safe.

       The filter accepts the following options:

       list_voices
	   If set to 1, list the names of the available voices and exit
	   immediately. Default value is 0.

       nb_samples, n
	   Set the maximum number of samples per frame. Default value is 512.

       textfile
	   Set the filename containing the text to speak.

       text
	   Set the text to speak.

       voice, v
	   Set the voice to use for the speech synthesis. Default value is
	   "kal". See also the list_voices option.

       Examples

       ·   Read from file speech.txt, and synthetize the text using the
	   standard flite voice:

		   flite=textfile=speech.txt

       ·   Read the specified text selecting the "slt" voice:

		   flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Input text to ffmpeg:

		   ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Make ffplay speak the specified text, using "flite" and the "lavfi"
	   device:

		   ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

       For more information about libflite, check:
       <http://www.speech.cs.cmu.edu/flite/>

   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, s
	   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, 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

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 to be employed in analysis / debugging
       tools.

VIDEO FILTERS
       When you configure your FFmpeg build, you can disable any of the
       existing filters using "--disable-filters".  The configure output will
       show the video filters included in your build.

       Below is a description of the currently available video filters.

   alphaextract
       Extract the alpha component from the input as a grayscale video. This
       is especially useful with the alphamerge filter.

   alphamerge
       Add or replace the alpha component of the primary input with the
       grayscale value of a second input. This is intended for use with
       alphaextract to allow the transmission or storage of frame sequences
       that have alpha in a format that doesn't support an alpha channel.

       For example, to reconstruct full frames from a normal YUV-encoded video
       and a separate video created with alphaextract, you might use:

	       movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

       Since this filter is designed for reconstruction, it operates on frame
       sequences without considering timestamps, and terminates when either
       input reaches end of stream. This will cause problems if your encoding
       pipeline drops frames. If you're trying to apply an image as an overlay
       to a video stream, consider the overlay filter instead.

   ass
       Same as the subtitles filter, except that it doesn't require libavcodec
       and libavformat to work. On the other hand, it is limited to ASS
       (Advanced Substation Alpha) subtitles files.

   bbox
       Compute the bounding box for the non-black pixels in the input frame
       luminance plane.

       This filter computes the bounding box containing all the pixels with a
       luminance value greater than the minimum allowed value.	The parameters
       describing the bounding box are printed on the filter log.

       The filter accepts the following option:

       min_val
	   Set the minimal luminance value. Default is 16.

   blackdetect
       Detect video intervals that are (almost) completely black. Can be
       useful to detect chapter transitions, commercials, or invalid
       recordings. Output lines contains the time for the start, end and
       duration of the detected black interval expressed in seconds.

       In order to display the output lines, you need to set the loglevel at
       least to the AV_LOG_INFO value.

       The filter accepts the following options:

       black_min_duration, d
	   Set the minimum detected black duration expressed in seconds. It
	   must be a non-negative floating point number.

	   Default value is 2.0.

       picture_black_ratio_th, pic_th
	   Set the threshold for considering a picture "black".	 Express the
	   minimum value for the ratio:

		   <nb_black_pixels> / <nb_pixels>

	   for which a picture is considered black.  Default value is 0.98.

       pixel_black_th, pix_th
	   Set the threshold for considering a pixel "black".

	   The threshold expresses the maximum pixel luminance value for which
	   a pixel is considered "black". The provided value is scaled
	   according to the following equation:

		   <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

	   luminance_range_size and luminance_minimum_value depend on the
	   input video format, the range is [0-255] for YUV full-range formats
	   and [16-235] for YUV non full-range formats.

	   Default value is 0.10.

       The following example sets the maximum pixel threshold to the minimum
       value, and detects only black intervals of 2 or more seconds:

	       blackdetect=d=2:pix_th=0.00

   blackframe
       Detect frames that are (almost) completely black. Can be useful to
       detect chapter transitions or commercials. Output lines consist of the
       frame number of the detected frame, the percentage of blackness, the
       position in the file if known or -1 and the timestamp in seconds.

       In order to display the output lines, you need to set the loglevel at
       least to the AV_LOG_INFO value.

       The filter accepts the following options:

       amount
	   Set the percentage of the pixels that have to be below the
	   threshold, defaults to 98.

       threshold, thresh
	   Set the threshold below which a pixel value is considered black,
	   defaults to 32.

   blend
       Blend two video frames into each other.

       It takes two input streams and outputs one stream, the first input is
       the "top" layer and second input is "bottom" layer.  Output terminates
       when shortest 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:

	   addition
	   and
	   average
	   burn
	   darken
	   difference
	   divide
	   dodge
	   exclusion
	   hardlight
	   lighten
	   multiply
	   negation
	   normal
	   or
	   overlay
	   phoenix
	   pinlight
	   reflect
	   screen
	   softlight
	   subtract
	   vividlight
	   xor
       c0_opacity
       c1_opacity
       c2_opacity
       c3_opacity
       all_opacity
	   Set blend opacity for specific pixel component or all pixel
	   components in case of all_opacity. Only used in combination with
	   pixel component blend modes.

       c0_expr
       c1_expr
       c2_expr
       c3_expr
       all_expr
	   Set blend expression for specific pixel component or all pixel
	   components in case of all_expr. Note that related mode options will
	   be ignored if those are set.

	   The expressions can use the following variables:

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

	   X
	   Y   the coordinates of the current sample

	   W
	   H   the width and height of currently filtered plane

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

	   T   Time of the current frame, expressed in seconds.

	   TOP, A
	       Value of pixel component at current location for first video
	       frame (top layer).

	   BOTTOM, B
	       Value of pixel component at current location for second video
	       frame (bottom layer).

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

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

       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 1x1 checkerboard effect:

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

   boxblur
       Apply boxblur algorithm to the input video.

       The filter accepts the following options:

       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
	       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 luma, chroma, and alpha radius set to
	   2:

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

       ·   Set luma radius to 2, alpha and chroma radius to 0:

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

       ·   Set luma and chroma radius 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

   colorbalance
       Modify intensity of primary colors (red, green and blue) of input
       frames.

       The filter allows an input frame to be adjusted in the shadows,
       midtones or highlights regions for the red-cyan, green-magenta or blue-
       yellow balance.

       A positive adjustment value shifts the balance towards the primary
       color, a negative value towards the complementary color.

       The filter accepts the following options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

	   Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

       Examples

       ·   Add red color cast to shadows:

		   colorbalance=rs=.3

   colorchannelmixer
       Adjust video input frames by re-mixing color channels.

       This filter modifies a color channel by adding the values associated to
       the other channels of the same pixels. For example if the value to
       modify is red, the output value will be:

	       <red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>

       The filter accepts the following options:

       rr
       rg
       rb
       ra  Adjust contribution of input red, green, blue and alpha channels
	   for output red channel.  Default is 1 for rr, and 0 for rg, rb and
	   ra.

       gr
       gg
       gb
       ga  Adjust contribution of input red, green, blue and alpha channels
	   for output green channel.  Default is 1 for gg, and 0 for gr, gb
	   and ga.

       br
       bg
       bb
       ba  Adjust contribution of input red, green, blue and alpha channels
	   for output blue channel.  Default is 1 for bb, and 0 for br, bg and
	   ba.

       ar
       ag
       ab
       aa  Adjust contribution of input red, green, blue and alpha channels
	   for output alpha channel.  Default is 1 for aa, and 0 for ar, ag
	   and ab.

	   Allowed ranges for options are "[-2.0, 2.0]".

       Examples

       ·   Convert source to grayscale:

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

   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

	   bt601
	       BT.601

	   smpte240m
	       SMPTE-240M

	   fcc FCC

       For example to convert from BT.601 to SMPTE-240M, use the command:

	       colormatrix=bt601:smpte240m

   copy
       Copy the input source unchanged to the output. Mainly useful for
       testing purposes.

   crop
       Crop the input video to given dimensions.

       The filter accepts the following options:

       w, out_w
	   Width of the output video. It defaults to "iw".  This expression is
	   evaluated only once during the filter configuration.

       h, out_h
	   Height of the output video. It defaults to "ih".  This expression
	   is evaluated only once during the filter configuration.

       x   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   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.

       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  same as in_w and in_h

       out_w
       out_h
	   the output (cropped) width and height

       ow
       oh  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 input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       t   timestamp expressed in seconds, 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

       ·   Appply 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)

   cropdetect
       Auto-detect crop size.

       Calculate necessary cropping parameters and prints the recommended
       parameters through the logging system. The detected dimensions
       correspond to the non-black area of the input video.

       The filter accepts the following options:

       limit
	   Set higher black value threshold, which can be optionally specified
	   from nothing (0) to everything (255). An intensity value greater to
	   the set value is considered non-black. Default value is 24.

       round
	   Set the value for which the width/height should be divisible by.
	   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. Default value is 16.

       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 return the largest area encountered
	   during playback.

   curves
       Apply color adjustments using curves.

       This filter is similar to the Adobe Photoshop and GIMP curves tools.
       Each component (red, green and blue) has its values defined by N key
       points tied from each other using a smooth curve. The x-axis represents
       the pixel values from the input frame, and the y-axis the new pixel
       values to be set for the output frame.

       By default, a component curve is defined by the two points (0;0) and
       (1;1). This creates a straight line where each original pixel value is
       "adjusted" to its own value, which means no change to the image.

       The filter allows you to redefine these two points and add some more. A
       new curve (using a natural cubic spline interpolation) will be define
       to pass smoothly through all these new coordinates. The new defined
       points needs to be strictly increasing over the x-axis, and their x and
       y values must be in the [0;1] interval.	If the computed curves
       happened to go outside the vector spaces, the values will be clipped
       accordingly.

       If there is no key point defined in "x=0", the filter will
       automatically insert a (0;0) point. In the same way, if there is no key
       point defined in "x=1", the filter will automatically insert a (1;1)
       point.

       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 (".asv") to import the settings
	   from.

       To avoid some filtergraph syntax conflicts, each key points list need
       to be defined using the following syntax: "x0/y0 x1/y1 x2/y2 ...".

       Examples

       ·   Increase slightly the middle level of blue:

		   curves=blue='0.5/0.58'

       ·   Vintage effect:

		   curves=r='0/0.11 .42/.51 1/0.95':g='0.50/0.48':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.asv':green='0.45/0.53'

   dctdnoiz
       Denoise frames using 2D DCT (frequency domain filtering).

       This filter is not designed for real time and can be extremely slow.

       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. Each block is of size
	   "16x16". 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 allow to process the whole input
	   width or height, a warning will be displayed and according borders
	   won't be denoised.

	   Default value is 15.

       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.

       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)'

   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.

   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).

       This filter accepts the following options:

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

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

       band, t
	   Specify the thickness of the fuzzy edge of the rectangle (added to
	   w and h). The default value is 4.

       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.

       Examples

       ·   Set a rectangle covering the area with top left corner coordinates
	   0,0 and size 100x77, setting a band of size 10:

		   delogo=x=0:y=0:w=100:h=77:band=10

   deshake
       Attempt to fix small changes in horizontal and/or vertical shift. This
       filter helps remove camera shake from hand-holding a camera, bumping a
       tripod, moving on a vehicle, etc.

       The filter accepts the following options:

       x
       y
       w
       h   Specify a rectangular area where to limit the search for motion
	   vectors.  If desired the search for motion vectors can be limited
	   to a rectangular area of the frame defined by its top left corner,
	   width and height. These parameters have the same meaning as the
	   drawbox filter which can be used to visualise the position of the
	   bounding box.

	   This is useful when simultaneous movement of subjects within the
	   frame might be confused for camera motion by the motion vector
	   search.

	   If any or all of x, y, w and h are set to -1 then the full frame is
	   used. This allows later options to be set without specifying the
	   bounding box for the motion vector search.

	   Default - search the whole frame.

       rx
       ry  Specify the maximum extent of movement in x and y directions in the
	   range 0-64 pixels. Default 16.

       edge
	   Specify how to generate pixels to fill blanks at the edge of the
	   frame. Available values are:

	   blank, 0
	       Fill zeroes at blank locations

	   original, 1
	       Original image at blank locations

	   clamp, 2
	       Extruded edge value at blank locations

	   mirror, 3
	       Mirrored edge at blank locations

	   Default value is mirror.

       blocksize
	   Specify the blocksize to use for motion search. Range 4-128 pixels,
	   default 8.

       contrast
	   Specify the contrast threshold for blocks. Only blocks with more
	   than the specified contrast (difference between darkest and
	   lightest pixels) will be considered. Range 1-255, default 125.

       search
	   Specify the search strategy. Available values are:

	   exhaustive, 0
	       Set exhaustive search

	   less, 1
	       Set less exhaustive search.

	   Default value is exhaustive.

       filename
	   If set then a detailed log of the motion search is written to the
	   specified file.

       opencl
	   If set to 1, specify using OpenCL capabilities, only available if
	   FFmpeg was configured with "--enable-opencl". Default value is 0.

   drawbox
       Draw a colored box on the input image.

       This filter accepts the following options:

       x
       y   The expressions which specify the top left corner coordinates of
	   the box. Default 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. Default to 0.

       color, c
	   Specify the color of the box to write, it can be the name of a
	   color (case insensitive match) or a 0xRRGGBB[AA] sequence. If the
	   special value "invert" is used, the box edge color is the same as
	   the video with inverted luma.

       thickness, t
	   The expression which sets the thickness of the box edge. Default
	   value is 3.

	   See below for the list of accepted constants.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y offset coordinates where the box is drawn.

       w
       h   The width and height of the drawn box.

       t   The thickness of the drawn box.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

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

		   drawbox

       ·   Draw a box with color red and an opacity of 50%:

		   drawbox=10:20:200:60:red@0.5

	   The previous example can be specified as:

		   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       ·   Fill the box with pink color:

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

       ·   Draw a 2-pixel red 2.40:1 mask:

		   drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

   drawgrid
       Draw a grid on the input image.

       This filter accepts the following options:

       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, it can be the name of a color (case
	   insensitive match) or a 0xRRGGBB[AA] sequence. If the special value
	   "invert" is used, the grid color is the same as the video with
	   inverted luma.  Note that you can append opacity value (in range of
	   0.0 - 1.0) to color name after @ sign.

       thickness, t
	   The expression which sets the thickness of the grid line. Default
	   value is 1.

	   See below for the list of accepted constants.

       The parameters for x, y, w and h and t are expressions containing the
       following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
	   The input grid cell width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y coordinates of some point of grid intersection (meant
	   to configure offset).

       w
       h   The width and height of the drawn cell.

       t   The thickness of the drawn cell.

	   These constants allow the x, y, w, h and t expressions to refer to
	   each other, so you may for example specify "y=x/dar" or "h=w/dar".

       Examples

       ·   Draw a grid with cell 100x100 pixels, thickness 2 pixels, with
	   color red and an opacity of 50%:

		   drawgrid=width=100:height=100:thickness=2:color=red@0.5

       ·   Draw a white 3x3 grid with an opacity of 50%:

		   drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

   drawtext
       Draw text string or text from specified file on top of video using the
       libfreetype library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libfreetype".

       Syntax

       The description of the accepted parameters follows.

       box Used to draw a box around text using background color.  Value
	   should be either 1 (enable) or 0 (disable).	The default value of
	   box is 0.

       boxcolor
	   The color to be used for drawing box around text.  Either a string
	   (e.g. "yellow") or in 0xRRGGBB[AA] format (e.g. "0xff00ff"),
	   possibly followed by an alpha specifier.  The default value of
	   boxcolor is "white".

       draw
	   Set an expression which specifies if the text should be drawn. If
	   the expression evaluates to 0, the text is not drawn. This is
	   useful for specifying that the text should be drawn only when
	   specific conditions are met.

	   Default value is "1".

	   See below for the list of accepted constants and functions.

       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.

       fix_bounds
	   If true, check and fix text coords to avoid clipping.

       fontcolor
	   The color to be used for drawing fonts.  Either a string (e.g.
	   "red") or in 0xRRGGBB[AA] format (e.g. "0xff000033"), possibly
	   followed by an alpha specifier.  The default value of fontcolor is
	   "black".

       fontfile
	   The font file to be used for drawing text. Path must be included.
	   This parameter is mandatory.

       fontsize
	   The font size to be used for drawing text.  The default value of
	   fontsize is 16.

       ft_load_flags
	   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 "render".

	   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.
	   It can be a color name (e.g. "yellow") or a string in the
	   0xRRGGBB[AA] form (e.g. "0xff00ff"), possibly followed by an alpha
	   specifier.  The default value of shadowcolor is "black".

       shadowx
       shadowy
	   The x and y offsets for the text shadow position with respect to
	   the position of the text. They can be either positive or negative
	   values. 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).

       text
	   The text string to be drawn. The text must be a sequence of UTF-8
	   encoded characters.	This parameter is mandatory if no file is
	   specified with the parameter textfile.

       textfile
	   A text file containing text to be drawn. The text must be a
	   sequence of UTF-8 encoded characters.

	   This parameter is mandatory if no text string is specified with the
	   parameter text.

	   If both text and textfile are specified, an error is thrown.

       reload
	   If set to 1, the textfile will be reloaded before each frame.  Be
	   sure to update it atomically, or it may be read partially, or even
	   fail.

       x
       y   The expressions which specify the offsets where text will be drawn
	   within the video frame. They are relative to the top/left border of
	   the output image.

	   The default value of x and y is "0".

	   See below for the list of accepted constants and functions.

       The parameters for x and y are expressions containing the following
       constants and functions:

       dar input display aspect ratio, it is the same as (w / h) * sar

       hsub
       vsub
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       line_h, lh
	   the height of each text line

       main_h, h, H
	   the input height

       main_w, w, W
	   the input width

       max_glyph_a, ascent
	   the maximum distance from the baseline to the highest/upper grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  It is a positive value, due to the grid's
	   orientation with the Y axis upwards.

       max_glyph_d, descent
	   the maximum distance from the baseline to the lowest grid
	   coordinate used to place a glyph outline point, for all the
	   rendered glyphs.  This is a negative value, due to the grid's
	   orientation, with the Y axis upwards.

       max_glyph_h
	   maximum glyph height, that is the maximum height for all the glyphs
	   contained in the rendered text, it is equivalent to ascent -
	   descent.

       max_glyph_w
	   maximum glyph width, that is the maximum width for all the glyphs
	   contained in the rendered text

       n   the number of input frame, starting from 0

       rand(min, max)
	   return a random number included between min and max

       sar input sample aspect ratio

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

       text_h, th
	   the height of the rendered text

       text_w, tw
	   the width of the rendered text

       x
       y   the x and y offset coordinates where the text is drawn.

	   These parameters allow the x and y expressions to refer each other,
	   so you can for example specify "y=x/dar".

       If libavfilter was built with "--enable-fontconfig", then fontfile can
       be a fontconfig pattern or omitted.

       Text expansion

       If expansion is set to "strftime", the filter recognizes strftime()
       sequences in the provided text and expands them accordingly. Check the
       documentation of strftime(). This feature is deprecated.

       If expansion is set to "none", the text is printed verbatim.

       If expansion is set to "normal" (which is the default), the following
       expansion mechanism is used.

       The backslash character '\', followed by any character, always expands
       to the second character.

       Sequence of the form "%{...}" are expanded. The text between the braces
       is a function name, possibly followed by arguments separated by ':'.
       If the arguments contain special characters or delimiters (':' or '}'),
       they should be escaped.

       Note that they probably must also be escaped as the value for the text
       option in the filter argument string and as the filter argument in the
       filtergraph description, and possibly also for the shell, that makes up
       to four levels of escaping; using a text file avoids these problems.

       The following functions are available:

       expr, e
	   The expression evaluation result.

	   It must take one argument specifying the expression to be
	   evaluated, which accepts the same constants and functions as the x
	   and y values. Note that not all constants should be used, for
	   example the text size is not known when evaluating the expression,
	   so the constants text_w and text_h will have an undefined value.

       gmtime
	   The time at which the filter is running, expressed in UTC.  It can
	   accept an argument: a strftime() format string.

       localtime
	   The time at which the filter is running, expressed in the local
	   time zone.  It can accept an argument: a strftime() format string.

       metadata
	   Frame metadata. It must take one argument specifying metadata key.

       n, frame_num
	   The frame number, starting from 0.

       pict_type
	   A 1 character description of the current picture type.

       pts The timestamp of the current frame, in seconds, with microsecond
	   accuracy.

       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-line_h)/2"

       ·   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:draw=lt(mod(t\,3)\,1):text='blink'"

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

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

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

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

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

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

   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 choosen 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".

       Example:

	       edgedetect=low=0.1:high=0.4

   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 fade-in/out effect to input video.

       This filter accepts the following options:

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

       start_frame, s
	   Specify the number of the start frame for starting to apply the
	   fade effect. Default is 0.

       nb_frames, n
	   The number of frames 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 completely black.  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 completely black.  If both duration and
	   nb_frames are specified, duration is used. Default is 0.

       Examples

       ·   Fade in first 30 frames of video:

		   fade=in:0:30

	   The command above is equivalent to:

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

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

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

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

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

       ·   Make first 5 frames black, then fade in from frame 5-24:

		   fade=in:5:20

       ·   Fade in alpha over first 25 frames of video:

		   fade=in:0:25:alpha=1

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

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

   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").

   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 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.

       This filter accepts the following options:

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

       Default value is tff.

       Transformation is achieved 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.

       This filter 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
       Buffer input images and send them when they are requested.

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

       The filter does not take parameters.

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

       This filter accepts the following parameters:

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

       Examples

       ·   Convert the input video to the format yuv420p

		   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.

       This filter accepts the following named parameters:

       fps Desired output frame rate. The default is 25.

       round
	   Rounding method.

	   Possible values are:

	   zero
	       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".

       Alternatively, the options can be specified as a flat string:
       fps[: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

   framestep
       Select one frame every N-th frame.

       This filter accepts the following option:

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

   frei0r
       Apply a frei0r effect to the input video.

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

       This filter accepts the following options:

       filter_name
	   The name to the frei0r effect to load. If the environment variable
	   FREI0R_PATH is defined, the frei0r effect is searched in each one
	   of the directories specified by the colon separated list in
	   FREIOR_PATH, otherwise in the standard frei0r paths, which are 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 (whose values are specified
       with "y" and "n"), a double, a color (specified by the syntax R/G/B, R,
       G, and B being float numbers from 0.0 to 1.0) or by an
       "av_parse_color()" color description), a position (specified by the
       syntax X/Y, X and Y being float numbers) and a string.

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

       Examples

       ·   Apply the distort0r effect, set the first two double parameters:

		   frei0r=filter_name=distort0r:filter_params=0.5|0.01

       ·   Apply the colordistance effect, take a color as first parameter:

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

       ·   Apply the perspective effect, specify 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>

   geq
       The filter accepts the following options:

       lum_expr, lum
	   Set the luminance expression.

       cb_expr, cb
	   Set the chrominance blue expression.

       cr_expr, cr
	   Set the chrominance red expression.

       alpha_expr, a
	   Set the alpha expression.

       red_expr, r
	   Set the red expression.

       green_expr, g
	   Set the green expression.

       blue_expr, b
	   Set the blue expression.

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

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

       The expressions can use the following variables and functions:

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

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

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

       T   Time of the current frame, expressed in seconds.

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

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

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

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

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

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

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

       Examples

       ·   Flip the image horizontally:

		   geq=p(W-X\,Y)

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

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

       ·   Generate a fancy enigmatic moving light:

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

       ·   Generate a quick emboss effect:

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

       ·   Modify RGB components depending on pixel position:

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

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

       This filter 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.

       This filter accepts the following options:

       strength
	   The maximum amount by which the filter will change any one pixel.
	   Also the threshold for detecting nearly flat regions. Acceptable
	   values range from .51 to 64, 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, default
	   value is 16, out-of-range values will be clipped to the valid
	   range.

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

       Examples

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

		   gradfun=3.5:8

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

		   gradfun=radius=8

   haldclut
       Apply a Hald CLUT to a video stream.

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

       The filter accepts the following options:

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

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

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

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

       Workflow examples

       Hald CLUT video stream

       Generate an identity Hald CLUT stream altered with various effects:

	       ffmpeg -f lavfi -i 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 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 distribution of color
       components in an image.

       The filter accepts the following options:

       mode
	   Set histogram mode.

	   It accepts the following values:

	   levels
	       standard histogram that display color components distribution
	       in an image.  Displays color graph for each color component.
	       Shows distribution of the Y, U, V, A or G, B, R components,
	       depending on input format, in current frame. Bellow each graph
	       is color component scale meter.

	   color
	       chroma values in vectorscope, if brighter more such chroma
	       values are distributed in an image.  Displays chroma values
	       (U/V color placement) in two dimensional graph (which is called
	       a vectorscope). It can be used to read of the hue and
	       saturation of the current frame. At a same time it is a
	       histogram.  The whiter a pixel in the vectorscope, the more
	       pixels of the input frame correspond to that pixel (that is the
	       more pixels have this chroma value).  The V component is
	       displayed on the horizontal (X) axis, with the leftmost side
	       being V = 0 and the rightmost side being V = 255.  The U
	       component is displayed on the vertical (Y) axis, with the top
	       representing U = 0 and the bottom representing U = 255.

	       The position of a white pixel in the graph corresponds to the
	       chroma value of a pixel of the input clip. So the graph can be
	       used to read of the hue (color flavor) and the saturation (the
	       dominance of the hue in the color).  As the hue of a color
	       changes, it moves around the square. At the center of the
	       square, the saturation is zero, which means that the
	       corresponding pixel has no color. If you increase the amount of
	       a specific color, while leaving the other colors unchanged, the
	       saturation increases, and you move towards the edge of the
	       square.

	   color2
	       chroma values in vectorscope, similar as "color" but actual
	       chroma values are displayed.

	   waveform
	       per row/column color component graph. In row mode graph in the
	       left side represents color component value 0 and right side
	       represents value = 255. In column mode top side represents
	       color component value = 0 and bottom side represents value =
	       255.

	   Default value is "levels".

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

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

       step
	   Set step for "waveform" mode. Smaller values are useful to find out
	   how much of same luminance values across input rows/columns are
	   distributed.	 Default value is 10. Allowed range is [1, 255].

       waveform_mode
	   Set mode for "waveform". Can be either "row", or "column".  Default
	   is "row".

       display_mode
	   Set display mode for "waveform" and "levels".  It accepts the
	   following values:

	   parade
	       Display separate graph for the color components side by side in
	       "row" waveform mode or one below other in "column" waveform
	       mode for "waveform" histogram mode. For "levels" histogram mode
	       per color component graphs are placed one bellow other.

	       This display mode in "waveform" histogram 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 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 make by making
	       adjustments to level the three waveforms.

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

	       This display mode in "waveform" histogram mode can make it
	       easier to spot the relative differences or similarities in
	       overlapping areas of the color components that are supposed to
	       be identical, such as neutral whites, grays, or blacks.

	   Default is "parade".

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

       Examples

       ·   Calculate and draw histogram:

		   ffplay -i input -vf histogram

   hqdn3d
       High precision/quality 3d denoise filter. This filter 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 float number which specifies spatial luma strength,
	   defaults to 4.0

       chroma_spatial
	   a non-negative float number which specifies spatial chroma
	   strength, defaults to 3.0*luma_spatial/4.0

       luma_tmp
	   a float number which specifies luma temporal strength, defaults to
	   6.0*luma_spatial/4.0

       chroma_tmp
	   a float number which specifies chroma temporal strength, defaults
	   to luma_tmp*chroma_spatial/luma_spatial

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

       This filter accepts the following options:

       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".

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

       The 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:

       s
       h
       H   Modify the hue and/or the saturation 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.

   idet
       Detect video interlacing type.

       This filter tries to detect if the input is interlaced or progressive,
       top or bottom field first.

       The filter accepts the following options:

       intl_thres
	   Set interlacing threshold.

       prog_thres
	   Set progressive threshold.

   il
       Deinterleave or interleave fields.

       This filter allows 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.

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

       lowpass
	   Enable (default) or disable the vertical lowpass filter to avoid
	   twitter interlacing and reduce moire patterns.

   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

   lut3d
       Apply a 3D LUT to an input video.

       The filter accepts the following options:

       file
	   Set the 3D LUT file name.

	   Currently supported formats:

	   3dl AfterEffects

	   cube
	       Iridas

	   dat DaVinci

	   m3d Pandora

       interp
	   Select interpolation mode.

	   Available values are:

	   nearest
	       Use values from the nearest defined point.

	   trilinear
	       Interpolate values using the 8 points defining a cube.

	   tetrahedral
	       Interpolate values using a tetrahedron.

   lut, lutrgb, lutyuv
       Compute a look-up table for binding each pixel component input value to
       an output value, and apply it to input video.

       lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB
       input video.

       These filters accept the following options:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha
	   component

       r   set red component expression

       g   set green component expression

       b   set blue component expression

       a   alpha component expression

       y   set Y/luminance component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

       Each of them specifies the expression to use for computing the lookup
       table for the corresponding pixel component values.

       The exact component associated to each of the c* options depends on the
       format in input.

       The lut filter requires either YUV or RGB pixel formats in input,
       lutrgb requires RGB pixel formats in input, and lutyuv requires YUV.

       The expressions can contain the following constants and functions:

       w
       h   the input width and height

       val input value for the pixel component

       clipval
	   the input value clipped in the minval-maxval range

       maxval
	   maximum value for the pixel component

       minval
	   minimum value for the pixel component

       negval
	   the negated value for the pixel component value clipped in the
	   minval-maxval range , it corresponds to the expression
	   "maxval-clipval+minval"

       clip(val)
	   the computed value in val clipped in the minval-maxval range

       gammaval(gamma)
	   the computed gamma correction value of the pixel component value
	   clipped in the minval-maxval range, corresponds to the expression
	   "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "val".

       Examples

       ·   Negate input video:

		   lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
		   lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

	   The above is the same as:

		   lutrgb="r=negval:g=negval:b=negval"
		   lutyuv="y=negval:u=negval:v=negval"

       ·   Negate luminance:

		   lutyuv=y=negval

       ·   Remove chroma components, turns the video into a graytone image:

		   lutyuv="u=128:v=128"

       ·   Apply a luma burning effect:

		   lutyuv="y=2*val"

       ·   Remove green and blue components:

		   lutrgb="g=0:b=0"

       ·   Set a constant alpha channel value on input:

		   format=rgba,lutrgb=a="maxval-minval/2"

       ·   Correct luminance gamma by a 0.5 factor:

		   lutyuv=y=gammaval(0.5)

       ·   Discard least significant bits of luma:

		   lutyuv=y='bitand(val, 128+64+32)'

   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.

   mp
       Apply an MPlayer filter to the input video.

       This filter provides a wrapper around most of the filters of
       MPlayer/MEncoder.

       This wrapper is considered experimental. Some of the wrapped filters
       may not work properly and we may drop support for them, as they will be
       implemented natively into FFmpeg. Thus you should avoid depending on
       them when writing portable scripts.

       The filters accepts the parameters: filter_name[:=]filter_params

       filter_name is the name of a supported MPlayer filter, filter_params is
       a string containing the parameters accepted by the named filter.

       The list of the currently supported filters follows:

       dint
       eq2
       eq
       fil
       fspp
       ilpack
       perspective
       phase
       pp7
       pullup
       qp
       softpulldown
       uspp

       The parameter syntax and behavior for the listed filters are the same
       of the corresponding MPlayer filters. For detailed instructions check
       the "VIDEO FILTERS" section in the MPlayer manual.

       Examples

       ·   Adjust gamma, brightness, contrast:

		   mp=eq2=1.0:2:0.5

       See also mplayer(1), <http://www.mplayerhq.hu/>.

   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 unregarding the
	   number of previous sequentially dropped frames.

	   Default value is 0.

       hi
       lo
       frac
	   Set the dropping threshold values.

	   Values for hi and lo are for 8x8 pixel blocks and represent actual
	   pixel value differences, so a threshold of 64 corresponds to 1 unit
	   of difference for each pixel, or the same spread out differently
	   over the block.

	   A frame is a candidate for dropping if no 8x8 blocks differ by more
	   than a threshold of hi, and if no more than frac blocks (1 meaning
	   the whole image) differ by more than a threshold of lo.

	   Default value for hi is 64*12, default value for lo is 64*5, and
	   default value for frac is 0.33.

   negate
       Negate input video.

       This filter accepts an integer in input, if non-zero it negates the
       alpha component (if available). The default value in input is 0.

   noformat
       Force libavfilter not to use any of the specified pixel formats for the
       input to the next filter.

       This filter accepts the following parameters:

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

       Examples

       ·   Force libavfilter to use a format different from yuv420p for the
	   input to the vflip filter:

		   noformat=pix_fmts=yuv420p,vflip

       ·   Convert the input video to any of the formats not contained in the
	   list:

		   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
	   Set noise seed for specific pixel component or all pixel components
	   in case of all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
	   Set noise strength for specific pixel component or all pixel
	   components in case all_strength. Default value is 0. Allowed range
	   is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
	   Set pixel component flags or set flags for all components if
	   all_flags.  Available values for component flags are:

	   a   averaged temporal noise (smoother)

	   p   mix random noise with a (semi)regular pattern

	   t   temporal noise (noise pattern changes between frames)

	   u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

	       noise=alls=20:allf=t+u

   null
       Pass the video source unchanged to the output.

   ocv
       Apply video transform using libopencv.

       To enable this filter install libopencv library and headers and
       configure FFmpeg with "--enable-libopencv".

       This filter 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://opencv.willowgarage.com/documentation/c/image_filtering.html>

       Follows the list of supported libopencv filters.

       dilate

       Dilate an image by using a specific structuring element.	 This filter
       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, and can be one of the values
       "rect", "cross", "ellipse", "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.

       Follow some example:

	       # use the default values
	       ocv=dilate

	       # dilate using a structuring element with a 5x5 cross, iterate two times
	       ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

	       # read the shape from the file diamond.shape, iterate two times
	       # the file diamond.shape may contain a pattern of characters like this:
	       #   *
	       #  ***
	       # *****
	       #  ***
	       #   *
	       # the specified cols and rows are ignored (but not the anchor point coordinates)
	       ocv=dilate:0x0+2x2/custom=diamond.shape|2

       erode

       Erode an image by using a specific structuring element.	This filter
       corresponds to the libopencv function "cvErode".

       The filter 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 can be one of the
       following values: "blur", "blur_no_scale", "median", "gaussian",
       "bilateral". The default value is "gaussian".

       param1, param2, param3, and param4 are parameters whose meanings depend
       on smooth type. param1 and param2 accept integer positive values or 0,
       param3 and param4 accept float 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".

   overlay
       Overlay one video on top of another.

       It takes two inputs and one output, the first input is the "main" video
       on which the second input is overlayed.

       This filter accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and y coordinates of the overlayed
	   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).

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

       format
	   Set the format for the output video.

	   It accepts the following values:

	   yuv420
	       force YUV420 output

	   yuv444
	       force YUV444 output

	   rgb force RGB output

	   Default value is yuv420.

       rgb (deprecated)
	   If set to 1, force the filter to accept inputs in the RGB color
	   space. Default value is 0. This option is deprecated, use format
	   instead.

       repeatlast
	   If set to 1, force the filter to draw the last overlay frame over
	   the main input until the end of the stream. A value of 0 disables
	   this behavior. Default value is 1.

       The x, and y expressions can contain the following parameters.

       main_w, W
       main_h, H
	   main input width and height

       overlay_w, w
       overlay_h, h
	   overlay input width and height

       x
       y   the computed values for x and y. They are evaluated for each new
	   frame.

       hsub
       vsub
	   horizontal and vertical chroma subsample values of the output
	   format. For example for the pixel format "yuv422p" hsub is 2 and
	   vsub is 1.

       n   the number of input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

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

       Note that the n, pos, t variables are available only when evaluation is
       done per frame, and will evaluate to NAN when eval is set to init.

       Be aware that frames are taken from each input video in timestamp
       order, hence, if their initial timestamps differ, it is a a good idea
       to pass the two inputs through a setpts=PTS-STARTPTS filter to have
       them begin in the same zero timestamp, as it does the example for the
       movie filter.

       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]
		   "

       ·   Chain several overlays in cascade:

		   nullsrc=s=200x200 [bg];
		   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
		   [in0] lutrgb=r=0, [bg]   overlay=0:0	    [mid0];
		   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
		   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
		   [in3] null,	     [mid2] overlay=100:100 [out0]

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
	   Set depth.

	   Larger depth values will denoise lower frequency components more,
	   but slow down filtering.

	   Must be an int in the range 8-16, default is 8.

       luma_strength, ls
	   Set luma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
	   Set chroma strength.

	   Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add paddings to the input image, and place the original input at the
       given coordinates x, y.

       This filter 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 an expression for the offsets where to place the input
	   image in 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.

       color
	   Specify the color of the padded area, it can be the name of a color
	   (case insensitive match) or a 0xRRGGBB[AA] sequence.

	   The default value of color is "black".

       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  same as in_w and in_h

       out_w
       out_h
	   the output width and height, that is the size of the padded area as
	   specified by the width and height expressions

       ow
       oh  same as out_w and out_h

       x
       y   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
	   horizontal and vertical chroma subsample values. For example for
	   the pixel format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   Add paddings with color "violet" to the input video. Output video
	   size is 640x480, 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 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"

   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.

   pp
       Enable the specified chain of postprocessing subfilters using
       libpostproc. This library should be automatically selected with a GPL
       build ("--enable-gpl").	Subfilters must be separated by '/' and can be
       disabled by prepending a '-'.  Each subfilter and some options have a
       short and a long name that can be used interchangeably, i.e. dr/dering
       are the same.

       The filters accept the following options:

       subfilters
	   Set postprocessing subfilters string.

       All subfilters share common options to determine their scope:

       a/autoq
	   Honor the quality commands for this subfilter.

       c/chrom
	   Do chrominance filtering, too (default).

       y/nochrom
	   Do luminance filtering only (no chrominance).

       n/noluma
	   Do chrominance filtering only (no luminance).

       These options can be appended after the subfilter name, separated by a
       '|'.

       Available subfilters are:

       hb/hdeblock[|difference[|flatness]]
	   Horizontal deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       vb/vdeblock[|difference[|flatness]]
	   Vertical deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       ha/hadeblock[|difference[|flatness]]
	   Accurate horizontal deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       va/vadeblock[|difference[|flatness]]
	   Accurate vertical deblocking filter

	   difference
	       Difference factor where higher values mean more deblocking
	       (default: 32).

	   flatness
	       Flatness threshold where lower values mean more deblocking
	       (default: 39).

       The horizontal and vertical deblocking filters share the difference and
       flatness values so you cannot set different horizontal and vertical
       thresholds.

       h1/x1hdeblock
	   Experimental horizontal deblocking filter

       v1/x1vdeblock
	   Experimental vertical deblocking filter

       dr/dering
	   Deringing filter

       tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise
       reducer
	   threshold1
	       larger -> stronger filtering

	   threshold2
	       larger -> stronger filtering

	   threshold3
	       larger -> stronger filtering

       al/autolevels[:f/fullyrange], automatic brightness / contrast
       correction
	   f/fullyrange
	       Stretch luminance to "0-255".

       lb/linblenddeint
	   Linear blend deinterlacing filter that deinterlaces the given block
	   by filtering all lines with a "(1 2 1)" filter.

       li/linipoldeint
	   Linear interpolating deinterlacing filter that deinterlaces the
	   given block by linearly interpolating every second line.

       ci/cubicipoldeint
	   Cubic interpolating deinterlacing filter deinterlaces the given
	   block by cubically interpolating every second line.

       md/mediandeint
	   Median deinterlacing filter that deinterlaces the given block by
	   applying a median filter to every second line.

       fd/ffmpegdeint
	   FFmpeg deinterlacing filter that deinterlaces the given block by
	   filtering every second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
	   Vertically applied FIR lowpass deinterlacing filter that
	   deinterlaces the given block by filtering all lines with a "(-1 2 6
	   2 -1)" filter.

       fq/forceQuant[|quantizer]
	   Overrides the quantizer table from the input with the constant
	   quantizer you specify.

	   quantizer
	       Quantizer to use

       de/default
	   Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
	   Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       ·   Apply horizontal and vertical deblocking, deringing and automatic
	   brightness/contrast:

		   pp=hb/vb/dr/al

       ·   Apply default filters without brightness/contrast correction:

		   pp=de/-al

       ·   Apply default filters and temporal denoiser:

		   pp=default/tmpnoise|1|2|3

       ·   Apply deblocking on luminance only, and switch vertical deblocking
	   on or off automatically depending on available CPU time:

		   pp=hb|y/vb|a

   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.

       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

       mse_avg
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames, averaged over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_g, mse_a
	   Mean Square Error pixel-by-pixel average difference of the compared
	   frames for the component specified by the suffix.

       psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
	   Peak Signal to Noise ratio of the compared frames for the component
	   specified by the suffix.

       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.

   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.

   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. 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 heigth

       out_w, ow
       out_h, oh
	   the output width and heigth, 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

       ·   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 choosen 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 0.1-4.0.
	   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 0.1-2.0 range.

       chroma_strength, cs
	   Set chroma maximum difference between pixels to still be
	   considered, must be a value in the 0.1-100.0 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.

       The filter accepts the following options:

       width, w
	   Set the output video width expression. Default value is "iw". See
	   below for the list of accepted constants.

       height, h
	   Set the output video height expression. Default value is "ih".  See
	   below for the list of accepted constants.

       interl
	   Set the interlacing. 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. If not explictly specified the filter
	   applies a bilinear scaling algorithm.

       size, s
	   Set the video size, the value must be a valid abbreviation or in
	   the form widthxheight.

       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  same as in_w and in_h

       out_w
       out_h
	   the output (cropped) width and height

       ow
       oh  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.

       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.

       If the value for w or h is 0, the respective input size is used for the
       output.

       If the value for w or h is -1, the scale filter will use, for the
       respective output size, a value that maintains the aspect ratio of the
       input image.

       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 for 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, but make 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, keep the same input
	   aspect ratio:

		   scale=w='min(500\, iw*3/2):h=-1'

   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.

       The filters accept the following options:

       r, ratio, dar ("setdar" only), sar ("setsar" only)
	   Set the aspect ratio used by the filter.

	   The parameter can be a floating point number string, an expression,
	   or a string of the form num:den, where num and den are the
	   numerator and denominator of the aspect ratio. If the parameter is
	   not specified, it is assumed the value "0".	In case the form
	   "num:den" is used, the ":" character should be escaped.

       max Set the maximum integer value to use for expressing numerator and
	   denominator when reducing the expressed aspect ratio to a rational.
	   Default value is 100.

       Examples

       ·   To change the display aspect ratio to 16:9, specify one of the
	   following:

		   setdar=dar=1.77777
		   setdar=dar=16/9
		   setdar=dar=1.77777

       ·   To change the sample aspect ratio to 10:11, specify:

		   setsar=sar=10/11

       ·   To set a display aspect ratio of 16:9, and specify a maximum
	   integer value of 1000 in the aspect ratio reduction, use the
	   command:

		   setdar=ratio=16/9:max=1000

   setfield
       Force field for the output video frame.

       The "setfield" filter marks the interlace type field for the output
       frames. It does not change the input frame, but only sets the
       corresponding property, which affects how the frame is treated by
       following filters (e.g. "fieldorder" or "yadif").

       The filter accepts the following options:

       mode
	   Available values are:

	   auto
	       Keep the same field property.

	   bff Mark the frame as bottom-field-first.

	   tff Mark the frame as top-field-first.

	   prog
	       Mark the frame as progressive.

   showinfo
       Show a line containing various information for each input video frame.
       The input video is not modified.

       The shown line contains a sequence of key/value pairs of the form
       key:value.

       A description of each shown parameter follows:

       n   sequential number of the input frame, starting from 0

       pts 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
	   Presentation TimeStamp of the input frame, expressed as a number of
	   seconds

       pos position of the frame in the input stream, -1 if this information
	   in unavailable and/or meaningless (for example in case of synthetic
	   video)

       fmt pixel format name

       sar sample aspect ratio of the input frame, expressed in the form
	   num/den

       s   size of the input frame, expressed in the form widthxheight

       i   interlaced mode ("P" for "progressive", "T" for top field first,
	   "B" for bottom field first)

       iskey
	   1 if the frame is a key frame, 0 otherwise

       type
	   picture type of the input frame ("I" for an I-frame, "P" for a
	   P-frame, "B" for a B-frame, "?" for unknown type).  Check also the
	   documentation of the "AVPictureType" enum and of the
	   "av_get_picture_type_char" function defined in libavutil/avutil.h.

       checksum
	   Adler-32 checksum (printed in hexadecimal) of all the planes of the
	   input frame

       plane_checksum
	   Adler-32 checksum (printed in hexadecimal) of each plane of the
	   input frame, expressed in the form "[c0 c1 c2 c3]"

   smartblur
       Blur the input video without impacting the outlines.

       The filter 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
	   1.0.

       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 1.0.

       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 0.

       If a chroma option is not explicitly set, the corresponding luma value
       is set.

   stereo3d
       Convert between different stereoscopic image formats.

       The filters accept the following options:

       in  Set stereoscopic image format of input.

	   Available values for input image formats are:

	   sbsl
	       side by side parallel (left eye left, right eye right)

	   sbsr
	       side by side crosseye (right eye left, left eye right)

	   sbs2l
	       side by side parallel with half width resolution (left eye
	       left, right eye right)

	   sbs2r
	       side by side crosseye with half width resolution (right eye
	       left, left eye right)

	   abl above-below (left eye above, right eye below)

	   abr above-below (right eye above, left eye below)

	   ab2l
	       above-below with half height resolution (left eye above, right
	       eye below)

	   ab2r
	       above-below with half height resolution (right eye above, left
	       eye below)

	   al  alternating frames (left eye first, right eye second)

	   ar  alternating frames (right eye first, left eye second)

	       Default value is sbsl.

       out Set stereoscopic image format of output.

	   Available values for output image formats are all the input formats
	   as well as:

	   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)

	   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)

	   ml  mono output (left eye only)

	   mr  mono output (right eye only)

	   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 bellow (left eye above, right eye
	   below) to side by side crosseye.

		   stereo3d=abl:sbsr

   spp
       Apply a simple postprocessing filter that compresses and decompresses
       the image at several (or - in the case of quality level 6 - all) shifts
       and average the results.

       The filter accepts the following options:

       quality
	   Set quality. This option defines the number of levels for
	   averaging. It accepts an integer in the range 0-6. If set to 0, the
	   filter will have no effect. A value of 6 means the higher quality.
	   For each increment of that value the speed drops by a factor of
	   approximately 2.  Default value is 3.

       qp  Force a constant quantization parameter. If not set, the filter
	   will use the QP from the video stream (if available).

       mode
	   Set thresholding mode. Available modes are:

	   hard
	       Set hard thresholding (default).

	   soft
	       Set soft thresholding (better de-ringing effect, but likely
	       blurrier).

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

   subtitles
       Draw subtitles on top of input video using the libass library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libass". This filter also requires a build with libavcodec
       and libavformat to convert the passed subtitles file to ASS (Advanced
       Substation Alpha) subtitles format.

       The filter accepts the following options:

       filename, f
	   Set the filename of the subtitle file to read. It must be
	   specified.

       original_size
	   Specify the size of the original video, the video for which the ASS
	   file was composed. Due to a misdesign in ASS aspect ratio
	   arithmetic, this is necessary to correctly scale the fonts if the
	   aspect ratio has been changed.

       charenc
	   Set subtitles input character encoding. "subtitles" filter only.
	   Only useful if not UTF-8.

       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

   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.

   swapuv
       Swap U & V plane.

   telecine
       Apply telecine process to the video.

       This filter accepts the following options:

       first_field
	   top, t
	       top field first

	   bottom, b
	       bottom field first The default value is "top".

       pattern
	   A string of numbers representing the pulldown pattern you wish to
	   apply.  The default value is 23.

	       Some typical patterns:

	       NTSC output (30i):
	       27.5p: 32222
	       24p: 23 (classic)
	       24p: 2332 (preferred)
	       20p: 33
	       18p: 334
	       16p: 3444

	       PAL output (25i):
	       27.5p: 12222
	       24p: 222222222223 ("Euro pulldown")
	       16.67p: 33
	       16p: 33333334

   thumbnail
       Select the most representative frame in a given sequence of consecutive
       frames.

       The filter accepts the following options:

       n   Set the frames batch size to analyze; in a set of n frames, the
	   filter will pick one of them, and then handle the next batch of n
	   frames until the end. Default is 100.

       Since the filter keeps track of the whole frames sequence, a bigger n
       value will result in a higher memory usage, so a high value is not
       recommended.

       Examples

       ·   Extract one picture each 50 frames:

		   thumbnail=50

       ·   Complete example of a thumbnail creation with ffmpeg:

		   ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png

   tile
       Tile several successive frames together.

       The filter accepts the following options:

       layout
	   Set the grid size (i.e. the number of lines and columns) in the
	   form "wxh".

       nb_frames
	   Set the maximum number of frames to render in the given area. It
	   must be less than or equal to wxh. The default value is 0, meaning
	   all the area will be used.

       margin
	   Set the outer border margin in pixels.

       padding
	   Set the inner border thickness (i.e. the number of pixels between
	   frames). For more advanced padding options (such as having
	   different values for the edges), refer to the pad video filter.

       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 accomodate 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.

	   drop_odd, 1
	       Only output even frames, odd frames are dropped, generating a
	       frame with unchanged height at half frame rate.

	   drop_even, 2
	       Only output odd frames, even frames are dropped, generating a
	       frame with unchanged height at half frame rate.

	   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.

	   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.

	   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.

	   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.

	   Numeric values are deprecated but are accepted for backward
	   compatibility reasons.

	   Default mode is "merge".

       flags
	   Specify flags influencing the filter process.

	   Available value for flags is:

	   low_pass_filter, vlfp
	       Enable 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.

	       Vertical low-pass filtering can only be enabled for mode
	       interleave_top and interleave_bottom.

   transpose
       Transpose rows with columns in the input video and optionally flip it.

       This filter accepts the following options:

       dir Specify the transposition direction.

	   Can assume the following values:

	   0, 4, cclock_flip
	       Rotate by 90 degrees counterclockwise and vertically flip
	       (default), that is:

		       L.R     L.l
		       . . ->  . .
		       l.r     R.r

	   1, 5, clock
	       Rotate by 90 degrees clockwise, that is:

		       L.R     l.L
		       . . ->  . .
		       l.r     r.R

	   2, 6, cclock
	       Rotate by 90 degrees counterclockwise, that is:

		       L.R     R.r
		       . . ->  . .
		       l.r     L.l

	   3, 7, clock_flip
	       Rotate by 90 degrees clockwise and vertically flip, that is:

		       L.R     r.R
		       . . ->  . .
		       l.r     l.L

	   For values between 4-7, the transposition is only done if the input
	   video geometry is portrait and not landscape. These values are
	   deprecated, the "passthrough" option should be used instead.

	   Numerical values are deprecated, and should be dropped in favor of
	   symbolic constants.

       passthrough
	   Do not apply the transposition if the input geometry matches the
	   one specified by the specified value. It accepts the following
	   values:

	   none
	       Always apply transposition.

	   portrait
	       Preserve portrait geometry (when height >= width).

	   landscape
	       Preserve landscape geometry (when width >= height).

	   Default value is "none".

       For example to rotate by 90 degrees clockwise and preserve portrait
       layout:

	       transpose=dir=1:passthrough=portrait

       The command above can also be specified as:

	       transpose=1:portrait

   trim
       Trim the input so that the output contains one continuous subpart of
       the input.

       This filter accepts the following options:

       start
	   Timestamp (in seconds) of the start of the kept section. I.e. the
	   frame with the timestamp start will be the first frame in the
	   output.

       end Timestamp (in seconds) 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
	   Same as start, except this option sets the start timestamp in
	   timebase units instead of seconds.

       end_pts
	   Same as end, except this option sets the end timestamp in timebase
	   units instead of seconds.

       duration
	   Maximum duration of the output in seconds.

       start_frame
	   Number of the first frame that should be passed to output.

       end_frame
	   Number of the first frame that should be dropped.

       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 that the output
       timestamps 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

   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 63, default value is 5.

       luma_msize_y, ly
	   Set the luma matrix vertical size. It must be an odd integer
	   between 3 and 63, default value is 5.

       luma_amount, la
	   Set the luma effect strength. It can be a float 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 63, default value is 5.

       chroma_msize_y, cy
	   Set the chroma matrix vertical size. It must be an odd integer
	   between 3 and 63, default value is 5.

       chroma_amount, ca
	   Set the chroma effect strength. It can be a float number,
	   reasonable values lay between -1.5 and 1.5.

	   Negative values will blur the input video, while positive values
	   will sharpen it, a value of zero will disable the effect.

	   Default value is 0.0.

       opencl
	   If set to 1, specify using OpenCL capabilities, only available if
	   FFmpeg was configured with "--enable-opencl". Default value is 0.

       All parameters are optional and default to the equivalent of the string
       '5:5:1.0:5:5:0.0'.

       Examples

       ·   Apply strong luma sharpen effect:

		   unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5

       ·   Apply strong blur of both luma and chroma parameters:

		   unsharp=7:7:-2:7:7:-2

   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 9.

       stepsize
	   Set stepsize of the search process. The region around minimum is
	   scanned with 1 pixel resolution. Default value is 6.

       mincontrast
	   Set minimum contrast. Below this value a local measurement field is
	   discarded. Must be a floating point value in the range 0-1. Default
	   value is 0.3.

       tripod
	   Set reference frame number for tripod mode.

	   If enabled, the motion of the frames is compared to a reference
	   frame in the filtered stream, identified by the specified number.
	   The idea is to compensate all movements in a more-or-less static
	   scene and keep the camera view absolutely still.

	   If set to 0, it is disabled. The frames are counted starting from
	   1.

       show
	   Show fields and transforms in the resulting frames. It accepts an
	   integer in the range 0-2. Default value is 0, which disables any
	   visualization.

       Examples

       ·   Use default values:

		   vidstabdetect

       ·   Analyze strongly shaky movie and put the results in file
	   mytransforms.trf:

		   vidstabdetect=shakiness=10:accuracy=15:result="mytransforms.trf"

       ·   Visualize the result of internal transformations in the resulting
	   video:

		   vidstabdetect=show=1

       ·   Analyze a video with medium shakiness using ffmpeg:

		   ffmpeg -i input -vf vidstabdetect=shakiness=5:show=1 dummy.avi

   vidstabtransform
       Video stabilization/deshaking: pass 2 of 2, see vidstabdetect for pass
       1.

       Read a file with transform information for each frame and
       apply/compensate them. Together with the vidstabdetect filter this can
       be used to deshake videos. See also
       <http://public.hronopik.de/vid.stab>. It is important to also use the
       unsharp filter, see below.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libvidstab".

       This filter accepts the following options:

       input
	   path to the file used to read the transforms (default:
	   transforms.trf)

       smoothing
	   number of frames (value*2 + 1) used for lowpass filtering the
	   camera movements (default: 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 values leads to a
	   smoother video, but limits the acceleration of the camera (pan/tilt
	   movements).

       maxshift
	   maximal number of pixels to translate frames (default: -1 no limit)

       maxangle
	   maximal angle in radians (degree*PI/180) to rotate frames (default:
	   -1 no limit)

       crop
	   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
	   0
		keep transforms normal (default)

	   1
		invert transforms

       relative
	   consider transforms as

	   0
		absolute

	   1
		relative to previous frame (default)

       zoom
	   percentage to zoom (default: 0)

	   >0
		 zoom in

	   <0
		 zoom out

       optzoom
	   if 1 then optimal zoom value is determined (default).  Optimal zoom
	   means no (or only little) border should be visible.	Note that the
	   value given at zoom is added to the one calculated here.

       interpol
	   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
	   virtual tripod mode means that the video is stabilized such that
	   the camera stays stationary. Use also "tripod" option of
	   vidstabdetect.

	   0   off (default)

	   1   virtual tripod mode: equivalent to "relative=0:smoothing=0"

       Examples

       ·   typical call with default default values:
	    (note the unsharp filter which is always recommended)

		   ffmpeg -i inp.mpeg -vf vidstabtransform,unsharp=5:5:0.8:3:3:0.4 inp_stabilized.mpeg

       ·   zoom in a bit more and load transform data from a given file

		   vidstabtransform=zoom=5:input="mytransforms.trf"

       ·   smoothen the video even more

		   vidstabtransform=smoothing=30

   vflip
       Flip the input video vertically.

       For example, to vertically flip a video with ffmpeg:

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

   vignette
       Make or reverse a natural vignetting effect.

       The filter accepts the following options:

       angle, a
	   Set lens angle expression as a number of radians.

	   The value is clipped in the "[0,PI/2]" range.

	   Default value: "PI/5"

       x0
       y0  Set center coordinates expressions. Respectively "w/2" and "h/2" by
	   default.

       mode
	   Set forward/backward mode.

	   Available modes are:

	   forward
	       The larger the distance from the central point, the darker the
	       image becomes.

	   backward
	       The larger the distance from the central point, the brighter
	       the image becomes.  This can be used to reverse a vignette
	       effect, though there is no automatic detection to extract the
	       lens angle and other settings (yet). It can also be used to
	       create a burning effect.

	   Default value is forward.

       eval
	   Set evaluation mode for the expressions (angle, x0, y0).

	   It accepts the following values:

	   init
	       Evaluate expressions only once during the filter
	       initialization.

	   frame
	       Evaluate expressions for each incoming frame. This is way
	       slower than the init mode since it requires all the scalers to
	       be re-computed, but it allows advanced dynamic expressions.

	   Default value is init.

       dither
	   Set dithering to reduce the circular banding effects. Default is 1
	   (enabled).

       aspect
	   Set vignette aspect. This setting allows 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

   yadif
       Deinterlace the input video ("yadif" means "yet another deinterlacing
       filter").

       This filter accepts the following options:

       mode
	   The interlacing mode to adopt, accepts one of the following values:

	   0, send_frame
	       output 1 frame for each frame

	   1, send_field
	       output 1 frame for each field

	   2, send_frame_nospatial
	       like "send_frame" but skip spatial interlacing check

	   3, send_field_nospatial
	       like "send_field" but skip spatial interlacing check

	   Default value is "send_frame".

       parity
	   The picture field parity assumed for the input interlaced video,
	   accepts one of the following values:

	   0, tff
	       assume top field first

	   1, bff
	       assume bottom field first

	   -1, auto
	       enable automatic detection

	   Default value is "auto".  If interlacing is unknown or decoder does
	   not export this information, top field first will be assumed.

       deint
	   Specify which frames to deinterlace. Accept one of the following
	   values:

	   0, all
	       deinterlace all frames

	   1, interlaced
	       only deinterlace frames marked as interlaced

	   Default value is "all".

VIDEO SOURCES
       Below is a description of the currently available video sources.

   buffer
       Buffer video frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular
       through the interface defined in libavfilter/vsrc_buffer.h.

       This source accepts the following options:

       video_size
	   Specify the size (width and height) of the buffered video frames.

       width
	   Input video width.

       height
	   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
	   Specify the sample aspect ratio assumed by the video frames.

       sws_param
	   Specify the optional parameters to be used for the scale filter
	   which is automatically inserted when an input change is detected in
	   the input size or format.

       For example:

	       buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1

       will instruct the source to accept video frames with size 320x240 and
       with format "yuv410p", assuming 1/24 as the timestamps timebase and
       square pixels (1:1 sample aspect ratio).	 Since the pixel format with
       name "yuv410p" corresponds to the number 6 (check the enum
       AVPixelFormat definition in libavutil/pixfmt.h), this example
       corresponds to:

	       buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1

       Alternatively, the options can be specified as a flat string, but this
       syntax is deprecated:

       width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]

   cellauto
       Create a pattern generated by an elementary cellular automaton.

       The initial state of the cellular automaton can be defined through the
       filename, and pattern options. If such options are not specified an
       initial state is created randomly.

       At each new frame a new row in the video is filled with the result of
       the cellular automaton next generation. The behavior when the whole
       frame is filled is defined by the scroll option.

       This source accepts the following options:

       filename, f
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified file.  In the file, each non-whitespace
	   character is considered an alive cell, a newline will terminate the
	   row, and further characters in the file will be ignored.

       pattern, p
	   Read the initial cellular automaton state, i.e. the starting row,
	   from the specified string.

	   Each non-whitespace character in the string is considered an alive
	   cell, a newline will terminate the row, and further characters in
	   the string will be ignored.

       rate, r
	   Set the video rate, that is the number of frames generated per
	   second.  Default is 25.

       random_fill_ratio, ratio
	   Set the random fill ratio for the initial cellular automaton row.
	   It is a floating point number value ranging from 0 to 1, defaults
	   to 1/PHI.

	   This option is ignored when a file or a pattern is specified.

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

       rule
	   Set the cellular automaton rule, it is a number ranging from 0 to
	   255.	 Default value is 110.

       size, s
	   Set the size of the output video.

	   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

   mandelbrot
       Generate a Mandelbrot set fractal, and progressively zoom towards the
       point specified with start_x and start_y.

       This source accepts the following options:

       end_pts
	   Set the terminal pts value. Default value is 400.

       end_scale
	   Set the terminal scale value.  Must be a floating point value.
	   Default value is 0.3.

       inner
	   Set the inner coloring mode, that is the algorithm used to draw the
	   Mandelbrot fractal internal region.

	   It shall assume one of the following values:

	   black
	       Set black mode.

	   convergence
	       Show time until convergence.

	   mincol
	       Set color based on point closest to the origin of the
	       iterations.

	   period
	       Set period mode.

	   Default value is mincol.

       bailout
	   Set the bailout value. Default value is 10.0.

       maxiter
	   Set the maximum of iterations performed by the rendering algorithm.
	   Default value is 7189.

       outer
	   Set outer coloring mode.  It shall assume one of following values:

	   iteration_count
	       Set iteration cound mode.

	   normalized_iteration_count
	       set normalized iteration count mode.

	   Default value is normalized_iteration_count.

       rate, r
	   Set frame rate, expressed as number of frames per second. Default
	   value is "25".

       size, s
	   Set frame size. 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 float number or
	   a valid video frame rate abbreviation. The default value is "25".

       duration, d
	   Set the video duration of the sourced video. The accepted syntax
	   is:

		   [-]HH:MM:SS[.m...]
		   [-]S+[.m...]

	   See also the function "av_parse_time()".

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       test, t
	   Set the number or the name of the test to perform. Supported tests
	   are:

	   dc_luma
	   dc_chroma
	   freq_luma
	   freq_chroma
	   amp_luma
	   amp_chroma
	   cbp
	   mv
	   ring1
	   ring2
	   all

	   Default value is "all", which will cycle through the list of all
	   tests.

       For example the following:

	       testsrc=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 options:

       size
	   The size of the video to generate, may be a string of the form
	   widthxheight or a frame size abbreviation.

       framerate
	   Framerate of the generated video, 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 section
	   frei0r in the description of the video filters.

       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 overlayed 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 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.

	   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.

       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

   color, haldclutsrc, nullsrc, rgbtestsrc, smptebars, smptehdbars, testsrc
       The "color" source provides an uniformly colored input.

       The "haldclutsrc" source provides an identity Hald CLUT. See also
       haldclut filter.

       The "nullsrc" source returns unprocessed video frames. It is mainly
       useful to be employed in analysis / debugging tools, or as the source
       for filters which ignore the input data.

       The "rgbtestsrc" source generates an RGB test pattern useful for
       detecting RGB vs BGR issues. You should see a red, green and blue
       stripe from top to bottom.

       The "smptebars" source generates a color bars pattern, based on the
       SMPTE Engineering Guideline EG 1-1990.

       The "smptehdbars" source generates a color bars pattern, based on the
       SMPTE RP 219-2002.

       The "testsrc" source generates a test video pattern, showing a color
       pattern, a scrolling gradient and a timestamp. This is mainly intended
       for testing purposes.

       The sources accept the following options:

       color, c
	   Specify the color of the source, only available in the "color"
	   source. It can be the name of a color (case insensitive match) or a
	   0xRRGGBB[AA] sequence, possibly followed by an alpha specifier. The
	   default value is "black".

       level
	   Specify the level of the Hald CLUT, only available in the
	   "haldclutsrc" source. A level of "N" generates a picture of "N*N*N"
	   by "N*N*N" pixels to be used as identity matrix for 3D lookup
	   tables. Each component is coded on a "1/(N*N)" scale.

       size, s
	   Specify the size of the sourced video, it may be a string of the
	   form widthxheight, or the name of a size abbreviation. The default
	   value is "320x240".

	   This option is not available with the "haldclutsrc" filter.

       rate, r
	   Specify the frame rate of the sourced video, as the number of
	   frames generated per second. It has to be a string in the format
	   frame_rate_num/frame_rate_den, an integer number, a float 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 video duration of the sourced video. The accepted syntax
	   is:

		   [-]HH[:MM[:SS[.m...]]]
		   [-]S+[.m...]

	   See also the function "av_parse_time()".

	   If not specified, or the expressed duration is negative, the video
	   is supposed to be generated forever.

       decimals, n
	   Set the number of decimals to show in the timestamp, only available
	   in the "testsrc" source.

	   The displayed timestamp value will correspond to the original
	   timestamp value multiplied by the power of 10 of the specified
	   value. Default value is 0.

       For example the following:

	       testsrc=duration=5.3:size=qcif:rate=10

       will generate a video with a duration of 5.3 seconds, with size 176x144
       and a frame rate of 10 frames per second.

       The following graph description will generate a red source with an
       opacity of 0.2, with size "qcif" and a frame rate of 10 frames per
       second.

	       color=c=red@0.2:s=qcif:r=10

       If the input content is to be ignored, "nullsrc" can be used. The
       following command generates noise in the luminance plane by employing
       the "geq" filter:

	       nullsrc=s=256x256, geq=random(1)*255:128:128

       Commands

       The "color" source supports the following commands:

       c, color
	   Set the color of the created image. Accepts the same syntax of the
	   corresponding color option.

VIDEO SINKS
       Below is a description of the currently available video sinks.

   buffersink
       Buffer video frames, and make them available to the end of the filter
       graph.

       This sink is mainly intended for a 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 to be employed in analysis / debugging
       tools.

MULTIMEDIA FILTERS
       Below is a description of the currently available multimedia filters.

   avectorscope
       Convert input audio to a video output, representing the audio vector
       scope.

       The filter is used to measure the difference between channels of stereo
       audio stream. A monoaural signal, consisting of identical left and
       right signal, results in straight vertical line. Any stereo separation
       is visible as a deviation from this line, creating a Lissajous figure.
       If the straight (or deviation from it) but horizontal line appears this
       indicates that the left and right channels are out of phase.

       The filter accepts the following options:

       mode, m
	   Set the vectorscope mode.

	   Available values are:

	   lissajous
	       Lissajous rotated by 45 degrees.

	   lissajous_xy
	       Same as above but not rotated.

	   Default value is lissajous.

       size, s
	   Set the video size for the output. Default value is "400x400".

       rate, r
	   Set the output frame rate. Default value is 25.

       rc
       gc
       bc  Specify the red, green and blue contrast. Default values are 40,
	   160 and 80.	Allowed range is "[0, 255]".

       rf
       gf
       bf  Specify the red, green and blue fade. Default values are 15, 10 and
	   5.  Allowed range is "[0, 255]".

       zoom
	   Set the zoom factor. Default value is 1. Allowed range is "[1,
	   10]".

       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]'

   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
	   video 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.

   ebur128
       EBU R128 scanner filter. This filter takes an audio stream as input and
       outputs it unchanged. By default, it logs a message at a frequency of
       10Hz with the Momentary loudness (identified by "M"), Short-term
       loudness ("S"), Integrated loudness ("I") and Loudness Range ("LRA").

       The filter also has a video output (see the video option) with a real
       time graph to observe the loudness evolution. The graphic contains the
       logged message mentioned above, so it is not printed anymore when this
       option is set, unless the verbose logging is set. The main graphing
       area contains the short-term loudness (3 seconds of analysis), and the
       gauge on the right is for the momentary loudness (400 milliseconds).

       More information about the Loudness Recommendation EBU R128 on
       <http://tech.ebu.ch/loudness>.

       The filter accepts the following options:

       video
	   Activate the video output. The audio stream is passed unchanged
	   whether this option is set or no. The video stream will be the
	   first output stream if activated. Default is 0.

       size
	   Set the video size. This option is for video only. Default and
	   minimum resolution is "640x480".

       meter
	   Set the EBU scale meter. Default is 9. Common values are 9 and 18,
	   respectively for EBU scale meter +9 and EBU scale meter +18. Any
	   other integer value between this range is allowed.

       metadata
	   Set metadata injection. If set to 1, the audio input will be
	   segmented into 100ms output frames, each of them containing various
	   loudness information in metadata.  All the metadata keys are
	   prefixed with "lavfi.r128.".

	   Default is 0.

       framelog
	   Force the frame logging level.

	   Available values are:

	   info
	       information logging level

	   verbose
	       verbose logging level

	   By default, the logging level is set to info. If the video or the
	   metadata options are set, it switches to verbose.

       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 a 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 drop 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 will send an end-of-stream signal.

       Also, depending on inputs synchronization, the filters will drop frames
       in case one input receives more frames than the other ones, and the
       queue is already filled.

       These filters accept the following options:

       nb_inputs, n
	   Set the number of different inputs, it is 2 by default.

       Examples

       ·   Interleave frames belonging to different streams using ffmpeg:

		   ffmpeg -i bambi.avi -i pr0n.mkv -filter_complex "[0:v][1:v] interleave" out.avi

       ·   Add flickering blur effect:

		   select='if(gt(random(0), 0.2), 1, 2)':n=2 [tmp], boxblur=2:2, [tmp] interleave

   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.

   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, NAN if undefined

       TB  timebase of the input timestamps

       pts the PTS (Presentation TimeStamp) of the filtered video frame,
	   expressed in TB units, NAN if undefined

       t   the PTS (Presentation TimeStamp) of the filtered video frame,
	   expressed in seconds, NAN if undefined

       prev_pts
	   the PTS of the previously filtered video frame, NAN if undefined

       prev_selected_pts
	   the PTS of the last previously filtered video frame, NAN if
	   undefined

       prev_selected_t
	   the PTS of the last previously selected video frame, NAN if
	   undefined

       start_pts
	   the PTS of the first video frame in the video, NAN if undefined

       start_t
	   the time of the first video frame in the video, NAN if undefined

       pict_type (video only)
	   the type of the filtered frame, can assume one of the following
	   values:

	   I
	   P
	   B
	   S
	   SI
	   SP
	   BI
       interlace_type (video only)
	   the frame interlace type, 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 1 if the filtered frame is a key-frame, 0 otherwise

       pos the position in the file of the filtered frame, -1 if the
	   information is not available (e.g. for synthetic video)

       scene (video only)
	   value between 0 and 1 to indicate a new scene; a low value reflects
	   a low probability for the current frame to introduce a new scene,
	   while a higher value means the current frame is more likely to be
	   one (see the example below)

       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='gte(t\,10)*lte(t\,20)'

       ·   Select only I frames contained in the 10-20 time interval:

		   select='gte(t\,10)*lte(t\,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

   sendcmd, asendcmd
       Send commands to filters in the filtergraph.

       These filters read commands to be sent to other filters in the
       filtergraph.

       "sendcmd" must be inserted between two video filters, "asendcmd" must
       be inserted between two audio filters, but apart from that they act the
       same way.

       The specification of commands can be provided in the filter arguments
       with the commands option, or in a file specified by the filename
       option.

       These filters accept the following options:

       commands, c
	   Set the commands to be read and sent to the other filters.

       filename, f
	   Set the filename of the commands to be read and sent to the other
	   filters.

       Commands syntax

       A commands description consists of a sequence of interval
       specifications, comprising a list of commands to be executed when a
       particular event related to that interval occurs. The occurring event
       is typically the current frame time entering or leaving a given time
       interval.

       An interval is specified by the following syntax:

	       <START>[-<END>] <COMMANDS>;

       The time interval is specified by the START and END times.  END is
       optional and defaults to the maximum time.

       The current frame time is considered within the specified interval if
       it is included in the interval [START, END), that is when the time is
       greater or equal to START and is lesser than END.

       COMMANDS consists of a sequence of one or more command specifications,
       separated by ",", relating to that interval.  The syntax of a command
       specification is given by:

	       [<FLAGS>] <TARGET> <COMMAND> <ARG>

       FLAGS is optional and specifies the type of events relating to the time
       interval which enable sending the specified command, and must be a non-
       null sequence of identifier flags separated by "+" or "|" and enclosed
       between "[" and "]".

       The following flags are recognized:

       enter
	   The command is sent when the current frame timestamp enters the
	   specified interval. In other words, the command is sent when the
	   previous frame timestamp was not in the given interval, and the
	   current is.

       leave
	   The command is sent when the current frame timestamp leaves the
	   specified interval. In other words, the command is sent when the
	   previous frame timestamp was in the given interval, and the current
	   is not.

       If FLAGS is not specified, a default value of "[enter]" is assumed.

       TARGET specifies the target of the command, usually the name of the
       filter class or a specific filter instance name.

       COMMAND specifies the name of the command for the target filter.

       ARG is optional and specifies the optional list of argument for the
       given COMMAND.

       Between one interval specification and another, whitespaces, or
       sequences of characters starting with "#" until the end of line, are
       ignored and can be used to annotate comments.

       A simplified BNF description of the commands specification syntax
       follows:

	       <COMMAND_FLAG>  ::= "enter" | "leave"
	       <COMMAND_FLAGS> ::= <COMMAND_FLAG> [(+|"|")<COMMAND_FLAG>]
	       <COMMAND>       ::= ["[" <COMMAND_FLAGS> "]"] <TARGET> <COMMAND> [<ARG>]
	       <COMMANDS>      ::= <COMMAND> [,<COMMANDS>]
	       <INTERVAL>      ::= <START>[-<END>] <COMMANDS>
	       <INTERVALS>     ::= <INTERVAL>[;<INTERVALS>]

       Examples

       ·   Specify audio tempo change at second 4:

		   asendcmd=c='4.0 atempo tempo 1.5',atempo

       ·   Specify a list of drawtext and hue commands in a file.

		   # show text in the interval 5-10
		   5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
			    [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';

		   # desaturate the image in the interval 15-20
		   15.0-20.0 [enter] hue s 0,
			     [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
			     [leave] hue s 1,
			     [leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';

		   # apply an exponential saturation fade-out effect, starting from time 25
		   25 [enter] hue s exp(25-t)

	   A filtergraph allowing to read and process the above command list
	   stored in a file test.cmd, can be specified with:

		   sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue

   setpts, asetpts
       Change the PTS (presentation timestamp) of the input frames.

       "setpts" works on video frames, "asetpts" on audio frames.

       This filter accepts the following options:

       expr
	   The expression which is evaluated for each frame to construct its
	   timestamp.

       The expression is evaluated through the eval API and can contain the
       following constants:

       FRAME_RATE
	   frame rate, only defined for constant frame-rate video

       PTS the presentation timestamp in input

       N   the count of the input frame for video or the number of consumed
	   samples, not including the current frame for audio, starting from
	   0.

       NB_CONSUMED_SAMPLES
	   the number of consumed samples, not including the current frame
	   (only audio)

       NB_SAMPLES, S
	   the number of samples in the current frame (only audio)

       SAMPLE_RATE, SR
	   audio sample rate

       STARTPTS
	   the PTS of the first frame

       STARTT
	   the time in seconds of the first frame

       INTERLACED
	   tell if the current frame is interlaced

       T   the time in seconds of the current frame

       TB  the time base

       POS original position in the file of the frame, or undefined if
	   undefined for the current frame

       PREV_INPTS
	   previous input PTS

       PREV_INT
	   previous input time in seconds

       PREV_OUTPTS
	   previous output PTS

       PREV_OUTT
	   previous output time in seconds

       RTCTIME
	   wallclock (RTC) time in microseconds. This is deprecated, use
	   time(0) instead.

       RTCSTART
	   wallclock (RTC) time at the start of the movie in microseconds

       Examples

       ·   Start counting PTS from zero

		   setpts=PTS-STARTPTS

       ·   Apply fast motion effect:

		   setpts=0.5*PTS

       ·   Apply slow motion effect:

		   setpts=2.0*PTS

       ·   Set fixed rate of 25 frames per second:

		   setpts=N/(25*TB)

       ·   Set fixed rate 25 fps with some jitter:

		   setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

       ·   Apply an offset of 10 seconds to the input PTS:

		   setpts=PTS+10/TB

       ·   Generate timestamps from a "live source" and rebase onto the
	   current timebase:

		   setpts='(RTCTIME - RTCSTART) / (TB * 1000000)'

       ·   Generate timestamps by counting samples:

		   asetpts=N/SR/TB

   settb, asettb
       Set the timebase to use for the output frames timestamps.  It is mainly
       useful for testing timebase configuration.

       This filter accepts the following options:

       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

   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. Default value is "640x512".

       slide
	   Specify if the spectrum should slide along the window. Default
	   value is 0.

       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 is displayed using the same 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

	   Default value is sqrt.

       saturation
	   Set saturation modifier for displayed colors. Negative values
	   provide alternative color scheme. 0 is no saturation at all.
	   Saturation must be in [-10.0, 10.0] range.  Default value is 1.

       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]'

   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. 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.

	   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".

       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]

   split, asplit
       Split input into several identical outputs.

       "asplit" works with audio input, "split" with video.

       The filter accepts a single parameter which specifies the number of
       outputs. If unspecified, it defaults to 2.

       Examples

       ·   Create two separate outputs from the same input:

		   [in] split [out0][out1]

       ·   To create 3 or more outputs, you need to specify the number of
	   outputs, like in:

		   [in] asplit=3 [out0][out1][out2]

       ·   Create two separate outputs from the same input, one cropped and
	   one padded:

		   [in] split [splitout1][splitout2];
		   [splitout1] crop=100:100:0:0	   [cropout];
		   [splitout2] pad=200:200:100:100 [padout];

       ·   Create 5 copies of the input audio with ffmpeg:

		   ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT

   zmq, azmq
       Receive commands sent through a libzmq client, and forward them to
       filters in the filtergraph.

       "zmq" and "azmq" work as a pass-through filters. "zmq" must be inserted
       between two video filters, "azmq" between two audio filters.

       To enable these filters you need to install the libzmq library and
       headers and configure FFmpeg with "--enable-libzmq".

       For more information about libzmq see: <http://www.zeromq.org/>

       The "zmq" and "azmq" filters work as a libzmq server, which receives
       messages sent through a network interface defined by the bind_address
       option.

       The received message must be in the form:

	       <TARGET> <COMMAND> [<ARG>]

       TARGET specifies the target of the command, usually the name of the
       filter class or a specific filter instance name.

       COMMAND specifies the name of the command for the target filter.

       ARG is optional and specifies the optional argument list for the given
       COMMAND.

       Upon reception, the message is processed and the corresponding command
       is injected into the filtergraph. Depending on the result, the filter
       will send a reply to the client, adopting the format:

	       <ERROR_CODE> <ERROR_REASON>
	       <MESSAGE>

       MESSAGE is optional.

       Examples

       Look at tools/zmqsend for an example of a zmq client which can be used
       to send commands processed by these filters.

       Consider the following filtergraph generated by ffplay

	       ffplay -dumpgraph 1 -f lavfi "
	       color=s=100x100:c=red  [l];
	       color=s=100x100:c=blue [r];
	       nullsrc=s=200x100, zmq [bg];
	       [bg][l]	 overlay      [bg+l];
	       [bg+l][r] overlay=x=100 "

       To change the color of the left side of the video, the following
       command can be used:

	       echo Parsed_color_0 c yellow | tools/zmqsend

       To change the right side:

	       echo Parsed_color_1 c pink | tools/zmqsend

MULTIMEDIA SOURCES
       Below is a description of the currently available multimedia sources.

   amovie
       This is the same as movie source, except it selects an audio stream by
       default.

   movie
       Read audio and/or video stream(s) from a movie container.

       This filter accepts the following options:

       filename
	   The name of the resource to read (not necessarily a file but also 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. 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 best suited video stream will be automatically selected.
	   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 less than 1, the stream will be read again and again.
	   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.

       This filter allows to overlay a second video on top of 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 as "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 as "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]

SEE ALSO
       ffmpeg(1), ffplay(1), ffprobe(1), ffserver(1), ffmpeg-utils(1),
       ffmpeg-scaler(1), ffmpeg-resampler(1), ffmpeg-codecs(1),
       ffmpeg-bitstream-filters(1), ffmpeg-formats(1), ffmpeg-devices(1),
       ffmpeg-protocols(1), ffmpeg-filters(1)

AUTHORS
       The FFmpeg developers.

       For details about the authorship, see the Git history of the project
       (git://source.ffmpeg.org/ffmpeg), e.g. by typing the command git log in
       the FFmpeg source directory, or browsing the online repository at
       <http://source.ffmpeg.org>.

       Maintainers for the specific components are listed in the file
       MAINTAINERS in the source code tree.

				  2013-10-21			 FFMPEG-ALL(1)
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