Provided by: ffmpeg_4.1.3-1_amd64 bug

NAME

       ffmpeg - ffmpeg video converter

SYNOPSIS

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

DESCRIPTION

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

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

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

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

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

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

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

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

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

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

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

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

       The format option may be needed for raw input files.

DETAILED DESCRIPTION

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

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

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

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

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

       Simple filtergraphs

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

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

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

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

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

       Complex filtergraphs

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

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

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

       The -lavfi option is equivalent to -filter_complex.

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

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

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

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

STREAM SELECTION

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

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

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

       Automatic stream selection

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

       It will select that stream based upon the following criteria:

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

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

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

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

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

       Manual stream selection

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

       Complex filtergraphs

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

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

       Stream handling

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

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

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

       They assume the following three input files.

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

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

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

       Example: automatic stream selection

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

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

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

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

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

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

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

       Example: automatic subtitles selection

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

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

       Example: unlabeled filtergraph outputs

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

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

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

       Example: labeled filtergraph outputs

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

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

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

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

       The command should be modified as follows,

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

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

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

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

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

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

OPTIONS

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

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

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

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

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

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

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

       Possible forms of stream specifiers are:

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

       stream_type[:stream_index]
           stream_type is one of following: 'v' or 'V' for video, 'a' for audio, 's' for
           subtitle, 'd' for data, and 't' for attachments. 'v' matches all video streams, 'V'
           only matches video streams which are not attached pictures, video thumbnails or cover
           arts.  If stream_index is given, then it matches stream number stream_index of this
           type. Otherwise, it matches all streams of this type.

       p:program_id[:stream_index] or p:program_id[:stream_type[:stream_index]] or
           p:program_id:m:key[:value] In first version, 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. In the second version, stream_type is one of
           following: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data. If
           stream_index is also given, then it matches stream number stream_index of this type in
           the program with the id program_id.  Otherwise, if only stream_type is given, it
           matches all streams of this type in the program with the id program_id.  In the third
           version matches streams in the program with the id program_id with the metadata tag
           key having the specified value. If value is not given, matches streams that contain
           the given tag with any value.

       #stream_id or i:stream_id
           Match the stream by stream id (e.g. PID in MPEG-TS container).

       m:key[:value]
           Matches streams with the metadata tag key having the specified value. If value is not
           given, matches streams that contain the given tag with any value.

       u   Matches streams with usable configuration, the codec must be defined and the essential
           information such as video dimension or audio sample rate must be present.

           Note that in ffmpeg, matching by metadata will only work properly for input files.

   Generic options
       These options are shared amongst the ff* tools.

       -L  Show license.

       -h, -?, -help, --help [arg]
           Show help. An optional parameter may be specified to print help about a specific item.
           If no argument is specified, only basic (non advanced) tool options are shown.

           Possible values of arg are:

           long
               Print advanced tool options in addition to the basic tool options.

           full
               Print complete list of options, including shared and private options for encoders,
               decoders, demuxers, muxers, filters, etc.

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

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

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

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

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

       -version
           Show version.

       -formats
           Show available formats (including devices).

       -demuxers
           Show available demuxers.

       -muxers
           Show available muxers.

       -devices
           Show available devices.

       -codecs
           Show all codecs known to libavcodec.

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

       -decoders
           Show available decoders.

       -encoders
           Show all available encoders.

       -bsfs
           Show available bitstream filters.

       -protocols
           Show available protocols.

       -filters
           Show available libavfilter filters.

       -pix_fmts
           Show available pixel formats.

       -sample_fmts
           Show available sample formats.

       -layouts
           Show channel names and standard channel layouts.

       -colors
           Show recognized color names.

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

                   ffmpeg -sources pulse,server=192.168.0.4

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

                   ffmpeg -sinks pulse,server=192.168.0.4

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

           The optional flags prefix can consist of the following values:

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

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

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

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

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

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

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

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

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

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

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

           debug, 48
               Show everything, including debugging information.

           trace, 56

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

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

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

                   ffmpeg [...] -loglevel +repeat

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

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

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

           The following options are recognized:

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

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

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

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

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

       -hide_banner
           Suppress printing banner.

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

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

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

           Possible flags for this option are:

           x86
               mmx
               mmxext
               sse
               sse2
               sse2slow
               sse3
               sse3slow
               ssse3
               atom
               sse4.1
               sse4.2
               avx
               avx2
               xop
               fma3
               fma4
               3dnow
               3dnowext
               bmi1
               bmi2
               cmov
           ARM
               armv5te
               armv6
               armv6t2
               vfp
               vfpv3
               neon
               setend
           AArch64
               armv8
               vfp
               neon
           PowerPC
               altivec
           Specific Processors
               pentium2
               pentium3
               pentium4
               k6
               k62
               athlon
               athlonxp
               k8

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

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

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

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

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

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

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

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

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

       -i url (input)
           input file url

       -y (global)
           Overwrite output files without asking.

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

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

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

           For example

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           To set the language of the first audio stream:

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

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

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

           The following dispositions are recognized:

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

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

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

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

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

           To add an embedded cover/thumbnail:

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

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

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

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

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

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

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

       -dn (output)
           Disable data recording. For full manual control see the "-map" option.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           See also the option "-fdebug ts".

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

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

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

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

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

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

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

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

                   ffmpeg -dump_attachment:t "" -i INPUT

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

       -noautorotate
           Disable automatically rotating video based on file metadata.

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

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

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

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

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

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

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

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

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

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

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

       -vn (output)
           Disable video recording. For full manual control see the "-map" option.

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

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

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

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

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

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

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

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

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

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

       -psnr
           Calculate PSNR of compressed frames.

       -vstats
           Dump video coding statistics to vstats_HHMMSS.log.

       -vstats_file file
           Dump video coding statistics to file.

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

           version = 1 :

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

           version > 1:

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

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

       -dc precision
           Intra_dc_precision.

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

       -qphist (global)
           Show QP histogram

       -vbsf bitstream_filter
           Deprecated see -bsf

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

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

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

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

                   -force_key_frames 0:05:00,chapters-0.1

           The expression in expr can contain the following constants:

           n   the number of current processed frame, starting from 0

           n_forced
               the number of forced frames

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

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

           t   the time of the current processed frame

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

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

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

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

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

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

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

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

           cuda
               device is the number of the CUDA device.

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

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

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

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

               auto
               sw
               hw
               auto_any
               hw_any
               hw2
               hw3
               hw4

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

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

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

               The strings usable as filters are:

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

               The indices and filters must together uniquely select a device.

               Examples:

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

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

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

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

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

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

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

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

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

           auto
               Automatically select the hardware acceleration method.

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

           dxva2
               Use DXVA2 (DirectX Video Acceleration) hardware acceleration.

           vaapi
               Use VAAPI (Video Acceleration API) hardware acceleration.

           qsv Use the Intel QuickSync Video acceleration for video transcoding.

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

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

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

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

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

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

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

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

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

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

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

       -an (output)
           Disable audio recording. For full manual control see the "-map" option.

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

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

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

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

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

       -absf bitstream_filter
           Deprecated, see -bsf

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

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

       -sn (output)
           Disable subtitle recording. For full manual control see the "-map" option.

       -sbsf bitstream_filter
           Deprecated, see -bsf

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

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

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

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

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

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

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

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

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

                   ffmpeg -i INPUT -map 0 output

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

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

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

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

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

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

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

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

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

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

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

           To pick the English audio stream:

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

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

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

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

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

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

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

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

                   ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           If metadata specifier is omitted, it defaults to global.

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

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

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

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

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

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

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

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

       -benchmark_all (global)
           Show benchmarking information during the encode.  Shows real, system and user time
           used in various steps (audio/video encode/decode).

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           1   Use the demuxer timebase.

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

           0   Use the decoder timebase.

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

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

           Default value is -1.

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

           0   Assign a default value according to the media type.

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

           -1  Use the input stream timebase when possible.

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

           >0  Use the provided number as the timebase.

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

           Default value is 0.

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

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

       -dts_delta_threshold
           Timestamp discontinuity delta threshold.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           For example, to overlay an image over video

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

           none
               Discard no frame.

           default
               Default, which discards no frames.

           noref
               Discard all non-reference frames.

           bidir
               Discard all bidirectional frames.

           nokey
               Discard all frames excepts keyframes.

           all Discard all frames.

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

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

       -xerror (global)
           Stop and exit on error

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

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

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

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

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

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

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

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

       ffpreset files

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

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

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

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

       avpreset files

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

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

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

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

EXAMPLES

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

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

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

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

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

   X11 grabbing
       Grab the X11 display with ffmpeg via

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

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

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

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

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

       Examples:

       ·   You can use YUV files as input:

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

           It will use the files:

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

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

       ·   You can input from a raw YUV420P file:

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

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

       ·   You can output to a raw YUV420P file:

                   ffmpeg -i mydivx.avi hugefile.yuv

       ·   You can set several input files and output files:

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

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

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

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

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

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

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

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

       ·   You can transcode decrypted VOBs:

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

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

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

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

           For extracting images from a video:

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

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

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

           For creating a video from many images:

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

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

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

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

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

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

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

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

       ·   To force CBR video output:

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

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

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

SYNTAX

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

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

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

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

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

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

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

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

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

       Examples

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

                   Crime d\'Amour

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

                   'Crime d'\''Amour'

       ·   Include leading or trailing whitespaces using quoting:

                   '  this string starts and ends with whitespaces  '

       ·   Escaping and quoting can be mixed together:

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

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

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

   Date
       The accepted syntax is:

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

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

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

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

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

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

       or

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

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

       In both expressions, the optional - indicates negative duration.

       Examples

       The following examples are all valid time duration:

       55  55 seconds

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

       23.189
           23.189 seconds

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

       The following abbreviations are recognized:

       ntsc
           720x480

       pal 720x576

       qntsc
           352x240

       qpal
           352x288

       sntsc
           640x480

       spal
           768x576

       film
           352x240

       ntsc-film
           352x240

       sqcif
           128x96

       qcif
           176x144

       cif 352x288

       4cif
           704x576

       16cif
           1408x1152

       qqvga
           160x120

       qvga
           320x240

       vga 640x480

       svga
           800x600

       xga 1024x768

       uxga
           1600x1200

       qxga
           2048x1536

       sxga
           1280x1024

       qsxga
           2560x2048

       hsxga
           5120x4096

       wvga
           852x480

       wxga
           1366x768

       wsxga
           1600x1024

       wuxga
           1920x1200

       woxga
           2560x1600

       wqsxga
           3200x2048

       wquxga
           3840x2400

       whsxga
           6400x4096

       whuxga
           7680x4800

       cga 320x200

       ega 640x350

       hd480
           852x480

       hd720
           1280x720

       hd1080
           1920x1080

       2k  2048x1080

       2kflat
           1998x1080

       2kscope
           2048x858

       4k  4096x2160

       4kflat
           3996x2160

       4kscope
           4096x1716

       nhd 640x360

       hqvga
           240x160

       wqvga
           400x240

       fwqvga
           432x240

       hvga
           480x320

       qhd 960x540

       2kdci
           2048x1080

       4kdci
           4096x2160

       uhd2160
           3840x2160

       uhd4320
           7680x4320

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

       The following abbreviations are recognized:

       ntsc
           30000/1001

       pal 25/1

       qntsc
           30000/1001

       qpal
           25/1

       sntsc
           30000/1001

       spal
           25/1

       film
           24/1

       ntsc-film
           24000/1001

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

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

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

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

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

       The string random will result in a random color.

       The following names of colors are recognized:

       AliceBlue
           0xF0F8FF

       AntiqueWhite
           0xFAEBD7

       Aqua
           0x00FFFF

       Aquamarine
           0x7FFFD4

       Azure
           0xF0FFFF

       Beige
           0xF5F5DC

       Bisque
           0xFFE4C4

       Black
           0x000000

       BlanchedAlmond
           0xFFEBCD

       Blue
           0x0000FF

       BlueViolet
           0x8A2BE2

       Brown
           0xA52A2A

       BurlyWood
           0xDEB887

       CadetBlue
           0x5F9EA0

       Chartreuse
           0x7FFF00

       Chocolate
           0xD2691E

       Coral
           0xFF7F50

       CornflowerBlue
           0x6495ED

       Cornsilk
           0xFFF8DC

       Crimson
           0xDC143C

       Cyan
           0x00FFFF

       DarkBlue
           0x00008B

       DarkCyan
           0x008B8B

       DarkGoldenRod
           0xB8860B

       DarkGray
           0xA9A9A9

       DarkGreen
           0x006400

       DarkKhaki
           0xBDB76B

       DarkMagenta
           0x8B008B

       DarkOliveGreen
           0x556B2F

       Darkorange
           0xFF8C00

       DarkOrchid
           0x9932CC

       DarkRed
           0x8B0000

       DarkSalmon
           0xE9967A

       DarkSeaGreen
           0x8FBC8F

       DarkSlateBlue
           0x483D8B

       DarkSlateGray
           0x2F4F4F

       DarkTurquoise
           0x00CED1

       DarkViolet
           0x9400D3

       DeepPink
           0xFF1493

       DeepSkyBlue
           0x00BFFF

       DimGray
           0x696969

       DodgerBlue
           0x1E90FF

       FireBrick
           0xB22222

       FloralWhite
           0xFFFAF0

       ForestGreen
           0x228B22

       Fuchsia
           0xFF00FF

       Gainsboro
           0xDCDCDC

       GhostWhite
           0xF8F8FF

       Gold
           0xFFD700

       GoldenRod
           0xDAA520

       Gray
           0x808080

       Green
           0x008000

       GreenYellow
           0xADFF2F

       HoneyDew
           0xF0FFF0

       HotPink
           0xFF69B4

       IndianRed
           0xCD5C5C

       Indigo
           0x4B0082

       Ivory
           0xFFFFF0

       Khaki
           0xF0E68C

       Lavender
           0xE6E6FA

       LavenderBlush
           0xFFF0F5

       LawnGreen
           0x7CFC00

       LemonChiffon
           0xFFFACD

       LightBlue
           0xADD8E6

       LightCoral
           0xF08080

       LightCyan
           0xE0FFFF

       LightGoldenRodYellow
           0xFAFAD2

       LightGreen
           0x90EE90

       LightGrey
           0xD3D3D3

       LightPink
           0xFFB6C1

       LightSalmon
           0xFFA07A

       LightSeaGreen
           0x20B2AA

       LightSkyBlue
           0x87CEFA

       LightSlateGray
           0x778899

       LightSteelBlue
           0xB0C4DE

       LightYellow
           0xFFFFE0

       Lime
           0x00FF00

       LimeGreen
           0x32CD32

       Linen
           0xFAF0E6

       Magenta
           0xFF00FF

       Maroon
           0x800000

       MediumAquaMarine
           0x66CDAA

       MediumBlue
           0x0000CD

       MediumOrchid
           0xBA55D3

       MediumPurple
           0x9370D8

       MediumSeaGreen
           0x3CB371

       MediumSlateBlue
           0x7B68EE

       MediumSpringGreen
           0x00FA9A

       MediumTurquoise
           0x48D1CC

       MediumVioletRed
           0xC71585

       MidnightBlue
           0x191970

       MintCream
           0xF5FFFA

       MistyRose
           0xFFE4E1

       Moccasin
           0xFFE4B5

       NavajoWhite
           0xFFDEAD

       Navy
           0x000080

       OldLace
           0xFDF5E6

       Olive
           0x808000

       OliveDrab
           0x6B8E23

       Orange
           0xFFA500

       OrangeRed
           0xFF4500

       Orchid
           0xDA70D6

       PaleGoldenRod
           0xEEE8AA

       PaleGreen
           0x98FB98

       PaleTurquoise
           0xAFEEEE

       PaleVioletRed
           0xD87093

       PapayaWhip
           0xFFEFD5

       PeachPuff
           0xFFDAB9

       Peru
           0xCD853F

       Pink
           0xFFC0CB

       Plum
           0xDDA0DD

       PowderBlue
           0xB0E0E6

       Purple
           0x800080

       Red 0xFF0000

       RosyBrown
           0xBC8F8F

       RoyalBlue
           0x4169E1

       SaddleBrown
           0x8B4513

       Salmon
           0xFA8072

       SandyBrown
           0xF4A460

       SeaGreen
           0x2E8B57

       SeaShell
           0xFFF5EE

       Sienna
           0xA0522D

       Silver
           0xC0C0C0

       SkyBlue
           0x87CEEB

       SlateBlue
           0x6A5ACD

       SlateGray
           0x708090

       Snow
           0xFFFAFA

       SpringGreen
           0x00FF7F

       SteelBlue
           0x4682B4

       Tan 0xD2B48C

       Teal
           0x008080

       Thistle
           0xD8BFD8

       Tomato
           0xFF6347

       Turquoise
           0x40E0D0

       Violet
           0xEE82EE

       Wheat
           0xF5DEB3

       White
           0xFFFFFF

       WhiteSmoke
           0xF5F5F5

       Yellow
           0xFFFF00

       YellowGreen
           0x9ACD32

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

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

       FL  front left

       FR  front right

       FC  front center

       LFE low frequency

       BL  back left

       BR  back right

       FLC front left-of-center

       FRC front right-of-center

       BC  back center

       SL  side left

       SR  side right

       TC  top center

       TFL top front left

       TFC top front center

       TFR top front right

       TBL top back left

       TBC top back center

       TBR top back right

       DL  downmix left

       DR  downmix right

       WL  wide left

       WR  wide right

       SDL surround direct left

       SDR surround direct right

       LFE2
           low frequency 2

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

       mono
           FC

       stereo
           FL+FR

       2.1 FL+FR+LFE

       3.0 FL+FR+FC

       3.0(back)
           FL+FR+BC

       4.0 FL+FR+FC+BC

       quad
           FL+FR+BL+BR

       quad(side)
           FL+FR+SL+SR

       3.1 FL+FR+FC+LFE

       5.0 FL+FR+FC+BL+BR

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

       4.1 FL+FR+FC+LFE+BC

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       downmix
           DL+DR

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

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

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

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

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

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

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

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

EXPRESSION EVALUATION

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

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

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

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

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

       The following functions are available:

       abs(x)
           Compute absolute value of x.

       acos(x)
           Compute arccosine of x.

       asin(x)
           Compute arcsine of x.

       atan(x)
           Compute arctangent of x.

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

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

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

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

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

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

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

       cos(x)
           Compute cosine of x.

       cosh(x)
           Compute hyperbolic cosine of x.

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

       log(x)
           Compute natural logarithm of x.

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

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

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

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

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

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

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

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

           Prints t with loglevel l

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

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

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

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

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

       sin(x)
           Compute sine of x.

       sinh(x)
           Compute hyperbolic sine of x.

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

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

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

       tan(x)
           Compute tangent of x.

       tanh(x)
           Compute hyperbolic tangent of x.

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

           When the series does not converge the result is undefined.

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

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

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

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

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

       The following constants are available:

       PI  area of the unit disc, approximately 3.14

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

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

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

       "*" works like AND

       "+" works like OR

       For example the construct:

               if (A AND B) then C

       is equivalent to:

               if(A*B, C)

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

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

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

       y   10^-24 / 2^-80

       z   10^-21 / 2^-70

       a   10^-18 / 2^-60

       f   10^-15 / 2^-50

       p   10^-12 / 2^-40

       n   10^-9 / 2^-30

       u   10^-6 / 2^-20

       m   10^-3 / 2^-10

       c   10^-2

       d   10^-1

       h   10^2

       k   10^3 / 2^10

       K   10^3 / 2^10

       M   10^6 / 2^20

       G   10^9 / 2^30

       T   10^12 / 2^40

       P   10^15 / 2^40

       E   10^18 / 2^50

       Z   10^21 / 2^60

       Y   10^24 / 2^70

CODEC OPTIONS

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

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

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

       The list of supported options follow:

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

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

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

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

           Possible values:

           mv4 Use four motion vector by macroblock (mpeg4).

           qpel
               Use 1/4 pel motion compensation.

           loop
               Use loop filter.

           qscale
               Use fixed qscale.

           pass1
               Use internal 2pass ratecontrol in first pass mode.

           pass2
               Use internal 2pass ratecontrol in second pass mode.

           gray
               Only decode/encode grayscale.

           emu_edge
               Do not draw edges.

           psnr
               Set error[?] variables during encoding.

           truncated
           ildct
               Use interlaced DCT.

           low_delay
               Force low delay.

           global_header
               Place global headers in extradata instead of every keyframe.

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

           aic Apply H263 advanced intra coding / mpeg4 ac prediction.

           cbp Deprecated, use mpegvideo private options instead.

           qprd
               Deprecated, use mpegvideo private options instead.

           ilme
               Apply interlaced motion estimation.

           cgop
               Use closed gop.

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

           Possible values:

           zero
               zero motion estimation (fastest)

           full
               full motion estimation (slowest)

           epzs
               EPZS motion estimation (default)

           esa esa motion estimation (alias for full)

           tesa
               tesa motion estimation

           dia dia motion estimation (alias for epzs)

           log log motion estimation

           phods
               phods motion estimation

           x1  X1 motion estimation

           hex hex motion estimation

           umh umh motion estimation

           iter
               iter motion estimation

       extradata_size integer
           Set extradata size.

       time_base rational number
           Set codec time base.

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

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

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

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

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

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

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

       frame_number integer
           Set the frame number.

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

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

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

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

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

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

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

           Default value is 0.

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

       rc_strategy integer (encoding,video)
           Set ratecontrol method.

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

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

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

           Possible values:

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

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

           ump4
               (autodetected if fourcc==UMP4)

           no_padding
               padding bug (autodetected)

           amv
           ac_vlc
               illegal vlc bug (autodetected per fourcc)

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

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

           edge
               edge padding bug (autodetected per fourcc/version)

           hpel_chroma
           dc_clip
           ms  Workaround various bugs in microsoft broken decoders.

           trunc
               trancated frames

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

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

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

           Possible values:

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

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

           normal
           unofficial
               allow unofficial extensions

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

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

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

           Possible values:

           crccheck
               verify embedded CRCs

           bitstream
               detect bitstream specification deviations

           buffer
               detect improper bitstream length

           explode
               abort decoding on minor error detection

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

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

           compliant
               consider all spec non compliancies as errors

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

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

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

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

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

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

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

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

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

       rc_buf_aggressivity float (encoding,video)
           Currently useless.

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

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

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

       dct integer (encoding,video)
           Set DCT algorithm.

           Possible values:

           auto
               autoselect a good one (default)

           fastint
               fast integer

           int accurate integer

           mmx
           altivec
           faan
               floating point AAN DCT

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

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

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

       p_mask float (encoding,video)
           Set inter masking.

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

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

           Possible values:

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

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

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

           Possible values:

           guess_mvs
               iterative motion vector (MV) search (slow)

           deblock
               use strong deblock filter for damaged MBs

           favor_inter
               favor predicting from the previous frame instead of the current

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

           Possible values:

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

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

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

           Possible values:

           pict
               picture info

           rc  rate control

           bitstream
           mb_type
               macroblock (MB) type

           qp  per-block quantization parameter (QP)

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

           skip
           startcode
           er  error recognition

           mmco
               memory management control operations (H.264)

           bugs
           buffers
               picture buffer allocations

           thread_ops
               threading operations

           nomc
               skip motion compensation

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

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

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

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

           vlc variable length coder / huffman coder

           ac  arithmetic coder

           raw raw (no encoding)

           rle run-length coder

           deflate
               deflate-based coder

       context integer (encoding,video)
           Set context model.

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

           Possible values:

           simple
               use mbcmp (default)

           bits
               use fewest bits

           rd  use best rate distortion

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

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

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

       nr integer (encoding,video)
           Set noise reduction.

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

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

           fast
               Allow non spec compliant speedup tricks.

           sgop
               Deprecated, use mpegvideo private options instead.

           noout
               Skip bitstream encoding.

           ignorecrop
               Ignore cropping information from sps.

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

           chunks
               Frame data might be split into multiple chunks.

           showall
               Show all frames before the first keyframe.

           skiprd
               Deprecated, use mpegvideo private options instead.

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

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

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

           Possible values:

           auto, 0
               automatically select the number of threads to set

           Default value is auto.

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

       mb_threshold integer (encoding,video)
           Set macroblock threshold.

       dc integer (encoding,video)
           Set intra_dc_precision.

       nssew integer (encoding,video)
           Set nsse weight.

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

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

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

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

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

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

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

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

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

           Possible values:

           sad sum of absolute differences, fast (default)

           sse sum of squared errors

           satd
               sum of absolute Hadamard transformed differences

           dct sum of absolute DCT transformed differences

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

           bit number of bits needed for the block

           rd  rate distortion optimal, slow

           zero
               0

           vsad
               sum of absolute vertical differences

           vsse
               sum of squared vertical differences

           nsse
               noise preserving sum of squared differences

           w53 5/3 wavelet, only used in snow

           w97 9/7 wavelet, only used in snow

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

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

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

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

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

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

           Possible values:

           none
               Discard no frame.

           default
               Discard useless frames like 0-sized frames.

           noref
               Discard all non-reference frames.

           bidir
               Discard all bidirectional frames.

           nokey
               Discard all frames excepts keyframes.

           all Discard all frames.

           Default value is default.

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

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

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

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

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

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

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

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

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

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

       request_channel_layout integer (decoding,audio)
           Possible values:

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

           bt709
               BT.709

           bt470m
               BT.470 M

           bt470bg
               BT.470 BG

           smpte170m
               SMPTE 170 M

           smpte240m
               SMPTE 240 M

           film
               Film

           bt2020
               BT.2020

           smpte428
           smpte428_1
               SMPTE ST 428-1

           smpte431
               SMPTE 431-2

           smpte432
               SMPTE 432-1

           jedec-p22
               JEDEC P22

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

           bt709
               BT.709

           gamma22
               BT.470 M

           gamma28
               BT.470 BG

           smpte170m
               SMPTE 170 M

           smpte240m
               SMPTE 240 M

           linear
               Linear

           log
           log100
               Log

           log_sqrt
           log316
               Log square root

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

           bt1361
           bt1361e
               BT.1361

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

           bt2020_10
           bt2020_10bit
               BT.2020 - 10 bit

           bt2020_12
           bt2020_12bit
               BT.2020 - 12 bit

           smpte2084
               SMPTE ST 2084

           smpte428
           smpte428_1
               SMPTE ST 428-1

           arib-std-b67
               ARIB STD-B67

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

           rgb RGB

           bt709
               BT.709

           fcc FCC

           bt470bg
               BT.470 BG

           smpte170m
               SMPTE 170 M

           smpte240m
               SMPTE 240 M

           ycocg
               YCOCG

           bt2020nc
           bt2020_ncl
               BT.2020 NCL

           bt2020c
           bt2020_cl
               BT.2020 CL

           smpte2085
               SMPTE 2085

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

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

           pc
           jpeg
               JPEG (2^n-1)

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

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

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

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

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

           Possible values:

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

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

           frame
               Decode more than one frame at once.

           Default value is slice+frame.

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

           Possible values:

           ma  Main Audio Service

           ef  Effects

           vi  Visually Impaired

           hi  Hearing Impaired

           di  Dialogue

           co  Commentary

           em  Emergency

           vo  Voice Over

           ka  Karaoke

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

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

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

           progressive
               Progressive video

           tt  Interlaced video, top field coded and displayed first

           bb  Interlaced video, bottom field coded and displayed first

           tb  Interlaced video, top coded first, bottom displayed first

           bt  Interlaced video, bottom coded first, top displayed first

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

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

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

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

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

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

DECODERS

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

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

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

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

VIDEO DECODERS

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

   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

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

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

AUDIO DECODERS

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

   ac3
       AC-3 audio decoder.

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

       AC-3 Decoder Options

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

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

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

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

   flac
       FLAC audio decoder.

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

       FLAC Decoder options

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

   ffwavesynth
       Internal wave synthesizer.

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

   libcelt
       libcelt decoder wrapper.

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

   libgsm
       libgsm decoder wrapper.

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

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

   libilbc
       libilbc decoder wrapper.

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

       Options

       The following option is supported by the libilbc wrapper.

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

   libopencore-amrnb
       libopencore-amrnb decoder wrapper.

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

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

   libopencore-amrwb
       libopencore-amrwb decoder wrapper.

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

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

   libopus
       libopus decoder wrapper.

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

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

SUBTITLES DECODERS

   dvbsub
       Options

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

           0   Never compute CLUT

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

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

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

       Options

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

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

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

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

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

       Options

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

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

       txt_format
           Specifies the format of the decoded subtitles.

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

           text
               Simple text based output without formatting.

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

       txt_left
           X offset of generated bitmaps, default is 0.

       txt_top
           Y offset of generated bitmaps, default is 0.

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

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

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

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

ENCODERS

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

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

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

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

AUDIO ENCODERS

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

   aac
       Advanced Audio Coding (AAC) encoder.

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

       Options

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

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

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

       aac_coder
           Set AAC encoder coding method. Possible values:

           twoloop
               Two loop searching (TLS) method.

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

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

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

           fast
               Constant quantizer method.

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

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

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

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

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

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

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

       profile
           Sets the encoding profile, possible values:

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

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

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

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

           If this option is unspecified it is set to aac_low.

   ac3 and ac3_fixed
       AC-3 audio encoders.

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

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

       AC-3 Metadata

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

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

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

       Metadata Control Options

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

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

           1   Metadata values can be changed before encoding each frame.

       Downmix Levels

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

           0.707
               Apply -3dB gain

           0.595
               Apply -4.5dB gain (default)

           0.500
               Apply -6dB gain

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

           0.707
               Apply -3dB gain

           0.500
               Apply -6dB gain (default)

           0.000
               Silence Surround Channel(s)

       Audio Production Information

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

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

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

           0
           notindicated
               Not Indicated (default)

           1
           large
               Large Room

           2
           small
               Small Room

       Other Metadata Options

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

           0
           off No Copyright Exists (default)

           1
           on  Copyright Exists

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

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

           0
           notindicated
               Not Indicated (default)

           1
           off Not Dolby Surround Encoded

           2
           on  Dolby Surround Encoded

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

           0
           off Not Original Source

           1
           on  Original Source (default)

       Extended Bitstream Information

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

       Extended Bitstream Information - Part 1

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

           0
           notindicated
               Not Indicated (default)

           1
           ltrt
               Lt/Rt Downmix Preferred

           2
           loro
               Lo/Ro Downmix Preferred

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

           1.414
               Apply +3dB gain

           1.189
               Apply +1.5dB gain

           1.000
               Apply 0dB gain

           0.841
               Apply -1.5dB gain

           0.707
               Apply -3.0dB gain

           0.595
               Apply -4.5dB gain (default)

           0.500
               Apply -6.0dB gain

           0.000
               Silence Center Channel

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

           0.841
               Apply -1.5dB gain

           0.707
               Apply -3.0dB gain

           0.595
               Apply -4.5dB gain

           0.500
               Apply -6.0dB gain (default)

           0.000
               Silence Surround Channel(s)

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

           1.414
               Apply +3dB gain

           1.189
               Apply +1.5dB gain

           1.000
               Apply 0dB gain

           0.841
               Apply -1.5dB gain

           0.707
               Apply -3.0dB gain

           0.595
               Apply -4.5dB gain (default)

           0.500
               Apply -6.0dB gain

           0.000
               Silence Center Channel

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

           0.841
               Apply -1.5dB gain

           0.707
               Apply -3.0dB gain

           0.595
               Apply -4.5dB gain

           0.500
               Apply -6.0dB gain (default)

           0.000
               Silence Surround Channel(s)

       Extended Bitstream Information - Part 2

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

           0
           notindicated
               Not Indicated (default)

           1
           on  Dolby Surround EX Off

           2
           off Dolby Surround EX On

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

           0
           notindicated
               Not Indicated (default)

           1
           on  Dolby Headphone Off

           2
           off Dolby Headphone On

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

           0
           standard
               Standard A/D Converter (default)

           1
           hdcd
               HDCD A/D Converter

       Other AC-3 Encoding Options

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

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

       Floating-Point-Only AC-3 Encoding Options

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

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

           -1
           auto
               Selected by Encoder (default)

           0
           off Disable Channel Coupling

           1
           on  Enable Channel Coupling

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

           -1
           auto
               Selected by Encoder (default)

   flac
       FLAC (Free Lossless Audio Codec) Encoder

       Options

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

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

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

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

       lpc_type
           Sets the first stage LPC algorithm

           none
               LPC is not used

           fixed
               fixed LPC coefficients

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

       min_partition_order
           The minimum partition order

       max_partition_order
           The maximum partition order

       prediction_order_method
           estimation
           2level
           4level
           8level
           search
               Bruteforce search

           log
       ch_mode
           Channel mode

           auto
               The mode is chosen automatically for each frame

           indep
               Channels are independently coded

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

       multi_dim_quant
           Multi Dimensional Quantization. If set to 1 then a 2nd stage LPC algorithm is applied
           after the first stage to finetune the coefficients. This is quite slow and slightly
           improves compression.

   opus
       Opus encoder.

       This is a native FFmpeg encoder for the Opus format. Currently its in development and only
       implements the CELT part of the codec. Its quality is usually worse and at best is equal
       to the libopus encoder.

       Options

       b   Set bit rate in bits/s. If unspecified it uses the number of channels and the layout
           to make a good guess.

       opus_delay
           Sets the maximum delay in milliseconds. Lower delays than 20ms will very quickly
           decrease quality.

   libfdk_aac
       libfdk-aac AAC (Advanced Audio Coding) encoder wrapper.

       The libfdk-aac library is based on the Fraunhofer FDK AAC code from the Android project.

       Requires the presence of the libfdk-aac headers and library during configuration. You need
       to explicitly configure the build with "--enable-libfdk-aac". The library is also
       incompatible with GPL, so if you allow the use of GPL, you should configure with
       "--enable-gpl --enable-nonfree --enable-libfdk-aac".

       This encoder is considered to produce output on par or worse at 128kbps to the the native
       FFmpeg AAC encoder but can often produce better sounding audio at identical or lower
       bitrates and has support for the AAC-HE profiles.

       VBR encoding, enabled through the vbr or flags +qscale options, is experimental and only
       works with some combinations of parameters.

       Support for encoding 7.1 audio is only available with libfdk-aac 0.1.3 or higher.

       For more information see the fdk-aac project at
       <http://sourceforge.net/p/opencore-amr/fdk-aac/>.

       Options

       The following options are mapped on the shared FFmpeg codec options.

       b   Set bit rate in bits/s. If the bitrate is not explicitly specified, it is
           automatically set to a suitable value depending on the selected profile.

           In case VBR mode is enabled the option is ignored.

       ar  Set audio sampling rate (in Hz).

       channels
           Set the number of audio channels.

       flags +qscale
           Enable fixed quality, VBR (Variable Bit Rate) mode.  Note that VBR is implicitly
           enabled when the vbr value is positive.

       cutoff
           Set cutoff frequency. If not specified (or explicitly set to 0) it will use a value
           automatically computed by the library. Default value is 0.

       profile
           Set audio profile.

           The following profiles are recognized:

           aac_low
               Low Complexity AAC (LC)

           aac_he
               High Efficiency AAC (HE-AAC)

           aac_he_v2
               High Efficiency AAC version 2 (HE-AACv2)

           aac_ld
               Low Delay AAC (LD)

           aac_eld
               Enhanced Low Delay AAC (ELD)

           If not specified it is set to aac_low.

       The following are private options of the libfdk_aac encoder.

       afterburner
           Enable afterburner feature if set to 1, disabled if set to 0. This improves the
           quality but also the required processing power.

           Default value is 1.

       eld_sbr
           Enable SBR (Spectral Band Replication) for ELD if set to 1, disabled if set to 0.

           Default value is 0.

       signaling
           Set SBR/PS signaling style.

           It can assume one of the following values:

           default
               choose signaling implicitly (explicit hierarchical by default, implicit if global
               header is disabled)

           implicit
               implicit backwards compatible signaling

           explicit_sbr
               explicit SBR, implicit PS signaling

           explicit_hierarchical
               explicit hierarchical signaling

           Default value is default.

       latm
           Output LATM/LOAS encapsulated data if set to 1, disabled if set to 0.

           Default value is 0.

       header_period
           Set StreamMuxConfig and PCE repetition period (in frames) for sending in-band
           configuration buffers within LATM/LOAS transport layer.

           Must be a 16-bits non-negative integer.

           Default value is 0.

       vbr Set VBR mode, from 1 to 5. 1 is lowest quality (though still pretty good) and 5 is
           highest quality. A value of 0 will disable VBR, and CBR (Constant Bit Rate) is
           enabled.

           Currently only the aac_low profile supports VBR encoding.

           VBR modes 1-5 correspond to roughly the following average bit rates:

           1   32 kbps/channel

           2   40 kbps/channel

           3   48-56 kbps/channel

           4   64 kbps/channel

           5   about 80-96 kbps/channel

           Default value is 0.

       Examples

       ·   Use ffmpeg to convert an audio file to VBR AAC in an M4A (MP4) container:

                   ffmpeg -i input.wav -codec:a libfdk_aac -vbr 3 output.m4a

       ·   Use ffmpeg to convert an audio file to CBR 64k kbps AAC, using the High-Efficiency AAC
           profile:

                   ffmpeg -i input.wav -c:a libfdk_aac -profile:a aac_he -b:a 64k output.m4a

   libmp3lame
       LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper.

       Requires the presence of the libmp3lame headers and library during configuration. You need
       to explicitly configure the build with "--enable-libmp3lame".

       See libshine for a fixed-point MP3 encoder, although with a lower quality.

       Options

       The following options are supported by the libmp3lame wrapper. The lame-equivalent of the
       options are listed in parentheses.

       b (-b)
           Set bitrate expressed in bits/s for CBR or ABR. LAME "bitrate" is expressed in
           kilobits/s.

       q (-V)
           Set constant quality setting for VBR. This option is valid only using the ffmpeg
           command-line tool. For library interface users, use global_quality.

       compression_level (-q)
           Set algorithm quality. Valid arguments are integers in the 0-9 range, with 0 meaning
           highest quality but slowest, and 9 meaning fastest while producing the worst quality.

       cutoff (--lowpass)
           Set lowpass cutoff frequency. If unspecified, the encoder dynamically adjusts the
           cutoff.

       reservoir
           Enable use of bit reservoir when set to 1. Default value is 1. LAME has this enabled
           by default, but can be overridden by use --nores option.

       joint_stereo (-m j)
           Enable the encoder to use (on a frame by frame basis) either L/R stereo or mid/side
           stereo. Default value is 1.

       abr (--abr)
           Enable the encoder to use ABR when set to 1. The lame --abr sets the target bitrate,
           while this options only tells FFmpeg to use ABR still relies on b to set bitrate.

   libopencore-amrnb
       OpenCORE Adaptive Multi-Rate Narrowband encoder.

       Requires the presence of the libopencore-amrnb headers and library during configuration.
       You need to explicitly configure the build with "--enable-libopencore-amrnb
       --enable-version3".

       This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can
       override it by setting strict to unofficial or lower.

       Options

       b   Set bitrate in bits per second. Only the following bitrates are supported, otherwise
           libavcodec will round to the nearest valid bitrate.

           4750
           5150
           5900
           6700
           7400
           7950
           10200
           12200
       dtx Allow discontinuous transmission (generate comfort noise) when set to 1. The default
           value is 0 (disabled).

   libopus
       libopus Opus Interactive Audio Codec encoder wrapper.

       Requires the presence of the libopus headers and library during configuration. You need to
       explicitly configure the build with "--enable-libopus".

       Option Mapping

       Most libopus options are modelled after the opusenc utility from opus-tools. The following
       is an option mapping chart describing options supported by the libopus wrapper, and their
       opusenc-equivalent in parentheses.

       b (bitrate)
           Set the bit rate in bits/s.  FFmpeg's b option is expressed in bits/s, while opusenc's
           bitrate in kilobits/s.

       vbr (vbr, hard-cbr, and cvbr)
           Set VBR mode. The FFmpeg vbr option has the following valid arguments, with the
           opusenc equivalent options in parentheses:

           off (hard-cbr)
               Use constant bit rate encoding.

           on (vbr)
               Use variable bit rate encoding (the default).

           constrained (cvbr)
               Use constrained variable bit rate encoding.

       compression_level (comp)
           Set encoding algorithm complexity. Valid options are integers in the 0-10 range. 0
           gives the fastest encodes but lower quality, while 10 gives the highest quality but
           slowest encoding. The default is 10.

       frame_duration (framesize)
           Set maximum frame size, or duration of a frame in milliseconds. The argument must be
           exactly the following: 2.5, 5, 10, 20, 40, 60. Smaller frame sizes achieve lower
           latency but less quality at a given bitrate.  Sizes greater than 20ms are only
           interesting at fairly low bitrates.  The default is 20ms.

       packet_loss (expect-loss)
           Set expected packet loss percentage. The default is 0.

       application (N.A.)
           Set intended application type. Valid options are listed below:

           voip
               Favor improved speech intelligibility.

           audio
               Favor faithfulness to the input (the default).

           lowdelay
               Restrict to only the lowest delay modes.

       cutoff (N.A.)
           Set cutoff bandwidth in Hz. The argument must be exactly one of the following: 4000,
           6000, 8000, 12000, or 20000, corresponding to narrowband, mediumband, wideband, super
           wideband, and fullband respectively. The default is 0 (cutoff disabled).

       mapping_family (mapping_family)
           Set channel mapping family to be used by the encoder. The default value of -1 uses
           mapping family 0 for mono and stereo inputs, and mapping family 1 otherwise. The
           default also disables the surround masking and LFE bandwidth optimzations in libopus,
           and requires that the input contains 8 channels or fewer.

           Other values include 0 for mono and stereo, 1 for surround sound with masking and LFE
           bandwidth optimizations, and 255 for independent streams with an unspecified channel
           layout.

       apply_phase_inv (N.A.) (requires libopus >= 1.2)
           If set to 0, disables the use of phase inversion for intensity stereo, improving the
           quality of mono downmixes, but slightly reducing normal stereo quality. The default is
           1 (phase inversion enabled).

   libshine
       Shine Fixed-Point MP3 encoder wrapper.

       Shine is a fixed-point MP3 encoder. It has a far better performance on platforms without
       an FPU, e.g. armel CPUs, and some phones and tablets.  However, as it is more targeted on
       performance than quality, it is not on par with LAME and other production-grade encoders
       quality-wise. Also, according to the project's homepage, this encoder may not be free of
       bugs as the code was written a long time ago and the project was dead for at least 5
       years.

       This encoder only supports stereo and mono input. This is also CBR-only.

       The original project (last updated in early 2007) is at
       <http://sourceforge.net/projects/libshine-fxp/>. We only support the updated fork by the
       Savonet/Liquidsoap project at <https://github.com/savonet/shine>.

       Requires the presence of the libshine headers and library during configuration. You need
       to explicitly configure the build with "--enable-libshine".

       See also libmp3lame.

       Options

       The following options are supported by the libshine wrapper. The shineenc-equivalent of
       the options are listed in parentheses.

       b (-b)
           Set bitrate expressed in bits/s for CBR. shineenc -b option is expressed in
           kilobits/s.

   libtwolame
       TwoLAME MP2 encoder wrapper.

       Requires the presence of the libtwolame headers and library during configuration. You need
       to explicitly configure the build with "--enable-libtwolame".

       Options

       The following options are supported by the libtwolame wrapper. The twolame-equivalent
       options follow the FFmpeg ones and are in parentheses.

       b (-b)
           Set bitrate expressed in bits/s for CBR. twolame b option is expressed in kilobits/s.
           Default value is 128k.

       q (-V)
           Set quality for experimental VBR support. Maximum value range is from -50 to 50,
           useful range is from -10 to 10. The higher the value, the better the quality. This
           option is valid only using the ffmpeg command-line tool. For library interface users,
           use global_quality.

       mode (--mode)
           Set the mode of the resulting audio. Possible values:

           auto
               Choose mode automatically based on the input. This is the default.

           stereo
               Stereo

           joint_stereo
               Joint stereo

           dual_channel
               Dual channel

           mono
               Mono

       psymodel (--psyc-mode)
           Set psychoacoustic model to use in encoding. The argument must be an integer between
           -1 and 4, inclusive. The higher the value, the better the quality. The default value
           is 3.

       energy_levels (--energy)
           Enable energy levels extensions when set to 1. The default value is 0 (disabled).

       error_protection (--protect)
           Enable CRC error protection when set to 1. The default value is 0 (disabled).

       copyright (--copyright)
           Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

       original (--original)
           Set MPEG audio original flag when set to 1. The default value is 0 (disabled).

   libvo-amrwbenc
       VisualOn Adaptive Multi-Rate Wideband encoder.

       Requires the presence of the libvo-amrwbenc headers and library during configuration. You
       need to explicitly configure the build with "--enable-libvo-amrwbenc --enable-version3".

       This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can
       override it by setting strict to unofficial or lower.

       Options

       b   Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec
           will round to the nearest valid bitrate.

           6600
           8850
           12650
           14250
           15850
           18250
           19850
           23050
           23850
       dtx Allow discontinuous transmission (generate comfort noise) when set to 1. The default
           value is 0 (disabled).

   libvorbis
       libvorbis encoder wrapper.

       Requires the presence of the libvorbisenc headers and library during configuration. You
       need to explicitly configure the build with "--enable-libvorbis".

       Options

       The following options are supported by the libvorbis wrapper. The oggenc-equivalent of the
       options are listed in parentheses.

       To get a more accurate and extensive documentation of the libvorbis options, consult the
       libvorbisenc's and oggenc's documentations.  See <http://xiph.org/vorbis/>,
       <http://wiki.xiph.org/Vorbis-tools>, and oggenc(1).

       b (-b)
           Set bitrate expressed in bits/s for ABR. oggenc -b is expressed in kilobits/s.

       q (-q)
           Set constant quality setting for VBR. The value should be a float number in the range
           of -1.0 to 10.0. The higher the value, the better the quality. The default value is
           3.0.

           This option is valid only using the ffmpeg command-line tool.  For library interface
           users, use global_quality.

       cutoff (--advanced-encode-option lowpass_frequency=N)
           Set cutoff bandwidth in Hz, a value of 0 disables cutoff. oggenc's related option is
           expressed in kHz. The default value is 0 (cutoff disabled).

       minrate (-m)
           Set minimum bitrate expressed in bits/s. oggenc -m is expressed in kilobits/s.

       maxrate (-M)
           Set maximum bitrate expressed in bits/s. oggenc -M is expressed in kilobits/s. This
           only has effect on ABR mode.

       iblock (--advanced-encode-option impulse_noisetune=N)
           Set noise floor bias for impulse blocks. The value is a float number from -15.0 to
           0.0. A negative bias instructs the encoder to pay special attention to the crispness
           of transients in the encoded audio. The tradeoff for better transient response is a
           higher bitrate.

   libwavpack
       A wrapper providing WavPack encoding through libwavpack.

       Only lossless mode using 32-bit integer samples is supported currently.

       Requires the presence of the libwavpack headers and library during configuration. You need
       to explicitly configure the build with "--enable-libwavpack".

       Note that a libavcodec-native encoder for the WavPack codec exists so users can encode
       audios with this codec without using this encoder. See wavpackenc.

       Options

       wavpack command line utility's corresponding options are listed in parentheses, if any.

       frame_size (--blocksize)
           Default is 32768.

       compression_level
           Set speed vs. compression tradeoff. Acceptable arguments are listed below:

           0 (-f)
               Fast mode.

           1   Normal (default) settings.

           2 (-h)
               High quality.

           3 (-hh)
               Very high quality.

           4-8 (-hh -xEXTRAPROC)
               Same as 3, but with extra processing enabled.

               4 is the same as -x2 and 8 is the same as -x6.

   mjpeg
       Motion JPEG encoder.

       Options

       huffman
           Set the huffman encoding strategy. Possible values:

           default
               Use the default huffman tables. This is the default strategy.

           optimal
               Compute and use optimal huffman tables.

   wavpack
       WavPack lossless audio encoder.

       This is a libavcodec-native WavPack encoder. There is also an encoder based on libwavpack,
       but there is virtually no reason to use that encoder.

       See also libwavpack.

       Options

       The equivalent options for wavpack command line utility are listed in parentheses.

       Shared options

       The following shared options are effective for this encoder. Only special notes about this
       particular encoder will be documented here. For the general meaning of the options, see
       the Codec Options chapter.

       frame_size (--blocksize)
           For this encoder, the range for this option is between 128 and 131072. Default is
           automatically decided based on sample rate and number of channel.

           For the complete formula of calculating default, see libavcodec/wavpackenc.c.

       compression_level (-f, -h, -hh, and -x)
           This option's syntax is consistent with libwavpack's.

       Private options

       joint_stereo (-j)
           Set whether to enable joint stereo. Valid values are:

           on (1)
               Force mid/side audio encoding.

           off (0)
               Force left/right audio encoding.

           auto
               Let the encoder decide automatically.

       optimize_mono
           Set whether to enable optimization for mono. This option is only effective for non-
           mono streams. Available values:

           on  enabled

           off disabled

VIDEO ENCODERS

       A description of some of the currently available video encoders follows.

   Hap
       Vidvox Hap video encoder.

       Options

       format integer
           Specifies the Hap format to encode.

           hap
           hap_alpha
           hap_q

           Default value is hap.

       chunks integer
           Specifies the number of chunks to split frames into, between 1 and 64. This permits
           multithreaded decoding of large frames, potentially at the cost of data-rate. The
           encoder may modify this value to divide frames evenly.

           Default value is 1.

       compressor integer
           Specifies the second-stage compressor to use. If set to none, chunks will be limited
           to 1, as chunked uncompressed frames offer no benefit.

           none
           snappy

           Default value is snappy.

   jpeg2000
       The native jpeg 2000 encoder is lossy by default, the "-q:v" option can be used to set the
       encoding quality. Lossless encoding can be selected with "-pred 1".

       Options

       format
           Can be set to either "j2k" or "jp2" (the default) that makes it possible to store non-
           rgb pix_fmts.

   libkvazaar
       Kvazaar H.265/HEVC encoder.

       Requires the presence of the libkvazaar headers and library during configuration. You need
       to explicitly configure the build with --enable-libkvazaar.

       Options

       b   Set target video bitrate in bit/s and enable rate control.

       kvazaar-params
           Set kvazaar parameters as a list of name=value pairs separated by commas (,). See
           kvazaar documentation for a list of options.

   libopenh264
       Cisco libopenh264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libopenh264 headers and library during
       configuration. You need to explicitly configure the build with "--enable-libopenh264". The
       library is detected using pkg-config.

       For more information about the library see <http://www.openh264.org>.

       Options

       The following FFmpeg global options affect the configurations of the libopenh264 encoder.

       b   Set the bitrate (as a number of bits per second).

       g   Set the GOP size.

       maxrate
           Set the max bitrate (as a number of bits per second).

       flags +global_header
           Set global header in the bitstream.

       slices
           Set the number of slices, used in parallelized encoding. Default value is 0. This is
           only used when slice_mode is set to fixed.

       slice_mode
           Set slice mode. Can assume one of the following possible values:

           fixed
               a fixed number of slices

           rowmb
               one slice per row of macroblocks

           auto
               automatic number of slices according to number of threads

           dyn dynamic slicing

           Default value is auto.

       loopfilter
           Enable loop filter, if set to 1 (automatically enabled). To disable set a value of 0.

       profile
           Set profile restrictions. If set to the value of main enable CABAC (set the
           "SEncParamExt.iEntropyCodingModeFlag" flag to 1).

       max_nal_size
           Set maximum NAL size in bytes.

       allow_skip_frames
           Allow skipping frames to hit the target bitrate if set to 1.

   libtheora
       libtheora Theora encoder wrapper.

       Requires the presence of the libtheora headers and library during configuration. You need
       to explicitly configure the build with "--enable-libtheora".

       For more information about the libtheora project see <http://www.theora.org/>.

       Options

       The following global options are mapped to internal libtheora options which affect the
       quality and the bitrate of the encoded stream.

       b   Set the video bitrate in bit/s for CBR (Constant Bit Rate) mode.  In case VBR
           (Variable Bit Rate) mode is enabled this option is ignored.

       flags
           Used to enable constant quality mode (VBR) encoding through the qscale flag, and to
           enable the "pass1" and "pass2" modes.

       g   Set the GOP size.

       global_quality
           Set the global quality as an integer in lambda units.

           Only relevant when VBR mode is enabled with "flags +qscale". The value is converted to
           QP units by dividing it by "FF_QP2LAMBDA", clipped in the [0 - 10] range, and then
           multiplied by 6.3 to get a value in the native libtheora range [0-63]. A higher value
           corresponds to a higher quality.

       q   Enable VBR mode when set to a non-negative value, and set constant quality value as a
           double floating point value in QP units.

           The value is clipped in the [0-10] range, and then multiplied by 6.3 to get a value in
           the native libtheora range [0-63].

           This option is valid only using the ffmpeg command-line tool. For library interface
           users, use global_quality.

       Examples

       ·   Set maximum constant quality (VBR) encoding with ffmpeg:

                   ffmpeg -i INPUT -codec:v libtheora -q:v 10 OUTPUT.ogg

       ·   Use ffmpeg to convert a CBR 1000 kbps Theora video stream:

                   ffmpeg -i INPUT -codec:v libtheora -b:v 1000k OUTPUT.ogg

   libvpx
       VP8/VP9 format supported through libvpx.

       Requires the presence of the libvpx headers and library during configuration.  You need to
       explicitly configure the build with "--enable-libvpx".

       Options

       The following options are supported by the libvpx wrapper. The vpxenc-equivalent options
       or values are listed in parentheses for easy migration.

       To reduce the duplication of documentation, only the private options and some others
       requiring special attention are documented here. For the documentation of the undocumented
       generic options, see the Codec Options chapter.

       To get more documentation of the libvpx options, invoke the command ffmpeg -h
       encoder=libvpx, ffmpeg -h encoder=libvpx-vp9 or vpxenc --help. Further information is
       available in the libvpx API documentation.

       b (target-bitrate)
           Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while
           vpxenc's target-bitrate is in kilobits/s.

       g (kf-max-dist)
       keyint_min (kf-min-dist)
       qmin (min-q)
       qmax (max-q)
       bufsize (buf-sz, buf-optimal-sz)
           Set ratecontrol buffer size (in bits). Note vpxenc's options are specified in
           milliseconds, the libvpx wrapper converts this value as follows: "buf-sz = bufsize *
           1000 / bitrate", "buf-optimal-sz = bufsize * 1000 / bitrate * 5 / 6".

       rc_init_occupancy (buf-initial-sz)
           Set number of bits which should be loaded into the rc buffer before decoding starts.
           Note vpxenc's option is specified in milliseconds, the libvpx wrapper converts this
           value as follows: "rc_init_occupancy * 1000 / bitrate".

       undershoot-pct
           Set datarate undershoot (min) percentage of the target bitrate.

       overshoot-pct
           Set datarate overshoot (max) percentage of the target bitrate.

       skip_threshold (drop-frame)
       qcomp (bias-pct)
       maxrate (maxsection-pct)
           Set GOP max bitrate in bits/s. Note vpxenc's option is specified as a percentage of
           the target bitrate, the libvpx wrapper converts this value as follows: "(maxrate * 100
           / bitrate)".

       minrate (minsection-pct)
           Set GOP min bitrate in bits/s. Note vpxenc's option is specified as a percentage of
           the target bitrate, the libvpx wrapper converts this value as follows: "(minrate * 100
           / bitrate)".

       minrate, maxrate, b end-usage=cbr
           "(minrate == maxrate == bitrate)".

       crf (end-usage=cq, cq-level)
       tune (tune)
           psnr (psnr)
           ssim (ssim)
       quality, deadline (deadline)
           best
               Use best quality deadline. Poorly named and quite slow, this option should be
               avoided as it may give worse quality output than good.

           good
               Use good quality deadline. This is a good trade-off between speed and quality when
               used with the cpu-used option.

           realtime
               Use realtime quality deadline.

       speed, cpu-used (cpu-used)
           Set quality/speed ratio modifier. Higher values speed up the encode at the cost of
           quality.

       nr (noise-sensitivity)
       static-thresh
           Set a change threshold on blocks below which they will be skipped by the encoder.

       slices (token-parts)
           Note that FFmpeg's slices option gives the total number of partitions, while vpxenc's
           token-parts is given as "log2(partitions)".

       max-intra-rate
           Set maximum I-frame bitrate as a percentage of the target bitrate. A value of 0 means
           unlimited.

       force_key_frames
           "VPX_EFLAG_FORCE_KF"

       Alternate reference frame related
           auto-alt-ref
               Enable use of alternate reference frames (2-pass only).

           arnr-max-frames
               Set altref noise reduction max frame count.

           arnr-type
               Set altref noise reduction filter type: backward, forward, centered.

           arnr-strength
               Set altref noise reduction filter strength.

           rc-lookahead, lag-in-frames (lag-in-frames)
               Set number of frames to look ahead for frametype and ratecontrol.

       error-resilient
           Enable error resiliency features.

       VP9-specific options
           lossless
               Enable lossless mode.

           tile-columns
               Set number of tile columns to use. Note this is given as "log2(tile_columns)". For
               example, 8 tile columns would be requested by setting the tile-columns option to
               3.

           tile-rows
               Set number of tile rows to use. Note this is given as "log2(tile_rows)".  For
               example, 4 tile rows would be requested by setting the tile-rows option to 2.

           frame-parallel
               Enable frame parallel decodability features.

           aq-mode
               Set adaptive quantization mode (0: off (default), 1: variance 2: complexity, 3:
               cyclic refresh, 4: equator360).

           colorspace color-space
               Set input color space. The VP9 bitstream supports signaling the following
               colorspaces:

               rgb sRGB
               bt709 bt709
               unspecified unknown
               bt470bg bt601
               smpte170m smpte170
               smpte240m smpte240
               bt2020_ncl bt2020
           row-mt boolean
               Enable row based multi-threading.

           tune-content
               Set content type: default (0), screen (1), film (2).

           corpus-complexity
               Corpus VBR mode is a variant of standard VBR where the complexity distribution
               midpoint is passed in rather than calculated for a specific clip or chunk.

               The valid range is [0, 10000]. 0 (default) uses standard VBR.

       For more information about libvpx see: <http://www.webmproject.org/>

   libwebp
       libwebp WebP Image encoder wrapper

       libwebp is Google's official encoder for WebP images. It can encode in either lossy or
       lossless mode. Lossy images are essentially a wrapper around a VP8 frame. Lossless images
       are a separate codec developed by Google.

       Pixel Format

       Currently, libwebp only supports YUV420 for lossy and RGB for lossless due to limitations
       of the format and libwebp. Alpha is supported for either mode.  Because of API
       limitations, if RGB is passed in when encoding lossy or YUV is passed in for encoding
       lossless, the pixel format will automatically be converted using functions from libwebp.
       This is not ideal and is done only for convenience.

       Options

       -lossless boolean
           Enables/Disables use of lossless mode. Default is 0.

       -compression_level integer
           For lossy, this is a quality/speed tradeoff. Higher values give better quality for a
           given size at the cost of increased encoding time. For lossless, this is a size/speed
           tradeoff. Higher values give smaller size at the cost of increased encoding time. More
           specifically, it controls the number of extra algorithms and compression tools used,
           and varies the combination of these tools. This maps to the method option in libwebp.
           The valid range is 0 to 6.  Default is 4.

       -qscale float
           For lossy encoding, this controls image quality, 0 to 100. For lossless encoding, this
           controls the effort and time spent at compressing more. The default value is 75. Note
           that for usage via libavcodec, this option is called global_quality and must be
           multiplied by FF_QP2LAMBDA.

       -preset type
           Configuration preset. This does some automatic settings based on the general type of
           the image.

           none
               Do not use a preset.

           default
               Use the encoder default.

           picture
               Digital picture, like portrait, inner shot

           photo
               Outdoor photograph, with natural lighting

           drawing
               Hand or line drawing, with high-contrast details

           icon
               Small-sized colorful images

           text
               Text-like

   libx264, libx264rgb
       x264 H.264/MPEG-4 AVC encoder wrapper.

       This encoder requires the presence of the libx264 headers and library during
       configuration. You need to explicitly configure the build with "--enable-libx264".

       libx264 supports an impressive number of features, including 8x8 and 4x4 adaptive spatial
       transform, adaptive B-frame placement, CAVLC/CABAC entropy coding, interlacing (MBAFF),
       lossless mode, psy optimizations for detail retention (adaptive quantization, psy-RD, psy-
       trellis).

       Many libx264 encoder options are mapped to FFmpeg global codec options, while unique
       encoder options are provided through private options. Additionally the x264opts and
       x264-params private options allows one to pass a list of key=value tuples as accepted by
       the libx264 "x264_param_parse" function.

       The x264 project website is at <http://www.videolan.org/developers/x264.html>.

       The libx264rgb encoder is the same as libx264, except it accepts packed RGB pixel formats
       as input instead of YUV.

       Supported Pixel Formats

       x264 supports 8- to 10-bit color spaces. The exact bit depth is controlled at x264's
       configure time. FFmpeg only supports one bit depth in one particular build. In other
       words, it is not possible to build one FFmpeg with multiple versions of x264 with
       different bit depths.

       Options

       The following options are supported by the libx264 wrapper. The x264-equivalent options or
       values are listed in parentheses for easy migration.

       To reduce the duplication of documentation, only the private options and some others
       requiring special attention are documented here. For the documentation of the undocumented
       generic options, see the Codec Options chapter.

       To get a more accurate and extensive documentation of the libx264 options, invoke the
       command x264 --fullhelp or consult the libx264 documentation.

       b (bitrate)
           Set bitrate in bits/s. Note that FFmpeg's b option is expressed in bits/s, while
           x264's bitrate is in kilobits/s.

       bf (bframes)
       g (keyint)
       qmin (qpmin)
           Minimum quantizer scale.

       qmax (qpmax)
           Maximum quantizer scale.

       qdiff (qpstep)
           Maximum difference between quantizer scales.

       qblur (qblur)
           Quantizer curve blur

       qcomp (qcomp)
           Quantizer curve compression factor

       refs (ref)
           Number of reference frames each P-frame can use. The range is from 0-16.

       sc_threshold (scenecut)
           Sets the threshold for the scene change detection.

       trellis (trellis)
           Performs Trellis quantization to increase efficiency. Enabled by default.

       nr  (nr)
       me_range (merange)
           Maximum range of the motion search in pixels.

       me_method (me)
           Set motion estimation method. Possible values in the decreasing order of speed:

           dia (dia)
           epzs (dia)
               Diamond search with radius 1 (fastest). epzs is an alias for dia.

           hex (hex)
               Hexagonal search with radius 2.

           umh (umh)
               Uneven multi-hexagon search.

           esa (esa)
               Exhaustive search.

           tesa (tesa)
               Hadamard exhaustive search (slowest).

       forced-idr
           Normally, when forcing a I-frame type, the encoder can select any type of I-frame.
           This option forces it to choose an IDR-frame.

       subq (subme)
           Sub-pixel motion estimation method.

       b_strategy (b-adapt)
           Adaptive B-frame placement decision algorithm. Use only on first-pass.

       keyint_min (min-keyint)
           Minimum GOP size.

       coder
           Set entropy encoder. Possible values:

           ac  Enable CABAC.

           vlc Enable CAVLC and disable CABAC. It generates the same effect as x264's --no-cabac
               option.

       cmp Set full pixel motion estimation comparison algorithm. Possible values:

           chroma
               Enable chroma in motion estimation.

           sad Ignore chroma in motion estimation. It generates the same effect as x264's
               --no-chroma-me option.

       threads (threads)
           Number of encoding threads.

       thread_type
           Set multithreading technique. Possible values:

           slice
               Slice-based multithreading. It generates the same effect as x264's
               --sliced-threads option.

           frame
               Frame-based multithreading.

       flags
           Set encoding flags. It can be used to disable closed GOP and enable open GOP by
           setting it to "-cgop". The result is similar to the behavior of x264's --open-gop
           option.

       rc_init_occupancy (vbv-init)
       preset (preset)
           Set the encoding preset.

       tune (tune)
           Set tuning of the encoding params.

       profile (profile)
           Set profile restrictions.

       fastfirstpass
           Enable fast settings when encoding first pass, when set to 1. When set to 0, it has
           the same effect of x264's --slow-firstpass option.

       crf (crf)
           Set the quality for constant quality mode.

       crf_max (crf-max)
           In CRF mode, prevents VBV from lowering quality beyond this point.

       qp (qp)
           Set constant quantization rate control method parameter.

       aq-mode (aq-mode)
           Set AQ method. Possible values:

           none (0)
               Disabled.

           variance (1)
               Variance AQ (complexity mask).

           autovariance (2)
               Auto-variance AQ (experimental).

       aq-strength (aq-strength)
           Set AQ strength, reduce blocking and blurring in flat and textured areas.

       psy Use psychovisual optimizations when set to 1. When set to 0, it has the same effect as
           x264's --no-psy option.

       psy-rd  (psy-rd)
           Set strength of psychovisual optimization, in psy-rd:psy-trellis format.

       rc-lookahead (rc-lookahead)
           Set number of frames to look ahead for frametype and ratecontrol.

       weightb
           Enable weighted prediction for B-frames when set to 1. When set to 0, it has the same
           effect as x264's --no-weightb option.

       weightp (weightp)
           Set weighted prediction method for P-frames. Possible values:

           none (0)
               Disabled

           simple (1)
               Enable only weighted refs

           smart (2)
               Enable both weighted refs and duplicates

       ssim (ssim)
           Enable calculation and printing SSIM stats after the encoding.

       intra-refresh (intra-refresh)
           Enable the use of Periodic Intra Refresh instead of IDR frames when set to 1.

       avcintra-class (class)
           Configure the encoder to generate AVC-Intra.  Valid values are 50,100 and 200

       bluray-compat (bluray-compat)
           Configure the encoder to be compatible with the bluray standard.  It is a shorthand
           for setting "bluray-compat=1 force-cfr=1".

       b-bias (b-bias)
           Set the influence on how often B-frames are used.

       b-pyramid (b-pyramid)
           Set method for keeping of some B-frames as references. Possible values:

           none (none)
               Disabled.

           strict (strict)
               Strictly hierarchical pyramid.

           normal (normal)
               Non-strict (not Blu-ray compatible).

       mixed-refs
           Enable the use of one reference per partition, as opposed to one reference per
           macroblock when set to 1. When set to 0, it has the same effect as x264's
           --no-mixed-refs option.

       8x8dct
           Enable adaptive spatial transform (high profile 8x8 transform) when set to 1. When set
           to 0, it has the same effect as x264's --no-8x8dct option.

       fast-pskip
           Enable early SKIP detection on P-frames when set to 1. When set to 0, it has the same
           effect as x264's --no-fast-pskip option.

       aud (aud)
           Enable use of access unit delimiters when set to 1.

       mbtree
           Enable use macroblock tree ratecontrol when set to 1. When set to 0, it has the same
           effect as x264's --no-mbtree option.

       deblock (deblock)
           Set loop filter parameters, in alpha:beta form.

       cplxblur (cplxblur)
           Set fluctuations reduction in QP (before curve compression).

       partitions (partitions)
           Set partitions to consider as a comma-separated list of. Possible values in the list:

           p8x8
               8x8 P-frame partition.

           p4x4
               4x4 P-frame partition.

           b8x8
               4x4 B-frame partition.

           i8x8
               8x8 I-frame partition.

           i4x4
               4x4 I-frame partition.  (Enabling p4x4 requires p8x8 to be enabled. Enabling i8x8
               requires adaptive spatial transform (8x8dct option) to be enabled.)

           none (none)
               Do not consider any partitions.

           all (all)
               Consider every partition.

       direct-pred (direct)
           Set direct MV prediction mode. Possible values:

           none (none)
               Disable MV prediction.

           spatial (spatial)
               Enable spatial predicting.

           temporal (temporal)
               Enable temporal predicting.

           auto (auto)
               Automatically decided.

       slice-max-size (slice-max-size)
           Set the limit of the size of each slice in bytes. If not specified but RTP payload
           size (ps) is specified, that is used.

       stats (stats)
           Set the file name for multi-pass stats.

       nal-hrd (nal-hrd)
           Set signal HRD information (requires vbv-bufsize to be set).  Possible values:

           none (none)
               Disable HRD information signaling.

           vbr (vbr)
               Variable bit rate.

           cbr (cbr)
               Constant bit rate (not allowed in MP4 container).

       x264opts (N.A.)
           Set any x264 option, see x264 --fullhelp for a list.

           Argument is a list of key=value couples separated by ":". In filter and psy-rd options
           that use ":" as a separator themselves, use "," instead. They accept it as well since
           long ago but this is kept undocumented for some reason.

           For example to specify libx264 encoding options with ffmpeg:

                   ffmpeg -i foo.mpg -c:v libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv

       a53cc boolean
           Import closed captions (which must be ATSC compatible format) into output.  Only the
           mpeg2 and h264 decoders provide these. Default is 1 (on).

       x264-params (N.A.)
           Override the x264 configuration using a :-separated list of key=value parameters.

           This option is functionally the same as the x264opts, but is duplicated for
           compatibility with the Libav fork.

           For example to specify libx264 encoding options with ffmpeg:

                   ffmpeg -i INPUT -c:v libx264 -x264-params level=30:bframes=0:weightp=0:\
                   cabac=0:ref=1:vbv-maxrate=768:vbv-bufsize=2000:analyse=all:me=umh:\
                   no-fast-pskip=1:subq=6:8x8dct=0:trellis=0 OUTPUT

       Encoding ffpresets for common usages are provided so they can be used with the general
       presets system (e.g. passing the pre option).

   libx265
       x265 H.265/HEVC encoder wrapper.

       This encoder requires the presence of the libx265 headers and library during
       configuration. You need to explicitly configure the build with --enable-libx265.

       Options

       preset
           Set the x265 preset.

       tune
           Set the x265 tune parameter.

       profile
           Set profile restrictions.

       crf Set the quality for constant quality mode.

       forced-idr
           Normally, when forcing a I-frame type, the encoder can select any type of I-frame.
           This option forces it to choose an IDR-frame.

       x265-params
           Set x265 options using a list of key=value couples separated by ":". See x265 --help
           for a list of options.

           For example to specify libx265 encoding options with -x265-params:

                   ffmpeg -i input -c:v libx265 -x265-params crf=26:psy-rd=1 output.mp4

   libxvid
       Xvid MPEG-4 Part 2 encoder wrapper.

       This encoder requires the presence of the libxvidcore headers and library during
       configuration. You need to explicitly configure the build with "--enable-libxvid
       --enable-gpl".

       The native "mpeg4" encoder supports the MPEG-4 Part 2 format, so users can encode to this
       format without this library.

       Options

       The following options are supported by the libxvid wrapper. Some of the following options
       are listed but are not documented, and correspond to shared codec options. See the Codec
       Options chapter for their documentation. The other shared options which are not listed
       have no effect for the libxvid encoder.

       b
       g
       qmin
       qmax
       mpeg_quant
       threads
       bf
       b_qfactor
       b_qoffset
       flags
           Set specific encoding flags. Possible values:

           mv4 Use four motion vector by macroblock.

           aic Enable high quality AC prediction.

           gray
               Only encode grayscale.

           gmc Enable the use of global motion compensation (GMC).

           qpel
               Enable quarter-pixel motion compensation.

           cgop
               Enable closed GOP.

           global_header
               Place global headers in extradata instead of every keyframe.

       trellis
       me_method
           Set motion estimation method. Possible values in decreasing order of speed and
           increasing order of quality:

           zero
               Use no motion estimation (default).

           phods
           x1
           log Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement
               for 16x16 blocks. x1 and log are aliases for phods.

           epzs
               Enable all of the things described above, plus advanced diamond zonal search for
               8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma
               planes.

           full
               Enable all of the things described above, plus extended 16x16 and 8x8 blocks
               search.

       mbd Set macroblock decision algorithm. Possible values in the increasing order of quality:

           simple
               Use macroblock comparing function algorithm (default).

           bits
               Enable rate distortion-based half pixel and quarter pixel refinement for 16x16
               blocks.

           rd  Enable all of the things described above, plus rate distortion-based half pixel
               and quarter pixel refinement for 8x8 blocks, and rate distortion-based search
               using square pattern.

       lumi_aq
           Enable lumi masking adaptive quantization when set to 1. Default is 0 (disabled).

       variance_aq
           Enable variance adaptive quantization when set to 1. Default is 0 (disabled).

           When combined with lumi_aq, the resulting quality will not be better than any of the
           two specified individually. In other words, the resulting quality will be the worse
           one of the two effects.

       ssim
           Set structural similarity (SSIM) displaying method. Possible values:

           off Disable displaying of SSIM information.

           avg Output average SSIM at the end of encoding to stdout. The format of showing the
               average SSIM is:

                       Average SSIM: %f

               For users who are not familiar with C, %f means a float number, or a decimal (e.g.
               0.939232).

           frame
               Output both per-frame SSIM data during encoding and average SSIM at the end of
               encoding to stdout. The format of per-frame information is:

                              SSIM: avg: %1.3f min: %1.3f max: %1.3f

               For users who are not familiar with C, %1.3f means a float number rounded to 3
               digits after the dot (e.g. 0.932).

       ssim_acc
           Set SSIM accuracy. Valid options are integers within the range of 0-4, while 0 gives
           the most accurate result and 4 computes the fastest.

   mpeg2
       MPEG-2 video encoder.

       Options

       seq_disp_ext integer
           Specifies if the encoder should write a sequence_display_extension to the output.

           -1
           auto
               Decide automatically to write it or not (this is the default) by checking if the
               data to be written is different from the default or unspecified values.

           0
           never
               Never write it.

           1
           always
               Always write it.

       video_format integer
           Specifies the video_format written into the sequence display extension indicating the
           source of the video pictures. The default is unspecified, can be component, pal, ntsc,
           secam or mac.  For maximum compatibility, use component.

   png
       PNG image encoder.

       Private options

       dpi integer
           Set physical density of pixels, in dots per inch, unset by default

       dpm integer
           Set physical density of pixels, in dots per meter, unset by default

   ProRes
       Apple ProRes encoder.

       FFmpeg contains 2 ProRes encoders, the prores-aw and prores-ks encoder.  The used encoder
       can be chosen with the "-vcodec" option.

       Private Options for prores-ks

       profile integer
           Select the ProRes profile to encode

           proxy
           lt
           standard
           hq
           4444
           4444xq
       quant_mat integer
           Select quantization matrix.

           auto
           default
           proxy
           lt
           standard
           hq

           If set to auto, the matrix matching the profile will be picked.  If not set, the
           matrix providing the highest quality, default, will be picked.

       bits_per_mb integer
           How many bits to allot for coding one macroblock. Different profiles use between 200
           and 2400 bits per macroblock, the maximum is 8000.

       mbs_per_slice integer
           Number of macroblocks in each slice (1-8); the default value (8) should be good in
           almost all situations.

       vendor string
           Override the 4-byte vendor ID.  A custom vendor ID like apl0 would claim the stream
           was produced by the Apple encoder.

       alpha_bits integer
           Specify number of bits for alpha component.  Possible values are 0, 8 and 16.  Use 0
           to disable alpha plane coding.

       Speed considerations

       In the default mode of operation the encoder has to honor frame constraints (i.e. not
       produce frames with size bigger than requested) while still making output picture as good
       as possible.  A frame containing a lot of small details is harder to compress and the
       encoder would spend more time searching for appropriate quantizers for each slice.

       Setting a higher bits_per_mb limit will improve the speed.

       For the fastest encoding speed set the qscale parameter (4 is the recommended value) and
       do not set a size constraint.

   QSV encoders
       The family of Intel QuickSync Video encoders (MPEG-2, H.264 and HEVC)

       The ratecontrol method is selected as follows:

       ·   When global_quality is specified, a quality-based mode is used.  Specifically this
           means either

           -   CQP - constant quantizer scale, when the qscale codec flag is also set (the
               -qscale ffmpeg option).

           -   LA_ICQ - intelligent constant quality with lookahead, when the look_ahead option
               is also set.

           -   ICQ -- intelligent constant quality otherwise.

       ·   Otherwise, a bitrate-based mode is used. For all of those, you should specify at least
           the desired average bitrate with the b option.

           -   LA - VBR with lookahead, when the look_ahead option is specified.

           -   VCM - video conferencing mode, when the vcm option is set.

           -   CBR - constant bitrate, when maxrate is specified and equal to the average
               bitrate.

           -   VBR - variable bitrate, when maxrate is specified, but is higher than the average
               bitrate.

           -   AVBR - average VBR mode, when maxrate is not specified. This mode is further
               configured by the avbr_accuracy and avbr_convergence options.

       Note that depending on your system, a different mode than the one you specified may be
       selected by the encoder. Set the verbosity level to verbose or higher to see the actual
       settings used by the QSV runtime.

       Additional libavcodec global options are mapped to MSDK options as follows:

       ·   g/gop_size -> GopPicSize

       ·   bf/max_b_frames+1 -> GopRefDist

       ·   rc_init_occupancy/rc_initial_buffer_occupancy -> InitialDelayInKB

       ·   slices -> NumSlice

       ·   refs -> NumRefFrame

       ·   b_strategy/b_frame_strategy -> BRefType

       ·   cgop/CLOSED_GOP codec flag -> GopOptFlag

       ·   For the CQP mode, the i_qfactor/i_qoffset and b_qfactor/b_qoffset set the difference
           between QPP and QPI, and QPP and QPB respectively.

       ·   Setting the coder option to the value vlc will make the H.264 encoder use CAVLC
           instead of CABAC.

   snow
       Options

       iterative_dia_size
           dia size for the iterative motion estimation

   VAAPI encoders
       Wrappers for hardware encoders accessible via VAAPI.

       These encoders only accept input in VAAPI hardware surfaces.  If you have input in
       software frames, use the hwupload filter to upload them to the GPU.

       The following standard libavcodec options are used:

       ·   g / gop_size

       ·   bf / max_b_frames

       ·   profile

           If not set, this will be determined automatically from the format of the input frames
           and the profiles supported by the driver.

       ·   level

       ·   b / bit_rate

       ·   maxrate / rc_max_rate

       ·   bufsize / rc_buffer_size

       ·   rc_init_occupancy / rc_initial_buffer_occupancy

       ·   compression_level

           Speed / quality tradeoff: higher values are faster / worse quality.

       ·   q / global_quality

           Size / quality tradeoff: higher values are smaller / worse quality.

       ·   qmin

       ·   qmax

       ·   i_qfactor / i_quant_factor

       ·   i_qoffset / i_quant_offset

       ·   b_qfactor / b_quant_factor

       ·   b_qoffset / b_quant_offset

       ·   slices

       All encoders support the following options:

       ·   low_power

           Some drivers/platforms offer a second encoder for some codecs intended to use less
           power than the default encoder; setting this option will attempt to use that encoder.
           Note that it may support a reduced feature set, so some other options may not be
           available in this mode.

       Each encoder also has its own specific options:

       h264_vaapi
           profile sets the value of profile_idc and the constraint_set*_flags.  level sets the
           value of level_idc.

           coder
               Set entropy encoder (default is cabac).  Possible values:

               ac
               cabac
                   Use CABAC.

               vlc
               cavlc
                   Use CAVLC.

           aud Include access unit delimiters in the stream (not included by default).

           sei Set SEI message types to include.  Some combination of the following values:

               identifier
                   Include a user_data_unregistered message containing information about the
                   encoder.

               timing
                   Include picture timing parameters (buffering_period and pic_timing messages).

               recovery_point
                   Include recovery points where appropriate (recovery_point messages).

       hevc_vaapi
           profile and level set the values of general_profile_idc and general_level_idc
           respectively.

           aud Include access unit delimiters in the stream (not included by default).

           tier
               Set general_tier_flag.  This may affect the level chosen for the stream if it is
               not explicitly specified.

           sei Set SEI message types to include.  Some combination of the following values:

               hdr Include HDR metadata if the input frames have it
                   (mastering_display_colour_volume and content_light_level messages).

       mjpeg_vaapi
           Only baseline DCT encoding is supported.  The encoder always uses the standard
           quantisation and huffman tables - global_quality scales the standard quantisation
           table (range 1-100).

           For YUV, 4:2:0, 4:2:2 and 4:4:4 subsampling modes are supported.  RGB is also
           supported, and will create an RGB JPEG.

           jfif
               Include JFIF header in each frame (not included by default).

           huffman
               Include standard huffman tables (on by default).  Turning this off will save a few
               hundred bytes in each output frame, but may lose compatibility with some JPEG
               decoders which don't fully handle MJPEG.

       mpeg2_vaapi
           profile and level set the value of profile_and_level_indication.

       vp8_vaapi
           B-frames are not supported.

           global_quality sets the q_idx used for non-key frames (range 0-127).

           loop_filter_level
           loop_filter_sharpness
               Manually set the loop filter parameters.

       vp9_vaapi
           global_quality sets the q_idx used for P-frames (range 0-255).

           loop_filter_level
           loop_filter_sharpness
               Manually set the loop filter parameters.

           B-frames are supported, but the output stream is always in encode order rather than
           display order.  If B-frames are enabled, it may be necessary to use the
           vp9_raw_reorder bitstream filter to modify the output stream to display frames in the
           correct order.

           Only normal frames are produced - the vp9_superframe bitstream filter may be required
           to produce a stream usable with all decoders.

   vc2
       SMPTE VC-2 (previously BBC Dirac Pro). This codec was primarily aimed at professional
       broadcasting but since it supports yuv420, yuv422 and yuv444 at 8 (limited range or full
       range), 10 or 12 bits, this makes it suitable for other tasks which require low overhead
       and low compression (like screen recording).

       Options

       b   Sets target video bitrate. Usually that's around 1:6 of the uncompressed video bitrate
           (e.g. for 1920x1080 50fps yuv422p10 that's around 400Mbps). Higher values (close to
           the uncompressed bitrate) turn on lossless compression mode.

       field_order
           Enables field coding when set (e.g. to tt - top field first) for interlaced inputs.
           Should increase compression with interlaced content as it splits the fields and
           encodes each separately.

       wavelet_depth
           Sets the total amount of wavelet transforms to apply, between 1 and 5 (default).
           Lower values reduce compression and quality. Less capable decoders may not be able to
           handle values of wavelet_depth over 3.

       wavelet_type
           Sets the transform type. Currently only 5_3 (LeGall) and 9_7 (Deslauriers-Dubuc) are
           implemented, with 9_7 being the one with better compression and thus is the default.

       slice_width
       slice_height
           Sets the slice size for each slice. Larger values result in better compression.  For
           compatibility with other more limited decoders use slice_width of 32 and slice_height
           of 8.

       tolerance
           Sets the undershoot tolerance of the rate control system in percent. This is to
           prevent an expensive search from being run.

       qm  Sets the quantization matrix preset to use by default or when wavelet_depth is set to
           5

           -   default Uses the default quantization matrix from the specifications, extended
               with values for the fifth level. This provides a good balance between keeping
               detail and omitting artifacts.

           -   flat Use a completely zeroed out quantization matrix. This increases PSNR but
               might reduce perception. Use in bogus benchmarks.

           -   color Reduces detail but attempts to preserve color at extremely low bitrates.

   libxavs2
       xavs2 AVS2-P2/IEEE1857.4 encoder wrapper.

       This encoder requires the presence of the libxavs2 headers and library during
       configuration. You need to explicitly configure the build with --enable-libxavs2.

       Options

       lcu_row_threads
           Set the number of parallel threads for rows from 1 to 8 (default 5).

       initial_qp
           Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set
           the initial qp for the first frame.

       qp  Set the xavs2 quantization parameter from 1 to 63 (default 34). This is used to set
           the qp value under constant-QP mode.

       max_qp
           Set the max qp for rate control from 1 to 63 (default 55).

       min_qp
           Set the min qp for rate control from 1 to 63 (default 20).

       speed_level
           Set the Speed level from 0 to 9 (default 0). Higher is better but slower.

       log_level
           Set the log level from -1 to 3 (default 0). -1: none, 0: error, 1: warning, 2: info,
           3: debug.

       xavs2-params
           Set xavs2 options using a list of key=value couples separated by ":".

           For example to specify libxavs2 encoding options with -xavs2-params:

                   ffmpeg -i input -c:v libxavs2 -xavs2-params preset_level=5 output.avs2

SUBTITLES ENCODERS

   dvdsub
       This codec encodes the bitmap subtitle format that is used in DVDs.  Typically they are
       stored in VOBSUB file pairs (*.idx + *.sub), and they can also be used in Matroska files.

       Options

       even_rows_fix
           When set to 1, enable a work-around that makes the number of pixel rows even in all
           subtitles.  This fixes a problem with some players that cut off the bottom row if the
           number is odd.  The work-around just adds a fully transparent row if needed.  The
           overhead is low, typically one byte per subtitle on average.

           By default, this work-around is disabled.

BITSTREAM FILTERS

       When you configure your FFmpeg build, all the supported bitstream filters are enabled by
       default. You can list all available ones using the configure option "--list-bsfs".

       You can disable all the bitstream filters using the configure option "--disable-bsfs", and
       selectively enable any bitstream filter using the option "--enable-bsf=BSF", or you can
       disable a particular bitstream filter using the option "--disable-bsf=BSF".

       The option "-bsfs" of the ff* tools will display the list of all the supported bitstream
       filters included in your build.

       The ff* tools have a -bsf option applied per stream, taking a comma-separated list of
       filters, whose parameters follow the filter name after a '='.

               ffmpeg -i INPUT -c:v copy -bsf:v filter1[=opt1=str1:opt2=str2][,filter2] OUTPUT

       Below is a description of the currently available bitstream filters, with their
       parameters, if any.

   aac_adtstoasc
       Convert MPEG-2/4 AAC ADTS to an MPEG-4 Audio Specific Configuration bitstream.

       This filter creates an MPEG-4 AudioSpecificConfig from an MPEG-2/4 ADTS header and removes
       the ADTS header.

       This filter is required for example when copying an AAC stream from a raw ADTS AAC or an
       MPEG-TS container to MP4A-LATM, to an FLV file, or to MOV/MP4 files and related formats
       such as 3GP or M4A. Please note that it is auto-inserted for MP4A-LATM and MOV/MP4 and
       related formats.

   av1_metadata
       Modify metadata embedded in an AV1 stream.

       td  Insert or remove temporal delimiter OBUs in all temporal units of the stream.

           insert
               Insert a TD at the beginning of every TU which does not already have one.

           remove
               Remove the TD from the beginning of every TU which has one.

       color_primaries
       transfer_characteristics
       matrix_coefficients
           Set the color description fields in the stream (see AV1 section 6.4.2).

       color_range
           Set the color range in the stream (see AV1 section 6.4.2; note that this cannot be set
           for streams using BT.709 primaries, sRGB transfer characteristic and identity (RGB)
           matrix coefficients).

           tv  Limited range.

           pc  Full range.

       chroma_sample_position
           Set the chroma sample location in the stream (see AV1 section 6.4.2).  This can only
           be set for 4:2:0 streams.

           vertical
               Left position (matching the default in MPEG-2 and H.264).

           colocated
               Top-left position.

       tick_rate
           Set the tick rate (num_units_in_display_tick / time_scale) in the timing info in the
           sequence header.

       num_ticks_per_picture
           Set the number of ticks in each picture, to indicate that the stream has a fixed
           framerate.  Ignored if tick_rate is not also set.

   chomp
       Remove zero padding at the end of a packet.

   dca_core
       Extract the core from a DCA/DTS stream, dropping extensions such as DTS-HD.

   dump_extra
       Add extradata to the beginning of the filtered packets.

       freq
           The additional argument specifies which packets should be filtered.  It accepts the
           values:

           k
           keyframe
               add extradata to all key packets

           e
           all add extradata to all packets

       If not specified it is assumed e.

       For example the following ffmpeg command forces a global header (thus disabling individual
       packet headers) in the H.264 packets generated by the "libx264" encoder, but corrects them
       by adding the header stored in extradata to the key packets:

               ffmpeg -i INPUT -map 0 -flags:v +global_header -c:v libx264 -bsf:v dump_extra out.ts

   eac3_core
       Extract the core from a E-AC-3 stream, dropping extra channels.

   extract_extradata
       Extract the in-band extradata.

       Certain codecs allow the long-term headers (e.g. MPEG-2 sequence headers, or H.264/HEVC
       (VPS/)SPS/PPS) to be transmitted either "in-band" (i.e. as a part of the bitstream
       containing the coded frames) or "out of band" (e.g. on the container level). This latter
       form is called "extradata" in FFmpeg terminology.

       This bitstream filter detects the in-band headers and makes them available as extradata.

       remove
           When this option is enabled, the long-term headers are removed from the bitstream
           after extraction.

   filter_units
       Remove units with types in or not in a given set from the stream.

       pass_types
           List of unit types or ranges of unit types to pass through while removing all others.
           This is specified as a '|'-separated list of unit type values or ranges of values with
           '-'.

       remove_types
           Identical to pass_types, except the units in the given set removed and all others
           passed through.

       Extradata is unchanged by this transformation, but note that if the stream contains inline
       parameter sets then the output may be unusable if they are removed.

       For example, to remove all non-VCL NAL units from an H.264 stream:

               ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=pass_types=1-5' OUTPUT

       To remove all AUDs, SEI and filler from an H.265 stream:

               ffmpeg -i INPUT -c:v copy -bsf:v 'filter_units=remove_types=35|38-40' OUTPUT

   hapqa_extract
       Extract Rgb or Alpha part of an HAPQA file, without recompression, in order to create an
       HAPQ or an HAPAlphaOnly file.

       texture
           Specifies the texture to keep.

           color
           alpha

       Convert HAPQA to HAPQ

               ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=color -tag:v HapY -metadata:s:v:0 encoder="HAPQ" hapq_file.mov

       Convert HAPQA to HAPAlphaOnly

               ffmpeg -i hapqa_inputfile.mov -c copy -bsf:v hapqa_extract=texture=alpha -tag:v HapA -metadata:s:v:0 encoder="HAPAlpha Only" hapalphaonly_file.mov

   h264_metadata
       Modify metadata embedded in an H.264 stream.

       aud Insert or remove AUD NAL units in all access units of the stream.

           insert
           remove
       sample_aspect_ratio
           Set the sample aspect ratio of the stream in the VUI parameters.

       video_format
       video_full_range_flag
           Set the video format in the stream (see H.264 section E.2.1 and table E-2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
           Set the colour description in the stream (see H.264 section E.2.1 and tables E-3, E-4
           and E-5).

       chroma_sample_loc_type
           Set the chroma sample location in the stream (see H.264 section E.2.1 and figure E-1).

       tick_rate
           Set the tick rate (num_units_in_tick / time_scale) in the VUI parameters.  This is the
           smallest time unit representable in the stream, and in many cases represents the field
           rate of the stream (double the frame rate).

       fixed_frame_rate_flag
           Set whether the stream has fixed framerate - typically this indicates that the
           framerate is exactly half the tick rate, but the exact meaning is dependent on
           interlacing and the picture structure (see H.264 section E.2.1 and table E-6).

       crop_left
       crop_right
       crop_top
       crop_bottom
           Set the frame cropping offsets in the SPS.  These values will replace the current ones
           if the stream is already cropped.

           These fields are set in pixels.  Note that some sizes may not be representable if the
           chroma is subsampled or the stream is interlaced (see H.264 section 7.4.2.1.1).

       sei_user_data
           Insert a string as SEI unregistered user data.  The argument must be of the form
           UUID+string, where the UUID is as hex digits possibly separated by hyphens, and the
           string can be anything.

           For example, 086f3693-b7b3-4f2c-9653-21492feee5b8+hello will insert the string
           ``hello'' associated with the given UUID.

       delete_filler
           Deletes both filler NAL units and filler SEI messages.

       level
           Set the level in the SPS.  Refer to H.264 section A.3 and tables A-1 to A-5.

           The argument must be the name of a level (for example, 4.2), a level_idc value (for
           example, 42), or the special name auto indicating that the filter should attempt to
           guess the level from the input stream properties.

   h264_mp4toannexb
       Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as
       defined in the Annex B of the ITU-T H.264 specification).

       This is required by some streaming formats, typically the MPEG-2 transport stream format
       (muxer "mpegts").

       For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg,
       you can use the command:

               ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts

       Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw H.264
       (muxer "h264") output formats.

   h264_redundant_pps
       This applies a specific fixup to some Blu-ray streams which contain redundant PPSs
       modifying irrelevant parameters of the stream which confuse other transformations which
       require correct extradata.

       A new single global PPS is created, and all of the redundant PPSs within the stream are
       removed.

   hevc_metadata
       Modify metadata embedded in an HEVC stream.

       aud Insert or remove AUD NAL units in all access units of the stream.

           insert
           remove
       sample_aspect_ratio
           Set the sample aspect ratio in the stream in the VUI parameters.

       video_format
       video_full_range_flag
           Set the video format in the stream (see H.265 section E.3.1 and table E.2).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
           Set the colour description in the stream (see H.265 section E.3.1 and tables E.3, E.4
           and E.5).

       chroma_sample_loc_type
           Set the chroma sample location in the stream (see H.265 section E.3.1 and figure E.1).

       tick_rate
           Set the tick rate in the VPS and VUI parameters (num_units_in_tick / time_scale).
           Combined with num_ticks_poc_diff_one, this can set a constant framerate in the stream.
           Note that it is likely to be overridden by container parameters when the stream is in
           a container.

       num_ticks_poc_diff_one
           Set poc_proportional_to_timing_flag in VPS and VUI and use this value to set
           num_ticks_poc_diff_one_minus1 (see H.265 sections 7.4.3.1 and E.3.1).  Ignored if
           tick_rate is not also set.

       crop_left
       crop_right
       crop_top
       crop_bottom
           Set the conformance window cropping offsets in the SPS.  These values will replace the
           current ones if the stream is already cropped.

           These fields are set in pixels.  Note that some sizes may not be representable if the
           chroma is subsampled (H.265 section 7.4.3.2.1).

   hevc_mp4toannexb
       Convert an HEVC/H.265 bitstream from length prefixed mode to start code prefixed mode (as
       defined in the Annex B of the ITU-T H.265 specification).

       This is required by some streaming formats, typically the MPEG-2 transport stream format
       (muxer "mpegts").

       For example to remux an MP4 file containing an HEVC stream to mpegts format with ffmpeg,
       you can use the command:

               ffmpeg -i INPUT.mp4 -codec copy -bsf:v hevc_mp4toannexb OUTPUT.ts

       Please note that this filter is auto-inserted for MPEG-TS (muxer "mpegts") and raw
       HEVC/H.265 (muxer "h265" or "hevc") output formats.

   imxdump
       Modifies the bitstream to fit in MOV and to be usable by the Final Cut Pro decoder. This
       filter only applies to the mpeg2video codec, and is likely not needed for Final Cut Pro 7
       and newer with the appropriate -tag:v.

       For example, to remux 30 MB/sec NTSC IMX to MOV:

               ffmpeg -i input.mxf -c copy -bsf:v imxdump -tag:v mx3n output.mov

   mjpeg2jpeg
       Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

       MJPEG is a video codec wherein each video frame is essentially a JPEG image. The
       individual frames can be extracted without loss, e.g. by

               ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg

       Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment
       required for decoding. Quoting from
       <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:

       Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or
       at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and
       *omitted* -- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it
       must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed
       extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to
       prepend the DHT segment to them, or else the decoder won't have any idea how to decompress
       the data. The exact table necessary is given in the OpenDML spec."

       This bitstream filter patches the header of frames extracted from an MJPEG stream
       (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG
       images.

               ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg
               exiftran -i -9 frame*.jpg
               ffmpeg -i frame_%d.jpg -c:v copy rotated.avi

   mjpegadump
       Add an MJPEG A header to the bitstream, to enable decoding by Quicktime.

   mov2textsub
       Extract a representable text file from MOV subtitles, stripping the metadata header from
       each subtitle packet.

       See also the text2movsub filter.

   mp3decomp
       Decompress non-standard compressed MP3 audio headers.

   mpeg2_metadata
       Modify metadata embedded in an MPEG-2 stream.

       display_aspect_ratio
           Set the display aspect ratio in the stream.

           The following fixed values are supported:

           4/3
           16/9
           221/100

           Any other value will result in square pixels being signalled instead (see H.262
           section 6.3.3 and table 6-3).

       frame_rate
           Set the frame rate in the stream.  This is constructed from a table of known values
           combined with a small multiplier and divisor - if the supplied value is not exactly
           representable, the nearest representable value will be used instead (see H.262 section
           6.3.3 and table 6-4).

       video_format
           Set the video format in the stream (see H.262 section 6.3.6 and table 6-6).

       colour_primaries
       transfer_characteristics
       matrix_coefficients
           Set the colour description in the stream (see H.262 section 6.3.6 and tables 6-7, 6-8
           and 6-9).

   mpeg4_unpack_bframes
       Unpack DivX-style packed B-frames.

       DivX-style packed B-frames are not valid MPEG-4 and were only a workaround for the broken
       Video for Windows subsystem.  They use more space, can cause minor AV sync issues, require
       more CPU power to decode (unless the player has some decoded picture queue to compensate
       the 2,0,2,0 frame per packet style) and cause trouble if copied into a standard container
       like mp4 or mpeg-ps/ts, because MPEG-4 decoders may not be able to decode them, since they
       are not valid MPEG-4.

       For example to fix an AVI file containing an MPEG-4 stream with DivX-style packed B-frames
       using ffmpeg, you can use the command:

               ffmpeg -i INPUT.avi -codec copy -bsf:v mpeg4_unpack_bframes OUTPUT.avi

   noise
       Damages the contents of packets or simply drops them without damaging the container. Can
       be used for fuzzing or testing error resilience/concealment.

       Parameters:

       amount
           A numeral string, whose value is related to how often output bytes will be modified.
           Therefore, values below or equal to 0 are forbidden, and the lower the more frequent
           bytes will be modified, with 1 meaning every byte is modified.

       dropamount
           A numeral string, whose value is related to how often packets will be dropped.
           Therefore, values below or equal to 0 are forbidden, and the lower the more frequent
           packets will be dropped, with 1 meaning every packet is dropped.

       The following example applies the modification to every byte but does not drop any
       packets.

               ffmpeg -i INPUT -c copy -bsf noise[=1] output.mkv

   null
       This bitstream filter passes the packets through unchanged.

   remove_extra
       Remove extradata from packets.

       It accepts the following parameter:

       freq
           Set which frame types to remove extradata from.

           k   Remove extradata from non-keyframes only.

           keyframe
               Remove extradata from keyframes only.

           e, all
               Remove extradata from all frames.

   text2movsub
       Convert text subtitles to MOV subtitles (as used by the "mov_text" codec) with metadata
       headers.

       See also the mov2textsub filter.

   trace_headers
       Log trace output containing all syntax elements in the coded stream headers (everything
       above the level of individual coded blocks).  This can be useful for debugging low-level
       stream issues.

       Supports H.264, H.265, MPEG-2 and VP9.

   vp9_metadata
       Modify metadata embedded in a VP9 stream.

       color_space
           Set the color space value in the frame header.

           unknown
           bt601
           bt709
           smpte170
           smpte240
           bt2020
           rgb
       color_range
           Set the color range value in the frame header.  Note that this cannot be set in RGB
           streams.

           tv
           pc

   vp9_superframe
       Merge VP9 invisible (alt-ref) frames back into VP9 superframes. This fixes merging of
       split/segmented VP9 streams where the alt-ref frame was split from its visible
       counterpart.

   vp9_superframe_split
       Split VP9 superframes into single frames.

   vp9_raw_reorder
       Given a VP9 stream with correct timestamps but possibly out of order, insert additional
       show-existing-frame packets to correct the ordering.

FORMAT OPTIONS

       The libavformat library provides some generic global options, which can be set on all the
       muxers and demuxers. In addition each muxer or demuxer may support so-called private
       options, which are specific for that component.

       Options may be set by specifying -option value in the FFmpeg tools, or by setting the
       value explicitly in the "AVFormatContext" options or using the libavutil/opt.h API for
       programmatic use.

       The list of supported options follows:

       avioflags flags (input/output)
           Possible values:

           direct
               Reduce buffering.

       probesize integer (input)
           Set probing size in bytes, i.e. the size of the data to analyze to get stream
           information. A higher value will enable detecting more information in case it is
           dispersed into the stream, but will increase latency. Must be an integer not lesser
           than 32. It is 5000000 by default.

       packetsize integer (output)
           Set packet size.

       fflags flags
           Set format flags. Some are implemented for a limited number of formats.

           Possible values for input files:

           discardcorrupt
               Discard corrupted packets.

           fastseek
               Enable fast, but inaccurate seeks for some formats.

           genpts
               Generate missing PTS if DTS is present.

           igndts
               Ignore DTS if PTS is set. Inert when nofillin is set.

           ignidx
               Ignore index.

           keepside (deprecated,inert)
           nobuffer
               Reduce the latency introduced by buffering during initial input streams analysis.

           nofillin
               Do not fill in missing values in packet fields that can be exactly calculated.

           noparse
               Disable AVParsers, this needs "+nofillin" too.

           sortdts
               Try to interleave output packets by DTS. At present, available only for AVIs with
               an index.

           Possible values for output files:

           autobsf
               Automatically apply bitstream filters as required by the output format. Enabled by
               default.

           bitexact
               Only write platform-, build- and time-independent data.  This ensures that file
               and data checksums are reproducible and match between platforms. Its primary use
               is for regression testing.

           flush_packets
               Write out packets immediately.

           latm (deprecated,inert)
           shortest
               Stop muxing at the end of the shortest stream.  It may be needed to increase
               max_interleave_delta to avoid flushing the longer streams before EOF.

       seek2any integer (input)
           Allow seeking to non-keyframes on demuxer level when supported if set to 1.  Default
           is 0.

       analyzeduration integer (input)
           Specify how many microseconds are analyzed to probe the input. A higher value will
           enable detecting more accurate information, but will increase latency. It defaults to
           5,000,000 microseconds = 5 seconds.

       cryptokey hexadecimal string (input)
           Set decryption key.

       indexmem integer (input)
           Set max memory used for timestamp index (per stream).

       rtbufsize integer (input)
           Set max memory used for buffering real-time frames.

       fdebug flags (input/output)
           Print specific debug info.

           Possible values:

           ts
       max_delay integer (input/output)
           Set maximum muxing or demuxing delay in microseconds.

       fpsprobesize integer (input)
           Set number of frames used to probe fps.

       audio_preload integer (output)
           Set microseconds by which audio packets should be interleaved earlier.

       chunk_duration integer (output)
           Set microseconds for each chunk.

       chunk_size integer (output)
           Set size in bytes for each chunk.

       err_detect, f_err_detect flags (input)
           Set error detection flags. "f_err_detect" is deprecated and should be used only via
           the ffmpeg tool.

           Possible values:

           crccheck
               Verify embedded CRCs.

           bitstream
               Detect bitstream specification deviations.

           buffer
               Detect improper bitstream length.

           explode
               Abort decoding on minor error detection.

           careful
               Consider things that violate the spec and have not been seen in the wild as
               errors.

           compliant
               Consider all spec non compliancies as errors.

           aggressive
               Consider things that a sane encoder should not do as an error.

       max_interleave_delta integer (output)
           Set maximum buffering duration for interleaving. The duration is expressed in
           microseconds, and defaults to 1000000 (1 second).

           To ensure all the streams are interleaved correctly, libavformat will wait until it
           has at least one packet for each stream before actually writing any packets to the
           output file. When some streams are "sparse" (i.e. there are large gaps between
           successive packets), this can result in excessive buffering.

           This field specifies the maximum difference between the timestamps of the first and
           the last packet in the muxing queue, above which libavformat will output a packet
           regardless of whether it has queued a packet for all the streams.

           If set to 0, libavformat will continue buffering packets until it has a packet for
           each stream, regardless of the maximum timestamp difference between the buffered
           packets.

       use_wallclock_as_timestamps integer (input)
           Use wallclock as timestamps if set to 1. Default is 0.

       avoid_negative_ts integer (output)
           Possible values:

           make_non_negative
               Shift timestamps to make them non-negative.  Also note that this affects only
               leading negative timestamps, and not non-monotonic negative timestamps.

           make_zero
               Shift timestamps so that the first timestamp is 0.

           auto (default)
               Enables shifting when required by the target format.

           disabled
               Disables shifting of timestamp.

           When shifting is enabled, all output timestamps are shifted by the same amount. Audio,
           video, and subtitles desynching and relative timestamp differences are preserved
           compared to how they would have been without shifting.

       skip_initial_bytes integer (input)
           Set number of bytes to skip before reading header and frames if set to 1.  Default is
           0.

       correct_ts_overflow integer (input)
           Correct single timestamp overflows if set to 1. Default is 1.

       flush_packets integer (output)
           Flush the underlying I/O stream after each packet. Default is -1 (auto), which means
           that the underlying protocol will decide, 1 enables it, and has the effect of reducing
           the latency, 0 disables it and may increase IO throughput in some cases.

       output_ts_offset offset (output)
           Set the output time offset.

           offset must be a time duration specification, see the Time duration section in the
           ffmpeg-utils(1) manual.

           The offset is added by the muxer to the output timestamps.

           Specifying a positive offset means that the corresponding streams are delayed bt the
           time duration specified in offset. Default value is 0 (meaning that no offset is
           applied).

       format_whitelist list (input)
           "," separated list of allowed demuxers. By default all are allowed.

       dump_separator string (input)
           Separator used to separate the fields printed on the command line about the Stream
           parameters.  For example to separate the fields with newlines and indention:

                   ffprobe -dump_separator "
                                             "  -i ~/videos/matrixbench_mpeg2.mpg

       max_streams integer (input)
           Specifies the maximum number of streams. This can be used to reject files that would
           require too many resources due to a large number of streams.

       skip_estimate_duration_from_pts bool (input)
           Skip estimation of input duration when calculated using PTS.  At present, applicable
           for MPEG-PS and MPEG-TS.

   Format stream specifiers
       Format stream specifiers allow selection of one or more streams that match specific
       properties.

       Possible forms of stream specifiers are:

       stream_index
           Matches the stream with this index.

       stream_type[:stream_index]
           stream_type is one of following: 'v' for video, 'a' for audio, 's' for subtitle, 'd'
           for data, and 't' for attachments. If stream_index is given, then it matches the
           stream number stream_index of this type. Otherwise, it matches all streams of this
           type.

       p:program_id[:stream_index]
           If stream_index is given, then it matches the stream with number stream_index in the
           program with the id program_id. Otherwise, it matches all streams in the program.

       #stream_id
           Matches the stream by a format-specific ID.

       The exact semantics of stream specifiers is defined by the
       "avformat_match_stream_specifier()" function declared in the libavformat/avformat.h
       header.

DEMUXERS

       Demuxers are configured elements in FFmpeg that can read the multimedia streams from a
       particular type of file.

       When you configure your FFmpeg build, all the supported demuxers are enabled by default.
       You can list all available ones using the configure option "--list-demuxers".

       You can disable all the demuxers using the configure option "--disable-demuxers", and
       selectively enable a single demuxer with the option "--enable-demuxer=DEMUXER", or disable
       it with the option "--disable-demuxer=DEMUXER".

       The option "-demuxers" of the ff* tools will display the list of enabled demuxers. Use
       "-formats" to view a combined list of enabled demuxers and muxers.

       The description of some of the currently available demuxers follows.

   aa
       Audible Format 2, 3, and 4 demuxer.

       This demuxer is used to demux Audible Format 2, 3, and 4 (.aa) files.

   applehttp
       Apple HTTP Live Streaming demuxer.

       This demuxer presents all AVStreams from all variant streams.  The id field is set to the
       bitrate variant index number. By setting the discard flags on AVStreams (by pressing 'a'
       or 'v' in ffplay), the caller can decide which variant streams to actually receive.  The
       total bitrate of the variant that the stream belongs to is available in a metadata key
       named "variant_bitrate".

   apng
       Animated Portable Network Graphics demuxer.

       This demuxer is used to demux APNG files.  All headers, but the PNG signature, up to (but
       not including) the first fcTL chunk are transmitted as extradata.  Frames are then split
       as being all the chunks between two fcTL ones, or between the last fcTL and IEND chunks.

       -ignore_loop bool
           Ignore the loop variable in the file if set.

       -max_fps int
           Maximum framerate in frames per second (0 for no limit).

       -default_fps int
           Default framerate in frames per second when none is specified in the file (0 meaning
           as fast as possible).

   asf
       Advanced Systems Format demuxer.

       This demuxer is used to demux ASF files and MMS network streams.

       -no_resync_search bool
           Do not try to resynchronize by looking for a certain optional start code.

   concat
       Virtual concatenation script demuxer.

       This demuxer reads a list of files and other directives from a text file and demuxes them
       one after the other, as if all their packets had been muxed together.

       The timestamps in the files are adjusted so that the first file starts at 0 and each next
       file starts where the previous one finishes. Note that it is done globally and may cause
       gaps if all streams do not have exactly the same length.

       All files must have the same streams (same codecs, same time base, etc.).

       The duration of each file is used to adjust the timestamps of the next file: if the
       duration is incorrect (because it was computed using the bit-rate or because the file is
       truncated, for example), it can cause artifacts. The "duration" directive can be used to
       override the duration stored in each file.

       Syntax

       The script is a text file in extended-ASCII, with one directive per line.  Empty lines,
       leading spaces and lines starting with '#' are ignored. The following directive is
       recognized:

       "file path"
           Path to a file to read; special characters and spaces must be escaped with backslash
           or single quotes.

           All subsequent file-related directives apply to that file.

       "ffconcat version 1.0"
           Identify the script type and version. It also sets the safe option to 1 if it was -1.

           To make FFmpeg recognize the format automatically, this directive must appear exactly
           as is (no extra space or byte-order-mark) on the very first line of the script.

       "duration dur"
           Duration of the file. This information can be specified from the file; specifying it
           here may be more efficient or help if the information from the file is not available
           or accurate.

           If the duration is set for all files, then it is possible to seek in the whole
           concatenated video.

       "inpoint timestamp"
           In point of the file. When the demuxer opens the file it instantly seeks to the
           specified timestamp. Seeking is done so that all streams can be presented successfully
           at In point.

           This directive works best with intra frame codecs, because for non-intra frame ones
           you will usually get extra packets before the actual In point and the decoded content
           will most likely contain frames before In point too.

           For each file, packets before the file In point will have timestamps less than the
           calculated start timestamp of the file (negative in case of the first file), and the
           duration of the files (if not specified by the "duration" directive) will be reduced
           based on their specified In point.

           Because of potential packets before the specified In point, packet timestamps may
           overlap between two concatenated files.

       "outpoint timestamp"
           Out point of the file. When the demuxer reaches the specified decoding timestamp in
           any of the streams, it handles it as an end of file condition and skips the current
           and all the remaining packets from all streams.

           Out point is exclusive, which means that the demuxer will not output packets with a
           decoding timestamp greater or equal to Out point.

           This directive works best with intra frame codecs and formats where all streams are
           tightly interleaved. For non-intra frame codecs you will usually get additional
           packets with presentation timestamp after Out point therefore the decoded content will
           most likely contain frames after Out point too. If your streams are not tightly
           interleaved you may not get all the packets from all streams before Out point and you
           may only will be able to decode the earliest stream until Out point.

           The duration of the files (if not specified by the "duration" directive) will be
           reduced based on their specified Out point.

       "file_packet_metadata key=value"
           Metadata of the packets of the file. The specified metadata will be set for each file
           packet. You can specify this directive multiple times to add multiple metadata
           entries.

       "stream"
           Introduce a stream in the virtual file.  All subsequent stream-related directives
           apply to the last introduced stream.  Some streams properties must be set in order to
           allow identifying the matching streams in the subfiles.  If no streams are defined in
           the script, the streams from the first file are copied.

       "exact_stream_id id"
           Set the id of the stream.  If this directive is given, the string with the
           corresponding id in the subfiles will be used.  This is especially useful for MPEG-PS
           (VOB) files, where the order of the streams is not reliable.

       Options

       This demuxer accepts the following option:

       safe
           If set to 1, reject unsafe file paths. A file path is considered safe if it does not
           contain a protocol specification and is relative and all components only contain
           characters from the portable character set (letters, digits, period, underscore and
           hyphen) and have no period at the beginning of a component.

           If set to 0, any file name is accepted.

           The default is 1.

           -1 is equivalent to 1 if the format was automatically probed and 0 otherwise.

       auto_convert
           If set to 1, try to perform automatic conversions on packet data to make the streams
           concatenable.  The default is 1.

           Currently, the only conversion is adding the h264_mp4toannexb bitstream filter to
           H.264 streams in MP4 format. This is necessary in particular if there are resolution
           changes.

       segment_time_metadata
           If set to 1, every packet will contain the lavf.concat.start_time and the
           lavf.concat.duration packet metadata values which are the start_time and the duration
           of the respective file segments in the concatenated output expressed in microseconds.
           The duration metadata is only set if it is known based on the concat file.  The
           default is 0.

       Examples

       ·   Use absolute filenames and include some comments:

                   # my first filename
                   file /mnt/share/file-1.wav
                   # my second filename including whitespace
                   file '/mnt/share/file 2.wav'
                   # my third filename including whitespace plus single quote
                   file '/mnt/share/file 3'\''.wav'

       ·   Allow for input format auto-probing, use safe filenames and set the duration of the
           first file:

                   ffconcat version 1.0

                   file file-1.wav
                   duration 20.0

                   file subdir/file-2.wav

   dash
       Dynamic Adaptive Streaming over HTTP demuxer.

       This demuxer presents all AVStreams found in the manifest.  By setting the discard flags
       on AVStreams the caller can decide which streams to actually receive.  Each stream mirrors
       the "id" and "bandwidth" properties from the "<Representation>" as metadata keys named
       "id" and "variant_bitrate" respectively.

   flv, live_flv
       Adobe Flash Video Format demuxer.

       This demuxer is used to demux FLV files and RTMP network streams. In case of live network
       streams, if you force format, you may use live_flv option instead of flv to survive
       timestamp discontinuities.

               ffmpeg -f flv -i myfile.flv ...
               ffmpeg -f live_flv -i rtmp://<any.server>/anything/key ....

       -flv_metadata bool
           Allocate the streams according to the onMetaData array content.

       -flv_ignore_prevtag bool
           Ignore the size of previous tag value.

       -flv_full_metadata bool
           Output all context of the onMetadata.

   gif
       Animated GIF demuxer.

       It accepts the following options:

       min_delay
           Set the minimum valid delay between frames in hundredths of seconds.  Range is 0 to
           6000. Default value is 2.

       max_gif_delay
           Set the maximum valid delay between frames in hundredth of seconds.  Range is 0 to
           65535. Default value is 65535 (nearly eleven minutes), the maximum value allowed by
           the specification.

       default_delay
           Set the default delay between frames in hundredths of seconds.  Range is 0 to 6000.
           Default value is 10.

       ignore_loop
           GIF files can contain information to loop a certain number of times (or infinitely).
           If ignore_loop is set to 1, then the loop setting from the input will be ignored and
           looping will not occur. If set to 0, then looping will occur and will cycle the number
           of times according to the GIF. Default value is 1.

       For example, with the overlay filter, place an infinitely looping GIF over another video:

               ffmpeg -i input.mp4 -ignore_loop 0 -i input.gif -filter_complex overlay=shortest=1 out.mkv

       Note that in the above example the shortest option for overlay filter is used to end the
       output video at the length of the shortest input file, which in this case is input.mp4 as
       the GIF in this example loops infinitely.

   hls
       HLS demuxer

       It accepts the following options:

       live_start_index
           segment index to start live streams at (negative values are from the end).

       allowed_extensions
           ',' separated list of file extensions that hls is allowed to access.

       max_reload
           Maximum number of times a insufficient list is attempted to be reloaded.  Default
           value is 1000.

       http_persistent
           Use persistent HTTP connections. Applicable only for HTTP streams.  Enabled by
           default.

       http_multiple
           Use multiple HTTP connections for downloading HTTP segments.  Enabled by default for
           HTTP/1.1 servers.

   image2
       Image file demuxer.

       This demuxer reads from a list of image files specified by a pattern.  The syntax and
       meaning of the pattern is specified by the option pattern_type.

       The pattern may contain a suffix which is used to automatically determine the format of
       the images contained in the files.

       The size, the pixel format, and the format of each image must be the same for all the
       files in the sequence.

       This demuxer accepts the following options:

       framerate
           Set the frame rate for the video stream. It defaults to 25.

       loop
           If set to 1, loop over the input. Default value is 0.

       pattern_type
           Select the pattern type used to interpret the provided filename.

           pattern_type accepts one of the following values.

           none
               Disable pattern matching, therefore the video will only contain the specified
               image. You should use this option if you do not want to create sequences from
               multiple images and your filenames may contain special pattern characters.

           sequence
               Select a sequence pattern type, used to specify a sequence of files indexed by
               sequential numbers.

               A sequence pattern may contain the string "%d" or "%0Nd", which specifies the
               position of the characters representing a sequential number in each filename
               matched by the pattern. If the form "%d0Nd" is used, the string representing the
               number in each filename is 0-padded and N is the total number of 0-padded digits
               representing the number. The literal character '%' can be specified in the pattern
               with the string "%%".

               If the sequence pattern contains "%d" or "%0Nd", the first filename of the file
               list specified by the pattern must contain a number inclusively contained between
               start_number and start_number+start_number_range-1, and all the following numbers
               must be sequential.

               For example the pattern "img-%03d.bmp" will match a sequence of filenames of the
               form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern
               "i%%m%%g-%d.jpg" will match a sequence of filenames of the form i%m%g-1.jpg,
               i%m%g-2.jpg, ..., i%m%g-10.jpg, etc.

               Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to
               convert a single image file img.jpeg you can employ the command:

                       ffmpeg -i img.jpeg img.png

           glob
               Select a glob wildcard pattern type.

               The pattern is interpreted like a "glob()" pattern. This is only selectable if
               libavformat was compiled with globbing support.

           glob_sequence (deprecated, will be removed)
               Select a mixed glob wildcard/sequence pattern.

               If your version of libavformat was compiled with globbing support, and the
               provided pattern contains at least one glob meta character among "%*?[]{}" that is
               preceded by an unescaped "%", the pattern is interpreted like a "glob()" pattern,
               otherwise it is interpreted like a sequence pattern.

               All glob special characters "%*?[]{}" must be prefixed with "%". To escape a
               literal "%" you shall use "%%".

               For example the pattern "foo-%*.jpeg" will match all the filenames prefixed by
               "foo-" and terminating with ".jpeg", and "foo-%?%?%?.jpeg" will match all the
               filenames prefixed with "foo-", followed by a sequence of three characters, and
               terminating with ".jpeg".

               This pattern type is deprecated in favor of glob and sequence.

           Default value is glob_sequence.

       pixel_format
           Set the pixel format of the images to read. If not specified the pixel format is
           guessed from the first image file in the sequence.

       start_number
           Set the index of the file matched by the image file pattern to start to read from.
           Default value is 0.

       start_number_range
           Set the index interval range to check when looking for the first image file in the
           sequence, starting from start_number. Default value is 5.

       ts_from_file
           If set to 1, will set frame timestamp to modification time of image file. Note that
           monotonity of timestamps is not provided: images go in the same order as without this
           option. Default value is 0.  If set to 2, will set frame timestamp to the modification
           time of the image file in nanosecond precision.

       video_size
           Set the video size of the images to read. If not specified the video size is guessed
           from the first image file in the sequence.

       Examples

       ·   Use ffmpeg for creating a video from the images in the file sequence img-001.jpeg,
           img-002.jpeg, ..., assuming an input frame rate of 10 frames per second:

                   ffmpeg -framerate 10 -i 'img-%03d.jpeg' out.mkv

       ·   As above, but start by reading from a file with index 100 in the sequence:

                   ffmpeg -framerate 10 -start_number 100 -i 'img-%03d.jpeg' out.mkv

       ·   Read images matching the "*.png" glob pattern , that is all the files terminating with
           the ".png" suffix:

                   ffmpeg -framerate 10 -pattern_type glob -i "*.png" out.mkv

   libgme
       The Game Music Emu library is a collection of video game music file emulators.

       See <http://code.google.com/p/game-music-emu/> for more information.

       Some files have multiple tracks. The demuxer will pick the first track by default. The
       track_index option can be used to select a different track. Track indexes start at 0. The
       demuxer exports the number of tracks as tracks meta data entry.

       For very large files, the max_size option may have to be adjusted.

   libopenmpt
       libopenmpt based module demuxer

       See <https://lib.openmpt.org/libopenmpt/> for more information.

       Some files have multiple subsongs (tracks) this can be set with the subsong option.

       It accepts the following options:

       subsong
           Set the subsong index. This can be either  'all', 'auto', or the index of the subsong.
           Subsong indexes start at 0. The default is 'auto'.

           The default value is to let libopenmpt choose.

       layout
           Set the channel layout. Valid values are 1, 2, and 4 channel layouts.  The default
           value is STEREO.

       sample_rate
           Set the sample rate for libopenmpt to output.  Range is from 1000 to INT_MAX. The
           value default is 48000.

   mov/mp4/3gp/QuickTime
       QuickTime / MP4 demuxer.

       This demuxer accepts the following options:

       enable_drefs
           Enable loading of external tracks, disabled by default.  Enabling this can
           theoretically leak information in some use cases.

       use_absolute_path
           Allows loading of external tracks via absolute paths, disabled by default.  Enabling
           this poses a security risk. It should only be enabled if the source is known to be non
           malicious.

   mpegts
       MPEG-2 transport stream demuxer.

       This demuxer accepts the following options:

       resync_size
           Set size limit for looking up a new synchronization. Default value is 65536.

       skip_unknown_pmt
           Skip PMTs for programs not defined in the PAT. Default value is 0.

       fix_teletext_pts
           Override teletext packet PTS and DTS values with the timestamps calculated from the
           PCR of the first program which the teletext stream is part of and is not discarded.
           Default value is 1, set this option to 0 if you want your teletext packet PTS and DTS
           values untouched.

       ts_packetsize
           Output option carrying the raw packet size in bytes.  Show the detected raw packet
           size, cannot be set by the user.

       scan_all_pmts
           Scan and combine all PMTs. The value is an integer with value from -1 to 1 (-1 means
           automatic setting, 1 means enabled, 0 means disabled). Default value is -1.

       merge_pmt_versions
           Re-use existing streams when a PMT's version is updated and elementary streams move to
           different PIDs. Default value is 0.

   mpjpeg
       MJPEG encapsulated in multi-part MIME demuxer.

       This demuxer allows reading of MJPEG, where each frame is represented as a part of
       multipart/x-mixed-replace stream.

       strict_mime_boundary
           Default implementation applies a relaxed standard to multi-part MIME boundary
           detection, to prevent regression with numerous existing endpoints not generating a
           proper MIME MJPEG stream. Turning this option on by setting it to 1 will result in a
           stricter check of the boundary value.

   rawvideo
       Raw video demuxer.

       This demuxer allows one to read raw video data. Since there is no header specifying the
       assumed video parameters, the user must specify them in order to be able to decode the
       data correctly.

       This demuxer accepts the following options:

       framerate
           Set input video frame rate. Default value is 25.

       pixel_format
           Set the input video pixel format. Default value is "yuv420p".

       video_size
           Set the input video size. This value must be specified explicitly.

       For example to read a rawvideo file input.raw with ffplay, assuming a pixel format of
       "rgb24", a video size of "320x240", and a frame rate of 10 images per second, use the
       command:

               ffplay -f rawvideo -pixel_format rgb24 -video_size 320x240 -framerate 10 input.raw

   sbg
       SBaGen script demuxer.

       This demuxer reads the script language used by SBaGen <http://uazu.net/sbagen/> to
       generate binaural beats sessions. A SBG script looks like that:

               -SE
               a: 300-2.5/3 440+4.5/0
               b: 300-2.5/0 440+4.5/3
               off: -
               NOW      == a
               +0:07:00 == b
               +0:14:00 == a
               +0:21:00 == b
               +0:30:00    off

       A SBG script can mix absolute and relative timestamps. If the script uses either only
       absolute timestamps (including the script start time) or only relative ones, then its
       layout is fixed, and the conversion is straightforward. On the other hand, if the script
       mixes both kind of timestamps, then the NOW reference for relative timestamps will be
       taken from the current time of day at the time the script is read, and the script layout
       will be frozen according to that reference. That means that if the script is directly
       played, the actual times will match the absolute timestamps up to the sound controller's
       clock accuracy, but if the user somehow pauses the playback or seeks, all times will be
       shifted accordingly.

   tedcaptions
       JSON captions used for <http://www.ted.com/>.

       TED does not provide links to the captions, but they can be guessed from the page. The
       file tools/bookmarklets.html from the FFmpeg source tree contains a bookmarklet to expose
       them.

       This demuxer accepts the following option:

       start_time
           Set the start time of the TED talk, in milliseconds. The default is 15000 (15s). It is
           used to sync the captions with the downloadable videos, because they include a 15s
           intro.

       Example: convert the captions to a format most players understand:

               ffmpeg -i http://www.ted.com/talks/subtitles/id/1/lang/en talk1-en.srt

MUXERS

       Muxers are configured elements in FFmpeg which allow writing multimedia streams to a
       particular type of file.

       When you configure your FFmpeg build, all the supported muxers are enabled by default. You
       can list all available muxers using the configure option "--list-muxers".

       You can disable all the muxers with the configure option "--disable-muxers" and
       selectively enable / disable single muxers with the options "--enable-muxer=MUXER" /
       "--disable-muxer=MUXER".

       The option "-muxers" of the ff* tools will display the list of enabled muxers. Use
       "-formats" to view a combined list of enabled demuxers and muxers.

       A description of some of the currently available muxers follows.

   aiff
       Audio Interchange File Format muxer.

       Options

       It accepts the following options:

       write_id3v2
           Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).

       id3v2_version
           Select ID3v2 version to write. Currently only version 3 and 4 (aka.  ID3v2.3 and
           ID3v2.4) are supported. The default is version 4.

   asf
       Advanced Systems Format muxer.

       Note that Windows Media Audio (wma) and Windows Media Video (wmv) use this muxer too.

       Options

       It accepts the following options:

       packet_size
           Set the muxer packet size. By tuning this setting you may reduce data fragmentation or
           muxer overhead depending on your source. Default value is 3200, minimum is 100,
           maximum is 64k.

   avi
       Audio Video Interleaved muxer.

       Options

       It accepts the following options:

       reserve_index_space
           Reserve the specified amount of bytes for the OpenDML master index of each stream
           within the file header. By default additional master indexes are embedded within the
           data packets if there is no space left in the first master index and are linked
           together as a chain of indexes. This index structure can cause problems for some use
           cases, e.g. third-party software strictly relying on the OpenDML index specification
           or when file seeking is slow. Reserving enough index space in the file header avoids
           these problems.

           The required index space depends on the output file size and should be about 16 bytes
           per gigabyte. When this option is omitted or set to zero the necessary index space is
           guessed.

       write_channel_mask
           Write the channel layout mask into the audio stream header.

           This option is enabled by default. Disabling the channel mask can be useful in
           specific scenarios, e.g. when merging multiple audio streams into one for
           compatibility with software that only supports a single audio stream in AVI (see the
           "amerge" section in the ffmpeg-filters manual).

   chromaprint
       Chromaprint fingerprinter

       This muxer feeds audio data to the Chromaprint library, which generates a fingerprint for
       the provided audio data. It takes a single signed native-endian 16-bit raw audio stream.

       Options

       silence_threshold
           Threshold for detecting silence, ranges from 0 to 32767. -1 for default (required for
           use with the AcoustID service).

       algorithm
           Algorithm index to fingerprint with.

       fp_format
           Format to output the fingerprint as. Accepts the following options:

           raw Binary raw fingerprint

           compressed
               Binary compressed fingerprint

           base64
               Base64 compressed fingerprint

   crc
       CRC (Cyclic Redundancy Check) testing format.

       This muxer computes and prints the Adler-32 CRC of all the input audio and video frames.
       By default audio frames are converted to signed 16-bit raw audio and video frames to raw
       video before computing the CRC.

       The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a
       hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input
       frames.

       See also the framecrc muxer.

       Examples

       For example to compute the CRC of the input, and store it in the file out.crc:

               ffmpeg -i INPUT -f crc out.crc

       You can print the CRC to stdout with the command:

               ffmpeg -i INPUT -f crc -

       You can select the output format of each frame with ffmpeg by specifying the audio and
       video codec and format. For example to compute the CRC of the input audio converted to PCM
       unsigned 8-bit and the input video converted to MPEG-2 video, use the command:

               ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -

   flv
       Adobe Flash Video Format muxer.

       This muxer accepts the following options:

       flvflags flags
           Possible values:

           aac_seq_header_detect
               Place AAC sequence header based on audio stream data.

           no_sequence_end
               Disable sequence end tag.

           no_metadata
               Disable metadata tag.

           no_duration_filesize
               Disable duration and filesize in metadata when they are equal to zero at the end
               of stream. (Be used to non-seekable living stream).

           add_keyframe_index
               Used to facilitate seeking; particularly for HTTP pseudo streaming.

   dash
       Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments and manifest files
       according to the MPEG-DASH standard ISO/IEC 23009-1:2014.

       For more information see:

       ·   ISO DASH Specification:
           <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

       ·   WebM DASH Specification:
           <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

       It creates a MPD manifest file and segment files for each stream.

       The segment filename might contain pre-defined identifiers used with SegmentTemplate as
       defined in section 5.3.9.4.4 of the standard. Available identifiers are
       "$RepresentationID$", "$Number$", "$Bandwidth$" and "$Time$".

               ffmpeg -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264
               -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline
               -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0
               -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1
               -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a"
               -f dash /path/to/out.mpd

       -min_seg_duration microseconds
           This is a deprecated option to set the segment length in microseconds, use
           seg_duration instead.

       -seg_duration duration
           Set the segment length in seconds (fractional value can be set). The value is treated
           as average segment duration when use_template is enabled and use_timeline is disabled
           and as minimum segment duration for all the other use cases.

       -window_size size
           Set the maximum number of segments kept in the manifest.

       -extra_window_size size
           Set the maximum number of segments kept outside of the manifest before removing from
           disk.

       -remove_at_exit remove
           Enable (1) or disable (0) removal of all segments when finished.

       -use_template template
           Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.

       -use_timeline timeline
           Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.

       -single_file single_file
           Enable (1) or disable (0) storing all segments in one file, accessed using byte
           ranges.

       -single_file_name file_name
           DASH-templated name to be used for baseURL. Implies single_file set to "1".

       -init_seg_name init_name
           DASH-templated name to used for the initialization segment. Default is
           "init-stream$RepresentationID$.m4s"

       -media_seg_name segment_name
           DASH-templated name to used for the media segments. Default is
           "chunk-stream$RepresentationID$-$Number%05d$.m4s"

       -utc_timing_url utc_url
           URL of the page that will return the UTC timestamp in ISO format. Example:
           "https://time.akamai.com/?iso"

       method method
           Use the given HTTP method to create output files. Generally set to PUT or POST.

       -http_user_agent user_agent
           Override User-Agent field in HTTP header. Applicable only for HTTP output.

       -http_persistent http_persistent
           Use persistent HTTP connections. Applicable only for HTTP output.

       -hls_playlist hls_playlist
           Generate HLS playlist files as well. The master playlist is generated with the
           filename master.m3u8.  One media playlist file is generated for each stream with
           filenames media_0.m3u8, media_1.m3u8, etc.

       -streaming streaming
           Enable (1) or disable (0) chunk streaming mode of output. In chunk streaming mode,
           each frame will be a moof fragment which forms a chunk.

       -adaptation_sets adaptation_sets
           Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with
           x and y being the IDs of the adaptation sets and a,b,c,d and e are the indices of the
           mapped streams.

           To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as
           stream identifier instead of IDs.

           When no assignment is defined, this defaults to an AdaptationSet for each stream.

       -timeout timeout
           Set timeout for socket I/O operations. Applicable only for HTTP output.

       -index_correction index_correction
           Enable (1) or Disable (0) segment index correction logic. Applicable only when
           use_template is enabled and use_timeline is disabled.

           When enabled, the logic monitors the flow of segment indexes. If a streams's segment
           index value is not at the expected real time position, then the logic corrects that
           index value.

           Typically this logic is needed in live streaming use cases. The network bandwidth
           fluctuations are common during long run streaming. Each fluctuation can cause the
           segment indexes fall behind the expected real time position.

       -format_options options_list
           Set container format (mp4/webm) options using a ":" separated list of key=value
           parameters. Values containing ":" special characters must be escaped.

       dash_segment_type dash_segment_type
           Possible values:

       mp4 If this flag is set, the dash segment files will be in in ISOBMFF format. This is the
           default format.

       webm
           If this flag is set, the dash segment files will be in in WebM format.

   framecrc
       Per-packet CRC (Cyclic Redundancy Check) testing format.

       This muxer computes and prints the Adler-32 CRC for each audio and video packet. By
       default audio frames are converted to signed 16-bit raw audio and video frames to raw
       video before computing the CRC.

       The output of the muxer consists of a line for each audio and video packet of the form:

               <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, 0x<CRC>

       CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.

       Examples

       For example to compute the CRC of the audio and video frames in INPUT, converted to raw
       audio and video packets, and store it in the file out.crc:

               ffmpeg -i INPUT -f framecrc out.crc

       To print the information to stdout, use the command:

               ffmpeg -i INPUT -f framecrc -

       With ffmpeg, you can select the output format to which the audio and video frames are
       encoded before computing the CRC for each packet by specifying the audio and video codec.
       For example, to compute the CRC of each decoded input audio frame converted to PCM
       unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the
       command:

               ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

       See also the crc muxer.

   framehash
       Per-packet hash testing format.

       This muxer computes and prints a cryptographic hash for each audio and video packet. This
       can be used for packet-by-packet equality checks without having to individually do a
       binary comparison on each.

       By default audio frames are converted to signed 16-bit raw audio and video frames to raw
       video before computing the hash, but the output of explicit conversions to other codecs
       can also be used. It uses the SHA-256 cryptographic hash function by default, but supports
       several other algorithms.

       The output of the muxer consists of a line for each audio and video packet of the form:

               <stream_index>, <packet_dts>, <packet_pts>, <packet_duration>, <packet_size>, <hash>

       hash is a hexadecimal number representing the computed hash for the packet.

       hash algorithm
           Use the cryptographic hash function specified by the string algorithm.  Supported
           values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320",
           "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384",
           "SHA512", "CRC32" and "adler32".

       Examples

       To compute the SHA-256 hash of the audio and video frames in INPUT, converted to raw audio
       and video packets, and store it in the file out.sha256:

               ffmpeg -i INPUT -f framehash out.sha256

       To print the information to stdout, using the MD5 hash function, use the command:

               ffmpeg -i INPUT -f framehash -hash md5 -

       See also the hash muxer.

   framemd5
       Per-packet MD5 testing format.

       This is a variant of the framehash muxer. Unlike that muxer, it defaults to using the MD5
       hash function.

       Examples

       To compute the MD5 hash of the audio and video frames in INPUT, converted to raw audio and
       video packets, and store it in the file out.md5:

               ffmpeg -i INPUT -f framemd5 out.md5

       To print the information to stdout, use the command:

               ffmpeg -i INPUT -f framemd5 -

       See also the framehash and md5 muxers.

   gif
       Animated GIF muxer.

       It accepts the following options:

       loop
           Set the number of times to loop the output. Use "-1" for no loop, 0 for looping
           indefinitely (default).

       final_delay
           Force the delay (expressed in centiseconds) after the last frame. Each frame ends with
           a delay until the next frame. The default is "-1", which is a special value to tell
           the muxer to re-use the previous delay. In case of a loop, you might want to customize
           this value to mark a pause for instance.

       For example, to encode a gif looping 10 times, with a 5 seconds delay between the loops:

               ffmpeg -i INPUT -loop 10 -final_delay 500 out.gif

       Note 1: if you wish to extract the frames into separate GIF files, you need to force the
       image2 muxer:

               ffmpeg -i INPUT -c:v gif -f image2 "out%d.gif"

       Note 2: the GIF format has a very large time base: the delay between two frames can
       therefore not be smaller than one centi second.

   hash
       Hash testing format.

       This muxer computes and prints a cryptographic hash of all the input audio and video
       frames. This can be used for equality checks without having to do a complete binary
       comparison.

       By default audio frames are converted to signed 16-bit raw audio and video frames to raw
       video before computing the hash, but the output of explicit conversions to other codecs
       can also be used. Timestamps are ignored. It uses the SHA-256 cryptographic hash function
       by default, but supports several other algorithms.

       The output of the muxer consists of a single line of the form: algo=hash, where algo is a
       short string representing the hash function used, and hash is a hexadecimal number
       representing the computed hash.

       hash algorithm
           Use the cryptographic hash function specified by the string algorithm.  Supported
           values include "MD5", "murmur3", "RIPEMD128", "RIPEMD160", "RIPEMD256", "RIPEMD320",
           "SHA160", "SHA224", "SHA256" (default), "SHA512/224", "SHA512/256", "SHA384",
           "SHA512", "CRC32" and "adler32".

       Examples

       To compute the SHA-256 hash of the input converted to raw audio and video, and store it in
       the file out.sha256:

               ffmpeg -i INPUT -f hash out.sha256

       To print an MD5 hash to stdout use the command:

               ffmpeg -i INPUT -f hash -hash md5 -

       See also the framehash muxer.

   hls
       Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming
       (HLS) specification.

       It creates a playlist file, and one or more segment files. The output filename specifies
       the playlist filename.

       By default, the muxer creates a file for each segment produced. These files have the same
       name as the playlist, followed by a sequential number and a .ts extension.

       Make sure to require a closed GOP when encoding and to set the GOP size to fit your
       segment time constraint.

       For example, to convert an input file with ffmpeg:

               ffmpeg -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8

       This example will produce the playlist, out.m3u8, and segment files: out0.ts, out1.ts,
       out2.ts, etc.

       See also the segment muxer, which provides a more generic and flexible implementation of a
       segmenter, and can be used to perform HLS segmentation.

       Options

       This muxer supports the following options:

       hls_init_time seconds
           Set the initial target segment length in seconds. Default value is 0.  Segment will be
           cut on the next key frame after this time has passed on the first m3u8 list.  After
           the initial playlist is filled ffmpeg will cut segments at duration equal to
           "hls_time"

       hls_time seconds
           Set the target segment length in seconds. Default value is 2.  Segment will be cut on
           the next key frame after this time has passed.

       hls_list_size size
           Set the maximum number of playlist entries. If set to 0 the list file will contain all
           the segments. Default value is 5.

       hls_delete_threshold size
           Set the number of unreferenced segments to keep on disk before "hls_flags
           delete_segments" deletes them. Increase this to allow continue clients to download
           segments which were recently referenced in the playlist. Default value is 1, meaning
           segments older than "hls_list_size+1" will be deleted.

       hls_ts_options options_list
           Set output format options using a :-separated list of key=value parameters. Values
           containing ":" special characters must be escaped.

       hls_wrap wrap
           This is a deprecated option, you can use "hls_list_size" and "hls_flags
           delete_segments" instead it

           This option is useful to avoid to fill the disk with many segment files, and limits
           the maximum number of segment files written to disk to wrap.

       hls_start_number_source
           Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") according to the
           specified source.  Unless "hls_flags single_file" is set, it also specifies source of
           starting sequence numbers of segment and subtitle filenames. In any case, if
           "hls_flags append_list" is set and read playlist sequence number is greater than the
           specified start sequence number, then that value will be used as start value.

           It accepts the following values:

           generic (default)
               Set the starting sequence numbers according to start_number option value.

           epoch
               The start number will be the seconds since epoch (1970-01-01 00:00:00)

           datetime
               The start number will be based on the current date/time as YYYYmmddHHMMSS. e.g.
               20161231235759.

       start_number number
           Start the playlist sequence number ("#EXT-X-MEDIA-SEQUENCE") from the specified number
           when hls_start_number_source value is generic. (This is the default case.)  Unless
           "hls_flags single_file" is set, it also specifies starting sequence numbers of segment
           and subtitle filenames.  Default value is 0.

       hls_allow_cache allowcache
           Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments.

       hls_base_url baseurl
           Append baseurl to every entry in the playlist.  Useful to generate playlists with
           absolute paths.

           Note that the playlist sequence number must be unique for each segment and it is not
           to be confused with the segment filename sequence number which can be cyclic, for
           example if the wrap option is specified.

       hls_segment_filename filename
           Set the segment filename. Unless "hls_flags single_file" is set, filename is used as a
           string format with the segment number:

                   ffmpeg -i in.nut -hls_segment_filename 'file%03d.ts' out.m3u8

           This example will produce the playlist, out.m3u8, and segment files: file000.ts,
           file001.ts, file002.ts, etc.

           filename may contain full path or relative path specification, but only the file name
           part without any path info will be contained in the m3u8 segment list.  Should a
           relative path be specified, the path of the created segment files will be relative to
           the current working directory.  When strftime_mkdir is set, the whole expanded value
           of filename will be written into the m3u8 segment list.

           When "var_stream_map" is set with two or more variant streams, the filename pattern
           must contain the string "%v", this string specifies the position of variant stream
           index in the generated segment file names.

                   ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
                     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
                     -hls_segment_filename 'file_%v_%03d.ts' out_%v.m3u8

           This example will produce the playlists segment file sets: file_0_000.ts,
           file_0_001.ts, file_0_002.ts, etc. and file_1_000.ts, file_1_001.ts, file_1_002.ts,
           etc.

           The string "%v" may be present in the filename or in the last directory name
           containing the file. If the string is present in the directory name, then sub-
           directories are created after expanding the directory name pattern. This enables
           creation of segments corresponding to different variant streams in subdirectories.

                   ffmpeg -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
                     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
                     -hls_segment_filename 'vs%v/file_%03d.ts' vs%v/out.m3u8

           This example will produce the playlists segment file sets: vs0/file_000.ts,
           vs0/file_001.ts, vs0/file_002.ts, etc. and vs1/file_000.ts, vs1/file_001.ts,
           vs1/file_002.ts, etc.

       use_localtime
           Same as strftime option, will be deprecated.

       strftime
           Use strftime() on filename to expand the segment filename with localtime.  The segment
           number is also available in this mode, but to use it, you need to specify
           second_level_segment_index hls_flag and %%d will be the specifier.

                   ffmpeg -i in.nut -strftime 1 -hls_segment_filename 'file-%Y%m%d-%s.ts' out.m3u8

           This example will produce the playlist, out.m3u8, and segment files:
           file-20160215-1455569023.ts, file-20160215-1455569024.ts, etc.  Note: On some
           systems/environments, the %s specifier is not available. See
             "strftime()" documentation.

                   ffmpeg -i in.nut -strftime 1 -hls_flags second_level_segment_index -hls_segment_filename 'file-%Y%m%d-%%04d.ts' out.m3u8

           This example will produce the playlist, out.m3u8, and segment files:
           file-20160215-0001.ts, file-20160215-0002.ts, etc.

       use_localtime_mkdir
           Same as strftime_mkdir option, will be deprecated .

       strftime_mkdir
           Used together with -strftime_mkdir, it will create all subdirectories which is
           expanded in filename.

                   ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y%m%d/file-%Y%m%d-%s.ts' out.m3u8

           This example will create a directory 201560215 (if it does not exist), and then
           produce the playlist, out.m3u8, and segment files:
           20160215/file-20160215-1455569023.ts, 20160215/file-20160215-1455569024.ts, etc.

                   ffmpeg -i in.nut -strftime 1 -strftime_mkdir 1 -hls_segment_filename '%Y/%m/%d/file-%Y%m%d-%s.ts' out.m3u8

           This example will create a directory hierarchy 2016/02/15 (if any of them do not
           exist), and then produce the playlist, out.m3u8, and segment files:
           2016/02/15/file-20160215-1455569023.ts, 2016/02/15/file-20160215-1455569024.ts, etc.

       hls_key_info_file key_info_file
           Use the information in key_info_file for segment encryption. The first line of
           key_info_file specifies the key URI written to the playlist. The key URL is used to
           access the encryption key during playback. The second line specifies the path to the
           key file used to obtain the key during the encryption process. The key file is read as
           a single packed array of 16 octets in binary format. The optional third line specifies
           the initialization vector (IV) as a hexadecimal string to be used instead of the
           segment sequence number (default) for encryption. Changes to key_info_file will result
           in segment encryption with the new key/IV and an entry in the playlist for the new key
           URI/IV if "hls_flags periodic_rekey" is enabled.

           Key info file format:

                   <key URI>
                   <key file path>
                   <IV> (optional)

           Example key URIs:

                   http://server/file.key
                   /path/to/file.key
                   file.key

           Example key file paths:

                   file.key
                   /path/to/file.key

           Example IV:

                   0123456789ABCDEF0123456789ABCDEF

           Key info file example:

                   http://server/file.key
                   /path/to/file.key
                   0123456789ABCDEF0123456789ABCDEF

           Example shell script:

                   #!/bin/sh
                   BASE_URL=${1:-'.'}
                   openssl rand 16 > file.key
                   echo $BASE_URL/file.key > file.keyinfo
                   echo file.key >> file.keyinfo
                   echo $(openssl rand -hex 16) >> file.keyinfo
                   ffmpeg -f lavfi -re -i testsrc -c:v h264 -hls_flags delete_segments \
                     -hls_key_info_file file.keyinfo out.m3u8

       -hls_enc enc
           Enable (1) or disable (0) the AES128 encryption.  When enabled every segment generated
           is encrypted and the encryption key is saved as playlist name.key.

       -hls_enc_key key
           Hex-coded 16byte key to encrypt the segments, by default it is randomly generated.

       -hls_enc_key_url keyurl
           If set, keyurl is prepended instead of baseurl to the key filename in the playlist.

       -hls_enc_iv iv
           Hex-coded 16byte initialization vector for every segment instead of the autogenerated
           ones.

       hls_segment_type flags
           Possible values:

           mpegts
               Output segment files in MPEG-2 Transport Stream format. This is compatible with
               all HLS versions.

           fmp4
               Output segment files in fragmented MP4 format, similar to MPEG-DASH.  fmp4 files
               may be used in HLS version 7 and above.

       hls_fmp4_init_filename filename
           Set filename to the fragment files header file, default filename is init.mp4.

           When "var_stream_map" is set with two or more variant streams, the filename pattern
           must contain the string "%v", this string specifies the position of variant stream
           index in the generated init file names.  The string "%v" may be present in the
           filename or in the last directory name containing the file. If the string is present
           in the directory name, then sub-directories are created after expanding the directory
           name pattern. This enables creation of init files corresponding to different variant
           streams in subdirectories.

       hls_flags flags
           Possible values:

           single_file
               If this flag is set, the muxer will store all segments in a single MPEG-TS file,
               and will use byte ranges in the playlist. HLS playlists generated with this way
               will have the version number 4.  For example:

                       ffmpeg -i in.nut -hls_flags single_file out.m3u8

               Will produce the playlist, out.m3u8, and a single segment file, out.ts.

           delete_segments
               Segment files removed from the playlist are deleted after a period of time equal
               to the duration of the segment plus the duration of the playlist.

           append_list
               Append new segments into the end of old segment list, and remove the
               "#EXT-X-ENDLIST" from the old segment list.

           round_durations
               Round the duration info in the playlist file segment info to integer values,
               instead of using floating point.

           discont_start
               Add the "#EXT-X-DISCONTINUITY" tag to the playlist, before the first segment's
               information.

           omit_endlist
               Do not append the "EXT-X-ENDLIST" tag at the end of the playlist.

           periodic_rekey
               The file specified by "hls_key_info_file" will be checked periodically and detect
               updates to the encryption info. Be sure to replace this file atomically, including
               the file containing the AES encryption key.

           independent_segments
               Add the "#EXT-X-INDEPENDENT-SEGMENTS" to playlists that has video segments and
               when all the segments of that playlist are guaranteed to start with a Key frame.

           split_by_time
               Allow segments to start on frames other than keyframes. This improves behavior on
               some players when the time between keyframes is inconsistent, but may make things
               worse on others, and can cause some oddities during seeking. This flag should be
               used with the "hls_time" option.

           program_date_time
               Generate "EXT-X-PROGRAM-DATE-TIME" tags.

           second_level_segment_index
               Makes it possible to use segment indexes as %%d in hls_segment_filename expression
               besides date/time values when strftime is on.  To get fixed width numbers with
               trailing zeroes, %%0xd format is available where x is the required width.

           second_level_segment_size
               Makes it possible to use segment sizes (counted in bytes) as %%s in
               hls_segment_filename expression besides date/time values when strftime is on.  To
               get fixed width numbers with trailing zeroes, %%0xs format is available where x is
               the required width.

           second_level_segment_duration
               Makes it possible to use segment duration (calculated  in microseconds) as %%t in
               hls_segment_filename expression besides date/time values when strftime is on.  To
               get fixed width numbers with trailing zeroes, %%0xt format is available where x is
               the required width.

                       ffmpeg -i sample.mpeg \
                          -f hls -hls_time 3 -hls_list_size 5 \
                          -hls_flags second_level_segment_index+second_level_segment_size+second_level_segment_duration \
                          -strftime 1 -strftime_mkdir 1 -hls_segment_filename "segment_%Y%m%d%H%M%S_%%04d_%%08s_%%013t.ts" stream.m3u8

               This will produce segments like this:
               segment_20170102194334_0003_00122200_0000003000000.ts,
               segment_20170102194334_0004_00120072_0000003000000.ts etc.

           temp_file
               Write segment data to filename.tmp and rename to filename only once the segment is
               complete. A webserver serving up segments can be configured to reject requests to
               *.tmp to prevent access to in-progress segments before they have been added to the
               m3u8 playlist.

       hls_playlist_type event
           Emit "#EXT-X-PLAYLIST-TYPE:EVENT" in the m3u8 header. Forces hls_list_size to 0; the
           playlist can only be appended to.

       hls_playlist_type vod
           Emit "#EXT-X-PLAYLIST-TYPE:VOD" in the m3u8 header. Forces hls_list_size to 0; the
           playlist must not change.

       method
           Use the given HTTP method to create the hls files.

                   ffmpeg -re -i in.ts -f hls -method PUT http://example.com/live/out.m3u8

           This example will upload all the mpegts segment files to the HTTP server using the
           HTTP PUT method, and update the m3u8 files every "refresh" times using the same
           method.  Note that the HTTP server must support the given method for uploading files.

       http_user_agent
           Override User-Agent field in HTTP header. Applicable only for HTTP output.

       var_stream_map
           Map string which specifies how to group the audio, video and subtitle streams into
           different variant streams. The variant stream groups are separated by space.  Expected
           string format is like this "a:0,v:0 a:1,v:1 ....". Here a:, v:, s: are the keys to
           specify audio, video and subtitle streams respectively.  Allowed values are 0 to 9
           (limited just based on practical usage).

           When there are two or more variant streams, the output filename pattern must contain
           the string "%v", this string specifies the position of variant stream index in the
           output media playlist filenames. The string "%v" may be present in the filename or in
           the last directory name containing the file. If the string is present in the directory
           name, then sub-directories are created after expanding the directory name pattern.
           This enables creation of variant streams in subdirectories.

                   ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
                     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
                     http://example.com/live/out_%v.m3u8

           This example creates two hls variant streams. The first variant stream will contain
           video stream of bitrate 1000k and audio stream of bitrate 64k and the second variant
           stream will contain video stream of bitrate 256k and audio stream of bitrate 32k.
           Here, two media playlist with file names out_0.m3u8 and out_1.m3u8 will be created.

                   ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k \
                     -map 0:v -map 0:a -map 0:v -f hls -var_stream_map "v:0 a:0 v:1" \
                     http://example.com/live/out_%v.m3u8

           This example creates three hls variant streams. The first variant stream will be a
           video only stream with video bitrate 1000k, the second variant stream will be an audio
           only stream with bitrate 64k and the third variant stream will be a video only stream
           with bitrate 256k. Here, three media playlist with file names out_0.m3u8, out_1.m3u8
           and out_2.m3u8 will be created.

                   ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
                     -map 0:v -map 0:a -map 0:v -map 0:a -f hls -var_stream_map "v:0,a:0 v:1,a:1" \
                     http://example.com/live/vs_%v/out.m3u8

           This example creates the variant streams in subdirectories. Here, the first media
           playlist is created at http://example.com/live/vs_0/out.m3u8 and the second one at
           http://example.com/live/vs_1/out.m3u8.

                   ffmpeg -re -i in.ts -b:a:0 32k -b:a:1 64k -b:v:0 1000k -b:v:1 3000k  \
                     -map 0:a -map 0:a -map 0:v -map 0:v -f hls \
                     -var_stream_map "a:0,agroup:aud_low a:1,agroup:aud_high v:0,agroup:aud_low v:1,agroup:aud_high" \
                     -master_pl_name master.m3u8 \
                     http://example.com/live/out_%v.m3u8

           This example creates two audio only and two video only variant streams. In addition to
           the #EXT-X-STREAM-INF tag for each variant stream in the master playlist, #EXT-X-MEDIA
           tag is also added for the two audio only variant streams and they are mapped to the
           two video only variant streams with audio group names 'aud_low' and 'aud_high'.

           By default, a single hls variant containing all the encoded streams is created.

       cc_stream_map
           Map string which specifies different closed captions groups and their attributes. The
           closed captions stream groups are separated by space.  Expected string format is like
           this "ccgroup:<group name>,instreamid:<INSTREAM-ID>,language:<language code> ....".
           'ccgroup' and 'instreamid' are mandatory attributes. 'language' is an optional
           attribute.  The closed captions groups configured using this option are mapped to
           different variant streams by providing the same 'ccgroup' name in the "var_stream_map"
           string. If "var_stream_map" is not set, then the first available ccgroup in
           "cc_stream_map" is mapped to the output variant stream. The examples for these two use
           cases are given below.

                   ffmpeg -re -i in.ts -b:v 1000k -b:a 64k -a53cc 1 -f hls \
                     -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en" \
                     -master_pl_name master.m3u8 \
                     http://example.com/live/out.m3u8

           This example adds "#EXT-X-MEDIA" tag with "TYPE=CLOSED-CAPTIONS" in the master
           playlist with group name 'cc', langauge 'en' (english) and INSTREAM-ID 'CC1'. Also, it
           adds "CLOSED-CAPTIONS" attribute with group name 'cc' for the output variant stream.

                   ffmpeg -re -i in.ts -b:v:0 1000k -b:v:1 256k -b:a:0 64k -b:a:1 32k \
                     -a53cc:0 1 -a53cc:1 1\
                     -map 0:v -map 0:a -map 0:v -map 0:a -f hls \
                     -cc_stream_map "ccgroup:cc,instreamid:CC1,language:en ccgroup:cc,instreamid:CC2,language:sp" \
                     -var_stream_map "v:0,a:0,ccgroup:cc v:1,a:1,ccgroup:cc" \
                     -master_pl_name master.m3u8 \
                     http://example.com/live/out_%v.m3u8

           This example adds two "#EXT-X-MEDIA" tags with "TYPE=CLOSED-CAPTIONS" in the master
           playlist for the INSTREAM-IDs 'CC1' and 'CC2'. Also, it adds "CLOSED-CAPTIONS"
           attribute with group name 'cc' for the two output variant streams.

       master_pl_name
           Create HLS master playlist with the given name.

                   ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 http://example.com/live/out.m3u8

           This example creates HLS master playlist with name master.m3u8 and it is published at
           http://example.com/live/

       master_pl_publish_rate
           Publish master play list repeatedly every after specified number of segment intervals.

                   ffmpeg -re -i in.ts -f hls -master_pl_name master.m3u8 \
                   -hls_time 2 -master_pl_publish_rate 30 http://example.com/live/out.m3u8

           This example creates HLS master playlist with name master.m3u8 and keep publishing it
           repeatedly every after 30 segments i.e. every after 60s.

       http_persistent
           Use persistent HTTP connections. Applicable only for HTTP output.

       timeout
           Set timeout for socket I/O operations. Applicable only for HTTP output.

   ico
       ICO file muxer.

       Microsoft's icon file format (ICO) has some strict limitations that should be noted:

       ·   Size cannot exceed 256 pixels in any dimension

       ·   Only BMP and PNG images can be stored

       ·   If a BMP image is used, it must be one of the following pixel formats:

                   BMP Bit Depth      FFmpeg Pixel Format
                   1bit               pal8
                   4bit               pal8
                   8bit               pal8
                   16bit              rgb555le
                   24bit              bgr24
                   32bit              bgra

       ·   If a BMP image is used, it must use the BITMAPINFOHEADER DIB header

       ·   If a PNG image is used, it must use the rgba pixel format

   image2
       Image file muxer.

       The image file muxer writes video frames to image files.

       The output filenames are specified by a pattern, which can be used to produce sequentially
       numbered series of files.  The pattern may contain the string "%d" or "%0Nd", this string
       specifies the position of the characters representing a numbering in the filenames. If the
       form "%0Nd" is used, the string representing the number in each filename is 0-padded to N
       digits. The literal character '%' can be specified in the pattern with the string "%%".

       If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will
       contain the number 1, all the following numbers will be sequential.

       The pattern may contain a suffix which is used to automatically determine the format of
       the image files to write.

       For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form
       img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.  The pattern "img%%-%d.jpg" will specify
       a sequence of filenames of the form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc.

       Examples

       The following example shows how to use ffmpeg for creating a sequence of files
       img-001.jpeg, img-002.jpeg, ..., taking one image every second from the input video:

               ffmpeg -i in.avi -vsync cfr -r 1 -f image2 'img-%03d.jpeg'

       Note that with ffmpeg, if the format is not specified with the "-f" option and the output
       filename specifies an image file format, the image2 muxer is automatically selected, so
       the previous command can be written as:

               ffmpeg -i in.avi -vsync cfr -r 1 'img-%03d.jpeg'

       Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to
       create a single image file img.jpeg from the start of the input video you can employ the
       command:

               ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg

       The strftime option allows you to expand the filename with date and time information.
       Check the documentation of the "strftime()" function for the syntax.

       For example to generate image files from the "strftime()" "%Y-%m-%d_%H-%M-%S" pattern, the
       following ffmpeg command can be used:

               ffmpeg -f v4l2 -r 1 -i /dev/video0 -f image2 -strftime 1 "%Y-%m-%d_%H-%M-%S.jpg"

       You can set the file name with current frame's PTS:

               ffmpeg -f v4l2 -r 1 -i /dev/video0 -copyts -f image2 -frame_pts true %d.jpg"

       Options

       frame_pts
           If set to 1, expand the filename with pts from pkt->pts.  Default value is 0.

       start_number
           Start the sequence from the specified number. Default value is 1.

       update
           If set to 1, the filename will always be interpreted as just a filename, not a
           pattern, and the corresponding file will be continuously overwritten with new images.
           Default value is 0.

       strftime
           If set to 1, expand the filename with date and time information from "strftime()".
           Default value is 0.

       The image muxer supports the .Y.U.V image file format. This format is special in that that
       each image frame consists of three files, for each of the YUV420P components. To read or
       write this image file format, specify the name of the '.Y' file. The muxer will
       automatically open the '.U' and '.V' files as required.

   matroska
       Matroska container muxer.

       This muxer implements the matroska and webm container specs.

       Metadata

       The recognized metadata settings in this muxer are:

       title
           Set title name provided to a single track.

       language
           Specify the language of the track in the Matroska languages form.

           The language can be either the 3 letters bibliographic ISO-639-2 (ISO 639-2/B) form
           (like "fre" for French), or a language code mixed with a country code for specialities
           in languages (like "fre-ca" for Canadian French).

       stereo_mode
           Set stereo 3D video layout of two views in a single video track.

           The following values are recognized:

           mono
               video is not stereo

           left_right
               Both views are arranged side by side, Left-eye view is on the left

           bottom_top
               Both views are arranged in top-bottom orientation, Left-eye view is at bottom

           top_bottom
               Both views are arranged in top-bottom orientation, Left-eye view is on top

           checkerboard_rl
               Each view is arranged in a checkerboard interleaved pattern, Left-eye view being
               first

           checkerboard_lr
               Each view is arranged in a checkerboard interleaved pattern, Right-eye view being
               first

           row_interleaved_rl
               Each view is constituted by a row based interleaving, Right-eye view is first row

           row_interleaved_lr
               Each view is constituted by a row based interleaving, Left-eye view is first row

           col_interleaved_rl
               Both views are arranged in a column based interleaving manner, Right-eye view is
               first column

           col_interleaved_lr
               Both views are arranged in a column based interleaving manner, Left-eye view is
               first column

           anaglyph_cyan_red
               All frames are in anaglyph format viewable through red-cyan filters

           right_left
               Both views are arranged side by side, Right-eye view is on the left

           anaglyph_green_magenta
               All frames are in anaglyph format viewable through green-magenta filters

           block_lr
               Both eyes laced in one Block, Left-eye view is first

           block_rl
               Both eyes laced in one Block, Right-eye view is first

       For example a 3D WebM clip can be created using the following command line:

               ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm

       Options

       This muxer supports the following options:

       reserve_index_space
           By default, this muxer writes the index for seeking (called cues in Matroska terms) at
           the end of the file, because it cannot know in advance how much space to leave for the
           index at the beginning of the file. However for some use cases -- e.g.  streaming
           where seeking is possible but slow -- it is useful to put the index at the beginning
           of the file.

           If this option is set to a non-zero value, the muxer will reserve a given amount of
           space in the file header and then try to write the cues there when the muxing
           finishes. If the available space does not suffice, muxing will fail. A safe size for
           most use cases should be about 50kB per hour of video.

           Note that cues are only written if the output is seekable and this option will have no
           effect if it is not.

   md5
       MD5 testing format.

       This is a variant of the hash muxer. Unlike that muxer, it defaults to using the MD5 hash
       function.

       Examples

       To compute the MD5 hash of the input converted to raw audio and video, and store it in the
       file out.md5:

               ffmpeg -i INPUT -f md5 out.md5

       You can print the MD5 to stdout with the command:

               ffmpeg -i INPUT -f md5 -

       See also the hash and framemd5 muxers.

   mov, mp4, ismv
       MOV/MP4/ISMV (Smooth Streaming) muxer.

       The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the
       metadata about all packets stored in one location (written at the end of the file, it can
       be moved to the start for better playback by adding faststart to the movflags, or using
       the qt-faststart tool). A fragmented file consists of a number of fragments, where packets
       and metadata about these packets are stored together. Writing a fragmented file has the
       advantage that the file is decodable even if the writing is interrupted (while a normal
       MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when
       writing very long files (since writing normal MOV/MP4 files stores info about every single
       packet in memory until the file is closed). The downside is that it is less compatible
       with other applications.

       Options

       Fragmentation is enabled by setting one of the AVOptions that define how to cut the file
       into fragments:

       -moov_size bytes
           Reserves space for the moov atom at the beginning of the file instead of placing the
           moov atom at the end. If the space reserved is insufficient, muxing will fail.

       -movflags frag_keyframe
           Start a new fragment at each video keyframe.

       -frag_duration duration
           Create fragments that are duration microseconds long.

       -frag_size size
           Create fragments that contain up to size bytes of payload data.

       -movflags frag_custom
           Allow the caller to manually choose when to cut fragments, by calling
           "av_write_frame(ctx, NULL)" to write a fragment with the packets written so far. (This
           is only useful with other applications integrating libavformat, not from ffmpeg.)

       -min_frag_duration duration
           Don't create fragments that are shorter than duration microseconds long.

       If more than one condition is specified, fragments are cut when one of the specified
       conditions is fulfilled. The exception to this is "-min_frag_duration", which has to be
       fulfilled for any of the other conditions to apply.

       Additionally, the way the output file is written can be adjusted through a few other
       options:

       -movflags empty_moov
           Write an initial moov atom directly at the start of the file, without describing any
           samples in it. Generally, an mdat/moov pair is written at the start of the file, as a
           normal MOV/MP4 file, containing only a short portion of the file. With this option
           set, there is no initial mdat atom, and the moov atom only describes the tracks but
           has a zero duration.

           This option is implicitly set when writing ismv (Smooth Streaming) files.

       -movflags separate_moof
           Write a separate moof (movie fragment) atom for each track. Normally, packets for all
           tracks are written in a moof atom (which is slightly more efficient), but with this
           option set, the muxer writes one moof/mdat pair for each track, making it easier to
           separate tracks.

           This option is implicitly set when writing ismv (Smooth Streaming) files.

       -movflags faststart
           Run a second pass moving the index (moov atom) to the beginning of the file.  This
           operation can take a while, and will not work in various situations such as fragmented
           output, thus it is not enabled by default.

       -movflags rtphint
           Add RTP hinting tracks to the output file.

       -movflags disable_chpl
           Disable Nero chapter markers (chpl atom).  Normally, both Nero chapters and a
           QuickTime chapter track are written to the file. With this option set, only the
           QuickTime chapter track will be written. Nero chapters can cause failures when the
           file is reprocessed with certain tagging programs, like mp3Tag 2.61a and iTunes 11.3,
           most likely other versions are affected as well.

       -movflags omit_tfhd_offset
           Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments
           to absolute byte positions in the file/streams.

       -movflags default_base_moof
           Similarly to the omit_tfhd_offset, this flag avoids writing the absolute
           base_data_offset field in tfhd atoms, but does so by using the new default-base-is-
           moof flag instead. This flag is new from 14496-12:2012. This may make the fragments
           easier to parse in certain circumstances (avoiding basing track fragment location
           calculations on the implicit end of the previous track fragment).

       -write_tmcd
           Specify "on" to force writing a timecode track, "off" to disable it and "auto" to
           write a timecode track only for mov and mp4 output (default).

       -movflags negative_cts_offsets
           Enables utilization of version 1 of the CTTS box, in which the CTS offsets can be
           negative. This enables the initial sample to have DTS/CTS of zero, and reduces the
           need for edit lists for some cases such as video tracks with B-frames. Additionally,
           eases conformance with the DASH-IF interoperability guidelines.

           This option is implicitly set when writing ismv (Smooth Streaming) files.

       -write_prft
           Write producer time reference box (PRFT) with a specified time source for the NTP
           field in the PRFT box. Set value as wallclock to specify timesource as wallclock time
           and pts to specify timesource as input packets' PTS values.

           Setting value to pts is applicable only for a live encoding use case, where PTS values
           are set as as wallclock time at the source. For example, an encoding use case with
           decklink capture source where video_pts and audio_pts are set to abs_wallclock.

       Example

       Smooth Streaming content can be pushed in real time to a publishing point on IIS with this
       muxer. Example:

               ffmpeg -re <<normal input/transcoding options>> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)

       Audible AAX

       Audible AAX files are encrypted M4B files, and they can be decrypted by specifying a 4
       byte activation secret.

               ffmpeg -activation_bytes 1CEB00DA -i test.aax -vn -c:a copy output.mp4

   mp3
       The MP3 muxer writes a raw MP3 stream with the following optional features:

       ·   An ID3v2 metadata header at the beginning (enabled by default). Versions 2.3 and 2.4
           are supported, the "id3v2_version" private option controls which one is used (3 or 4).
           Setting "id3v2_version" to 0 disables the ID3v2 header completely.

           The muxer supports writing attached pictures (APIC frames) to the ID3v2 header.  The
           pictures are supplied to the muxer in form of a video stream with a single packet.
           There can be any number of those streams, each will correspond to a single APIC frame.
           The stream metadata tags title and comment map to APIC description and picture type
           respectively. See <http://id3.org/id3v2.4.0-frames> for allowed picture types.

           Note that the APIC frames must be written at the beginning, so the muxer will buffer
           the audio frames until it gets all the pictures. It is therefore advised to provide
           the pictures as soon as possible to avoid excessive buffering.

       ·   A Xing/LAME frame right after the ID3v2 header (if present). It is enabled by default,
           but will be written only if the output is seekable. The "write_xing" private option
           can be used to disable it.  The frame contains various information that may be useful
           to the decoder, like the audio duration or encoder delay.

       ·   A legacy ID3v1 tag at the end of the file (disabled by default). It may be enabled
           with the "write_id3v1" private option, but as its capabilities are very limited, its
           usage is not recommended.

       Examples:

       Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

               ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

       To attach a picture to an mp3 file select both the audio and the picture stream with
       "map":

               ffmpeg -i input.mp3 -i cover.png -c copy -map 0 -map 1
               -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3

       Write a "clean" MP3 without any extra features:

               ffmpeg -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

   mpegts
       MPEG transport stream muxer.

       This muxer implements ISO 13818-1 and part of ETSI EN 300 468.

       The recognized metadata settings in mpegts muxer are "service_provider" and
       "service_name". If they are not set the default for "service_provider" is FFmpeg and the
       default for "service_name" is Service01.

       Options

       The muxer options are:

       mpegts_transport_stream_id integer
           Set the transport_stream_id. This identifies a transponder in DVB.  Default is 0x0001.

       mpegts_original_network_id integer
           Set the original_network_id. This is unique identifier of a network in DVB. Its main
           use is in the unique identification of a service through the path Original_Network_ID,
           Transport_Stream_ID. Default is 0x0001.

       mpegts_service_id integer
           Set the service_id, also known as program in DVB. Default is 0x0001.

       mpegts_service_type integer
           Set the program service_type. Default is "digital_tv".  Accepts the following options:

           hex_value
               Any hexdecimal value between 0x01 to 0xff as defined in ETSI 300 468.

           digital_tv
               Digital TV service.

           digital_radio
               Digital Radio service.

           teletext
               Teletext service.

           advanced_codec_digital_radio
               Advanced Codec Digital Radio service.

           mpeg2_digital_hdtv
               MPEG2 Digital HDTV service.

           advanced_codec_digital_sdtv
               Advanced Codec Digital SDTV service.

           advanced_codec_digital_hdtv
               Advanced Codec Digital HDTV service.

       mpegts_pmt_start_pid integer
           Set the first PID for PMT. Default is 0x1000. Max is 0x1f00.

       mpegts_start_pid integer
           Set the first PID for data packets. Default is 0x0100. Max is 0x0f00.

       mpegts_m2ts_mode boolean
           Enable m2ts mode if set to 1. Default value is "-1" which disables m2ts mode.

       muxrate integer
           Set a constant muxrate. Default is VBR.

       pes_payload_size integer
           Set minimum PES packet payload in bytes. Default is 2930.

       mpegts_flags flags
           Set mpegts flags. Accepts the following options:

           resend_headers
               Reemit PAT/PMT before writing the next packet.

           latm
               Use LATM packetization for AAC.

           pat_pmt_at_frames
               Reemit PAT and PMT at each video frame.

           system_b
               Conform to System B (DVB) instead of System A (ATSC).

           initial_discontinuity
               Mark the initial packet of each stream as discontinuity.

       resend_headers integer
           Reemit PAT/PMT before writing the next packet. This option is deprecated: use
           mpegts_flags instead.

       mpegts_copyts boolean
           Preserve original timestamps, if value is set to 1. Default value is "-1", which
           results in shifting timestamps so that they start from 0.

       omit_video_pes_length boolean
           Omit the PES packet length for video packets. Default is 1 (true).

       pcr_period integer
           Override the default PCR retransmission time in milliseconds. Ignored if variable
           muxrate is selected. Default is 20.

       pat_period double
           Maximum time in seconds between PAT/PMT tables.

       sdt_period double
           Maximum time in seconds between SDT tables.

       tables_version integer
           Set PAT, PMT and SDT version (default 0, valid values are from 0 to 31, inclusively).
           This option allows updating stream structure so that standard consumer may detect the
           change. To do so, reopen output "AVFormatContext" (in case of API usage) or restart
           ffmpeg instance, cyclically changing tables_version value:

                   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
                   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
                   ...
                   ffmpeg -i source3.ts -codec copy -f mpegts -tables_version 31 udp://1.1.1.1:1111
                   ffmpeg -i source1.ts -codec copy -f mpegts -tables_version 0 udp://1.1.1.1:1111
                   ffmpeg -i source2.ts -codec copy -f mpegts -tables_version 1 udp://1.1.1.1:1111
                   ...

       Example

               ffmpeg -i file.mpg -c copy \
                    -mpegts_original_network_id 0x1122 \
                    -mpegts_transport_stream_id 0x3344 \
                    -mpegts_service_id 0x5566 \
                    -mpegts_pmt_start_pid 0x1500 \
                    -mpegts_start_pid 0x150 \
                    -metadata service_provider="Some provider" \
                    -metadata service_name="Some Channel" \
                    out.ts

   mxf, mxf_d10
       MXF muxer.

       Options

       The muxer options are:

       store_user_comments bool
           Set if user comments should be stored if available or never.  IRT D-10 does not allow
           user comments. The default is thus to write them for mxf but not for mxf_d10

   null
       Null muxer.

       This muxer does not generate any output file, it is mainly useful for testing or
       benchmarking purposes.

       For example to benchmark decoding with ffmpeg you can use the command:

               ffmpeg -benchmark -i INPUT -f null out.null

       Note that the above command does not read or write the out.null file, but specifying the
       output file is required by the ffmpeg syntax.

       Alternatively you can write the command as:

               ffmpeg -benchmark -i INPUT -f null -

   nut
       -syncpoints flags
           Change the syncpoint usage in nut:

           default use the normal low-overhead seeking aids.
           none do not use the syncpoints at all, reducing the overhead but making the stream
           non-seekable;
                   Use of this option is not recommended, as the resulting files are very damage
                   sensitive and seeking is not possible. Also in general the overhead from
                   syncpoints is negligible. Note, -C<write_index> 0 can be used to disable
                   all growing data tables, allowing to mux endless streams with limited memory
                   and without these disadvantages.

           timestamped extend the syncpoint with a wallclock field.

           The none and timestamped flags are experimental.

       -write_index bool
           Write index at the end, the default is to write an index.

               ffmpeg -i INPUT -f_strict experimental -syncpoints none - | processor

   ogg
       Ogg container muxer.

       -page_duration duration
           Preferred page duration, in microseconds. The muxer will attempt to create pages that
           are approximately duration microseconds long. This allows the user to compromise
           between seek granularity and container overhead. The default is 1 second. A value of 0
           will fill all segments, making pages as large as possible. A value of 1 will
           effectively use 1 packet-per-page in most situations, giving a small seek granularity
           at the cost of additional container overhead.

       -serial_offset value
           Serial value from which to set the streams serial number.  Setting it to different and
           sufficiently large values ensures that the produced ogg files can be safely chained.

   segment, stream_segment, ssegment
       Basic stream segmenter.

       This muxer outputs streams to a number of separate files of nearly fixed duration. Output
       filename pattern can be set in a fashion similar to image2, or by using a "strftime"
       template if the strftime option is enabled.

       "stream_segment" is a variant of the muxer used to write to streaming output formats, i.e.
       which do not require global headers, and is recommended for outputting e.g. to MPEG
       transport stream segments.  "ssegment" is a shorter alias for "stream_segment".

       Every segment starts with a keyframe of the selected reference stream, which is set
       through the reference_stream option.

       Note that if you want accurate splitting for a video file, you need to make the input key
       frames correspond to the exact splitting times expected by the segmenter, or the segment
       muxer will start the new segment with the key frame found next after the specified start
       time.

       The segment muxer works best with a single constant frame rate video.

       Optionally it can generate a list of the created segments, by setting the option
       segment_list. The list type is specified by the segment_list_type option. The entry
       filenames in the segment list are set by default to the basename of the corresponding
       segment files.

       See also the hls muxer, which provides a more specific implementation for HLS
       segmentation.

       Options

       The segment muxer supports the following options:

       increment_tc 1|0
           if set to 1, increment timecode between each segment If this is selected, the input
           need to have a timecode in the first video stream. Default value is 0.

       reference_stream specifier
           Set the reference stream, as specified by the string specifier.  If specifier is set
           to "auto", the reference is chosen automatically. Otherwise it must be a stream
           specifier (see the ``Stream specifiers'' chapter in the ffmpeg manual) which specifies
           the reference stream. The default value is "auto".

       segment_format format
           Override the inner container format, by default it is guessed by the filename
           extension.

       segment_format_options options_list
           Set output format options using a :-separated list of key=value parameters. Values
           containing the ":" special character must be escaped.

       segment_list name
           Generate also a listfile named name. If not specified no listfile is generated.

       segment_list_flags flags
           Set flags affecting the segment list generation.

           It currently supports the following flags:

           cache
               Allow caching (only affects M3U8 list files).

           live
               Allow live-friendly file generation.

       segment_list_size size
           Update the list file so that it contains at most size segments. If 0 the list file
           will contain all the segments. Default value is 0.

       segment_list_entry_prefix prefix
           Prepend prefix to each entry. Useful to generate absolute paths.  By default no prefix
           is applied.

       segment_list_type type
           Select the listing format.

           The following values are recognized:

           flat
               Generate a flat list for the created segments, one segment per line.

           csv, ext
               Generate a list for the created segments, one segment per line, each line matching
               the format (comma-separated values):

                       <segment_filename>,<segment_start_time>,<segment_end_time>

               segment_filename is the name of the output file generated by the muxer according
               to the provided pattern. CSV escaping (according to RFC4180) is applied if
               required.

               segment_start_time and segment_end_time specify the segment start and end time
               expressed in seconds.

               A list file with the suffix ".csv" or ".ext" will auto-select this format.

               ext is deprecated in favor or csv.

           ffconcat
               Generate an ffconcat file for the created segments. The resulting file can be read
               using the FFmpeg concat demuxer.

               A list file with the suffix ".ffcat" or ".ffconcat" will auto-select this format.

           m3u8
               Generate an extended M3U8 file, version 3, compliant with
               <http://tools.ietf.org/id/draft-pantos-http-live-streaming>.

               A list file with the suffix ".m3u8" will auto-select this format.

           If not specified the type is guessed from the list file name suffix.

       segment_time time
           Set segment duration to time, the value must be a duration specification. Default
           value is "2". See also the segment_times option.

           Note that splitting may not be accurate, unless you force the reference stream key-
           frames at the given time. See the introductory notice and the examples below.

       segment_atclocktime 1|0
           If set to "1" split at regular clock time intervals starting from 00:00 o'clock. The
           time value specified in segment_time is used for setting the length of the splitting
           interval.

           For example with segment_time set to "900" this makes it possible to create files at
           12:00 o'clock, 12:15, 12:30, etc.

           Default value is "0".

       segment_clocktime_offset duration
           Delay the segment splitting times with the specified duration when using
           segment_atclocktime.

           For example with segment_time set to "900" and segment_clocktime_offset set to "300"
           this makes it possible to create files at 12:05, 12:20, 12:35, etc.

           Default value is "0".

       segment_clocktime_wrap_duration duration
           Force the segmenter to only start a new segment if a packet reaches the muxer within
           the specified duration after the segmenting clock time. This way you can make the
           segmenter more resilient to backward local time jumps, such as leap seconds or
           transition to standard time from daylight savings time.

           Default is the maximum possible duration which means starting a new segment regardless
           of the elapsed time since the last clock time.

       segment_time_delta delta
           Specify the accuracy time when selecting the start time for a segment, expressed as a
           duration specification. Default value is "0".

           When delta is specified a key-frame will start a new segment if its PTS satisfies the
           relation:

                   PTS >= start_time - time_delta

           This option is useful when splitting video content, which is always split at GOP
           boundaries, in case a key frame is found just before the specified split time.

           In particular may be used in combination with the ffmpeg option force_key_frames. The
           key frame times specified by force_key_frames may not be set accurately because of
           rounding issues, with the consequence that a key frame time may result set just before
           the specified time. For constant frame rate videos a value of 1/(2*frame_rate) should
           address the worst case mismatch between the specified time and the time set by
           force_key_frames.

       segment_times times
           Specify a list of split points. times contains a list of comma separated duration
           specifications, in increasing order. See also the segment_time option.

       segment_frames frames
           Specify a list of split video frame numbers. frames contains a list of comma separated
           integer numbers, in increasing order.

           This option specifies to start a new segment whenever a reference stream key frame is
           found and the sequential number (starting from 0) of the frame is greater or equal to
           the next value in the list.

       segment_wrap limit
           Wrap around segment index once it reaches limit.

       segment_start_number number
           Set the sequence number of the first segment. Defaults to 0.

       strftime 1|0
           Use the "strftime" function to define the name of the new segments to write. If this
           is selected, the output segment name must contain a "strftime" function template.
           Default value is 0.

       break_non_keyframes 1|0
           If enabled, allow segments to start on frames other than keyframes. This improves
           behavior on some players when the time between keyframes is inconsistent, but may make
           things worse on others, and can cause some oddities during seeking. Defaults to 0.

       reset_timestamps 1|0
           Reset timestamps at the beginning of each segment, so that each segment will start
           with near-zero timestamps. It is meant to ease the playback of the generated segments.
           May not work with some combinations of muxers/codecs. It is set to 0 by default.

       initial_offset offset
           Specify timestamp offset to apply to the output packet timestamps. The argument must
           be a time duration specification, and defaults to 0.

       write_empty_segments 1|0
           If enabled, write an empty segment if there are no packets during the period a segment
           would usually span. Otherwise, the segment will be filled with the next packet
           written. Defaults to 0.

       Make sure to require a closed GOP when encoding and to set the GOP size to fit your
       segment time constraint.

       Examples

       ·   Remux the content of file in.mkv to a list of segments out-000.nut, out-001.nut, etc.,
           and write the list of generated segments to out.list:

                   ffmpeg -i in.mkv -codec hevc -flags +cgop -g 60 -map 0 -f segment -segment_list out.list out%03d.nut

       ·   Segment input and set output format options for the output segments:

                   ffmpeg -i in.mkv -f segment -segment_time 10 -segment_format_options movflags=+faststart out%03d.mp4

       ·   Segment the input file according to the split points specified by the segment_times
           option:

                   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut

       ·   Use the ffmpeg force_key_frames option to force key frames in the input at the
           specified location, together with the segment option segment_time_delta to account for
           possible roundings operated when setting key frame times.

                   ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -codec:v mpeg4 -codec:a pcm_s16le -map 0 \
                   -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut

           In order to force key frames on the input file, transcoding is required.

       ·   Segment the input file by splitting the input file according to the frame numbers
           sequence specified with the segment_frames option:

                   ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_frames 100,200,300,500,800 out%03d.nut

       ·   Convert the in.mkv to TS segments using the "libx264" and "aac" encoders:

                   ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a aac -f ssegment -segment_list out.list out%03d.ts

       ·   Segment the input file, and create an M3U8 live playlist (can be used as live HLS
           source):

                   ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \
                   -segment_list_flags +live -segment_time 10 out%03d.mkv

   smoothstreaming
       Smooth Streaming muxer generates a set of files (Manifest, chunks) suitable for serving
       with conventional web server.

       window_size
           Specify the number of fragments kept in the manifest. Default 0 (keep all).

       extra_window_size
           Specify the number of fragments kept outside of the manifest before removing from
           disk. Default 5.

       lookahead_count
           Specify the number of lookahead fragments. Default 2.

       min_frag_duration
           Specify the minimum fragment duration (in microseconds). Default 5000000.

       remove_at_exit
           Specify whether to remove all fragments when finished. Default 0 (do not remove).

   fifo
       The fifo pseudo-muxer allows the separation of encoding and muxing by using first-in-
       first-out queue and running the actual muxer in a separate thread. This is especially
       useful in combination with the tee muxer and can be used to send data to several
       destinations with different reliability/writing speed/latency.

       API users should be aware that callback functions (interrupt_callback, io_open and
       io_close) used within its AVFormatContext must be thread-safe.

       The behavior of the fifo muxer if the queue fills up or if the output fails is selectable,

       ·   output can be transparently restarted with configurable delay between retries based on
           real time or time of the processed stream.

       ·   encoding can be blocked during temporary failure, or continue transparently dropping
           packets in case fifo queue fills up.

       fifo_format
           Specify the format name. Useful if it cannot be guessed from the output name suffix.

       queue_size
           Specify size of the queue (number of packets). Default value is 60.

       format_opts
           Specify format options for the underlying muxer. Muxer options can be specified as a
           list of key=value pairs separated by ':'.

       drop_pkts_on_overflow bool
           If set to 1 (true), in case the fifo queue fills up, packets will be dropped rather
           than blocking the encoder. This makes it possible to continue streaming without
           delaying the input, at the cost of omitting part of the stream. By default this option
           is set to 0 (false), so in such cases the encoder will be blocked until the muxer
           processes some of the packets and none of them is lost.

       attempt_recovery bool
           If failure occurs, attempt to recover the output. This is especially useful when used
           with network output, since it makes it possible to restart streaming transparently.
           By default this option is set to 0 (false).

       max_recovery_attempts
           Sets maximum number of successive unsuccessful recovery attempts after which the
           output fails permanently. By default this option is set to 0 (unlimited).

       recovery_wait_time duration
           Waiting time before the next recovery attempt after previous unsuccessful recovery
           attempt. Default value is 5 seconds.

       recovery_wait_streamtime bool
           If set to 0 (false), the real time is used when waiting for the recovery attempt (i.e.
           the recovery will be attempted after at least recovery_wait_time seconds).  If set to
           1 (true), the time of the processed stream is taken into account instead (i.e. the
           recovery will be attempted after at least recovery_wait_time seconds of the stream is
           omitted).  By default, this option is set to 0 (false).

       recover_any_error bool
           If set to 1 (true), recovery will be attempted regardless of type of the error causing
           the failure. By default this option is set to 0 (false) and in case of certain
           (usually permanent) errors the recovery is not attempted even when attempt_recovery is
           set to 1.

       restart_with_keyframe bool
           Specify whether to wait for the keyframe after recovering from queue overflow or
           failure. This option is set to 0 (false) by default.

       Examples

       ·   Stream something to rtmp server, continue processing the stream at real-time rate even
           in case of temporary failure (network outage) and attempt to recover streaming every
           second indefinitely.

                   ffmpeg -re -i ... -c:v libx264 -c:a aac -f fifo -fifo_format flv -map 0:v -map 0:a
                     -drop_pkts_on_overflow 1 -attempt_recovery 1 -recovery_wait_time 1 rtmp://example.com/live/stream_name

   tee
       The tee muxer can be used to write the same data to several outputs, such as files or
       streams.  It can be used, for example, to stream a video over a network and save it to
       disk at the same time.

       It is different from specifying several outputs to the ffmpeg command-line tool. With the
       tee muxer, the audio and video data will be encoded only once.  With conventional multiple
       outputs, multiple encoding operations in parallel are initiated, which can be a very
       expensive process. The tee muxer is not useful when using the libavformat API directly
       because it is then possible to feed the same packets to several muxers directly.

       Since the tee muxer does not represent any particular output format, ffmpeg cannot auto-
       select output streams. So all streams intended for output must be specified using "-map".
       See the examples below.

       Some encoders may need different options depending on the output format; the auto-
       detection of this can not work with the tee muxer, so they need to be explicitly
       specified.  The main example is the global_header flag.

       The slave outputs are specified in the file name given to the muxer, separated by '|'. If
       any of the slave name contains the '|' separator, leading or trailing spaces or any
       special character, those must be escaped (see the "Quoting and escaping" section in the
       ffmpeg-utils(1) manual).

       Options

       use_fifo bool
           If set to 1, slave outputs will be processed in separate threads using the fifo muxer.
           This allows to compensate for different speed/latency/reliability of outputs and setup
           transparent recovery. By default this feature is turned off.

       fifo_options
           Options to pass to fifo pseudo-muxer instances. See fifo.

       Muxer options can be specified for each slave by prepending them as a list of key=value
       pairs separated by ':', between square brackets. If the options values contain a special
       character or the ':' separator, they must be escaped; note that this is a second level
       escaping.

       The following special options are also recognized:

       f   Specify the format name. Required if it cannot be guessed from the output URL.

       bsfs[/spec]
           Specify a list of bitstream filters to apply to the specified output.

           It is possible to specify to which streams a given bitstream filter applies, by
           appending a stream specifier to the option separated by "/". spec must be a stream
           specifier (see Format stream specifiers).

           If the stream specifier is not specified, the bitstream filters will be applied to all
           streams in the output. This will cause that output operation to fail if the output
           contains streams to which the bitstream filter cannot be applied e.g.
           "h264_mp4toannexb" being applied to an output containing an audio stream.

           Options for a bitstream filter must be specified in the form of "opt=value".

           Several bitstream filters can be specified, separated by ",".

       use_fifo bool
           This allows to override tee muxer use_fifo option for individual slave muxer.

       fifo_options
           This allows to override tee muxer fifo_options for individual slave muxer.  See fifo.

       select
           Select the streams that should be mapped to the slave output, specified by a stream
           specifier. If not specified, this defaults to all the mapped streams. This will cause
           that output operation to fail if the output format does not accept all mapped streams.

           You may use multiple stream specifiers separated by commas (",") e.g.: "a:0,v"

       onfail
           Specify behaviour on output failure. This can be set to either "abort" (which is
           default) or "ignore". "abort" will cause whole process to fail in case of failure on
           this slave output. "ignore" will ignore failure on this output, so other outputs will
           continue without being affected.

       Examples

       ·   Encode something and both archive it in a WebM file and stream it as MPEG-TS over UDP:

                   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
                     "archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       ·   As above, but continue streaming even if output to local file fails (for example local
           drive fills up):

                   ffmpeg -i ... -c:v libx264 -c:a mp2 -f tee -map 0:v -map 0:a
                     "[onfail=ignore]archive-20121107.mkv|[f=mpegts]udp://10.0.1.255:1234/"

       ·   Use ffmpeg to encode the input, and send the output to three different destinations.
           The "dump_extra" bitstream filter is used to add extradata information to all the
           output video keyframes packets, as requested by the MPEG-TS format. The select option
           is applied to out.aac in order to make it contain only audio packets.

                   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
                          -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=a]out.aac"

       ·   As above, but select only stream "a:1" for the audio output. Note that a second level
           escaping must be performed, as ":" is a special character used to separate options.

                   ffmpeg -i ... -map 0 -flags +global_header -c:v libx264 -c:a aac
                          -f tee "[bsfs/v=dump_extra=freq=keyframe]out.ts|[movflags=+faststart]out.mp4|[select=\'a:1\']out.aac"

   webm_dash_manifest
       WebM DASH Manifest muxer.

       This muxer implements the WebM DASH Manifest specification to generate the DASH manifest
       XML. It also supports manifest generation for DASH live streams.

       For more information see:

       ·   WebM DASH Specification:
           <https://sites.google.com/a/webmproject.org/wiki/adaptive-streaming/webm-dash-specification>

       ·   ISO DASH Specification:
           <http://standards.iso.org/ittf/PubliclyAvailableStandards/c065274_ISO_IEC_23009-1_2014.zip>

       Options

       This muxer supports the following options:

       adaptation_sets
           This option has the following syntax: "id=x,streams=a,b,c id=y,streams=d,e" where x
           and y are the unique identifiers of the adaptation sets and a,b,c,d and e are the
           indices of the corresponding audio and video streams. Any number of adaptation sets
           can be added using this option.

       live
           Set this to 1 to create a live stream DASH Manifest. Default: 0.

       chunk_start_index
           Start index of the first chunk. This will go in the startNumber attribute of the
           SegmentTemplate element in the manifest. Default: 0.

       chunk_duration_ms
           Duration of each chunk in milliseconds. This will go in the duration attribute of the
           SegmentTemplate element in the manifest. Default: 1000.

       utc_timing_url
           URL of the page that will return the UTC timestamp in ISO format. This will go in the
           value attribute of the UTCTiming element in the manifest.  Default: None.

       time_shift_buffer_depth
           Smallest time (in seconds) shifting buffer for which any Representation is guaranteed
           to be available. This will go in the timeShiftBufferDepth attribute of the MPD
           element. Default: 60.

       minimum_update_period
           Minimum update period (in seconds) of the manifest. This will go in the
           minimumUpdatePeriod attribute of the MPD element. Default: 0.

       Example

               ffmpeg -f webm_dash_manifest -i video1.webm \
                      -f webm_dash_manifest -i video2.webm \
                      -f webm_dash_manifest -i audio1.webm \
                      -f webm_dash_manifest -i audio2.webm \
                      -map 0 -map 1 -map 2 -map 3 \
                      -c copy \
                      -f webm_dash_manifest \
                      -adaptation_sets "id=0,streams=0,1 id=1,streams=2,3" \
                      manifest.xml

   webm_chunk
       WebM Live Chunk Muxer.

       This muxer writes out WebM headers and chunks as separate files which can be consumed by
       clients that support WebM Live streams via DASH.

       Options

       This muxer supports the following options:

       chunk_start_index
           Index of the first chunk (defaults to 0).

       header
           Filename of the header where the initialization data will be written.

       audio_chunk_duration
           Duration of each audio chunk in milliseconds (defaults to 5000).

       Example

               ffmpeg -f v4l2 -i /dev/video0 \
                      -f alsa -i hw:0 \
                      -map 0:0 \
                      -c:v libvpx-vp9 \
                      -s 640x360 -keyint_min 30 -g 30 \
                      -f webm_chunk \
                      -header webm_live_video_360.hdr \
                      -chunk_start_index 1 \
                      webm_live_video_360_%d.chk \
                      -map 1:0 \
                      -c:a libvorbis \
                      -b:a 128k \
                      -f webm_chunk \
                      -header webm_live_audio_128.hdr \
                      -chunk_start_index 1 \
                      -audio_chunk_duration 1000 \
                      webm_live_audio_128_%d.chk

METADATA

       FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text
       file and then load it back using the metadata muxer/demuxer.

       The file format is as follows:

       1.  A file consists of a header and a number of metadata tags divided into sections, each
           on its own line.

       2.  The header is a ;FFMETADATA string, followed by a version number (now 1).

       3.  Metadata tags are of the form key=value

       4.  Immediately after header follows global metadata

       5.  After global metadata there may be sections with per-stream/per-chapter metadata.

       6.  A section starts with the section name in uppercase (i.e. STREAM or CHAPTER) in
           brackets ([, ]) and ends with next section or end of file.

       7.  At the beginning of a chapter section there may be an optional timebase to be used for
           start/end values. It must be in form TIMEBASE=num/den, where num and den are integers.
           If the timebase is missing then start/end times are assumed to be in milliseconds.

           Next a chapter section must contain chapter start and end times in form START=num,
           END=num, where num is a positive integer.

       8.  Empty lines and lines starting with ; or # are ignored.

       9.  Metadata keys or values containing special characters (=, ;, #, \ and a newline) must
           be escaped with a backslash \.

       10. Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the
           tag (in the example above key is foo , value is
            bar).

       A ffmetadata file might look like this:

               ;FFMETADATA1
               title=bike\\shed
               ;this is a comment
               artist=FFmpeg troll team

               [CHAPTER]
               TIMEBASE=1/1000
               START=0
               #chapter ends at 0:01:00
               END=60000
               title=chapter \#1
               [STREAM]
               title=multi\
               line

       By using the ffmetadata muxer and demuxer it is possible to extract metadata from an input
       file to an ffmetadata file, and then transcode the file into an output file with the
       edited ffmetadata file.

       Extracting an ffmetadata file with ffmpeg goes as follows:

               ffmpeg -i INPUT -f ffmetadata FFMETADATAFILE

       Reinserting edited metadata information from the FFMETADATAFILE file can be done as:

               ffmpeg -i INPUT -i FFMETADATAFILE -map_metadata 1 -codec copy OUTPUT

PROTOCOL OPTIONS

       The libavformat library provides some generic global options, which can be set on all the
       protocols. In addition each protocol may support so-called private options, which are
       specific for that component.

       Options may be set by specifying -option value in the FFmpeg tools, or by setting the
       value explicitly in the "AVFormatContext" options or using the libavutil/opt.h API for
       programmatic use.

       The list of supported options follows:

       protocol_whitelist list (input)
           Set a ","-separated list of allowed protocols. "ALL" matches all protocols. Protocols
           prefixed by "-" are disabled.  All protocols are allowed by default but protocols used
           by an another protocol (nested protocols) are restricted to a per protocol subset.

PROTOCOLS

       Protocols are configured elements in FFmpeg that enable access to resources that require
       specific protocols.

       When you configure your FFmpeg build, all the supported protocols are enabled by default.
       You can list all available ones using the configure option "--list-protocols".

       You can disable all the protocols using the configure option "--disable-protocols", and
       selectively enable a protocol using the option "--enable-protocol=PROTOCOL", or you can
       disable a particular protocol using the option "--disable-protocol=PROTOCOL".

       The option "-protocols" of the ff* tools will display the list of supported protocols.

       All protocols accept the following options:

       rw_timeout
           Maximum time to wait for (network) read/write operations to complete, in microseconds.

       A description of the currently available protocols follows.

   async
       Asynchronous data filling wrapper for input stream.

       Fill data in a background thread, to decouple I/O operation from demux thread.

               async:<URL>
               async:http://host/resource
               async:cache:http://host/resource

   bluray
       Read BluRay playlist.

       The accepted options are:

       angle
           BluRay angle

       chapter
           Start chapter (1...N)

       playlist
           Playlist to read (BDMV/PLAYLIST/?????.mpls)

       Examples:

       Read longest playlist from BluRay mounted to /mnt/bluray:

               bluray:/mnt/bluray

       Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:

               -playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray

   cache
       Caching wrapper for input stream.

       Cache the input stream to temporary file. It brings seeking capability to live streams.

               cache:<URL>

   concat
       Physical concatenation protocol.

       Read and seek from many resources in sequence as if they were a unique resource.

       A URL accepted by this protocol has the syntax:

               concat:<URL1>|<URL2>|...|<URLN>

       where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one
       possibly specifying a distinct protocol.

       For example to read a sequence of files split1.mpeg, split2.mpeg, split3.mpeg with ffplay
       use the command:

               ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg

       Note that you may need to escape the character "|" which is special for many shells.

   crypto
       AES-encrypted stream reading protocol.

       The accepted options are:

       key Set the AES decryption key binary block from given hexadecimal representation.

       iv  Set the AES decryption initialization vector binary block from given hexadecimal
           representation.

       Accepted URL formats:

               crypto:<URL>
               crypto+<URL>

   data
       Data in-line in the URI. See <http://en.wikipedia.org/wiki/Data_URI_scheme>.

       For example, to convert a GIF file given inline with ffmpeg:

               ffmpeg -i "data:image/gif;base64,R0lGODdhCAAIAMIEAAAAAAAA//8AAP//AP///////////////ywAAAAACAAIAAADF0gEDLojDgdGiJdJqUX02iB4E8Q9jUMkADs=" smiley.png

   file
       File access protocol.

       Read from or write to a file.

       A file URL can have the form:

               file:<filename>

       where filename is the path of the file to read.

       An URL that does not have a protocol prefix will be assumed to be a file URL. Depending on
       the build, an URL that looks like a Windows path with the drive letter at the beginning
       will also be assumed to be a file URL (usually not the case in builds for unix-like
       systems).

       For example to read from a file input.mpeg with ffmpeg use the command:

               ffmpeg -i file:input.mpeg output.mpeg

       This protocol accepts the following options:

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       blocksize
           Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which
           results in not limiting the requested block size.  Setting this value reasonably low
           improves user termination request reaction time, which is valuable for files on slow
           medium.

   ftp
       FTP (File Transfer Protocol).

       Read from or write to remote resources using FTP protocol.

       Following syntax is required.

               ftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
           Set timeout in microseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       ftp-anonymous-password
           Password used when login as anonymous user. Typically an e-mail address should be
           used.

       ftp-write-seekable
           Control seekability of connection during encoding. If set to 1 the resource is
           supposed to be seekable, if set to 0 it is assumed not to be seekable. Default value
           is 0.

       NOTE: Protocol can be used as output, but it is recommended to not do it, unless special
       care is taken (tests, customized server configuration etc.). Different FTP servers behave
       in different way during seek operation. ff* tools may produce incomplete content due to
       server limitations.

       This protocol accepts the following options:

       follow
           If set to 1, the protocol will retry reading at the end of the file, allowing reading
           files that still are being written. In order for this to terminate, you either need to
           use the rw_timeout option, or use the interrupt callback (for API users).

   gopher
       Gopher protocol.

   hls
       Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8
       playlists describing the segments can be remote HTTP resources or local files, accessed
       using the standard file protocol.  The nested protocol is declared by specifying "+proto"
       after the hls URI scheme name, where proto is either "file" or "http".

               hls+http://host/path/to/remote/resource.m3u8
               hls+file://path/to/local/resource.m3u8

       Using this protocol is discouraged - the hls demuxer should work just as well (if not,
       please report the issues) and is more complete.  To use the hls demuxer instead, simply
       use the direct URLs to the m3u8 files.

   http
       HTTP (Hyper Text Transfer Protocol).

       This protocol accepts the following options:

       seekable
           Control seekability of connection. If set to 1 the resource is supposed to be
           seekable, if set to 0 it is assumed not to be seekable, if set to -1 it will try to
           autodetect if it is seekable. Default value is -1.

       chunked_post
           If set to 1 use chunked Transfer-Encoding for posts, default is 1.

       content_type
           Set a specific content type for the POST messages or for listen mode.

       http_proxy
           set HTTP proxy to tunnel through e.g. http://example.com:1234

       headers
           Set custom HTTP headers, can override built in default headers. The value must be a
           string encoding the headers.

       multiple_requests
           Use persistent connections if set to 1, default is 0.

       post_data
           Set custom HTTP post data.

       referer
           Set the Referer header. Include 'Referer: URL' header in HTTP request.

       user_agent
           Override the User-Agent header. If not specified the protocol will use a string
           describing the libavformat build. ("Lavf/<version>")

       user-agent
           This is a deprecated option, you can use user_agent instead it.

       timeout
           Set timeout in microseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       reconnect_at_eof
           If set then eof is treated like an error and causes reconnection, this is useful for
           live / endless streams.

       reconnect_streamed
           If set then even streamed/non seekable streams will be reconnected on errors.

       reconnect_delay_max
           Sets the maximum delay in seconds after which to give up reconnecting

       mime_type
           Export the MIME type.

       http_version
           Exports the HTTP response version number. Usually "1.0" or "1.1".

       icy If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports
           this, the metadata has to be retrieved by the application by reading the
           icy_metadata_headers and icy_metadata_packet options.  The default is 1.

       icy_metadata_headers
           If the server supports ICY metadata, this contains the ICY-specific HTTP reply
           headers, separated by newline characters.

       icy_metadata_packet
           If the server supports ICY metadata, and icy was set to 1, this contains the last non-
           empty metadata packet sent by the server. It should be polled in regular intervals by
           applications interested in mid-stream metadata updates.

       cookies
           Set the cookies to be sent in future requests. The format of each cookie is the same
           as the value of a Set-Cookie HTTP response field. Multiple cookies can be delimited by
           a newline character.

       offset
           Set initial byte offset.

       end_offset
           Try to limit the request to bytes preceding this offset.

       method
           When used as a client option it sets the HTTP method for the request.

           When used as a server option it sets the HTTP method that is going to be expected from
           the client(s).  If the expected and the received HTTP method do not match the client
           will be given a Bad Request response.  When unset the HTTP method is not checked for
           now. This will be replaced by autodetection in the future.

       listen
           If set to 1 enables experimental HTTP server. This can be used to send data when used
           as an output option, or read data from a client with HTTP POST when used as an input
           option.  If set to 2 enables experimental multi-client HTTP server. This is not yet
           implemented in ffmpeg.c and thus must not be used as a command line option.

                   # Server side (sending):
                   ffmpeg -i somefile.ogg -c copy -listen 1 -f ogg http://<server>:<port>

                   # Client side (receiving):
                   ffmpeg -i http://<server>:<port> -c copy somefile.ogg

                   # Client can also be done with wget:
                   wget http://<server>:<port> -O somefile.ogg

                   # Server side (receiving):
                   ffmpeg -listen 1 -i http://<server>:<port> -c copy somefile.ogg

                   # Client side (sending):
                   ffmpeg -i somefile.ogg -chunked_post 0 -c copy -f ogg http://<server>:<port>

                   # Client can also be done with wget:
                   wget --post-file=somefile.ogg http://<server>:<port>

       HTTP Cookies

       Some HTTP requests will be denied unless cookie values are passed in with the request. The
       cookies option allows these cookies to be specified. At the very least, each cookie must
       specify a value along with a path and domain.  HTTP requests that match both the domain
       and path will automatically include the cookie value in the HTTP Cookie header field.
       Multiple cookies can be delimited by a newline.

       The required syntax to play a stream specifying a cookie is:

               ffplay -cookies "nlqptid=nltid=tsn; path=/; domain=somedomain.com;" http://somedomain.com/somestream.m3u8

   Icecast
       Icecast protocol (stream to Icecast servers)

       This protocol accepts the following options:

       ice_genre
           Set the stream genre.

       ice_name
           Set the stream name.

       ice_description
           Set the stream description.

       ice_url
           Set the stream website URL.

       ice_public
           Set if the stream should be public.  The default is 0 (not public).

       user_agent
           Override the User-Agent header. If not specified a string of the form "Lavf/<version>"
           will be used.

       password
           Set the Icecast mountpoint password.

       content_type
           Set the stream content type. This must be set if it is different from audio/mpeg.

       legacy_icecast
           This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT
           method but the SOURCE method.

               icecast://[<username>[:<password>]@]<server>:<port>/<mountpoint>

   mmst
       MMS (Microsoft Media Server) protocol over TCP.

   mmsh
       MMS (Microsoft Media Server) protocol over HTTP.

       The required syntax is:

               mmsh://<server>[:<port>][/<app>][/<playpath>]

   md5
       MD5 output protocol.

       Computes the MD5 hash of the data to be written, and on close writes this to the
       designated output or stdout if none is specified. It can be used to test muxers without
       writing an actual file.

       Some examples follow.

               # Write the MD5 hash of the encoded AVI file to the file output.avi.md5.
               ffmpeg -i input.flv -f avi -y md5:output.avi.md5

               # Write the MD5 hash of the encoded AVI file to stdout.
               ffmpeg -i input.flv -f avi -y md5:

       Note that some formats (typically MOV) require the output protocol to be seekable, so they
       will fail with the MD5 output protocol.

   pipe
       UNIX pipe access protocol.

       Read and write from UNIX pipes.

       The accepted syntax is:

               pipe:[<number>]

       number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1
       for stdout, 2 for stderr).  If number is not specified, by default the stdout file
       descriptor will be used for writing, stdin for reading.

       For example to read from stdin with ffmpeg:

               cat test.wav | ffmpeg -i pipe:0
               # ...this is the same as...
               cat test.wav | ffmpeg -i pipe:

       For writing to stdout with ffmpeg:

               ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi
               # ...this is the same as...
               ffmpeg -i test.wav -f avi pipe: | cat > test.avi

       This protocol accepts the following options:

       blocksize
           Set I/O operation maximum block size, in bytes. Default value is "INT_MAX", which
           results in not limiting the requested block size.  Setting this value reasonably low
           improves user termination request reaction time, which is valuable if data
           transmission is slow.

       Note that some formats (typically MOV), require the output protocol to be seekable, so
       they will fail with the pipe output protocol.

   prompeg
       Pro-MPEG Code of Practice #3 Release 2 FEC protocol.

       The Pro-MPEG CoP#3 FEC is a 2D parity-check forward error correction mechanism for MPEG-2
       Transport Streams sent over RTP.

       This protocol must be used in conjunction with the "rtp_mpegts" muxer and the "rtp"
       protocol.

       The required syntax is:

               -f rtp_mpegts -fec prompeg=<option>=<val>... rtp://<hostname>:<port>

       The destination UDP ports are "port + 2" for the column FEC stream and "port + 4" for the
       row FEC stream.

       This protocol accepts the following options:

       l=n The number of columns (4-20, LxD <= 100)

       d=n The number of rows (4-20, LxD <= 100)

       Example usage:

               -f rtp_mpegts -fec prompeg=l=8:d=4 rtp://<hostname>:<port>

   rtmp
       Real-Time Messaging Protocol.

       The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a
       TCP/IP network.

       The required syntax is:

               rtmp://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

       The accepted parameters are:

       username
           An optional username (mostly for publishing).

       password
           An optional password (mostly for publishing).

       server
           The address of the RTMP server.

       port
           The number of the TCP port to use (by default is 1935).

       app It is the name of the application to access. It usually corresponds to the path where
           the application is installed on the RTMP server (e.g. /ondemand/, /flash/live/, etc.).
           You can override the value parsed from the URI through the "rtmp_app" option, too.

       playpath
           It is the path or name of the resource to play with reference to the application
           specified in app, may be prefixed by "mp4:". You can override the value parsed from
           the URI through the "rtmp_playpath" option, too.

       listen
           Act as a server, listening for an incoming connection.

       timeout
           Maximum time to wait for the incoming connection. Implies listen.

       Additionally, the following parameters can be set via command line options (or in code via
       "AVOption"s):

       rtmp_app
           Name of application to connect on the RTMP server. This option overrides the parameter
           specified in the URI.

       rtmp_buffer
           Set the client buffer time in milliseconds. The default is 3000.

       rtmp_conn
           Extra arbitrary AMF connection parameters, parsed from a string, e.g. like "B:1
           S:authMe O:1 NN:code:1.23 NS:flag:ok O:0".  Each value is prefixed by a single
           character denoting the type, B for Boolean, N for number, S for string, O for object,
           or Z for null, followed by a colon. For Booleans the data must be either 0 or 1 for
           FALSE or TRUE, respectively.  Likewise for Objects the data must be 0 or 1 to end or
           begin an object, respectively. Data items in subobjects may be named, by prefixing the
           type with 'N' and specifying the name before the value (i.e. "NB:myFlag:1"). This
           option may be used multiple times to construct arbitrary AMF sequences.

       rtmp_flashver
           Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.
           (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

       rtmp_flush_interval
           Number of packets flushed in the same request (RTMPT only). The default is 10.

       rtmp_live
           Specify that the media is a live stream. No resuming or seeking in live streams is
           possible. The default value is "any", which means the subscriber first tries to play
           the live stream specified in the playpath. If a live stream of that name is not found,
           it plays the recorded stream. The other possible values are "live" and "recorded".

       rtmp_pageurl
           URL of the web page in which the media was embedded. By default no value will be sent.

       rtmp_playpath
           Stream identifier to play or to publish. This option overrides the parameter specified
           in the URI.

       rtmp_subscribe
           Name of live stream to subscribe to. By default no value will be sent.  It is only
           sent if the option is specified or if rtmp_live is set to live.

       rtmp_swfhash
           SHA256 hash of the decompressed SWF file (32 bytes).

       rtmp_swfsize
           Size of the decompressed SWF file, required for SWFVerification.

       rtmp_swfurl
           URL of the SWF player for the media. By default no value will be sent.

       rtmp_swfverify
           URL to player swf file, compute hash/size automatically.

       rtmp_tcurl
           URL of the target stream. Defaults to proto://host[:port]/app.

       For example to read with ffplay a multimedia resource named "sample" from the application
       "vod" from an RTMP server "myserver":

               ffplay rtmp://myserver/vod/sample

       To publish to a password protected server, passing the playpath and app names separately:

               ffmpeg -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

   rtmpe
       Encrypted Real-Time Messaging Protocol.

       The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia
       content within standard cryptographic primitives, consisting of Diffie-Hellman key
       exchange and HMACSHA256, generating a pair of RC4 keys.

   rtmps
       Real-Time Messaging Protocol over a secure SSL connection.

       The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across
       an encrypted connection.

   rtmpt
       Real-Time Messaging Protocol tunneled through HTTP.

       The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming
       multimedia content within HTTP requests to traverse firewalls.

   rtmpte
       Encrypted Real-Time Messaging Protocol tunneled through HTTP.

       The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for
       streaming multimedia content within HTTP requests to traverse firewalls.

   rtmpts
       Real-Time Messaging Protocol tunneled through HTTPS.

       The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming
       multimedia content within HTTPS requests to traverse firewalls.

   libsmbclient
       libsmbclient permits one to manipulate CIFS/SMB network resources.

       Following syntax is required.

               smb://[[domain:]user[:password@]]server[/share[/path[/file]]]

       This protocol accepts the following options.

       timeout
           Set timeout in milliseconds of socket I/O operations used by the underlying low level
           operation. By default it is set to -1, which means that the timeout is not specified.

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       workgroup
           Set the workgroup used for making connections. By default workgroup is not specified.

       For more information see: <http://www.samba.org/>.

   libssh
       Secure File Transfer Protocol via libssh

       Read from or write to remote resources using SFTP protocol.

       Following syntax is required.

               sftp://[user[:password]@]server[:port]/path/to/remote/resource.mpeg

       This protocol accepts the following options.

       timeout
           Set timeout of socket I/O operations used by the underlying low level operation. By
           default it is set to -1, which means that the timeout is not specified.

       truncate
           Truncate existing files on write, if set to 1. A value of 0 prevents truncating.
           Default value is 1.

       private_key
           Specify the path of the file containing private key to use during authorization.  By
           default libssh searches for keys in the ~/.ssh/ directory.

       Example: Play a file stored on remote server.

               ffplay sftp://user:password@server_address:22/home/user/resource.mpeg

   librtmp rtmp, rtmpe, rtmps, rtmpt, rtmpte
       Real-Time Messaging Protocol and its variants supported through librtmp.

       Requires the presence of the librtmp headers and library during configuration. You need to
       explicitly configure the build with "--enable-librtmp". If enabled this will replace the
       native RTMP protocol.

       This protocol provides most client functions and a few server functions needed to support
       RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and
       tunneled variants of these encrypted types (RTMPTE, RTMPTS).

       The required syntax is:

               <rtmp_proto>://<server>[:<port>][/<app>][/<playpath>] <options>

       where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte",
       "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the
       same meaning as specified for the RTMP native protocol.  options contains a list of space-
       separated options of the form key=val.

       See the librtmp manual page (man 3 librtmp) for more information.

       For example, to stream a file in real-time to an RTMP server using ffmpeg:

               ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream

       To play the same stream using ffplay:

               ffplay "rtmp://myserver/live/mystream live=1"

   rtp
       Real-time Transport Protocol.

       The required syntax for an RTP URL is: rtp://hostname[:port][?option=val...]

       port specifies the RTP port to use.

       The following URL options are supported:

       ttl=n
           Set the TTL (Time-To-Live) value (for multicast only).

       rtcpport=n
           Set the remote RTCP port to n.

       localrtpport=n
           Set the local RTP port to n.

       localrtcpport=n'
           Set the local RTCP port to n.

       pkt_size=n
           Set max packet size (in bytes) to n.

       connect=0|1
           Do a "connect()" on the UDP socket (if set to 1) or not (if set to 0).

       sources=ip[,ip]
           List allowed source IP addresses.

       block=ip[,ip]
           List disallowed (blocked) source IP addresses.

       write_to_source=0|1
           Send packets to the source address of the latest received packet (if set to 1) or to a
           default remote address (if set to 0).

       localport=n
           Set the local RTP port to n.

           This is a deprecated option. Instead, localrtpport should be used.

       Important notes:

       1.  If rtcpport is not set the RTCP port will be set to the RTP port value plus 1.

       2.  If localrtpport (the local RTP port) is not set any available port will be used for
           the local RTP and RTCP ports.

       3.  If localrtcpport (the local RTCP port) is not set it will be set to the local RTP port
           value plus 1.

   rtsp
       Real-Time Streaming Protocol.

       RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The
       demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g.
       Apple and Microsoft) and Real-RTSP (with data transferred over RDT).

       The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it
       (currently Darwin Streaming Server and Mischa Spiegelmock's
       <https://github.com/revmischa/rtsp-server>).

       The required syntax for a RTSP url is:

               rtsp://<hostname>[:<port>]/<path>

       Options can be set on the ffmpeg/ffplay command line, or set in code via "AVOption"s or in
       "avformat_open_input".

       The following options are supported.

       initial_pause
           Do not start playing the stream immediately if set to 1. Default value is 0.

       rtsp_transport
           Set RTSP transport protocols.

           It accepts the following values:

           udp Use UDP as lower transport protocol.

           tcp Use TCP (interleaving within the RTSP control channel) as lower transport
               protocol.

           udp_multicast
               Use UDP multicast as lower transport protocol.

           http
               Use HTTP tunneling as lower transport protocol, which is useful for passing
               proxies.

           Multiple lower transport protocols may be specified, in that case they are tried one
           at a time (if the setup of one fails, the next one is tried).  For the muxer, only the
           tcp and udp options are supported.

       rtsp_flags
           Set RTSP flags.

           The following values are accepted:

           filter_src
               Accept packets only from negotiated peer address and port.

           listen
               Act as a server, listening for an incoming connection.

           prefer_tcp
               Try TCP for RTP transport first, if TCP is available as RTSP RTP transport.

           Default value is none.

       allowed_media_types
           Set media types to accept from the server.

           The following flags are accepted:

           video
           audio
           data

           By default it accepts all media types.

       min_port
           Set minimum local UDP port. Default value is 5000.

       max_port
           Set maximum local UDP port. Default value is 65000.

       timeout
           Set maximum timeout (in seconds) to wait for incoming connections.

           A value of -1 means infinite (default). This option implies the rtsp_flags set to
           listen.

       reorder_queue_size
           Set number of packets to buffer for handling of reordered packets.

       stimeout
           Set socket TCP I/O timeout in microseconds.

       user-agent
           Override User-Agent header. If not specified, it defaults to the libavformat
           identifier string.

       When receiving data over UDP, the demuxer tries to reorder received packets (since they
       may arrive out of order, or packets may get lost totally). This can be disabled by setting
       the maximum demuxing delay to zero (via the "max_delay" field of AVFormatContext).

       When watching multi-bitrate Real-RTSP streams with ffplay, the streams to display can be
       chosen with "-vst" n and "-ast" n for video and audio respectively, and can be switched on
       the fly by pressing "v" and "a".

       Examples

       The following examples all make use of the ffplay and ffmpeg tools.

       ·   Watch a stream over UDP, with a max reordering delay of 0.5 seconds:

                   ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

       ·   Watch a stream tunneled over HTTP:

                   ffplay -rtsp_transport http rtsp://server/video.mp4

       ·   Send a stream in realtime to a RTSP server, for others to watch:

                   ffmpeg -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

       ·   Receive a stream in realtime:

                   ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp <output>

   sap
       Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in
       libavformat, it is a muxer and demuxer.  It is used for signalling of RTP streams, by
       announcing the SDP for the streams regularly on a separate port.

       Muxer

       The syntax for a SAP url given to the muxer is:

               sap://<destination>[:<port>][?<options>]

       The RTP packets are sent to destination on port port, or to port 5004 if no port is
       specified.  options is a "&"-separated list. The following options are supported:

       announce_addr=address
           Specify the destination IP address for sending the announcements to.  If omitted, the
           announcements are sent to the commonly used SAP announcement multicast address
           224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.

       announce_port=port
           Specify the port to send the announcements on, defaults to 9875 if not specified.

       ttl=ttl
           Specify the time to live value for the announcements and RTP packets, defaults to 255.

       same_port=0|1
           If set to 1, send all RTP streams on the same port pair. If zero (the default), all
           streams are sent on unique ports, with each stream on a port 2 numbers higher than the
           previous.  VLC/Live555 requires this to be set to 1, to be able to receive the stream.
           The RTP stack in libavformat for receiving requires all streams to be sent on unique
           ports.

       Example command lines follow.

       To broadcast a stream on the local subnet, for watching in VLC:

               ffmpeg -re -i <input> -f sap sap://224.0.0.255?same_port=1

       Similarly, for watching in ffplay:

               ffmpeg -re -i <input> -f sap sap://224.0.0.255

       And for watching in ffplay, over IPv6:

               ffmpeg -re -i <input> -f sap sap://[ff0e::1:2:3:4]

       Demuxer

       The syntax for a SAP url given to the demuxer is:

               sap://[<address>][:<port>]

       address is the multicast address to listen for announcements on, if omitted, the default
       224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if
       omitted.

       The demuxers listens for announcements on the given address and port.  Once an
       announcement is received, it tries to receive that particular stream.

       Example command lines follow.

       To play back the first stream announced on the normal SAP multicast address:

               ffplay sap://

       To play back the first stream announced on one the default IPv6 SAP multicast address:

               ffplay sap://[ff0e::2:7ffe]

   sctp
       Stream Control Transmission Protocol.

       The accepted URL syntax is:

               sctp://<host>:<port>[?<options>]

       The protocol accepts the following options:

       listen
           If set to any value, listen for an incoming connection. Outgoing connection is done by
           default.

       max_streams
           Set the maximum number of streams. By default no limit is set.

   srt
       Haivision Secure Reliable Transport Protocol via libsrt.

       The supported syntax for a SRT URL is:

               srt://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       or

               <options> srt://<hostname>:<port>

       options contains a list of '-key val' options.

       This protocol accepts the following options.

       connect_timeout
           Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges)
           with the default connect timeout of 3 seconds. This option applies to the caller and
           rendezvous connection modes. The connect timeout is 10 times the value set for the
           rendezvous mode (which can be used as a workaround for this connection problem with
           earlier versions).

       ffs=bytes
           Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and
           you should set it to not less than recv_buffer_size and mss. The default value is
           relatively large, therefore unless you set a very large receiver buffer, you do not
           need to change this option. Default value is 25600.

       inputbw=bytes/seconds
           Sender nominal input rate, in bytes per seconds. Used along with oheadbw, when maxbw
           is set to relative (0), to calculate maximum sending rate when recovery packets are
           sent along with the main media stream: inputbw * (100 + oheadbw) / 100 if inputbw is
           not set while maxbw is set to relative (0), the actual input rate is evaluated inside
           the library. Default value is 0.

       iptos=tos
           IP Type of Service. Applies to sender only. Default value is 0xB8.

       ipttl=ttl
           IP Time To Live. Applies to sender only. Default value is 64.

       latency
           Timestamp-based Packet Delivery Delay.  Used to absorb bursts of missed packet
           retransmissions.  This flag sets both rcvlatency and peerlatency to the same value.
           Note that prior to version 1.3.0 this is the only flag to set the latency, however
           this is effectively equivalent to setting peerlatency, when side is sender and
           rcvlatency when side is receiver, and the bidirectional stream sending is not
           supported.

       listen_timeout
           Set socket listen timeout.

       maxbw=bytes/seconds
           Maximum sending bandwidth, in bytes per seconds.  -1 infinite (CSRTCC limit is 30mbps)
           0 relative to input rate (see inputbw) >0 absolute limit value Default value is 0
           (relative)

       mode=caller|listener|rendezvous
           Connection mode.  caller opens client connection.  listener starts server to listen
           for incoming connections.  rendezvous use Rendez-Vous connection mode.  Default value
           is caller.

       mss=bytes
           Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using
           a packet counter assuming fully filled packets. The smallest MSS between the peers is
           used. This is 1500 by default in the overall internet.  This is the maximum size of
           the UDP packet and can be only decreased, unless you have some unusual dedicated
           network settings. Default value is 1500.

       nakreport=1|0
           If set to 1, Receiver will send `UMSG_LOSSREPORT` messages periodically until a lost
           packet is retransmitted or intentionally dropped. Default value is 1.

       oheadbw=percents
           Recovery bandwidth overhead above input rate, in percents.  See inputbw. Default value
           is 25%.

       passphrase=string
           HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The
           passphrase is the shared secret between the sender and the receiver. It is used to
           generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function).
           It is used only if pbkeylen is non-zero. It is used on the receiver only if the
           received data is encrypted.  The configured passphrase cannot be recovered (write-
           only).

       payload_size=bytes
           Sets the maximum declared size of a packet transferred during the single call to the
           sending function in Live mode. Use 0 if this value isn't used (which is default in
           file mode).  Default is -1 (automatic), which typically means MPEG-TS; if you are
           going to use SRT to send any different kind of payload, such as, for example, wrapping
           a live stream in very small frames, then you can use a bigger maximum frame size,
           though not greater than 1456 bytes.

       pkt_size=bytes
           Alias for payload_size.

       peerlatency
           The latency value (as described in rcvlatency) that is set by the sender side as a
           minimum value for the receiver.

       pbkeylen=bytes
           Sender encryption key length, in bytes.  Only can be set to 0, 16, 24 and 32.  Enable
           sender encryption if not 0.  Not required on receiver (set to 0), key size obtained
           from sender in HaiCrypt handshake.  Default value is 0.

       rcvlatency
           The time that should elapse since the moment when the packet was sent and the moment
           when it's delivered to the receiver application in the receiving function.  This time
           should be a buffer time large enough to cover the time spent for sending, unexpectedly
           extended RTT time, and the time needed to retransmit the lost UDP packet. The
           effective latency value will be the maximum of this options' value and the value of
           peerlatency set by the peer side. Before version 1.3.0 this option is only available
           as latency.

       recv_buffer_size=bytes
           Set UDP receive buffer size, expressed in bytes.

       send_buffer_size=bytes
           Set UDP send buffer size, expressed in bytes.

       rw_timeout
           Set raise error timeout for read/write optations.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       tlpktdrop=1|0
           Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not
           been delivered in time and delivers the following packets to the application when
           their time-to-play has come. It also sends a fake ACK to the sender. When enabled on
           sender and enabled on the receiving peer, the sender drops the older packets that have
           no chance of being delivered in time. It was automatically enabled in the sender if
           the receiver supports it.

       sndbuf=bytes
           Set send buffer size, expressed in bytes.

       rcvbuf=bytes
           Set receive buffer size, expressed in bytes.

           Receive buffer must not be greater than ffs.

       lossmaxttl=packets
           The value up to which the Reorder Tolerance may grow. When Reorder Tolerance is > 0,
           then packet loss report is delayed until that number of packets come in. Reorder
           Tolerance increases every time a "belated" packet has come, but it wasn't due to
           retransmission (that is, when UDP packets tend to come out of order), with the
           difference between the latest sequence and this packet's sequence, and not more than
           the value of this option. By default it's 0, which means that this mechanism is turned
           off, and the loss report is always sent immediately upon experiencing a "gap" in
           sequences.

       minversion
           The minimum SRT version that is required from the peer. A connection to a peer that
           does not satisfy the minimum version requirement will be rejected.

           The version format in hex is 0xXXYYZZ for x.y.z in human readable form.

       streamid=string
           A string limited to 512 characters that can be set on the socket prior to connecting.
           This stream ID will be able to be retrieved by the listener side from the socket that
           is returned from srt_accept and was connected by a socket with that set stream ID. SRT
           does not enforce any special interpretation of the contents of this string.  This
           option doesnXt make sense in Rendezvous connection; the result might be that simply
           one side will override the value from the other side and itXs the matter of luck which
           one would win

       smoother=live|file
           The type of Smoother used for the transmission for that socket, which is responsible
           for the transmission and congestion control. The Smoother type must be exactly the
           same on both connecting parties, otherwise the connection is rejected.

       messageapi=1|0
           When set, this socket uses the Message API, otherwise it uses Buffer API. Note that in
           live mode (see transtype) thereXs only message API available. In File mode you can
           chose to use one of two modes:

           Stream API (default, when this option is false). In this mode you may send as many
           data as you wish with one sending instruction, or even use dedicated functions that
           read directly from a file. The internal facility will take care of any speed and
           congestion control. When receiving, you can also receive as many data as desired, the
           data not extracted will be waiting for the next call. There is no boundary between
           data portions in the Stream mode.

           Message API. In this mode your single sending instruction passes exactly one piece of
           data that has boundaries (a message). Contrary to Live mode, this message may span
           across multiple UDP packets and the only size limitation is that it shall fit as a
           whole in the sending buffer. The receiver shall use as large buffer as necessary to
           receive the message, otherwise the message will not be given up. When the message is
           not complete (not all packets received or there was a packet loss) it will not be
           given up.

       transtype=live|file
           Sets the transmission type for the socket, in particular, setting this option sets
           multiple other parameters to their default values as required for a particular
           transmission type.

           live: Set options as for live transmission. In this mode, you should send by one
           sending instruction only so many data that fit in one UDP packet, and limited to the
           value defined first in payload_size (1316 is default in this mode). There is no speed
           control in this mode, only the bandwidth control, if configured, in order to not
           exceed the bandwidth with the overhead transmission (retransmitted and control
           packets).

           file: Set options as for non-live transmission. See messageapi for further
           explanations

       For more information see: <https://github.com/Haivision/srt>.

   srtp
       Secure Real-time Transport Protocol.

       The accepted options are:

       srtp_in_suite
       srtp_out_suite
           Select input and output encoding suites.

           Supported values:

           AES_CM_128_HMAC_SHA1_80
           SRTP_AES128_CM_HMAC_SHA1_80
           AES_CM_128_HMAC_SHA1_32
           SRTP_AES128_CM_HMAC_SHA1_32
       srtp_in_params
       srtp_out_params
           Set input and output encoding parameters, which are expressed by a base64-encoded
           representation of a binary block. The first 16 bytes of this binary block are used as
           master key, the following 14 bytes are used as master salt.

   subfile
       Virtually extract a segment of a file or another stream.  The underlying stream must be
       seekable.

       Accepted options:

       start
           Start offset of the extracted segment, in bytes.

       end End offset of the extracted segment, in bytes.  If set to 0, extract till end of file.

       Examples:

       Extract a chapter from a DVD VOB file (start and end sectors obtained externally and
       multiplied by 2048):

               subfile,,start,153391104,end,268142592,,:/media/dvd/VIDEO_TS/VTS_08_1.VOB

       Play an AVI file directly from a TAR archive:

               subfile,,start,183241728,end,366490624,,:archive.tar

       Play a MPEG-TS file from start offset till end:

               subfile,,start,32815239,end,0,,:video.ts

   tee
       Writes the output to multiple protocols. The individual outputs are separated by |

               tee:file://path/to/local/this.avi|file://path/to/local/that.avi

   tcp
       Transmission Control Protocol.

       The required syntax for a TCP url is:

               tcp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       The list of supported options follows.

       listen=1|0
           Listen for an incoming connection. Default value is 0.

       timeout=microseconds
           Set raise error timeout, expressed in microseconds.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       listen_timeout=milliseconds
           Set listen timeout, expressed in milliseconds.

       recv_buffer_size=bytes
           Set receive buffer size, expressed bytes.

       send_buffer_size=bytes
           Set send buffer size, expressed bytes.

       tcp_nodelay=1|0
           Set TCP_NODELAY to disable Nagle's algorithm. Default value is 0.

       tcp_mss=bytes
           Set maximum segment size for outgoing TCP packets, expressed in bytes.

       The following example shows how to setup a listening TCP connection with ffmpeg, which is
       then accessed with ffplay:

               ffmpeg -i <input> -f <format> tcp://<hostname>:<port>?listen
               ffplay tcp://<hostname>:<port>

   tls
       Transport Layer Security (TLS) / Secure Sockets Layer (SSL)

       The required syntax for a TLS/SSL url is:

               tls://<hostname>:<port>[?<options>]

       The following parameters can be set via command line options (or in code via "AVOption"s):

       ca_file, cafile=filename
           A file containing certificate authority (CA) root certificates to treat as trusted. If
           the linked TLS library contains a default this might not need to be specified for
           verification to work, but not all libraries and setups have defaults built in.  The
           file must be in OpenSSL PEM format.

       tls_verify=1|0
           If enabled, try to verify the peer that we are communicating with.  Note, if using
           OpenSSL, this currently only makes sure that the peer certificate is signed by one of
           the root certificates in the CA database, but it does not validate that the
           certificate actually matches the host name we are trying to connect to. (With other
           backends, the host name is validated as well.)

           This is disabled by default since it requires a CA database to be provided by the
           caller in many cases.

       cert_file, cert=filename
           A file containing a certificate to use in the handshake with the peer.  (When
           operating as server, in listen mode, this is more often required by the peer, while
           client certificates only are mandated in certain setups.)

       key_file, key=filename
           A file containing the private key for the certificate.

       listen=1|0
           If enabled, listen for connections on the provided port, and assume the server role in
           the handshake instead of the client role.

       Example command lines:

       To create a TLS/SSL server that serves an input stream.

               ffmpeg -i <input> -f <format> tls://<hostname>:<port>?listen&cert=<server.crt>&key=<server.key>

       To play back a stream from the TLS/SSL server using ffplay:

               ffplay tls://<hostname>:<port>

   udp
       User Datagram Protocol.

       The required syntax for an UDP URL is:

               udp://<hostname>:<port>[?<options>]

       options contains a list of &-separated options of the form key=val.

       In case threading is enabled on the system, a circular buffer is used to store the
       incoming data, which allows one to reduce loss of data due to UDP socket buffer overruns.
       The fifo_size and overrun_nonfatal options are related to this buffer.

       The list of supported options follows.

       buffer_size=size
           Set the UDP maximum socket buffer size in bytes. This is used to set either the
           receive or send buffer size, depending on what the socket is used for.  Default is
           64KB.  See also fifo_size.

       bitrate=bitrate
           If set to nonzero, the output will have the specified constant bitrate if the input
           has enough packets to sustain it.

       burst_bits=bits
           When using bitrate this specifies the maximum number of bits in packet bursts.

       localport=port
           Override the local UDP port to bind with.

       localaddr=addr
           Local IP address of a network interface used for sending packets or joining multicast
           groups.

       pkt_size=size
           Set the size in bytes of UDP packets.

       reuse=1|0
           Explicitly allow or disallow reusing UDP sockets.

       ttl=ttl
           Set the time to live value (for multicast only).

       connect=1|0
           Initialize the UDP socket with "connect()". In this case, the destination address
           can't be changed with ff_udp_set_remote_url later.  If the destination address isn't
           known at the start, this option can be specified in ff_udp_set_remote_url, too.  This
           allows finding out the source address for the packets with getsockname, and makes
           writes return with AVERROR(ECONNREFUSED) if "destination unreachable" is received.
           For receiving, this gives the benefit of only receiving packets from the specified
           peer address/port.

       sources=address[,address]
           Only receive packets sent from the specified addresses. In case of multicast, also
           subscribe to multicast traffic coming from these addresses only.

       block=address[,address]
           Ignore packets sent from the specified addresses. In case of multicast, also exclude
           the source addresses in the multicast subscription.

       fifo_size=units
           Set the UDP receiving circular buffer size, expressed as a number of packets with size
           of 188 bytes. If not specified defaults to 7*4096.

       overrun_nonfatal=1|0
           Survive in case of UDP receiving circular buffer overrun. Default value is 0.

       timeout=microseconds
           Set raise error timeout, expressed in microseconds.

           This option is only relevant in read mode: if no data arrived in more than this time
           interval, raise error.

       broadcast=1|0
           Explicitly allow or disallow UDP broadcasting.

           Note that broadcasting may not work properly on networks having a broadcast storm
           protection.

       Examples

       ·   Use ffmpeg to stream over UDP to a remote endpoint:

                   ffmpeg -i <input> -f <format> udp://<hostname>:<port>

       ·   Use ffmpeg to stream in mpegts format over UDP using 188 sized UDP packets, using a
           large input buffer:

                   ffmpeg -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

       ·   Use ffmpeg to receive over UDP from a remote endpoint:

                   ffmpeg -i udp://[<multicast-address>]:<port> ...

   unix
       Unix local socket

       The required syntax for a Unix socket URL is:

               unix://<filepath>

       The following parameters can be set via command line options (or in code via "AVOption"s):

       timeout
           Timeout in ms.

       listen
           Create the Unix socket in listening mode.

DEVICE OPTIONS

       The libavdevice library provides the same interface as libavformat. Namely, an input
       device is considered like a demuxer, and an output device like a muxer, and the interface
       and generic device options are the same provided by libavformat (see the ffmpeg-formats
       manual).

       In addition each input or output device may support so-called private options, which are
       specific for that component.

       Options may be set by specifying -option value in the FFmpeg tools, or by setting the
       value explicitly in the device "AVFormatContext" options or using the libavutil/opt.h API
       for programmatic use.

INPUT DEVICES

       Input devices are configured elements in FFmpeg which enable accessing the data coming
       from a multimedia device attached to your system.

       When you configure your FFmpeg build, all the supported input devices are enabled by
       default. You can list all available ones using the configure option "--list-indevs".

       You can disable all the input devices using the configure option "--disable-indevs", and
       selectively enable an input device using the option "--enable-indev=INDEV", or you can
       disable a particular input device using the option "--disable-indev=INDEV".

       The option "-devices" of the ff* tools will display the list of supported input devices.

       A description of the currently available input devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) input device.

       To enable this input device during configuration you need libasound installed on your
       system.

       This device allows capturing from an ALSA device. The name of the device to capture has to
       be an ALSA card identifier.

       An ALSA identifier has the syntax:

               hw:<CARD>[,<DEV>[,<SUBDEV>]]

       where the DEV and SUBDEV components are optional.

       The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device
       number and subdevice number (-1 means any).

       To see the list of cards currently recognized by your system check the files
       /proc/asound/cards and /proc/asound/devices.

       For example to capture with ffmpeg from an ALSA device with card id 0, you may run the
       command:

               ffmpeg -f alsa -i hw:0 alsaout.wav

       For more information see: <http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html>

       Options

       sample_rate
           Set the sample rate in Hz. Default is 48000.

       channels
           Set the number of channels. Default is 2.

   android_camera
       Android camera input device.

       This input devices uses the Android Camera2 NDK API which is available on devices with API
       level 24+. The availability of android_camera is autodetected during configuration.

       This device allows capturing from all cameras on an Android device, which are integrated
       into the Camera2 NDK API.

       The available cameras are enumerated internally and can be selected with the camera_index
       parameter. The input file string is discarded.

       Generally the back facing camera has index 0 while the front facing camera has index 1.

       Options

       video_size
           Set the video size given as a string such as 640x480 or hd720.  Falls back to the
           first available configuration reported by Android if requested video size is not
           available or by default.

       framerate
           Set the video framerate.  Falls back to the first available configuration reported by
           Android if requested framerate is not available or by default (-1).

       camera_index
           Set the index of the camera to use. Default is 0.

       input_queue_size
           Set the maximum number of frames to buffer. Default is 5.

   avfoundation
       AVFoundation input device.

       AVFoundation is the currently recommended framework by Apple for streamgrabbing on OSX >=
       10.7 as well as on iOS.

       The input filename has to be given in the following syntax:

               -i "[[VIDEO]:[AUDIO]]"

       The first entry selects the video input while the latter selects the audio input.  The
       stream has to be specified by the device name or the device index as shown by the device
       list.  Alternatively, the video and/or audio input device can be chosen by index using the

           B<-video_device_index E<lt>INDEXE<gt>>

       and/or

           B<-audio_device_index E<lt>INDEXE<gt>>

       , overriding any device name or index given in the input filename.

       All available devices can be enumerated by using -list_devices true, listing all device
       names and corresponding indices.

       There are two device name aliases:

       "default"
           Select the AVFoundation default device of the corresponding type.

       "none"
           Do not record the corresponding media type.  This is equivalent to specifying an empty
           device name or index.

       Options

       AVFoundation supports the following options:

       -list_devices <TRUE|FALSE>
           If set to true, a list of all available input devices is given showing all device
           names and indices.

       -video_device_index <INDEX>
           Specify the video device by its index. Overrides anything given in the input filename.

       -audio_device_index <INDEX>
           Specify the audio device by its index. Overrides anything given in the input filename.

       -pixel_format <FORMAT>
           Request the video device to use a specific pixel format.  If the specified format is
           not supported, a list of available formats is given and the first one in this list is
           used instead. Available pixel formats are: "monob, rgb555be, rgb555le, rgb565be,
           rgb565le, rgb24, bgr24, 0rgb, bgr0, 0bgr, rgb0,
            bgr48be, uyvy422, yuva444p, yuva444p16le, yuv444p, yuv422p16, yuv422p10, yuv444p10,
            yuv420p, nv12, yuyv422, gray"

       -framerate
           Set the grabbing frame rate. Default is "ntsc", corresponding to a frame rate of
           "30000/1001".

       -video_size
           Set the video frame size.

       -capture_cursor
           Capture the mouse pointer. Default is 0.

       -capture_mouse_clicks
           Capture the screen mouse clicks. Default is 0.

       Examples

       ·   Print the list of AVFoundation supported devices and exit:

                   $ ffmpeg -f avfoundation -list_devices true -i ""

       ·   Record video from video device 0 and audio from audio device 0 into out.avi:

                   $ ffmpeg -f avfoundation -i "0:0" out.avi

       ·   Record video from video device 2 and audio from audio device 1 into out.avi:

                   $ ffmpeg -f avfoundation -video_device_index 2 -i ":1" out.avi

       ·   Record video from the system default video device using the pixel format bgr0 and do
           not record any audio into out.avi:

                   $ ffmpeg -f avfoundation -pixel_format bgr0 -i "default:none" out.avi

   bktr
       BSD video input device.

       Options

       framerate
           Set the frame rate.

       video_size
           Set the video frame size. Default is "vga".

       standard
           Available values are:

           pal
           ntsc
           secam
           paln
           palm
           ntscj

   decklink
       The decklink input device provides capture capabilities for Blackmagic DeckLink devices.

       To enable this input device, you need the Blackmagic DeckLink SDK and you need to
       configure with the appropriate "--extra-cflags" and "--extra-ldflags".  On Windows, you
       need to run the IDL files through widl.

       DeckLink is very picky about the formats it supports. Pixel format of the input can be set
       with raw_format.  Framerate and video size must be determined for your device with
       -list_formats 1. Audio sample rate is always 48 kHz and the number of channels can be 2, 8
       or 16. Note that all audio channels are bundled in one single audio track.

       Options

       list_devices
           If set to true, print a list of devices and exit.  Defaults to false. Alternatively
           you can use the "-sources" option of ffmpeg to list the available input devices.

       list_formats
           If set to true, print a list of supported formats and exit.  Defaults to false.

       format_code <FourCC>
           This sets the input video format to the format given by the FourCC. To see the
           supported values of your device(s) use list_formats.  Note that there is a FourCC 'pal
           ' that can also be used as pal (3 letters).  Default behavior is autodetection of the
           input video format, if the hardware supports it.

       bm_v210
           This is a deprecated option, you can use raw_format instead.  If set to 1, video is
           captured in 10 bit v210 instead of uyvy422. Not all Blackmagic devices support this
           option.

       raw_format
           Set the pixel format of the captured video.  Available values are:

           uyvy422
           yuv422p10
           argb
           bgra
           rgb10
       teletext_lines
           If set to nonzero, an additional teletext stream will be captured from the vertical
           ancillary data. Both SD PAL (576i) and HD (1080i or 1080p) sources are supported. In
           case of HD sources, OP47 packets are decoded.

           This option is a bitmask of the SD PAL VBI lines captured, specifically lines 6 to 22,
           and lines 318 to 335. Line 6 is the LSB in the mask. Selected lines which do not
           contain teletext information will be ignored. You can use the special all constant to
           select all possible lines, or standard to skip lines 6, 318 and 319, which are not
           compatible with all receivers.

           For SD sources, ffmpeg needs to be compiled with "--enable-libzvbi". For HD sources,
           on older (pre-4K) DeckLink card models you have to capture in 10 bit mode.

       channels
           Defines number of audio channels to capture. Must be 2, 8 or 16.  Defaults to 2.

       duplex_mode
           Sets the decklink device duplex mode. Must be unset, half or full.  Defaults to unset.

       timecode_format
           Timecode type to include in the frame and video stream metadata. Must be none,
           rp188vitc, rp188vitc2, rp188ltc, rp188any, vitc, vitc2, or serial. Defaults to none
           (not included).

       video_input
           Sets the video input source. Must be unset, sdi, hdmi, optical_sdi, component,
           composite or s_video.  Defaults to unset.

       audio_input
           Sets the audio input source. Must be unset, embedded, aes_ebu, analog, analog_xlr,
           analog_rca or microphone. Defaults to unset.

       video_pts
           Sets the video packet timestamp source. Must be video, audio, reference, wallclock or
           abs_wallclock.  Defaults to video.

       audio_pts
           Sets the audio packet timestamp source. Must be video, audio, reference, wallclock or
           abs_wallclock.  Defaults to audio.

       draw_bars
           If set to true, color bars are drawn in the event of a signal loss.  Defaults to true.

       queue_size
           Sets maximum input buffer size in bytes. If the buffering reaches this value, incoming
           frames will be dropped.  Defaults to 1073741824.

       audio_depth
           Sets the audio sample bit depth. Must be 16 or 32.  Defaults to 16.

       decklink_copyts
           If set to true, timestamps are forwarded as they are without removing the initial
           offset.  Defaults to false.

       timestamp_align
           Capture start time alignment in seconds. If set to nonzero, input frames are dropped
           till the system timestamp aligns with configured value.  Alignment difference of up to
           one frame duration is tolerated.  This is useful for maintaining input synchronization
           across N different hardware devices deployed for 'N-way' redundancy. The system time
           of different hardware devices should be synchronized with protocols such as NTP or
           PTP, before using this option.  Note that this method is not foolproof. In some border
           cases input synchronization may not happen due to thread scheduling jitters in the OS.
           Either sync could go wrong by 1 frame or in a rarer case timestamp_align seconds.
           Defaults to 0.

       Examples

       ·   List input devices:

                   ffmpeg -f decklink -list_devices 1 -i dummy

       ·   List supported formats:

                   ffmpeg -f decklink -list_formats 1 -i 'Intensity Pro'

       ·   Capture video clip at 1080i50:

                   ffmpeg -format_code Hi50 -f decklink -i 'Intensity Pro' -c:a copy -c:v copy output.avi

       ·   Capture video clip at 1080i50 10 bit:

                   ffmpeg -bm_v210 1 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

       ·   Capture video clip at 1080i50 with 16 audio channels:

                   ffmpeg -channels 16 -format_code Hi50 -f decklink -i 'UltraStudio Mini Recorder' -c:a copy -c:v copy output.avi

   dshow
       Windows DirectShow input device.

       DirectShow support is enabled when FFmpeg is built with the mingw-w64 project.  Currently
       only audio and video devices are supported.

       Multiple devices may be opened as separate inputs, but they may also be opened on the same
       input, which should improve synchronism between them.

       The input name should be in the format:

               <TYPE>=<NAME>[:<TYPE>=<NAME>]

       where TYPE can be either audio or video, and NAME is the device's name or alternative
       name..

       Options

       If no options are specified, the device's defaults are used.  If the device does not
       support the requested options, it will fail to open.

       video_size
           Set the video size in the captured video.

       framerate
           Set the frame rate in the captured video.

       sample_rate
           Set the sample rate (in Hz) of the captured audio.

       sample_size
           Set the sample size (in bits) of the captured audio.

       channels
           Set the number of channels in the captured audio.

       list_devices
           If set to true, print a list of devices and exit.

       list_options
           If set to true, print a list of selected device's options and exit.

       video_device_number
           Set video device number for devices with the same name (starts at 0, defaults to 0).

       audio_device_number
           Set audio device number for devices with the same name (starts at 0, defaults to 0).

       pixel_format
           Select pixel format to be used by DirectShow. This may only be set when the video
           codec is not set or set to rawvideo.

       audio_buffer_size
           Set audio device buffer size in milliseconds (which can directly impact latency,
           depending on the device).  Defaults to using the audio device's default buffer size
           (typically some multiple of 500ms).  Setting this value too low can degrade
           performance.  See also
           <http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx>

       video_pin_name
           Select video capture pin to use by name or alternative name.

       audio_pin_name
           Select audio capture pin to use by name or alternative name.

       crossbar_video_input_pin_number
           Select video input pin number for crossbar device. This will be routed to the crossbar
           device's Video Decoder output pin.  Note that changing this value can affect future
           invocations (sets a new default) until system reboot occurs.

       crossbar_audio_input_pin_number
           Select audio input pin number for crossbar device. This will be routed to the crossbar
           device's Audio Decoder output pin.  Note that changing this value can affect future
           invocations (sets a new default) until system reboot occurs.

       show_video_device_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to change video filter properties and configurations manually.  Note
           that for crossbar devices, adjusting values in this dialog may be needed at times to
           toggle between PAL (25 fps) and NTSC (29.97) input frame rates, sizes, interlacing,
           etc.  Changing these values can enable different scan rates/frame rates and avoiding
           green bars at the bottom, flickering scan lines, etc.  Note that with some devices,
           changing these properties can also affect future invocations (sets new defaults) until
           system reboot occurs.

       show_audio_device_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to change audio filter properties and configurations manually.

       show_video_crossbar_connection_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to manually modify crossbar pin routings, when it opens a video device.

       show_audio_crossbar_connection_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to manually modify crossbar pin routings, when it opens an audio device.

       show_analog_tv_tuner_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to manually modify TV channels and frequencies.

       show_analog_tv_tuner_audio_dialog
           If set to true, before capture starts, popup a display dialog to the end user,
           allowing them to manually modify TV audio (like mono vs. stereo, Language A,B or C).

       audio_device_load
           Load an audio capture filter device from file instead of searching it by name. It may
           load additional parameters too, if the filter supports the serialization of its
           properties to.  To use this an audio capture source has to be specified, but it can be
           anything even fake one.

       audio_device_save
           Save the currently used audio capture filter device and its parameters (if the filter
           supports it) to a file.  If a file with the same name exists it will be overwritten.

       video_device_load
           Load a video capture filter device from file instead of searching it by name. It may
           load additional parameters too, if the filter supports the serialization of its
           properties to.  To use this a video capture source has to be specified, but it can be
           anything even fake one.

       video_device_save
           Save the currently used video capture filter device and its parameters (if the filter
           supports it) to a file.  If a file with the same name exists it will be overwritten.

       Examples

       ·   Print the list of DirectShow supported devices and exit:

                   $ ffmpeg -list_devices true -f dshow -i dummy

       ·   Open video device Camera:

                   $ ffmpeg -f dshow -i video="Camera"

       ·   Open second video device with name Camera:

                   $ ffmpeg -f dshow -video_device_number 1 -i video="Camera"

       ·   Open video device Camera and audio device Microphone:

                   $ ffmpeg -f dshow -i video="Camera":audio="Microphone"

       ·   Print the list of supported options in selected device and exit:

                   $ ffmpeg -list_options true -f dshow -i video="Camera"

       ·   Specify pin names to capture by name or alternative name, specify alternative device
           name:

                   $ ffmpeg -f dshow -audio_pin_name "Audio Out" -video_pin_name 2 -i video=video="@device_pnp_\\?\pci#ven_1a0a&dev_6200&subsys_62021461&rev_01#4&e2c7dd6&0&00e1#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\{ca465100-deb0-4d59-818f-8c477184adf6}":audio="Microphone"

       ·   Configure a crossbar device, specifying crossbar pins, allow user to adjust video
           capture properties at startup:

                   $ ffmpeg -f dshow -show_video_device_dialog true -crossbar_video_input_pin_number 0
                        -crossbar_audio_input_pin_number 3 -i video="AVerMedia BDA Analog Capture":audio="AVerMedia BDA Analog Capture"

   fbdev
       Linux framebuffer input device.

       The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics
       on a computer monitor, typically on the console. It is accessed through a file device
       node, usually /dev/fb0.

       For more detailed information read the file Documentation/fb/framebuffer.txt included in
       the Linux source tree.

       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

       To record from the framebuffer device /dev/fb0 with ffmpeg:

               ffmpeg -f fbdev -framerate 10 -i /dev/fb0 out.avi

       You can take a single screenshot image with the command:

               ffmpeg -f fbdev -framerate 1 -i /dev/fb0 -frames:v 1 screenshot.jpeg

       Options

       framerate
           Set the frame rate. Default is 25.

   gdigrab
       Win32 GDI-based screen capture device.

       This device allows you to capture a region of the display on Windows.

       There are two options for the input filename:

               desktop

       or

               title=<window_title>

       The first option will capture the entire desktop, or a fixed region of the desktop. The
       second option will instead capture the contents of a single window, regardless of its
       position on the screen.

       For example, to grab the entire desktop using ffmpeg:

               ffmpeg -f gdigrab -framerate 6 -i desktop out.mpg

       Grab a 640x480 region at position "10,20":

               ffmpeg -f gdigrab -framerate 6 -offset_x 10 -offset_y 20 -video_size vga -i desktop out.mpg

       Grab the contents of the window named "Calculator"

               ffmpeg -f gdigrab -framerate 6 -i title=Calculator out.mpg

       Options

       draw_mouse
           Specify whether to draw the mouse pointer. Use the value 0 to not draw the pointer.
           Default value is 1.

       framerate
           Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
           "30000/1001".

       show_region
           Show grabbed region on screen.

           If show_region is specified with 1, then the grabbing region will be indicated on
           screen. With this option, it is easy to know what is being grabbed if only a portion
           of the screen is grabbed.

           Note that show_region is incompatible with grabbing the contents of a single window.

           For example:

                   ffmpeg -f gdigrab -show_region 1 -framerate 6 -video_size cif -offset_x 10 -offset_y 20 -i desktop out.mpg

       video_size
           Set the video frame size. The default is to capture the full screen if desktop is
           selected, or the full window size if title=window_title is selected.

       offset_x
           When capturing a region with video_size, set the distance from the left edge of the
           screen or desktop.

           Note that the offset calculation is from the top left corner of the primary monitor on
           Windows. If you have a monitor positioned to the left of your primary monitor, you
           will need to use a negative offset_x value to move the region to that monitor.

       offset_y
           When capturing a region with video_size, set the distance from the top edge of the
           screen or desktop.

           Note that the offset calculation is from the top left corner of the primary monitor on
           Windows. If you have a monitor positioned above your primary monitor, you will need to
           use a negative offset_y value to move the region to that monitor.

   iec61883
       FireWire DV/HDV input device using libiec61883.

       To enable this input device, you need libiec61883, libraw1394 and libavc1394 installed on
       your system. Use the configure option "--enable-libiec61883" to compile with the device
       enabled.

       The iec61883 capture device supports capturing from a video device connected via IEEE1394
       (FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default
       DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was
       removed.

       Specify the FireWire port to be used as input file, or "auto" to choose the first port
       connected.

       Options

       dvtype
           Override autodetection of DV/HDV. This should only be used if auto detection does not
           work, or if usage of a different device type should be prohibited. Treating a DV
           device as HDV (or vice versa) will not work and result in undefined behavior.  The
           values auto, dv and hdv are supported.

       dvbuffer
           Set maximum size of buffer for incoming data, in frames. For DV, this is an exact
           value. For HDV, it is not frame exact, since HDV does not have a fixed frame size.

       dvguid
           Select the capture device by specifying its GUID. Capturing will only be performed
           from the specified device and fails if no device with the given GUID is found. This is
           useful to select the input if multiple devices are connected at the same time.  Look
           at /sys/bus/firewire/devices to find out the GUIDs.

       Examples

       ·   Grab and show the input of a FireWire DV/HDV device.

                   ffplay -f iec61883 -i auto

       ·   Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000
           packets if the source is HDV.

                   ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg

   jack
       JACK input device.

       To enable this input device during configuration you need libjack installed on your
       system.

       A JACK input device creates one or more JACK writable clients, one for each audio channel,
       with name client_name:input_N, where client_name is the name provided by the application,
       and N is a number which identifies the channel.  Each writable client will send the
       acquired data to the FFmpeg input device.

       Once you have created one or more JACK readable clients, you need to connect them to one
       or more JACK writable clients.

       To connect or disconnect JACK clients you can use the jack_connect and jack_disconnect
       programs, or do it through a graphical interface, for example with qjackctl.

       To list the JACK clients and their properties you can invoke the command jack_lsp.

       Follows an example which shows how to capture a JACK readable client with ffmpeg.

               # Create a JACK writable client with name "ffmpeg".
               $ ffmpeg -f jack -i ffmpeg -y out.wav

               # Start the sample jack_metro readable client.
               $ jack_metro -b 120 -d 0.2 -f 4000

               # List the current JACK clients.
               $ jack_lsp -c
               system:capture_1
               system:capture_2
               system:playback_1
               system:playback_2
               ffmpeg:input_1
               metro:120_bpm

               # Connect metro to the ffmpeg writable client.
               $ jack_connect metro:120_bpm ffmpeg:input_1

       For more information read: <http://jackaudio.org/>

       Options

       channels
           Set the number of channels. Default is 2.

   kmsgrab
       KMS video input device.

       Captures the KMS scanout framebuffer associated with a specified CRTC or plane as a DRM
       object that can be passed to other hardware functions.

       Requires either DRM master or CAP_SYS_ADMIN to run.

       If you don't understand what all of that means, you probably don't want this.  Look at
       x11grab instead.

       Options

       device
           DRM device to capture on.  Defaults to /dev/dri/card0.

       format
           Pixel format of the framebuffer.  Defaults to bgr0.

       format_modifier
           Format modifier to signal on output frames.  This is necessary to import correctly
           into some APIs, but can't be autodetected.  See the libdrm documentation for possible
           values.

       crtc_id
           KMS CRTC ID to define the capture source.  The first active plane on the given CRTC
           will be used.

       plane_id
           KMS plane ID to define the capture source.  Defaults to the first active plane found
           if neither crtc_id nor plane_id are specified.

       framerate
           Framerate to capture at.  This is not synchronised to any page flipping or framebuffer
           changes - it just defines the interval at which the framebuffer is sampled.  Sampling
           faster than the framebuffer update rate will generate independent frames with the same
           content.  Defaults to 30.

       Examples

       ·   Capture from the first active plane, download the result to normal frames and encode.
           This will only work if the framebuffer is both linear and mappable - if not, the
           result may be scrambled or fail to download.

                   ffmpeg -f kmsgrab -i - -vf 'hwdownload,format=bgr0' output.mp4

       ·   Capture from CRTC ID 42 at 60fps, map the result to VAAPI, convert to NV12 and encode
           as H.264.

                   ffmpeg -crtc_id 42 -framerate 60 -f kmsgrab -i - -vf 'hwmap=derive_device=vaapi,scale_vaapi=w=1920:h=1080:format=nv12' -c:v h264_vaapi output.mp4

   lavfi
       Libavfilter input virtual device.

       This input device reads data from the open output pads of a libavfilter filtergraph.

       For each filtergraph open output, the input device will create a corresponding stream
       which is mapped to the generated output. Currently only video data is supported. The
       filtergraph is specified through the option graph.

       Options

       graph
           Specify the filtergraph to use as input. Each video open output must be labelled by a
           unique string of the form "outN", where N is a number starting from 0 corresponding to
           the mapped input stream generated by the device.  The first unlabelled output is
           automatically assigned to the "out0" label, but all the others need to be specified
           explicitly.

           The suffix "+subcc" can be appended to the output label to create an extra stream with
           the closed captions packets attached to that output (experimental; only for EIA-608 /
           CEA-708 for now).  The subcc streams are created after all the normal streams, in the
           order of the corresponding stream.  For example, if there is "out19+subcc",
           "out7+subcc" and up to "out42", the stream #43 is subcc for stream #7 and stream #44
           is subcc for stream #19.

           If not specified defaults to the filename specified for the input device.

       graph_file
           Set the filename of the filtergraph to be read and sent to the other filters. Syntax
           of the filtergraph is the same as the one specified by the option graph.

       dumpgraph
           Dump graph to stderr.

       Examples

       ·   Create a color video stream and play it back with ffplay:

                   ffplay -f lavfi -graph "color=c=pink [out0]" dummy

       ·   As the previous example, but use filename for specifying the graph description, and
           omit the "out0" label:

                   ffplay -f lavfi color=c=pink

       ·   Create three different video test filtered sources and play them:

                   ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

       ·   Read an audio stream from a file using the amovie source and play it back with ffplay:

                   ffplay -f lavfi "amovie=test.wav"

       ·   Read an audio stream and a video stream and play it back with ffplay:

                   ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

       ·   Dump decoded frames to images and closed captions to a file (experimental):

                   ffmpeg -f lavfi -i "movie=test.ts[out0+subcc]" -map v frame%08d.png -map s -c copy -f rawvideo subcc.bin

   libcdio
       Audio-CD input device based on libcdio.

       To enable this input device during configuration you need libcdio installed on your
       system. It requires the configure option "--enable-libcdio".

       This device allows playing and grabbing from an Audio-CD.

       For example to copy with ffmpeg the entire Audio-CD in /dev/sr0, you may run the command:

               ffmpeg -f libcdio -i /dev/sr0 cd.wav

       Options

       speed
           Set drive reading speed. Default value is 0.

           The speed is specified CD-ROM speed units. The speed is set through the libcdio
           "cdio_cddap_speed_set" function. On many CD-ROM drives, specifying a value too large
           will result in using the fastest speed.

       paranoia_mode
           Set paranoia recovery mode flags. It accepts one of the following values:

           disable
           verify
           overlap
           neverskip
           full

           Default value is disable.

           For more information about the available recovery modes, consult the paranoia project
           documentation.

   libdc1394
       IIDC1394 input device, based on libdc1394 and libraw1394.

       Requires the configure option "--enable-libdc1394".

   libndi_newtek
       The libndi_newtek input device provides capture capabilities for using NDI (Network Device
       Interface, standard created by NewTek).

       Input filename is a NDI source name that could be found by sending -find_sources 1 to
       command line - it has no specific syntax but human-readable formatted.

       To enable this input device, you need the NDI SDK and you need to configure with the
       appropriate "--extra-cflags" and "--extra-ldflags".

       Options

       find_sources
           If set to true, print a list of found/available NDI sources and exit.  Defaults to
           false.

       wait_sources
           Override time to wait until the number of online sources have changed.  Defaults to
           0.5.

       allow_video_fields
           When this flag is false, all video that you receive will be progressive.  Defaults to
           true.

       extra_ips
           If is set to list of comma separated ip addresses, scan for sources not only using
           mDNS but also use unicast ip addresses specified by this list.

       Examples

       ·   List input devices:

                   ffmpeg -f libndi_newtek -find_sources 1 -i dummy

       ·   List local and remote input devices:

                   ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -find_sources 1 -i dummy

       ·   Restream to NDI:

                   ffmpeg -f libndi_newtek -i "DEV-5.INTERNAL.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2

       ·   Restream remote NDI to local NDI:

                   ffmpeg -f libndi_newtek -extra_ips "192.168.10.10" -i "DEV-5.REMOTE.M1STEREO.TV (NDI_SOURCE_NAME_1)" -f libndi_newtek -y NDI_SOURCE_NAME_2

   openal
       The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1
       implementation.

       To enable this input device during configuration, you need OpenAL headers and libraries
       installed on your system, and need to configure FFmpeg with "--enable-openal".

       OpenAL headers and libraries should be provided as part of your OpenAL implementation, or
       as an additional download (an SDK). Depending on your installation you may need to specify
       additional flags via the "--extra-cflags" and "--extra-ldflags" for allowing the build
       system to locate the OpenAL headers and libraries.

       An incomplete list of OpenAL implementations follows:

       Creative
           The official Windows implementation, providing hardware acceleration with supported
           devices and software fallback.  See <http://openal.org/>.

       OpenAL Soft
           Portable, open source (LGPL) software implementation. Includes backends for the most
           common sound APIs on the Windows, Linux, Solaris, and BSD operating systems.  See
           <http://kcat.strangesoft.net/openal.html>.

       Apple
           OpenAL is part of Core Audio, the official Mac OS X Audio interface.  See
           <http://developer.apple.com/technologies/mac/audio-and-video.html>

       This device allows one to capture from an audio input device handled through OpenAL.

       You need to specify the name of the device to capture in the provided filename. If the
       empty string is provided, the device will automatically select the default device. You can
       get the list of the supported devices by using the option list_devices.

       Options

       channels
           Set the number of channels in the captured audio. Only the values 1 (monaural) and 2
           (stereo) are currently supported.  Defaults to 2.

       sample_size
           Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are
           currently supported. Defaults to 16.

       sample_rate
           Set the sample rate (in Hz) of the captured audio.  Defaults to 44.1k.

       list_devices
           If set to true, print a list of devices and exit.  Defaults to false.

       Examples

       Print the list of OpenAL supported devices and exit:

               $ ffmpeg -list_devices true -f openal -i dummy out.ogg

       Capture from the OpenAL device DR-BT101 via PulseAudio:

               $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg

       Capture from the default device (note the empty string '' as filename):

               $ ffmpeg -f openal -i '' out.ogg

       Capture from two devices simultaneously, writing to two different files, within the same
       ffmpeg command:

               $ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg

       Note: not all OpenAL implementations support multiple simultaneous capture - try the
       latest OpenAL Soft if the above does not work.

   oss
       Open Sound System input device.

       The filename to provide to the input device is the device node representing the OSS input
       device, and is usually set to /dev/dsp.

       For example to grab from /dev/dsp using ffmpeg use the command:

               ffmpeg -f oss -i /dev/dsp /tmp/oss.wav

       For more information about OSS see: <http://manuals.opensound.com/usersguide/dsp.html>

       Options

       sample_rate
           Set the sample rate in Hz. Default is 48000.

       channels
           Set the number of channels. Default is 2.

   pulse
       PulseAudio input device.

       To enable this output device you need to configure FFmpeg with "--enable-libpulse".

       The filename to provide to the input device is a source device or the string "default"

       To list the PulseAudio source devices and their properties you can invoke the command
       pactl list sources.

       More information about PulseAudio can be found on <http://www.pulseaudio.org>.

       Options

       server
           Connect to a specific PulseAudio server, specified by an IP address.  Default server
           is used when not provided.

       name
           Specify the application name PulseAudio will use when showing active clients, by
           default it is the "LIBAVFORMAT_IDENT" string.

       stream_name
           Specify the stream name PulseAudio will use when showing active streams, by default it
           is "record".

       sample_rate
           Specify the samplerate in Hz, by default 48kHz is used.

       channels
           Specify the channels in use, by default 2 (stereo) is set.

       frame_size
           Specify the number of bytes per frame, by default it is set to 1024.

       fragment_size
           Specify the minimal buffering fragment in PulseAudio, it will affect the audio
           latency. By default it is unset.

       wallclock
           Set the initial PTS using the current time. Default is 1.

       Examples

       Record a stream from default device:

               ffmpeg -f pulse -i default /tmp/pulse.wav

   sndio
       sndio input device.

       To enable this input device during configuration you need libsndio installed on your
       system.

       The filename to provide to the input device is the device node representing the sndio
       input device, and is usually set to /dev/audio0.

       For example to grab from /dev/audio0 using ffmpeg use the command:

               ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav

       Options

       sample_rate
           Set the sample rate in Hz. Default is 48000.

       channels
           Set the number of channels. Default is 2.

   video4linux2, v4l2
       Video4Linux2 input video device.

       "v4l2" can be used as alias for "video4linux2".

       If FFmpeg is built with v4l-utils support (by using the "--enable-libv4l2" configure
       option), it is possible to use it with the "-use_libv4l2" input device option.

       The name of the device to grab is a file device node, usually Linux systems tend to
       automatically create such nodes when the device (e.g. an USB webcam) is plugged into the
       system, and has a name of the kind /dev/videoN, where N is a number associated to the
       device.

       Video4Linux2 devices usually support a limited set of widthxheight sizes and frame rates.
       You can check which are supported using -list_formats all for Video4Linux2 devices.  Some
       devices, like TV cards, support one or more standards. It is possible to list all the
       supported standards using -list_standards all.

       The time base for the timestamps is 1 microsecond. Depending on the kernel version and
       configuration, the timestamps may be derived from the real time clock (origin at the Unix
       Epoch) or the monotonic clock (origin usually at boot time, unaffected by NTP or manual
       changes to the clock). The -timestamps abs or -ts abs option can be used to force
       conversion into the real time clock.

       Some usage examples of the video4linux2 device with ffmpeg and ffplay:

       ·   List supported formats for a video4linux2 device:

                   ffplay -f video4linux2 -list_formats all /dev/video0

       ·   Grab and show the input of a video4linux2 device:

                   ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0

       ·   Grab and record the input of a video4linux2 device, leave the frame rate and size as
           previously set:

                   ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

       For more information about Video4Linux, check <http://linuxtv.org/>.

       Options

       standard
           Set the standard. Must be the name of a supported standard. To get a list of the
           supported standards, use the list_standards option.

       channel
           Set the input channel number. Default to -1, which means using the previously selected
           channel.

       video_size
           Set the video frame size. The argument must be a string in the form WIDTHxHEIGHT or a
           valid size abbreviation.

       pixel_format
           Select the pixel format (only valid for raw video input).

       input_format
           Set the preferred pixel format (for raw video) or a codec name.  This option allows
           one to select the input format, when several are available.

       framerate
           Set the preferred video frame rate.

       list_formats
           List available formats (supported pixel formats, codecs, and frame sizes) and exit.

           Available values are:

           all Show all available (compressed and non-compressed) formats.

           raw Show only raw video (non-compressed) formats.

           compressed
               Show only compressed formats.

       list_standards
           List supported standards and exit.

           Available values are:

           all Show all supported standards.

       timestamps, ts
           Set type of timestamps for grabbed frames.

           Available values are:

           default
               Use timestamps from the kernel.

           abs Use absolute timestamps (wall clock).

           mono2abs
               Force conversion from monotonic to absolute timestamps.

           Default value is "default".

       use_libv4l2
           Use libv4l2 (v4l-utils) conversion functions. Default is 0.

   vfwcap
       VfW (Video for Windows) capture input device.

       The filename passed as input is the capture driver number, ranging from 0 to 9. You may
       use "list" as filename to print a list of drivers. Any other filename will be interpreted
       as device number 0.

       Options

       video_size
           Set the video frame size.

       framerate
           Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
           "30000/1001".

   x11grab
       X11 video input device.

       To enable this input device during configuration you need libxcb installed on your system.
       It will be automatically detected during configuration.

       This device allows one to capture a region of an X11 display.

       The filename passed as input has the syntax:

               [<hostname>]:<display_number>.<screen_number>[+<x_offset>,<y_offset>]

       hostname:display_number.screen_number specifies the X11 display name of the screen to grab
       from. hostname can be omitted, and defaults to "localhost". The environment variable
       DISPLAY contains the default display name.

       x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left
       border of the X11 screen. They default to 0.

       Check the X11 documentation (e.g. man X) for more detailed information.

       Use the xdpyinfo program for getting basic information about the properties of your X11
       display (e.g. grep for "name" or "dimensions").

       For example to grab from :0.0 using ffmpeg:

               ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0 out.mpg

       Grab at position "10,20":

               ffmpeg -f x11grab -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

       Options

       draw_mouse
           Specify whether to draw the mouse pointer. A value of 0 specifies not to draw the
           pointer. Default value is 1.

       follow_mouse
           Make the grabbed area follow the mouse. The argument can be "centered" or a number of
           pixels PIXELS.

           When it is specified with "centered", the grabbing region follows the mouse pointer
           and keeps the pointer at the center of region; otherwise, the region follows only when
           the mouse pointer reaches within PIXELS (greater than zero) to the edge of region.

           For example:

                   ffmpeg -f x11grab -follow_mouse centered -framerate 25 -video_size cif -i :0.0 out.mpg

           To follow only when the mouse pointer reaches within 100 pixels to edge:

                   ffmpeg -f x11grab -follow_mouse 100 -framerate 25 -video_size cif -i :0.0 out.mpg

       framerate
           Set the grabbing frame rate. Default value is "ntsc", corresponding to a frame rate of
           "30000/1001".

       show_region
           Show grabbed region on screen.

           If show_region is specified with 1, then the grabbing region will be indicated on
           screen. With this option, it is easy to know what is being grabbed if only a portion
           of the screen is grabbed.

       region_border
           Set the region border thickness if -show_region 1 is used.  Range is 1 to 128 and
           default is 3 (XCB-based x11grab only).

           For example:

                   ffmpeg -f x11grab -show_region 1 -framerate 25 -video_size cif -i :0.0+10,20 out.mpg

           With follow_mouse:

                   ffmpeg -f x11grab -follow_mouse centered -show_region 1 -framerate 25 -video_size cif -i :0.0 out.mpg

       video_size
           Set the video frame size. Default value is "vga".

       grab_x
       grab_y
           Set the grabbing region coordinates. They are expressed as offset from the top left
           corner of the X11 window and correspond to the x_offset and y_offset parameters in the
           device name. The default value for both options is 0.

OUTPUT DEVICES

       Output devices are configured elements in FFmpeg that can write multimedia data to an
       output device attached to your system.

       When you configure your FFmpeg build, all the supported output devices are enabled by
       default. You can list all available ones using the configure option "--list-outdevs".

       You can disable all the output devices using the configure option "--disable-outdevs", and
       selectively enable an output device using the option "--enable-outdev=OUTDEV", or you can
       disable a particular input device using the option "--disable-outdev=OUTDEV".

       The option "-devices" of the ff* tools will display the list of enabled output devices.

       A description of the currently available output devices follows.

   alsa
       ALSA (Advanced Linux Sound Architecture) output device.

       Examples

       ·   Play a file on default ALSA device:

                   ffmpeg -i INPUT -f alsa default

       ·   Play a file on soundcard 1, audio device 7:

                   ffmpeg -i INPUT -f alsa hw:1,7

   caca
       CACA output device.

       This output device allows one to show a video stream in CACA window.  Only one CACA window
       is allowed per application, so you can have only one instance of this output device in an
       application.

       To enable this output device you need to configure FFmpeg with "--enable-libcaca".
       libcaca is a graphics library that outputs text instead of pixels.

       For more information about libcaca, check: <http://caca.zoy.org/wiki/libcaca>

       Options

       window_title
           Set the CACA window title, if not specified default to the filename specified for the
           output device.

       window_size
           Set the CACA window size, can be a string of the form widthxheight or a video size
           abbreviation.  If not specified it defaults to the size of the input video.

       driver
           Set display driver.

       algorithm
           Set dithering algorithm. Dithering is necessary because the picture being rendered has
           usually far more colours than the available palette.  The accepted values are listed
           with "-list_dither algorithms".

       antialias
           Set antialias method. Antialiasing smoothens the rendered image and avoids the
           commonly seen staircase effect.  The accepted values are listed with "-list_dither
           antialiases".

       charset
           Set which characters are going to be used when rendering text.  The accepted values
           are listed with "-list_dither charsets".

       color
           Set color to be used when rendering text.  The accepted values are listed with
           "-list_dither colors".

       list_drivers
           If set to true, print a list of available drivers and exit.

       list_dither
           List available dither options related to the argument.  The argument must be one of
           "algorithms", "antialiases", "charsets", "colors".

       Examples

       ·   The following command shows the ffmpeg output is an CACA window, forcing its size to
           80x25:

                   ffmpeg -i INPUT -c:v rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca -

       ·   Show the list of available drivers and exit:

                   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true -

       ·   Show the list of available dither colors and exit:

                   ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors -

   decklink
       The decklink output device provides playback capabilities for Blackmagic DeckLink devices.

       To enable this output device, you need the Blackmagic DeckLink SDK and you need to
       configure with the appropriate "--extra-cflags" and "--extra-ldflags".  On Windows, you
       need to run the IDL files through widl.

       DeckLink is very picky about the formats it supports. Pixel format is always uyvy422,
       framerate, field order and video size must be determined for your device with
       -list_formats 1. Audio sample rate is always 48 kHz.

       Options

       list_devices
           If set to true, print a list of devices and exit.  Defaults to false. Alternatively
           you can use the "-sinks" option of ffmpeg to list the available output devices.

       list_formats
           If set to true, print a list of supported formats and exit.  Defaults to false.

       preroll
           Amount of time to preroll video in seconds.  Defaults to 0.5.

       duplex_mode
           Sets the decklink device duplex mode. Must be unset, half or full.  Defaults to unset.

       Examples

       ·   List output devices:

                   ffmpeg -i test.avi -f decklink -list_devices 1 dummy

       ·   List supported formats:

                   ffmpeg -i test.avi -f decklink -list_formats 1 'DeckLink Mini Monitor'

       ·   Play video clip:

                   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 'DeckLink Mini Monitor'

       ·   Play video clip with non-standard framerate or video size:

                   ffmpeg -i test.avi -f decklink -pix_fmt uyvy422 -s 720x486 -r 24000/1001 'DeckLink Mini Monitor'

   fbdev
       Linux framebuffer output device.

       The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics
       on a computer monitor, typically on the console. It is accessed through a file device
       node, usually /dev/fb0.

       For more detailed information read the file Documentation/fb/framebuffer.txt included in
       the Linux source tree.

       Options

       xoffset
       yoffset
           Set x/y coordinate of top left corner. Default is 0.

       Examples

       Play a file on framebuffer device /dev/fb0.  Required pixel format depends on current
       framebuffer settings.

               ffmpeg -re -i INPUT -c:v rawvideo -pix_fmt bgra -f fbdev /dev/fb0

       See also <http://linux-fbdev.sourceforge.net/>, and fbset(1).

   libndi_newtek
       The libndi_newtek output device provides playback capabilities for using NDI (Network
       Device Interface, standard created by NewTek).

       Output filename is a NDI name.

       To enable this output device, you need the NDI SDK and you need to configure with the
       appropriate "--extra-cflags" and "--extra-ldflags".

       NDI uses uyvy422 pixel format natively, but also supports bgra, bgr0, rgba and rgb0.

       Options

       reference_level
           The audio reference level in dB. This specifies how many dB above the reference level
           (+4dBU) is the full range of 16 bit audio.  Defaults to 0.

       clock_video
           These specify whether video "clock" themselves.  Defaults to false.

       clock_audio
           These specify whether audio "clock" themselves.  Defaults to false.

       Examples

       ·   Play video clip:

                   ffmpeg -i "udp://@239.1.1.1:10480?fifo_size=1000000&overrun_nonfatal=1" -vf "scale=720:576,fps=fps=25,setdar=dar=16/9,format=pix_fmts=uyvy422" -f libndi_newtek NEW_NDI1

   opengl
       OpenGL output device.

       To enable this output device you need to configure FFmpeg with "--enable-opengl".

       This output device allows one to render to OpenGL context.  Context may be provided by
       application or default SDL window is created.

       When device renders to external context, application must implement handlers for following
       messages: "AV_DEV_TO_APP_CREATE_WINDOW_BUFFER" - create OpenGL context on current thread.
       "AV_DEV_TO_APP_PREPARE_WINDOW_BUFFER" - make OpenGL context current.
       "AV_DEV_TO_APP_DISPLAY_WINDOW_BUFFER" - swap buffers.
       "AV_DEV_TO_APP_DESTROY_WINDOW_BUFFER" - destroy OpenGL context.  Application is also
       required to inform a device about current resolution by sending
       "AV_APP_TO_DEV_WINDOW_SIZE" message.

       Options

       background
           Set background color. Black is a default.

       no_window
           Disables default SDL window when set to non-zero value.  Application must provide
           OpenGL context and both "window_size_cb" and "window_swap_buffers_cb" callbacks when
           set.

       window_title
           Set the SDL window title, if not specified default to the filename specified for the
           output device.  Ignored when no_window is set.

       window_size
           Set preferred window size, can be a string of the form widthxheight or a video size
           abbreviation.  If not specified it defaults to the size of the input video, downscaled
           according to the aspect ratio.  Mostly usable when no_window is not set.

       Examples

       Play a file on SDL window using OpenGL rendering:

               ffmpeg  -i INPUT -f opengl "window title"

   oss
       OSS (Open Sound System) output device.

   pulse
       PulseAudio output device.

       To enable this output device you need to configure FFmpeg with "--enable-libpulse".

       More information about PulseAudio can be found on <http://www.pulseaudio.org>

       Options

       server
           Connect to a specific PulseAudio server, specified by an IP address.  Default server
           is used when not provided.

       name
           Specify the application name PulseAudio will use when showing active clients, by
           default it is the "LIBAVFORMAT_IDENT" string.

       stream_name
           Specify the stream name PulseAudio will use when showing active streams, by default it
           is set to the specified output name.

       device
           Specify the device to use. Default device is used when not provided.  List of output
           devices can be obtained with command pactl list sinks.

       buffer_size
       buffer_duration
           Control the size and duration of the PulseAudio buffer. A small buffer gives more
           control, but requires more frequent updates.

           buffer_size specifies size in bytes while buffer_duration specifies duration in
           milliseconds.

           When both options are provided then the highest value is used (duration is
           recalculated to bytes using stream parameters). If they are set to 0 (which is
           default), the device will use the default PulseAudio duration value. By default
           PulseAudio set buffer duration to around 2 seconds.

       prebuf
           Specify pre-buffering size in bytes. The server does not start with playback before at
           least prebuf bytes are available in the buffer. By default this option is initialized
           to the same value as buffer_size or buffer_duration (whichever is bigger).

       minreq
           Specify minimum request size in bytes. The server does not request less than minreq
           bytes from the client, instead waits until the buffer is free enough to request more
           bytes at once. It is recommended to not set this option, which will initialize this to
           a value that is deemed sensible by the server.

       Examples

       Play a file on default device on default server:

               ffmpeg  -i INPUT -f pulse "stream name"

   sdl
       SDL (Simple DirectMedia Layer) output device.

       This output device allows one to show a video stream in an SDL window. Only one SDL window
       is allowed per application, so you can have only one instance of this output device in an
       application.

       To enable this output device you need libsdl installed on your system when configuring
       your build.

       For more information about SDL, check: <http://www.libsdl.org/>

       Options

       window_title
           Set the SDL window title, if not specified default to the filename specified for the
           output device.

       icon_title
           Set the name of the iconified SDL window, if not specified it is set to the same value
           of window_title.

       window_size
           Set the SDL window size, can be a string of the form widthxheight or a video size
           abbreviation.  If not specified it defaults to the size of the input video, downscaled
           according to the aspect ratio.

       window_x
       window_y
           Set the position of the window on the screen.

       window_fullscreen
           Set fullscreen mode when non-zero value is provided.  Default value is zero.

       window_enable_quit
           Enable quit action (using window button or keyboard key) when non-zero value is
           provided.  Default value is 1 (enable quit action)

       Interactive commands

       The window created by the device can be controlled through the following interactive
       commands.

       q, ESC
           Quit the device immediately.

       Examples

       The following command shows the ffmpeg output is an SDL window, forcing its size to the
       qcif format:

               ffmpeg -i INPUT -c:v rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output"

   sndio
       sndio audio output device.

   v4l2
       Video4Linux2 output device.

   xv
       XV (XVideo) output device.

       This output device allows one to show a video stream in a X Window System window.

       Options

       display_name
           Specify the hardware display name, which determines the display and communications
           domain to be used.

           The display name or DISPLAY environment variable can be a string in the format
           hostname[:number[.screen_number]].

           hostname specifies the name of the host machine on which the display is physically
           attached. number specifies the number of the display server on that host machine.
           screen_number specifies the screen to be used on that server.

           If unspecified, it defaults to the value of the DISPLAY environment variable.

           For example, "dual-headed:0.1" would specify screen 1 of display 0 on the machine
           named ``dual-headed''.

           Check the X11 specification for more detailed information about the display name
           format.

       window_id
           When set to non-zero value then device doesn't create new window, but uses existing
           one with provided window_id. By default this options is set to zero and device creates
           its own window.

       window_size
           Set the created window size, can be a string of the form widthxheight or a video size
           abbreviation. If not specified it defaults to the size of the input video.  Ignored
           when window_id is set.

       window_x
       window_y
           Set the X and Y window offsets for the created window. They are both set to 0 by
           default. The values may be ignored by the window manager.  Ignored when window_id is
           set.

       window_title
           Set the window title, if not specified default to the filename specified for the
           output device. Ignored when window_id is set.

       For more information about XVideo see <http://www.x.org/>.

       Examples

       ·   Decode, display and encode video input with ffmpeg at the same time:

                   ffmpeg -i INPUT OUTPUT -f xv display

       ·   Decode and display the input video to multiple X11 windows:

                   ffmpeg -i INPUT -f xv normal -vf negate -f xv negated

RESAMPLER OPTIONS

       The audio resampler supports the following named options.

       Options may be set by specifying -option value in the FFmpeg tools, option=value for the
       aresample filter, by setting the value explicitly in the "SwrContext" options or using the
       libavutil/opt.h API for programmatic use.

       ich, in_channel_count
           Set the number of input channels. Default value is 0. Setting this value is not
           mandatory if the corresponding channel layout in_channel_layout is set.

       och, out_channel_count
           Set the number of output channels. Default value is 0. Setting this value is not
           mandatory if the corresponding channel layout out_channel_layout is set.

       uch, used_channel_count
           Set the number of used input channels. Default value is 0. This option is only used
           for special remapping.

       isr, in_sample_rate
           Set the input sample rate. Default value is 0.

       osr, out_sample_rate
           Set the output sample rate. Default value is 0.

       isf, in_sample_fmt
           Specify the input sample format. It is set by default to "none".

       osf, out_sample_fmt
           Specify the output sample format. It is set by default to "none".

       tsf, internal_sample_fmt
           Set the internal sample format. Default value is "none".  This will automatically be
           chosen when it is not explicitly set.

       icl, in_channel_layout
       ocl, out_channel_layout
           Set the input/output channel layout.

           See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

       clev, center_mix_level
           Set the center mix level. It is a value expressed in deciBel, and must be in the
           interval [-32,32].

       slev, surround_mix_level
           Set the surround mix level. It is a value expressed in deciBel, and must be in the
           interval [-32,32].

       lfe_mix_level
           Set LFE mix into non LFE level. It is used when there is a LFE input but no LFE
           output. It is a value expressed in deciBel, and must be in the interval [-32,32].

       rmvol, rematrix_volume
           Set rematrix volume. Default value is 1.0.

       rematrix_maxval
           Set maximum output value for rematrixing.  This can be used to prevent clipping vs.
           preventing volume reduction.  A value of 1.0 prevents clipping.

       flags, swr_flags
           Set flags used by the converter. Default value is 0.

           It supports the following individual flags:

           res force resampling, this flag forces resampling to be used even when the input and
               output sample rates match.

       dither_scale
           Set the dither scale. Default value is 1.

       dither_method
           Set dither method. Default value is 0.

           Supported values:

           rectangular
               select rectangular dither

           triangular
               select triangular dither

           triangular_hp
               select triangular dither with high pass

           lipshitz
               select Lipshitz noise shaping dither.

           shibata
               select Shibata noise shaping dither.

           low_shibata
               select low Shibata noise shaping dither.

           high_shibata
               select high Shibata noise shaping dither.

           f_weighted
               select f-weighted noise shaping dither

           modified_e_weighted
               select modified-e-weighted noise shaping dither

           improved_e_weighted
               select improved-e-weighted noise shaping dither

       resampler
           Set resampling engine. Default value is swr.

           Supported values:

           swr select the native SW Resampler; filter options precision and cheby are not
               applicable in this case.

           soxr
               select the SoX Resampler (where available); compensation, and filter options
               filter_size, phase_shift, exact_rational, filter_type & kaiser_beta, are not
               applicable in this case.

       filter_size
           For swr only, set resampling filter size, default value is 32.

       phase_shift
           For swr only, set resampling phase shift, default value is 10, and must be in the
           interval [0,30].

       linear_interp
           Use linear interpolation when enabled (the default). Disable it if you want to
           preserve speed instead of quality when exact_rational fails.

       exact_rational
           For swr only, when enabled, try to use exact phase_count based on input and output
           sample rate. However, if it is larger than "1 << phase_shift", the phase_count will be
           "1 << phase_shift" as fallback. Default is enabled.

       cutoff
           Set cutoff frequency (swr: 6dB point; soxr: 0dB point) ratio; must be a float value
           between 0 and 1.  Default value is 0.97 with swr, and 0.91 with soxr (which, with a
           sample-rate of 44100, preserves the entire audio band to 20kHz).

       precision
           For soxr only, the precision in bits to which the resampled signal will be calculated.
           The default value of 20 (which, with suitable dithering, is appropriate for a
           destination bit-depth of 16) gives SoX's 'High Quality'; a value of 28 gives SoX's
           'Very High Quality'.

       cheby
           For soxr only, selects passband rolloff none (Chebyshev) & higher-precision
           approximation for 'irrational' ratios. Default value is 0.

       async
           For swr only, simple 1 parameter audio sync to timestamps using stretching, squeezing,
           filling and trimming. Setting this to 1 will enable filling and trimming, larger
           values represent the maximum amount in samples that the data may be stretched or
           squeezed for each second.  Default value is 0, thus no compensation is applied to make
           the samples match the audio timestamps.

       first_pts
           For swr only, assume the first pts should be this value. The time unit is 1 / sample
           rate.  This allows for padding/trimming at the start of stream. By default, no
           assumption is made about the first frame's expected pts, so no padding or trimming is
           done. For example, this could be set to 0 to pad the beginning with silence if an
           audio stream starts after the video stream or to trim any samples with a negative pts
           due to encoder delay.

       min_comp
           For swr only, set the minimum difference between timestamps and audio data (in
           seconds) to trigger stretching/squeezing/filling or trimming of the data to make it
           match the timestamps. The default is that stretching/squeezing/filling and trimming is
           disabled (min_comp = "FLT_MAX").

       min_hard_comp
           For swr only, set the minimum difference between timestamps and audio data (in
           seconds) to trigger adding/dropping samples to make it match the timestamps.  This
           option effectively is a threshold to select between hard (trim/fill) and soft
           (squeeze/stretch) compensation. Note that all compensation is by default disabled
           through min_comp.  The default is 0.1.

       comp_duration
           For swr only, set duration (in seconds) over which data is stretched/squeezed to make
           it match the timestamps. Must be a non-negative double float value, default value is
           1.0.

       max_soft_comp
           For swr only, set maximum factor by which data is stretched/squeezed to make it match
           the timestamps. Must be a non-negative double float value, default value is 0.

       matrix_encoding
           Select matrixed stereo encoding.

           It accepts the following values:

           none
               select none

           dolby
               select Dolby

           dplii
               select Dolby Pro Logic II

           Default value is "none".

       filter_type
           For swr only, select resampling filter type. This only affects resampling operations.

           It accepts the following values:

           cubic
               select cubic

           blackman_nuttall
               select Blackman Nuttall windowed sinc

           kaiser
               select Kaiser windowed sinc

       kaiser_beta
           For swr only, set Kaiser window beta value. Must be a double float value in the
           interval [2,16], default value is 9.

       output_sample_bits
           For swr only, set number of used output sample bits for dithering. Must be an integer
           in the interval [0,64], default value is 0, which means it's not used.

SCALER OPTIONS

       The video scaler supports the following named options.

       Options may be set by specifying -option value in the FFmpeg tools. For programmatic use,
       they can be set explicitly in the "SwsContext" options or through the libavutil/opt.h API.

       sws_flags
           Set the scaler flags. This is also used to set the scaling algorithm. Only a single
           algorithm should be selected. Default value is bicubic.

           It accepts the following values:

           fast_bilinear
               Select fast bilinear scaling algorithm.

           bilinear
               Select bilinear scaling algorithm.

           bicubic
               Select bicubic scaling algorithm.

           experimental
               Select experimental scaling algorithm.

           neighbor
               Select nearest neighbor rescaling algorithm.

           area
               Select averaging area rescaling algorithm.

           bicublin
               Select bicubic scaling algorithm for the luma component, bilinear for chroma
               components.

           gauss
               Select Gaussian rescaling algorithm.

           sinc
               Select sinc rescaling algorithm.

           lanczos
               Select Lanczos rescaling algorithm.

           spline
               Select natural bicubic spline rescaling algorithm.

           print_info
               Enable printing/debug logging.

           accurate_rnd
               Enable accurate rounding.

           full_chroma_int
               Enable full chroma interpolation.

           full_chroma_inp
               Select full chroma input.

           bitexact
               Enable bitexact output.

       srcw
           Set source width.

       srch
           Set source height.

       dstw
           Set destination width.

       dsth
           Set destination height.

       src_format
           Set source pixel format (must be expressed as an integer).

       dst_format
           Set destination pixel format (must be expressed as an integer).

       src_range
           Select source range.

       dst_range
           Select destination range.

       param0, param1
           Set scaling algorithm parameters. The specified values are specific of some scaling
           algorithms and ignored by others. The specified values are floating point number
           values.

       sws_dither
           Set the dithering algorithm. Accepts one of the following values. Default value is
           auto.

           auto
               automatic choice

           none
               no dithering

           bayer
               bayer dither

           ed  error diffusion dither

           a_dither
               arithmetic dither, based using addition

           x_dither
               arithmetic dither, based using xor (more random/less apparent patterning that
               a_dither).

       alphablend
           Set the alpha blending to use when the input has alpha but the output does not.
           Default value is none.

           uniform_color
               Blend onto a uniform background color

           checkerboard
               Blend onto a checkerboard

           none
               No blending

FILTERING INTRODUCTION

       Filtering in FFmpeg is enabled through the libavfilter library.

       In libavfilter, a filter can have multiple inputs and multiple outputs.  To illustrate the
       sorts of things that are possible, we consider the following filtergraph.

                               [main]
               input --> split ---------------------> overlay --> output
                           |                             ^
                           |[tmp]                  [flip]|
                           +-----> crop --> vflip -------+

       This filtergraph splits the input stream in two streams, then sends one stream through the
       crop filter and the vflip filter, before merging it back with the other stream by
       overlaying it on top. You can use the following command to achieve this:

               ffmpeg -i INPUT -vf "split [main][tmp]; [tmp] crop=iw:ih/2:0:0, vflip [flip]; [main][flip] overlay=0:H/2" OUTPUT

       The result will be that the top half of the video is mirrored onto the bottom half of the
       output video.

       Filters in the same linear chain are separated by commas, and distinct linear chains of
       filters are separated by semicolons. In our example, crop,vflip are in one linear chain,
       split and overlay are separately in another. The points where the linear chains join are
       labelled by names enclosed in square brackets. In the example, the split filter generates
       two outputs that are associated to the labels [main] and [tmp].

       The stream sent to the second output of split, labelled as [tmp], is processed through the
       crop filter, which crops away the lower half part of the video, and then vertically
       flipped. The overlay filter takes in input the first unchanged output of the split filter
       (which was labelled as [main]), and overlay on its lower half the output generated by the
       crop,vflip filterchain.

       Some filters take in input a list of parameters: they are specified after the filter name
       and an equal sign, and are separated from each other by a colon.

       There exist so-called source filters that do not have an audio/video input, and sink
       filters that will not have audio/video output.

GRAPH

       The graph2dot program included in the FFmpeg tools directory can be used to parse a
       filtergraph description and issue a corresponding textual representation in the dot
       language.

       Invoke the command:

               graph2dot -h

       to see how to use graph2dot.

       You can then pass the dot description to the dot program (from the graphviz suite of
       programs) and obtain a graphical representation of the filtergraph.

       For example the sequence of commands:

               echo <GRAPH_DESCRIPTION> | \
               tools/graph2dot -o graph.tmp && \
               dot -Tpng graph.tmp -o graph.png && \
               display graph.png

       can be used to create and display an image representing the graph described by the
       GRAPH_DESCRIPTION string. Note that this string must be a complete self-contained graph,
       with its inputs and outputs explicitly defined.  For example if your command line is of
       the form:

               ffmpeg -i infile -vf scale=640:360 outfile

       your GRAPH_DESCRIPTION string will need to be of the form:

               nullsrc,scale=640:360,nullsink

       you may also need to set the nullsrc parameters and add a format filter in order to
       simulate a specific input file.

FILTERGRAPH DESCRIPTION

       A filtergraph is a directed graph of connected filters. It can contain cycles, and there
       can be multiple links between a pair of filters. Each link has one input pad on one side
       connecting it to one filter from which it takes its input, and one output pad on the other
       side connecting it to one filter accepting its output.

       Each filter in a filtergraph is an instance of a filter class registered in the
       application, which defines the features and the number of input and output pads of the
       filter.

       A filter with no input pads is called a "source", and a filter with no output pads is
       called a "sink".

   Filtergraph syntax
       A filtergraph has a textual representation, which is recognized by the -filter/-vf/-af and
       -filter_complex options in ffmpeg and -vf/-af in ffplay, and by the
       "avfilter_graph_parse_ptr()" function defined in libavfilter/avfilter.h.

       A filterchain consists of a sequence of connected filters, each one connected to the
       previous one in the sequence. A filterchain is represented by a list of ","-separated
       filter descriptions.

       A filtergraph consists of a sequence of filterchains. A sequence of filterchains is
       represented by a list of ";"-separated filterchain descriptions.

       A filter is represented by a string of the form:
       [in_link_1]...[in_link_N]filter_name@id=arguments[out_link_1]...[out_link_M]

       filter_name is the name of the filter class of which the described filter is an instance
       of, and has to be the name of one of the filter classes registered in the program
       optionally followed by "@id".  The name of the filter class is optionally followed by a
       string "=arguments".

       arguments is a string which contains the parameters used to initialize the filter
       instance. It may have one of two forms:

       ·   A ':'-separated list of key=value pairs.

       ·   A ':'-separated list of value. In this case, the keys are assumed to be the option
           names in the order they are declared. E.g. the "fade" filter declares three options in
           this order -- type, start_frame and nb_frames. Then the parameter list in:0:30 means
           that the value in is assigned to the option type, 0 to start_frame and 30 to
           nb_frames.

       ·   A ':'-separated list of mixed direct value and long key=value pairs. The direct value
           must precede the key=value pairs, and follow the same constraints order of the
           previous point. The following key=value pairs can be set in any preferred order.

       If the option value itself is a list of items (e.g. the "format" filter takes a list of
       pixel formats), the items in the list are usually separated by |.

       The list of arguments can be quoted using the character ' as initial and ending mark, and
       the character \ for escaping the characters within the quoted text; otherwise the argument
       string is considered terminated when the next special character (belonging to the set
       []=;,) is encountered.

       The name and arguments of the filter are optionally preceded and followed by a list of
       link labels.  A link label allows one to name a link and associate it to a filter output
       or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter
       input pads, the following labels out_link_1 ... out_link_M, are associated to the output
       pads.

       When two link labels with the same name are found in the filtergraph, a link between the
       corresponding input and output pad is created.

       If an output pad is not labelled, it is linked by default to the first unlabelled input
       pad of the next filter in the filterchain.  For example in the filterchain

               nullsrc, split[L1], [L2]overlay, nullsink

       the split filter instance has two output pads, and the overlay filter instance two input
       pads. The first output pad of split is labelled "L1", the first input pad of overlay is
       labelled "L2", and the second output pad of split is linked to the second input pad of
       overlay, which are both unlabelled.

       In a filter description, if the input label of the first filter is not specified, "in" is
       assumed; if the output label of the last filter is not specified, "out" is assumed.

       In a complete filterchain all the unlabelled filter input and output pads must be
       connected. A filtergraph is considered valid if all the filter input and output pads of
       all the filterchains are connected.

       Libavfilter will automatically insert scale filters where format conversion is required.
       It is possible to specify swscale flags for those automatically inserted scalers by
       prepending "sws_flags=flags;" to the filtergraph description.

       Here is a BNF description of the filtergraph syntax:

               <NAME>             ::= sequence of alphanumeric characters and '_'
               <FILTER_NAME>      ::= <NAME>["@"<NAME>]
               <LINKLABEL>        ::= "[" <NAME> "]"
               <LINKLABELS>       ::= <LINKLABEL> [<LINKLABELS>]
               <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
               <FILTER>           ::= [<LINKLABELS>] <FILTER_NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
               <FILTERCHAIN>      ::= <FILTER> [,<FILTERCHAIN>]
               <FILTERGRAPH>      ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

   Notes on filtergraph escaping
       Filtergraph description composition entails several levels of escaping. See the "Quoting
       and escaping" section in the ffmpeg-utils(1) manual for more information about the
       employed escaping procedure.

       A first level escaping affects the content of each filter option value, which may contain
       the special character ":" used to separate values, or one of the escaping characters "\'".

       A second level escaping affects the whole filter description, which may contain the
       escaping characters "\'" or the special characters "[],;" used by the filtergraph
       description.

       Finally, when you specify a filtergraph on a shell commandline, you need to perform a
       third level escaping for the shell special characters contained within it.

       For example, consider the following string to be embedded in the drawtext filter
       description text value:

               this is a 'string': may contain one, or more, special characters

       This string contains the "'" special escaping character, and the ":" special character, so
       it needs to be escaped in this way:

               text=this is a \'string\'\: may contain one, or more, special characters

       A second level of escaping is required when embedding the filter description in a
       filtergraph description, in order to escape all the filtergraph special characters. Thus
       the example above becomes:

               drawtext=text=this is a \\\'string\\\'\\: may contain one\, or more\, special characters

       (note that in addition to the "\'" escaping special characters, also "," needs to be
       escaped).

       Finally an additional level of escaping is needed when writing the filtergraph description
       in a shell command, which depends on the escaping rules of the adopted shell. For example,
       assuming that "\" is special and needs to be escaped with another "\", the previous string
       will finally result in:

               -vf "drawtext=text=this is a \\\\\\'string\\\\\\'\\\\: may contain one\\, or more\\, special characters"

TIMELINE EDITING

       Some filters support a generic enable option. For the filters supporting timeline editing,
       this option can be set to an expression which is evaluated before sending a frame to the
       filter. If the evaluation is non-zero, the filter will be enabled, otherwise the frame
       will be sent unchanged to the next filter in the filtergraph.

       The expression accepts the following values:

       t   timestamp expressed in seconds, NAN if the input timestamp is unknown

       n   sequential number of the input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       w
       h   width and height of the input frame if video

       Additionally, these filters support an enable command that can be used to re-define the
       expression.

       Like any other filtering option, the enable option follows the same rules.

       For example, to enable a blur filter (smartblur) from 10 seconds to 3 minutes, and a
       curves filter starting at 3 seconds:

               smartblur = enable='between(t,10,3*60)',
               curves    = enable='gte(t,3)' : preset=cross_process

       See "ffmpeg -filters" to view which filters have timeline support.

OPTIONS FOR FILTERS WITH SEVERAL INPUTS

       Some filters with several inputs support a common set of options.  These options can only
       be set by name, not with the short notation.

       eof_action
           The action to take when EOF is encountered on the secondary input; it accepts one of
           the following values:

           repeat
               Repeat the last frame (the default).

           endall
               End both streams.

           pass
               Pass the main input through.

       shortest
           If set to 1, force the output to terminate when the shortest input terminates. Default
           value is 0.

       repeatlast
           If set to 1, force the filter to extend the last frame of secondary streams until the
           end of the primary stream. A value of 0 disables this behavior.  Default value is 1.

AUDIO FILTERS

       When you configure your FFmpeg build, you can disable any of the existing filters using
       "--disable-filters".  The configure output will show the audio filters included in your
       build.

       Below is a description of the currently available audio filters.

   acompressor
       A compressor is mainly used to reduce the dynamic range of a signal.  Especially modern
       music is mostly compressed at a high ratio to improve the overall loudness. It's done to
       get the highest attention of a listener, "fatten" the sound and bring more "power" to the
       track.  If a signal is compressed too much it may sound dull or "dead" afterwards or it
       may start to "pump" (which could be a powerful effect but can also destroy a track
       completely).  The right compression is the key to reach a professional sound and is the
       high art of mixing and mastering. Because of its complex settings it may take a long time
       to get the right feeling for this kind of effect.

       Compression is done by detecting the volume above a chosen level "threshold" and dividing
       it by the factor set with "ratio".  So if you set the threshold to -12dB and your signal
       reaches -6dB a ratio of 2:1 will result in a signal at -9dB. Because an exact manipulation
       of the signal would cause distortion of the waveform the reduction can be levelled over
       the time. This is done by setting "Attack" and "Release".  "attack" determines how long
       the signal has to rise above the threshold before any reduction will occur and "release"
       sets the time the signal has to fall below the threshold to reduce the reduction again.
       Shorter signals than the chosen attack time will be left untouched.  The overall reduction
       of the signal can be made up afterwards with the "makeup" setting. So compressing the
       peaks of a signal about 6dB and raising the makeup to this level results in a signal twice
       as loud than the source. To gain a softer entry in the compression the "knee" flattens the
       hard edge at the threshold in the range of the chosen decibels.

       The filter accepts the following options:

       level_in
           Set input gain. Default is 1. Range is between 0.015625 and 64.

       threshold
           If a signal of stream rises above this level it will affect the gain reduction.  By
           default it is 0.125. Range is between 0.00097563 and 1.

       ratio
           Set a ratio by which the signal is reduced. 1:2 means that if the level rose 4dB above
           the threshold, it will be only 2dB above after the reduction.  Default is 2. Range is
           between 1 and 20.

       attack
           Amount of milliseconds the signal has to rise above the threshold before gain
           reduction starts. Default is 20. Range is between 0.01 and 2000.

       release
           Amount of milliseconds the signal has to fall below the threshold before reduction is
           decreased again. Default is 250. Range is between 0.01 and 9000.

       makeup
           Set the amount by how much signal will be amplified after processing.  Default is 1.
           Range is from 1 to 64.

       knee
           Curve the sharp knee around the threshold to enter gain reduction more softly.
           Default is 2.82843. Range is between 1 and 8.

       link
           Choose if the "average" level between all channels of input stream or the
           louder("maximum") channel of input stream affects the reduction. Default is "average".

       detection
           Should the exact signal be taken in case of "peak" or an RMS one in case of "rms".
           Default is "rms" which is mostly smoother.

       mix How much to use compressed signal in output. Default is 1.  Range is between 0 and 1.

   acontrast
       Simple audio dynamic range commpression/expansion filter.

       The filter accepts the following options:

       contrast
           Set contrast. Default is 33. Allowed range is between 0 and 100.

   acopy
       Copy the input audio source unchanged to the output. This is mainly useful for testing
       purposes.

   acrossfade
       Apply cross fade from one input audio stream to another input audio stream.  The cross
       fade is applied for specified duration near the end of first stream.

       The filter accepts the following options:

       nb_samples, ns
           Specify the number of samples for which the cross fade effect has to last.  At the end
           of the cross fade effect the first input audio will be completely silent. Default is
           44100.

       duration, d
           Specify the duration of the cross fade effect. See the Time duration section in the
           ffmpeg-utils(1) manual for the accepted syntax.  By default the duration is determined
           by nb_samples.  If set this option is used instead of nb_samples.

       overlap, o
           Should first stream end overlap with second stream start. Default is enabled.

       curve1
           Set curve for cross fade transition for first stream.

       curve2
           Set curve for cross fade transition for second stream.

           For description of available curve types see afade filter description.

       Examples

       ·   Cross fade from one input to another:

                   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:c1=exp:c2=exp output.flac

       ·   Cross fade from one input to another but without overlapping:

                   ffmpeg -i first.flac -i second.flac -filter_complex acrossfade=d=10:o=0:c1=exp:c2=exp output.flac

   acrossover
       Split audio stream into several bands.

       This filter splits audio stream into two or more frequency ranges.  Summing all streams
       back will give flat output.

       The filter accepts the following options:

       split
           Set split frequencies. Those must be positive and increasing.

       order
           Set filter order, can be 2nd, 4th or 8th.  Default is 4th.

   acrusher
       Reduce audio bit resolution.

       This filter is bit crusher with enhanced functionality. A bit crusher is used to audibly
       reduce number of bits an audio signal is sampled with. This doesn't change the bit depth
       at all, it just produces the effect. Material reduced in bit depth sounds more harsh and
       "digital".  This filter is able to even round to continuous values instead of discrete bit
       depths.  Additionally it has a D/C offset which results in different crushing of the lower
       and the upper half of the signal.  An Anti-Aliasing setting is able to produce "softer"
       crushing sounds.

       Another feature of this filter is the logarithmic mode.  This setting switches from linear
       distances between bits to logarithmic ones.  The result is a much more "natural" sounding
       crusher which doesn't gate low signals for example. The human ear has a logarithmic
       perception, so this kind of crushing is much more pleasant.  Logarithmic crushing is also
       able to get anti-aliased.

       The filter accepts the following options:

       level_in
           Set level in.

       level_out
           Set level out.

       bits
           Set bit reduction.

       mix Set mixing amount.

       mode
           Can be linear: "lin" or logarithmic: "log".

       dc  Set DC.

       aa  Set anti-aliasing.

       samples
           Set sample reduction.

       lfo Enable LFO. By default disabled.

       lforange
           Set LFO range.

       lforate
           Set LFO rate.

   acue
       Delay audio filtering until a given wallclock timestamp. See the cue filter.

   adeclick
       Remove impulsive noise from input audio.

       Samples detected as impulsive noise are replaced by interpolated samples using
       autoregressive modelling.

       w   Set window size, in milliseconds. Allowed range is from 10 to 100. Default value is 55
           milliseconds.  This sets size of window which will be processed at once.

       o   Set window overlap, in percentage of window size. Allowed range is from 50 to 95.
           Default value is 75 percent.  Setting this to a very high value increases impulsive
           noise removal but makes whole process much slower.

       a   Set autoregression order, in percentage of window size. Allowed range is from 0 to 25.
           Default value is 2 percent. This option also controls quality of interpolated samples
           using neighbour good samples.

       t   Set threshold value. Allowed range is from 1 to 100.  Default value is 2.  This
           controls the strength of impulsive noise which is going to be removed.  The lower
           value, the more samples will be detected as impulsive noise.

       b   Set burst fusion, in percentage of window size. Allowed range is 0 to 10. Default
           value is 2.  If any two samples deteced as noise are spaced less than this value then
           any sample inbetween those two samples will be also detected as noise.

       m   Set overlap method.

           It accepts the following values:

           a   Select overlap-add method. Even not interpolated samples are slightly changed with
               this method.

           s   Select overlap-save method. Not interpolated samples remain unchanged.

           Default value is "a".

   adeclip
       Remove clipped samples from input audio.

       Samples detected as clipped are replaced by interpolated samples using autoregressive
       modelling.

       w   Set window size, in milliseconds. Allowed range is from 10 to 100.  Default value is
           55 milliseconds.  This sets size of window which will be processed at once.

       o   Set window overlap, in percentage of window size. Allowed range is from 50 to 95.
           Default value is 75 percent.

       a   Set autoregression order, in percentage of window size. Allowed range is from 0 to 25.
           Default value is 8 percent. This option also controls quality of interpolated samples
           using neighbour good samples.

       t   Set threshold value. Allowed range is from 1 to 100.  Default value is 10. Higher
           values make clip detection less aggressive.

       n   Set size of histogram used to detect clips. Allowed range is from 100 to 9999.
           Default value is 1000. Higher values make clip detection less aggressive.

       m   Set overlap method.

           It accepts the following values:

           a   Select overlap-add method. Even not interpolated samples are slightly changed with
               this method.

           s   Select overlap-save method. Not interpolated samples remain unchanged.

           Default value is "a".

   adelay
       Delay one or more audio channels.

       Samples in delayed channel are filled with silence.

       The filter accepts the following option:

       delays
           Set list of delays in milliseconds for each channel separated by '|'.  Unused delays
           will be silently ignored. If number of given delays is smaller than number of channels
           all remaining channels will not be delayed.  If you want to delay exact number of
           samples, append 'S' to number.

       Examples

       ·   Delay first channel by 1.5 seconds, the third channel by 0.5 seconds and leave the
           second channel (and any other channels that may be present) unchanged.

                   adelay=1500|0|500

       ·   Delay second channel by 500 samples, the third channel by 700 samples and leave the
           first channel (and any other channels that may be present) unchanged.

                   adelay=0|500S|700S

   aderivative, aintegral
       Compute derivative/integral of audio stream.

       Applying both filters one after another produces original audio.

   aecho
       Apply echoing to the input audio.

       Echoes are reflected sound and can occur naturally amongst mountains (and sometimes large
       buildings) when talking or shouting; digital echo effects emulate this behaviour and are
       often used to help fill out the sound of a single instrument or vocal. The time difference
       between the original signal and the reflection is the "delay", and the loudness of the
       reflected signal is the "decay".  Multiple echoes can have different delays and decays.

       A description of the accepted parameters follows.

       in_gain
           Set input gain of reflected signal. Default is 0.6.

       out_gain
           Set output gain of reflected signal. Default is 0.3.

       delays
           Set list of time intervals in milliseconds between original signal and reflections
           separated by '|'. Allowed range for each "delay" is "(0 - 90000.0]".  Default is 1000.

       decays
           Set list of loudness of reflected signals separated by '|'.  Allowed range for each
           "decay" is "(0 - 1.0]".  Default is 0.5.

       Examples

       ·   Make it sound as if there are twice as many instruments as are actually playing:

                   aecho=0.8:0.88:60:0.4

       ·   If delay is very short, then it sound like a (metallic) robot playing music:

                   aecho=0.8:0.88:6:0.4

       ·   A longer delay will sound like an open air concert in the mountains:

                   aecho=0.8:0.9:1000:0.3

       ·   Same as above but with one more mountain:

                   aecho=0.8:0.9:1000|1800:0.3|0.25

   aemphasis
       Audio emphasis filter creates or restores material directly taken from LPs or emphased CDs
       with different filter curves. E.g. to store music on vinyl the signal has to be altered by
       a filter first to even out the disadvantages of this recording medium.  Once the material
       is played back the inverse filter has to be applied to restore the distortion of the
       frequency response.

       The filter accepts the following options:

       level_in
           Set input gain.

       level_out
           Set output gain.

       mode
           Set filter mode. For restoring material use "reproduction" mode, otherwise use
           "production" mode. Default is "reproduction" mode.

       type
           Set filter type. Selects medium. Can be one of the following:

           col select Columbia.

           emi select EMI.

           bsi select BSI (78RPM).

           riaa
               select RIAA.

           cd  select Compact Disc (CD).

           50fm
               select 50Xs (FM).

           75fm
               select 75Xs (FM).

           50kf
               select 50Xs (FM-KF).

           75kf
               select 75Xs (FM-KF).

   aeval
       Modify an audio signal according to the specified expressions.

       This filter accepts one or more expressions (one for each channel), which are evaluated
       and used to modify a corresponding audio signal.

       It accepts the following parameters:

       exprs
           Set the '|'-separated expressions list for each separate channel. If the number of
           input channels is greater than the number of expressions, the last specified
           expression is used for the remaining output channels.

       channel_layout, c
           Set output channel layout. If not specified, the channel layout is specified by the
           number of expressions. If set to same, it will use by default the same input channel
           layout.

       Each expression in exprs can contain the following constants and functions:

       ch  channel number of the current expression

       n   number of the evaluated sample, starting from 0

       s   sample rate

       t   time of the evaluated sample expressed in seconds

       nb_in_channels
       nb_out_channels
           input and output number of channels

       val(CH)
           the value of input channel with number CH

       Note: this filter is slow. For faster processing you should use a dedicated filter.

       Examples

       ·   Half volume:

                   aeval=val(ch)/2:c=same

       ·   Invert phase of the second channel:

                   aeval=val(0)|-val(1)

   afade
       Apply fade-in/out effect to input audio.

       A description of the accepted parameters follows.

       type, t
           Specify the effect type, can be either "in" for fade-in, or "out" for a fade-out
           effect. Default is "in".

       start_sample, ss
           Specify the number of the start sample for starting to apply the fade effect. Default
           is 0.

       nb_samples, ns
           Specify the number of samples for which the fade effect has to last. At the end of the
           fade-in effect the output audio will have the same volume as the input audio, at the
           end of the fade-out transition the output audio will be silence. Default is 44100.

       start_time, st
           Specify the start time of the fade effect. Default is 0.  The value must be specified
           as a time duration; see the Time duration section in the ffmpeg-utils(1) manual for
           the accepted syntax.  If set this option is used instead of start_sample.

       duration, d
           Specify the duration of the fade effect. See the Time duration section in the
           ffmpeg-utils(1) manual for the accepted syntax.  At the end of the fade-in effect the
           output audio will have the same volume as the input audio, at the end of the fade-out
           transition the output audio will be silence.  By default the duration is determined by
           nb_samples.  If set this option is used instead of nb_samples.

       curve
           Set curve for fade transition.

           It accepts the following values:

           tri select triangular, linear slope (default)

           qsin
               select quarter of sine wave

           hsin
               select half of sine wave

           esin
               select exponential sine wave

           log select logarithmic

           ipar
               select inverted parabola

           qua select quadratic

           cub select cubic

           squ select square root

           cbr select cubic root

           par select parabola

           exp select exponential

           iqsin
               select inverted quarter of sine wave

           ihsin
               select inverted half of sine wave

           dese
               select double-exponential seat

           desi
               select double-exponential sigmoid

           losi
               select logistic sigmoid

       Examples

       ·   Fade in first 15 seconds of audio:

                   afade=t=in:ss=0:d=15

       ·   Fade out last 25 seconds of a 900 seconds audio:

                   afade=t=out:st=875:d=25

   afftdn
       Denoise audio samples with FFT.

       A description of the accepted parameters follows.

       nr  Set the noise reduction in dB, allowed range is 0.01 to 97.  Default value is 12 dB.

       nf  Set the noise floor in dB, allowed range is -80 to -20.  Default value is -50 dB.

       nt  Set the noise type.

           It accepts the following values:

           w   Select white noise.

           v   Select vinyl noise.

           s   Select shellac noise.

           c   Select custom noise, defined in "bn" option.

               Default value is white noise.

       bn  Set custom band noise for every one of 15 bands.  Bands are separated by ' ' or '|'.

       rf  Set the residual floor in dB, allowed range is -80 to -20.  Default value is -38 dB.

       tn  Enable noise tracking. By default is disabled.  With this enabled, noise floor is
           automatically adjusted.

       tr  Enable residual tracking. By default is disabled.

       om  Set the output mode.

           It accepts the following values:

           i   Pass input unchanged.

           o   Pass noise filtered out.

           n   Pass only noise.

               Default value is o.

       Commands

       This filter supports the following commands:

       sample_noise, sn
           Start or stop measuring noise profile.  Syntax for the command is : "start" or "stop"
           string.  After measuring noise profile is stopped it will be automatically applied in
           filtering.

       noise_reduction, nr
           Change noise reduction. Argument is single float number.  Syntax for the command is :
           "noise_reduction"

       noise_floor, nf
           Change noise floor. Argument is single float number.  Syntax for the command is :
           "noise_floor"

       output_mode, om
           Change output mode operation.  Syntax for the command is : "i", "o" or "n" string.

   afftfilt
       Apply arbitrary expressions to samples in frequency domain.

       real
           Set frequency domain real expression for each separate channel separated by '|'.
           Default is "1".  If the number of input channels is greater than the number of
           expressions, the last specified expression is used for the remaining output channels.

       imag
           Set frequency domain imaginary expression for each separate channel separated by '|'.
           If not set, real option is used.

           Each expression in real and imag can contain the following constants:

           sr  sample rate

           b   current frequency bin number

           nb  number of available bins

           ch  channel number of the current expression

           chs number of channels

           pts current frame pts

       win_size
           Set window size.

           It accepts the following values:

           w16
           w32
           w64
           w128
           w256
           w512
           w1024
           w2048
           w4096
           w8192
           w16384
           w32768
           w65536

           Default is "w4096"

       win_func
           Set window function. Default is "hann".

       overlap
           Set window overlap. If set to 1, the recommended overlap for selected window function
           will be picked. Default is 0.75.

       Examples

       ·   Leave almost only low frequencies in audio:

                   afftfilt="1-clip((b/nb)*b,0,1)"

   afir
       Apply an arbitrary Frequency Impulse Response filter.

       This filter is designed for applying long FIR filters, up to 60 seconds long.

       It can be used as component for digital crossover filters, room equalization, cross talk
       cancellation, wavefield synthesis, auralization, ambiophonics and ambisonics.

       This filter uses second stream as FIR coefficients.  If second stream holds single
       channel, it will be used for all input channels in first stream, otherwise number of
       channels in second stream must be same as number of channels in first stream.

       It accepts the following parameters:

       dry Set dry gain. This sets input gain.

       wet Set wet gain. This sets final output gain.

       length
           Set Impulse Response filter length. Default is 1, which means whole IR is processed.

       gtype
           Enable applying gain measured from power of IR.

           Set which approach to use for auto gain measurement.

           none
               Do not apply any gain.

           peak
               select peak gain, very conservative approach. This is default value.

           dc  select DC gain, limited application.

           gn  select gain to noise approach, this is most popular one.

       irgain
           Set gain to be applied to IR coefficients before filtering.  Allowed range is 0 to 1.
           This gain is applied after any gain applied with gtype option.

       irfmt
           Set format of IR stream. Can be "mono" or "input".  Default is "input".

       maxir
           Set max allowed Impulse Response filter duration in seconds. Default is 30 seconds.
           Allowed range is 0.1 to 60 seconds.

       response
           Show IR frequency reponse, magnitude(magenta) and phase(green) and group delay(yellow)
           in additional video stream.  By default it is disabled.

       channel
           Set for which IR channel to display frequency response. By default is first channel
           displayed. This option is used only when response is enabled.

       size
           Set video stream size. This option is used only when response is enabled.

       Examples

       ·   Apply reverb to stream using mono IR file as second input, complete command using
           ffmpeg:

                   ffmpeg -i input.wav -i middle_tunnel_1way_mono.wav -lavfi afir output.wav

   aformat
       Set output format constraints for the input audio. The framework will negotiate the most
       appropriate format to minimize conversions.

       It accepts the following parameters:

       sample_fmts
           A '|'-separated list of requested sample formats.

       sample_rates
           A '|'-separated list of requested sample rates.

       channel_layouts
           A '|'-separated list of requested channel layouts.

           See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

       If a parameter is omitted, all values are allowed.

       Force the output to either unsigned 8-bit or signed 16-bit stereo

               aformat=sample_fmts=u8|s16:channel_layouts=stereo

   agate
       A gate is mainly used to reduce lower parts of a signal. This kind of signal processing
       reduces disturbing noise between useful signals.

       Gating is done by detecting the volume below a chosen level threshold and dividing it by
       the factor set with ratio. The bottom of the noise floor is set via range. Because an
       exact manipulation of the signal would cause distortion of the waveform the reduction can
       be levelled over time. This is done by setting attack and release.

       attack determines how long the signal has to fall below the threshold before any reduction
       will occur and release sets the time the signal has to rise above the threshold to reduce
       the reduction again.  Shorter signals than the chosen attack time will be left untouched.

       level_in
           Set input level before filtering.  Default is 1. Allowed range is from 0.015625 to 64.

       range
           Set the level of gain reduction when the signal is below the threshold.  Default is
           0.06125. Allowed range is from 0 to 1.

       threshold
           If a signal rises above this level the gain reduction is released.  Default is 0.125.
           Allowed range is from 0 to 1.

       ratio
           Set a ratio by which the signal is reduced.  Default is 2. Allowed range is from 1 to
           9000.

       attack
           Amount of milliseconds the signal has to rise above the threshold before gain
           reduction stops.  Default is 20 milliseconds. Allowed range is from 0.01 to 9000.

       release
           Amount of milliseconds the signal has to fall below the threshold before the reduction
           is increased again. Default is 250 milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
           Set amount of amplification of signal after processing.  Default is 1. Allowed range
           is from 1 to 64.

       knee
           Curve the sharp knee around the threshold to enter gain reduction more softly.
           Default is 2.828427125. Allowed range is from 1 to 8.

       detection
           Choose if exact signal should be taken for detection or an RMS like one.  Default is
           "rms". Can be "peak" or "rms".

       link
           Choose if the average level between all channels or the louder channel affects the
           reduction.  Default is "average". Can be "average" or "maximum".

   aiir
       Apply an arbitrary Infinite Impulse Response filter.

       It accepts the following parameters:

       z   Set numerator/zeros coefficients.

       p   Set denominator/poles coefficients.

       k   Set channels gains.

       dry_gain
           Set input gain.

       wet_gain
           Set output gain.

       f   Set coefficients format.

           tf  transfer function

           zp  Z-plane zeros/poles, cartesian (default)

           pr  Z-plane zeros/poles, polar radians

           pd  Z-plane zeros/poles, polar degrees

       r   Set kind of processing.  Can be "d" - direct or "s" - serial cascading. Defauls is
           "s".

       e   Set filtering precision.

           dbl double-precision floating-point (default)

           flt single-precision floating-point

           i32 32-bit integers

           i16 16-bit integers

       response
           Show IR frequency reponse, magnitude and phase in additional video stream.  By default
           it is disabled.

       channel
           Set for which IR channel to display frequency response. By default is first channel
           displayed. This option is used only when response is enabled.

       size
           Set video stream size. This option is used only when response is enabled.

       Coefficients in "tf" format are separated by spaces and are in ascending order.

       Coefficients in "zp" format are separated by spaces and order of coefficients doesn't
       matter. Coefficients in "zp" format are complex numbers with i imaginary unit.

       Different coefficients and gains can be provided for every channel, in such case use '|'
       to separate coefficients or gains. Last provided coefficients will be used for all
       remaining channels.

       Examples

       ·   Apply 2 pole elliptic notch at arround 5000Hz for 48000 Hz sample rate:

                   aiir=k=1:z=7.957584807809675810E-1 -2.575128568908332300 3.674839853930788710 -2.57512875289799137 7.957586296317130880E-1:p=1 -2.86950072432325953 3.63022088054647218 -2.28075678147272232 6.361362326477423500E-1:f=tf:r=d

       ·   Same as above but in "zp" format:

                   aiir=k=0.79575848078096756:z=0.80918701+0.58773007i 0.80918701-0.58773007i 0.80884700+0.58784055i 0.80884700-0.58784055i:p=0.63892345+0.59951235i 0.63892345-0.59951235i 0.79582691+0.44198673i 0.79582691-0.44198673i:f=zp:r=s

   alimiter
       The limiter prevents an input signal from rising over a desired threshold.  This limiter
       uses lookahead technology to prevent your signal from distorting.  It means that there is
       a small delay after the signal is processed. Keep in mind that the delay it produces is
       the attack time you set.

       The filter accepts the following options:

       level_in
           Set input gain. Default is 1.

       level_out
           Set output gain. Default is 1.

       limit
           Don't let signals above this level pass the limiter. Default is 1.

       attack
           The limiter will reach its attenuation level in this amount of time in milliseconds.
           Default is 5 milliseconds.

       release
           Come back from limiting to attenuation 1.0 in this amount of milliseconds.  Default is
           50 milliseconds.

       asc When gain reduction is always needed ASC takes care of releasing to an average
           reduction level rather than reaching a reduction of 0 in the release time.

       asc_level
           Select how much the release time is affected by ASC, 0 means nearly no changes in
           release time while 1 produces higher release times.

       level
           Auto level output signal. Default is enabled.  This normalizes audio back to 0dB if
           enabled.

       Depending on picked setting it is recommended to upsample input 2x or 4x times with
       aresample before applying this filter.

   allpass
       Apply a two-pole all-pass filter with central frequency (in Hz) frequency, and filter-
       width width.  An all-pass filter changes the audio's frequency to phase relationship
       without changing its frequency to amplitude relationship.

       The filter accepts the following options:

       frequency, f
           Set frequency in Hz.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change allpass frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change allpass width_type.  Syntax for the command is : "width_type"

       width, w
           Change allpass width.  Syntax for the command is : "width"

   aloop
       Loop audio samples.

       The filter accepts the following options:

       loop
           Set the number of loops. Setting this value to -1 will result in infinite loops.
           Default is 0.

       size
           Set maximal number of samples. Default is 0.

       start
           Set first sample of loop. Default is 0.

   amerge
       Merge two or more audio streams into a single multi-channel stream.

       The filter accepts the following options:

       inputs
           Set the number of inputs. Default is 2.

       If the channel layouts of the inputs are disjoint, and therefore compatible, the channel
       layout of the output will be set accordingly and the channels will be reordered as
       necessary. If the channel layouts of the inputs are not disjoint, the output will have all
       the channels of the first input then all the channels of the second input, in that order,
       and the channel layout of the output will be the default value corresponding to the total
       number of channels.

       For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR,
       then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3,
       b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second
       input).

       On the other hand, if both input are in stereo, the output channels will be in the default
       order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or
       may not be the expected value.

       All inputs must have the same sample rate, and format.

       If inputs do not have the same duration, the output will stop with the shortest.

       Examples

       ·   Merge two mono files into a stereo stream:

                   amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge

       ·   Multiple merges assuming 1 video stream and 6 audio streams in input.mkv:

                   ffmpeg -i input.mkv -filter_complex "[0:1][0:2][0:3][0:4][0:5][0:6] amerge=inputs=6" -c:a pcm_s16le output.mkv

   amix
       Mixes multiple audio inputs into a single output.

       Note that this filter only supports float samples (the amerge and pan audio filters
       support many formats). If the amix input has integer samples then aresample will be
       automatically inserted to perform the conversion to float samples.

       For example

               ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT

       will mix 3 input audio streams to a single output with the same duration as the first
       input and a dropout transition time of 3 seconds.

       It accepts the following parameters:

       inputs
           The number of inputs. If unspecified, it defaults to 2.

       duration
           How to determine the end-of-stream.

           longest
               The duration of the longest input. (default)

           shortest
               The duration of the shortest input.

           first
               The duration of the first input.

       dropout_transition
           The transition time, in seconds, for volume renormalization when an input stream ends.
           The default value is 2 seconds.

       weights
           Specify weight of each input audio stream as sequence.  Each weight is separated by
           space. By default all inputs have same weight.

   amultiply
       Multiply first audio stream with second audio stream and store result in output audio
       stream. Multiplication is done by multiplying each sample from first stream with sample at
       same position from second stream.

       With this element-wise multiplication one can create amplitude fades and amplitude
       modulations.

   anequalizer
       High-order parametric multiband equalizer for each channel.

       It accepts the following parameters:

       params
           This option string is in format: "cchn f=cf w=w g=g t=f | ..."  Each equalizer band is
           separated by '|'.

           chn Set channel number to which equalization will be applied.  If input doesn't have
               that channel the entry is ignored.

           f   Set central frequency for band.  If input doesn't have that frequency the entry is
               ignored.

           w   Set band width in hertz.

           g   Set band gain in dB.

           t   Set filter type for band, optional, can be:

               0   Butterworth, this is default.

               1   Chebyshev type 1.

               2   Chebyshev type 2.

       curves
           With this option activated frequency response of anequalizer is displayed in video
           stream.

       size
           Set video stream size. Only useful if curves option is activated.

       mgain
           Set max gain that will be displayed. Only useful if curves option is activated.
           Setting this to a reasonable value makes it possible to display gain which is derived
           from neighbour bands which are too close to each other and thus produce higher gain
           when both are activated.

       fscale
           Set frequency scale used to draw frequency response in video output.  Can be linear or
           logarithmic. Default is logarithmic.

       colors
           Set color for each channel curve which is going to be displayed in video stream.  This
           is list of color names separated by space or by '|'.  Unrecognised or missing colors
           will be replaced by white color.

       Examples

       ·   Lower gain by 10 of central frequency 200Hz and width 100 Hz for first 2 channels
           using Chebyshev type 1 filter:

                   anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1

       Commands

       This filter supports the following commands:

       change
           Alter existing filter parameters.  Syntax for the commands is :
           "fN|f=freq|w=width|g=gain"

           fN is existing filter number, starting from 0, if no such filter is available error is
           returned.  freq set new frequency parameter.  width set new width parameter in herz.
           gain set new gain parameter in dB.

           Full filter invocation with asendcmd may look like this: asendcmd=c='4.0 anequalizer
           change 0|f=200|w=50|g=1',anequalizer=...

   anull
       Pass the audio source unchanged to the output.

   apad
       Pad the end of an audio stream with silence.

       This can be used together with ffmpeg -shortest to extend audio streams to the same length
       as the video stream.

       A description of the accepted options follows.

       packet_size
           Set silence packet size. Default value is 4096.

       pad_len
           Set the number of samples of silence to add to the end. After the value is reached,
           the stream is terminated. This option is mutually exclusive with whole_len.

       whole_len
           Set the minimum total number of samples in the output audio stream. If the value is
           longer than the input audio length, silence is added to the end, until the value is
           reached. This option is mutually exclusive with pad_len.

       If neither the pad_len nor the whole_len option is set, the filter will add silence to the
       end of the input stream indefinitely.

       Examples

       ·   Add 1024 samples of silence to the end of the input:

                   apad=pad_len=1024

       ·   Make sure the audio output will contain at least 10000 samples, pad the input with
           silence if required:

                   apad=whole_len=10000

       ·   Use ffmpeg to pad the audio input with silence, so that the video stream will always
           result the shortest and will be converted until the end in the output file when using
           the shortest option:

                   ffmpeg -i VIDEO -i AUDIO -filter_complex "[1:0]apad" -shortest OUTPUT

   aphaser
       Add a phasing effect to the input audio.

       A phaser filter creates series of peaks and troughs in the frequency spectrum.  The
       position of the peaks and troughs are modulated so that they vary over time, creating a
       sweeping effect.

       A description of the accepted parameters follows.

       in_gain
           Set input gain. Default is 0.4.

       out_gain
           Set output gain. Default is 0.74

       delay
           Set delay in milliseconds. Default is 3.0.

       decay
           Set decay. Default is 0.4.

       speed
           Set modulation speed in Hz. Default is 0.5.

       type
           Set modulation type. Default is triangular.

           It accepts the following values:

           triangular, t
           sinusoidal, s

   apulsator
       Audio pulsator is something between an autopanner and a tremolo.  But it can produce funny
       stereo effects as well. Pulsator changes the volume of the left and right channel based on
       a LFO (low frequency oscillator) with different waveforms and shifted phases.  This filter
       have the ability to define an offset between left and right channel. An offset of 0 means
       that both LFO shapes match each other.  The left and right channel are altered equally - a
       conventional tremolo.  An offset of 50% means that the shape of the right channel is
       exactly shifted in phase (or moved backwards about half of the frequency) - pulsator acts
       as an autopanner. At 1 both curves match again. Every setting in between moves the phase
       shift gapless between all stages and produces some "bypassing" sounds with sine and
       triangle waveforms. The more you set the offset near 1 (starting from the 0.5) the faster
       the signal passes from the left to the right speaker.

       The filter accepts the following options:

       level_in
           Set input gain. By default it is 1. Range is [0.015625 - 64].

       level_out
           Set output gain. By default it is 1. Range is [0.015625 - 64].

       mode
           Set waveform shape the LFO will use. Can be one of: sine, triangle, square, sawup or
           sawdown. Default is sine.

       amount
           Set modulation. Define how much of original signal is affected by the LFO.

       offset_l
           Set left channel offset. Default is 0. Allowed range is [0 - 1].

       offset_r
           Set right channel offset. Default is 0.5. Allowed range is [0 - 1].

       width
           Set pulse width. Default is 1. Allowed range is [0 - 2].

       timing
           Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz.

       bpm Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing is set to
           bpm.

       ms  Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing is set to
           ms.

       hz  Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used if timing
           is set to hz.

   aresample
       Resample the input audio to the specified parameters, using the libswresample library. If
       none are specified then the filter will automatically convert between its input and
       output.

       This filter is also able to stretch/squeeze the audio data to make it match the timestamps
       or to inject silence / cut out audio to make it match the timestamps, do a combination of
       both or do neither.

       The filter accepts the syntax [sample_rate:]resampler_options, where sample_rate expresses
       a sample rate and resampler_options is a list of key=value pairs, separated by ":". See
       the "Resampler Options" section in the ffmpeg-resampler(1) manual for the complete list of
       supported options.

       Examples

       ·   Resample the input audio to 44100Hz:

                   aresample=44100

       ·   Stretch/squeeze samples to the given timestamps, with a maximum of 1000 samples per
           second compensation:

                   aresample=async=1000

   areverse
       Reverse an audio clip.

       Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

       Examples

       ·   Take the first 5 seconds of a clip, and reverse it.

                   atrim=end=5,areverse

   asetnsamples
       Set the number of samples per each output audio frame.

       The last output packet may contain a different number of samples, as the filter will flush
       all the remaining samples when the input audio signals its end.

       The filter accepts the following options:

       nb_out_samples, n
           Set the number of frames per each output audio frame. The number is intended as the
           number of samples per each channel.  Default value is 1024.

       pad, p
           If set to 1, the filter will pad the last audio frame with zeroes, so that the last
           frame will contain the same number of samples as the previous ones. Default value is
           1.

       For example, to set the number of per-frame samples to 1234 and disable padding for the
       last frame, use:

               asetnsamples=n=1234:p=0

   asetrate
       Set the sample rate without altering the PCM data.  This will result in a change of speed
       and pitch.

       The filter accepts the following options:

       sample_rate, r
           Set the output sample rate. Default is 44100 Hz.

   ashowinfo
       Show a line containing various information for each input audio frame.  The input audio is
       not modified.

       The shown line contains a sequence of key/value pairs of the form key:value.

       The following values are shown in the output:

       n   The (sequential) number of the input frame, starting from 0.

       pts The presentation timestamp of the input frame, in time base units; the time base
           depends on the filter input pad, and is usually 1/sample_rate.

       pts_time
           The presentation timestamp of the input frame in seconds.

       pos position of the frame in the input stream, -1 if this information in unavailable
           and/or meaningless (for example in case of synthetic audio)

       fmt The sample format.

       chlayout
           The channel layout.

       rate
           The sample rate for the audio frame.

       nb_samples
           The number of samples (per channel) in the frame.

       checksum
           The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio,
           the data is treated as if all the planes were concatenated.

       plane_checksums
           A list of Adler-32 checksums for each data plane.

   astats
       Display time domain statistical information about the audio channels.  Statistics are
       calculated and displayed for each audio channel and, where applicable, an overall figure
       is also given.

       It accepts the following option:

       length
           Short window length in seconds, used for peak and trough RMS measurement.  Default is
           0.05 (50 milliseconds). Allowed range is "[0.01 - 10]".

       metadata
           Set metadata injection. All the metadata keys are prefixed with "lavfi.astats.X",
           where "X" is channel number starting from 1 or string "Overall". Default is disabled.

           Available keys for each channel are: DC_offset Min_level Max_level Min_difference
           Max_difference Mean_difference RMS_difference Peak_level RMS_peak RMS_trough
           Crest_factor Flat_factor Peak_count Bit_depth Dynamic_range Zero_crossings
           Zero_crossings_rate

           and for Overall: DC_offset Min_level Max_level Min_difference Max_difference
           Mean_difference RMS_difference Peak_level RMS_level RMS_peak RMS_trough Flat_factor
           Peak_count Bit_depth Number_of_samples

           For example full key look like this "lavfi.astats.1.DC_offset" or this
           "lavfi.astats.Overall.Peak_count".

           For description what each key means read below.

       reset
           Set number of frame after which stats are going to be recalculated.  Default is
           disabled.

       A description of each shown parameter follows:

       DC offset
           Mean amplitude displacement from zero.

       Min level
           Minimal sample level.

       Max level
           Maximal sample level.

       Min difference
           Minimal difference between two consecutive samples.

       Max difference
           Maximal difference between two consecutive samples.

       Mean difference
           Mean difference between two consecutive samples.  The average of each difference
           between two consecutive samples.

       RMS difference
           Root Mean Square difference between two consecutive samples.

       Peak level dB
       RMS level dB
           Standard peak and RMS level measured in dBFS.

       RMS peak dB
       RMS trough dB
           Peak and trough values for RMS level measured over a short window.

       Crest factor
           Standard ratio of peak to RMS level (note: not in dB).

       Flat factor
           Flatness (i.e. consecutive samples with the same value) of the signal at its peak
           levels (i.e. either Min level or Max level).

       Peak count
           Number of occasions (not the number of samples) that the signal attained either Min
           level or Max level.

       Bit depth
           Overall bit depth of audio. Number of bits used for each sample.

       Dynamic range
           Measured dynamic range of audio in dB.

       Zero crossings
           Number of points where the waveform crosses the zero level axis.

       Zero crossings rate
           Rate of Zero crossings and number of audio samples.

   atempo
       Adjust audio tempo.

       The filter accepts exactly one parameter, the audio tempo. If not specified then the
       filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 100.0] range.

       Note that tempo greater than 2 will skip some samples rather than blend them in.  If for
       any reason this is a concern it is always possible to daisy-chain several instances of
       atempo to achieve the desired product tempo.

       Examples

       ·   Slow down audio to 80% tempo:

                   atempo=0.8

       ·   To speed up audio to 300% tempo:

                   atempo=3

       ·   To speed up audio to 300% tempo by daisy-chaining two atempo instances:

                   atempo=sqrt(3),atempo=sqrt(3)

   atrim
       Trim the input so that the output contains one continuous subpart of the input.

       It accepts the following parameters:

       start
           Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with
           the timestamp start will be the first sample in the output.

       end Specify time of the first audio sample that will be dropped, i.e. the audio sample
           immediately preceding the one with the timestamp end will be the last sample in the
           output.

       start_pts
           Same as start, except this option sets the start timestamp in samples instead of
           seconds.

       end_pts
           Same as end, except this option sets the end timestamp in samples instead of seconds.

       duration
           The maximum duration of the output in seconds.

       start_sample
           The number of the first sample that should be output.

       end_sample
           The number of the first sample that should be dropped.

       start, end, and duration are expressed as time duration specifications; see the Time
       duration section in the ffmpeg-utils(1) manual.

       Note that the first two sets of the start/end options and the duration option look at the
       frame timestamp, while the _sample options simply count the samples that pass through the
       filter. So start/end_pts and start/end_sample will give different results when the
       timestamps are wrong, inexact or do not start at zero. Also note that this filter does not
       modify the timestamps. If you wish to have the output timestamps start at zero, insert the
       asetpts filter after the atrim filter.

       If multiple start or end options are set, this filter tries to be greedy and keep all
       samples that match at least one of the specified constraints. To keep only the part that
       matches all the constraints at once, chain multiple atrim filters.

       The defaults are such that all the input is kept. So it is possible to set e.g.  just the
       end values to keep everything before the specified time.

       Examples:

       ·   Drop everything except the second minute of input:

                   ffmpeg -i INPUT -af atrim=60:120

       ·   Keep only the first 1000 samples:

                   ffmpeg -i INPUT -af atrim=end_sample=1000

   bandpass
       Apply a two-pole Butterworth band-pass filter with central frequency frequency, and
       (3dB-point) band-width width.  The csg option selects a constant skirt gain (peak gain =
       Q) instead of the default: constant 0dB peak gain.  The filter roll off at 6dB per octave
       (20dB per decade).

       The filter accepts the following options:

       frequency, f
           Set the filter's central frequency. Default is 3000.

       csg Constant skirt gain if set to 1. Defaults to 0.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change bandpass frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change bandpass width_type.  Syntax for the command is : "width_type"

       width, w
           Change bandpass width.  Syntax for the command is : "width"

   bandreject
       Apply a two-pole Butterworth band-reject filter with central frequency frequency, and
       (3dB-point) band-width width.  The filter roll off at 6dB per octave (20dB per decade).

       The filter accepts the following options:

       frequency, f
           Set the filter's central frequency. Default is 3000.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change bandreject frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change bandreject width_type.  Syntax for the command is : "width_type"

       width, w
           Change bandreject width.  Syntax for the command is : "width"

   bass, lowshelf
       Boost or cut the bass (lower) frequencies of the audio using a two-pole shelving filter
       with a response similar to that of a standard hi-fi's tone-controls. This is also known as
       shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
           Give the gain at 0 Hz. Its useful range is about -20 (for a large cut) to +20 (for a
           large boost).  Beware of clipping when using a positive gain.

       frequency, f
           Set the filter's central frequency and so can be used to extend or reduce the
           frequency range to be boosted or cut.  The default value is 100 Hz.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Determine how steep is the filter's shelf transition.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change bass frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change bass width_type.  Syntax for the command is : "width_type"

       width, w
           Change bass width.  Syntax for the command is : "width"

       gain, g
           Change bass gain.  Syntax for the command is : "gain"

   biquad
       Apply a biquad IIR filter with the given coefficients.  Where b0, b1, b2 and a0, a1, a2
       are the numerator and denominator coefficients respectively.  and channels, c specify
       which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       a0
       a1
       a2
       b0
       b1
       b2  Change biquad parameter.  Syntax for the command is : "value"

   bs2b
       Bauer stereo to binaural transformation, which improves headphone listening of stereo
       audio records.

       To enable compilation of this filter you need to configure FFmpeg with "--enable-libbs2b".

       It accepts the following parameters:

       profile
           Pre-defined crossfeed level.

           default
               Default level (fcut=700, feed=50).

           cmoy
               Chu Moy circuit (fcut=700, feed=60).

           jmeier
               Jan Meier circuit (fcut=650, feed=95).

       fcut
           Cut frequency (in Hz).

       feed
           Feed level (in Hz).

   channelmap
       Remap input channels to new locations.

       It accepts the following parameters:

       map Map channels from input to output. The argument is a '|'-separated list of mappings,
           each in the "in_channel-out_channel" or in_channel form. in_channel can be either the
           name of the input channel (e.g. FL for front left) or its index in the input channel
           layout.  out_channel is the name of the output channel or its index in the output
           channel layout. If out_channel is not given then it is implicitly an index, starting
           with zero and increasing by one for each mapping.

       channel_layout
           The channel layout of the output stream.

       If no mapping is present, the filter will implicitly map input channels to output
       channels, preserving indices.

       Examples

       ·   For example, assuming a 5.1+downmix input MOV file,

                   ffmpeg -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wav

           will create an output WAV file tagged as stereo from the downmix channels of the
           input.

       ·   To fix a 5.1 WAV improperly encoded in AAC's native channel order

                   ffmpeg -i in.wav -filter 'channelmap=1|2|0|5|3|4:5.1' out.wav

   channelsplit
       Split each channel from an input audio stream into a separate output stream.

       It accepts the following parameters:

       channel_layout
           The channel layout of the input stream. The default is "stereo".

       channels
           A channel layout describing the channels to be extracted as separate output streams or
           "all" to extract each input channel as a separate stream. The default is "all".

           Choosing channels not present in channel layout in the input will result in an error.

       Examples

       ·   For example, assuming a stereo input MP3 file,

                   ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv

           will create an output Matroska file with two audio streams, one containing only the
           left channel and the other the right channel.

       ·   Split a 5.1 WAV file into per-channel files:

                   ffmpeg -i in.wav -filter_complex
                   'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]'
                   -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]'
                   front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]'
                   side_right.wav

       ·   Extract only LFE from a 5.1 WAV file:

                   ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1:channels=LFE[LFE]'
                   -map '[LFE]' lfe.wav

   chorus
       Add a chorus effect to the audio.

       Can make a single vocal sound like a chorus, but can also be applied to instrumentation.

       Chorus resembles an echo effect with a short delay, but whereas with echo the delay is
       constant, with chorus, it is varied using using sinusoidal or triangular modulation.  The
       modulation depth defines the range the modulated delay is played before or after the
       delay. Hence the delayed sound will sound slower or faster, that is the delayed sound
       tuned around the original one, like in a chorus where some vocals are slightly off key.

       It accepts the following parameters:

       in_gain
           Set input gain. Default is 0.4.

       out_gain
           Set output gain. Default is 0.4.

       delays
           Set delays. A typical delay is around 40ms to 60ms.

       decays
           Set decays.

       speeds
           Set speeds.

       depths
           Set depths.

       Examples

       ·   A single delay:

                   chorus=0.7:0.9:55:0.4:0.25:2

       ·   Two delays:

                   chorus=0.6:0.9:50|60:0.4|0.32:0.25|0.4:2|1.3

       ·   Fuller sounding chorus with three delays:

                   chorus=0.5:0.9:50|60|40:0.4|0.32|0.3:0.25|0.4|0.3:2|2.3|1.3

   compand
       Compress or expand the audio's dynamic range.

       It accepts the following parameters:

       attacks
       decays
           A list of times in seconds for each channel over which the instantaneous level of the
           input signal is averaged to determine its volume. attacks refers to increase of volume
           and decays refers to decrease of volume. For most situations, the attack time
           (response to the audio getting louder) should be shorter than the decay time, because
           the human ear is more sensitive to sudden loud audio than sudden soft audio. A typical
           value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds.  If
           specified number of attacks & decays is lower than number of channels, the last set
           attack/decay will be used for all remaining channels.

       points
           A list of points for the transfer function, specified in dB relative to the maximum
           possible signal amplitude. Each key points list must be defined using the following
           syntax: "x0/y0|x1/y1|x2/y2|...." or "x0/y0 x1/y1 x2/y2 ...."

           The input values must be in strictly increasing order but the transfer function does
           not have to be monotonically rising. The point "0/0" is assumed but may be overridden
           (by "0/out-dBn"). Typical values for the transfer function are "-70/-70|-60/-20|1/0".

       soft-knee
           Set the curve radius in dB for all joints. It defaults to 0.01.

       gain
           Set the additional gain in dB to be applied at all points on the transfer function.
           This allows for easy adjustment of the overall gain.  It defaults to 0.

       volume
           Set an initial volume, in dB, to be assumed for each channel when filtering starts.
           This permits the user to supply a nominal level initially, so that, for example, a
           very large gain is not applied to initial signal levels before the companding has
           begun to operate. A typical value for audio which is initially quiet is -90 dB. It
           defaults to 0.

       delay
           Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed
           before being fed to the volume adjuster. Specifying a delay approximately equal to the
           attack/decay times allows the filter to effectively operate in predictive rather than
           reactive mode. It defaults to 0.

       Examples

       ·   Make music with both quiet and loud passages suitable for listening to in a noisy
           environment:

                   compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2

           Another example for audio with whisper and explosion parts:

                   compand=0|0:1|1:-90/-900|-70/-70|-30/-9|0/-3:6:0:0:0

       ·   A noise gate for when the noise is at a lower level than the signal:

                   compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1

       ·   Here is another noise gate, this time for when the noise is at a higher level than the
           signal (making it, in some ways, similar to squelch):

                   compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1

       ·   2:1 compression starting at -6dB:

                   compand=points=-80/-80|-6/-6|0/-3.8|20/3.5

       ·   2:1 compression starting at -9dB:

                   compand=points=-80/-80|-9/-9|0/-5.3|20/2.9

       ·   2:1 compression starting at -12dB:

                   compand=points=-80/-80|-12/-12|0/-6.8|20/1.9

       ·   2:1 compression starting at -18dB:

                   compand=points=-80/-80|-18/-18|0/-9.8|20/0.7

       ·   3:1 compression starting at -15dB:

                   compand=points=-80/-80|-15/-15|0/-10.8|20/-5.2

       ·   Compressor/Gate:

                   compand=points=-80/-105|-62/-80|-15.4/-15.4|0/-12|20/-7.6

       ·   Expander:

                   compand=attacks=0:points=-80/-169|-54/-80|-49.5/-64.6|-41.1/-41.1|-25.8/-15|-10.8/-4.5|0/0|20/8.3

       ·   Hard limiter at -6dB:

                   compand=attacks=0:points=-80/-80|-6/-6|20/-6

       ·   Hard limiter at -12dB:

                   compand=attacks=0:points=-80/-80|-12/-12|20/-12

       ·   Hard noise gate at -35 dB:

                   compand=attacks=0:points=-80/-115|-35.1/-80|-35/-35|20/20

       ·   Soft limiter:

                   compand=attacks=0:points=-80/-80|-12.4/-12.4|-6/-8|0/-6.8|20/-2.8

   compensationdelay
       Compensation Delay Line is a metric based delay to compensate differing positions of
       microphones or speakers.

       For example, you have recorded guitar with two microphones placed in different location.
       Because the front of sound wave has fixed speed in normal conditions, the phasing of
       microphones can vary and depends on their location and interposition. The best sound mix
       can be achieved when these microphones are in phase (synchronized). Note that distance of
       ~30 cm between microphones makes one microphone to capture signal in antiphase to another
       microphone. That makes the final mix sounding moody.  This filter helps to solve phasing
       problems by adding different delays to each microphone track and make them synchronized.

       The best result can be reached when you take one track as base and synchronize other
       tracks one by one with it.  Remember that synchronization/delay tolerance depends on
       sample rate, too.  Higher sample rates will give more tolerance.

       It accepts the following parameters:

       mm  Set millimeters distance. This is compensation distance for fine tuning.  Default is
           0.

       cm  Set cm distance. This is compensation distance for tightening distance setup.  Default
           is 0.

       m   Set meters distance. This is compensation distance for hard distance setup.  Default
           is 0.

       dry Set dry amount. Amount of unprocessed (dry) signal.  Default is 0.

       wet Set wet amount. Amount of processed (wet) signal.  Default is 1.

       temp
           Set temperature degree in Celsius. This is the temperature of the environment.
           Default is 20.

   crossfeed
       Apply headphone crossfeed filter.

       Crossfeed is the process of blending the left and right channels of stereo audio
       recording.  It is mainly used to reduce extreme stereo separation of low frequencies.

       The intent is to produce more speaker like sound to the listener.

       The filter accepts the following options:

       strength
           Set strength of crossfeed. Default is 0.2. Allowed range is from 0 to 1.  This sets
           gain of low shelf filter for side part of stereo image.  Default is -6dB. Max allowed
           is -30db when strength is set to 1.

       range
           Set soundstage wideness. Default is 0.5. Allowed range is from 0 to 1.  This sets cut
           off frequency of low shelf filter. Default is cut off near 1550 Hz. With range set to
           1 cut off frequency is set to 2100 Hz.

       level_in
           Set input gain. Default is 0.9.

       level_out
           Set output gain. Default is 1.

   crystalizer
       Simple algorithm to expand audio dynamic range.

       The filter accepts the following options:

       i   Sets the intensity of effect (default: 2.0). Must be in range between 0.0 (unchanged
           sound) to 10.0 (maximum effect).

       c   Enable clipping. By default is enabled.

   dcshift
       Apply a DC shift to the audio.

       This can be useful to remove a DC offset (caused perhaps by a hardware problem in the
       recording chain) from the audio. The effect of a DC offset is reduced headroom and hence
       volume. The astats filter can be used to determine if a signal has a DC offset.

       shift
           Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift the
           audio.

       limitergain
           Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is used to
           prevent clipping.

   drmeter
       Measure audio dynamic range.

       DR values of 14 and higher is found in very dynamic material. DR of 8 to 13 is found in
       transition material. And anything less that 8 have very poor dynamics and is very
       compressed.

       The filter accepts the following options:

       length
           Set window length in seconds used to split audio into segments of equal length.
           Default is 3 seconds.

   dynaudnorm
       Dynamic Audio Normalizer.

       This filter applies a certain amount of gain to the input audio in order to bring its peak
       magnitude to a target level (e.g. 0 dBFS). However, in contrast to more "simple"
       normalization algorithms, the Dynamic Audio Normalizer *dynamically* re-adjusts the gain
       factor to the input audio.  This allows for applying extra gain to the "quiet" sections of
       the audio while avoiding distortions or clipping the "loud" sections. In other words: The
       Dynamic Audio Normalizer will "even out" the volume of quiet and loud sections, in the
       sense that the volume of each section is brought to the same target level. Note, however,
       that the Dynamic Audio Normalizer achieves this goal *without* applying "dynamic range
       compressing". It will retain 100% of the dynamic range *within* each section of the audio
       file.

       f   Set the frame length in milliseconds. In range from 10 to 8000 milliseconds.  Default
           is 500 milliseconds.  The Dynamic Audio Normalizer processes the input audio in small
           chunks, referred to as frames. This is required, because a peak magnitude has no
           meaning for just a single sample value. Instead, we need to determine the peak
           magnitude for a contiguous sequence of sample values. While a "standard" normalizer
           would simply use the peak magnitude of the complete file, the Dynamic Audio Normalizer
           determines the peak magnitude individually for each frame. The length of a frame is
           specified in milliseconds. By default, the Dynamic Audio Normalizer uses a frame
           length of 500 milliseconds, which has been found to give good results with most files.
           Note that the exact frame length, in number of samples, will be determined
           automatically, based on the sampling rate of the individual input audio file.

       g   Set the Gaussian filter window size. In range from 3 to 301, must be odd number.
           Default is 31.  Probably the most important parameter of the Dynamic Audio Normalizer
           is the "window size" of the Gaussian smoothing filter. The filter's window size is
           specified in frames, centered around the current frame. For the sake of simplicity,
           this must be an odd number. Consequently, the default value of 31 takes into account
           the current frame, as well as the 15 preceding frames and the 15 subsequent frames.
           Using a larger window results in a stronger smoothing effect and thus in less gain
           variation, i.e. slower gain adaptation. Conversely, using a smaller window results in
           a weaker smoothing effect and thus in more gain variation, i.e. faster gain
           adaptation.  In other words, the more you increase this value, the more the Dynamic
           Audio Normalizer will behave like a "traditional" normalization filter. On the
           contrary, the more you decrease this value, the more the Dynamic Audio Normalizer will
           behave like a dynamic range compressor.

       p   Set the target peak value. This specifies the highest permissible magnitude level for
           the normalized audio input. This filter will try to approach the target peak magnitude
           as closely as possible, but at the same time it also makes sure that the normalized
           signal will never exceed the peak magnitude.  A frame's maximum local gain factor is
           imposed directly by the target peak magnitude. The default value is 0.95 and thus
           leaves a headroom of 5%*.  It is not recommended to go above this value.

       m   Set the maximum gain factor. In range from 1.0 to 100.0. Default is 10.0.  The Dynamic
           Audio Normalizer determines the maximum possible (local) gain factor for each input
           frame, i.e. the maximum gain factor that does not result in clipping or distortion.
           The maximum gain factor is determined by the frame's highest magnitude sample.
           However, the Dynamic Audio Normalizer additionally bounds the frame's maximum gain
           factor by a predetermined (global) maximum gain factor. This is done in order to avoid
           excessive gain factors in "silent" or almost silent frames. By default, the maximum
           gain factor is 10.0, For most inputs the default value should be sufficient and it
           usually is not recommended to increase this value. Though, for input with an extremely
           low overall volume level, it may be necessary to allow even higher gain factors. Note,
           however, that the Dynamic Audio Normalizer does not simply apply a "hard" threshold
           (i.e. cut off values above the threshold).  Instead, a "sigmoid" threshold function
           will be applied. This way, the gain factors will smoothly approach the threshold
           value, but never exceed that value.

       r   Set the target RMS. In range from 0.0 to 1.0. Default is 0.0 - disabled.  By default,
           the Dynamic Audio Normalizer performs "peak" normalization.  This means that the
           maximum local gain factor for each frame is defined (only) by the frame's highest
           magnitude sample. This way, the samples can be amplified as much as possible without
           exceeding the maximum signal level, i.e. without clipping. Optionally, however, the
           Dynamic Audio Normalizer can also take into account the frame's root mean square,
           abbreviated RMS. In electrical engineering, the RMS is commonly used to determine the
           power of a time-varying signal. It is therefore considered that the RMS is a better
           approximation of the "perceived loudness" than just looking at the signal's peak
           magnitude. Consequently, by adjusting all frames to a constant RMS value, a uniform
           "perceived loudness" can be established. If a target RMS value has been specified, a
           frame's local gain factor is defined as the factor that would result in exactly that
           RMS value.  Note, however, that the maximum local gain factor is still restricted by
           the frame's highest magnitude sample, in order to prevent clipping.

       n   Enable channels coupling. By default is enabled.  By default, the Dynamic Audio
           Normalizer will amplify all channels by the same amount. This means the same gain
           factor will be applied to all channels, i.e.  the maximum possible gain factor is
           determined by the "loudest" channel.  However, in some recordings, it may happen that
           the volume of the different channels is uneven, e.g. one channel may be "quieter" than
           the other one(s).  In this case, this option can be used to disable the channel
           coupling. This way, the gain factor will be determined independently for each channel,
           depending only on the individual channel's highest magnitude sample. This allows for
           harmonizing the volume of the different channels.

       c   Enable DC bias correction. By default is disabled.  An audio signal (in the time
           domain) is a sequence of sample values.  In the Dynamic Audio Normalizer these sample
           values are represented in the -1.0 to 1.0 range, regardless of the original input
           format. Normally, the audio signal, or "waveform", should be centered around the zero
           point.  That means if we calculate the mean value of all samples in a file, or in a
           single frame, then the result should be 0.0 or at least very close to that value. If,
           however, there is a significant deviation of the mean value from 0.0, in either
           positive or negative direction, this is referred to as a DC bias or DC offset. Since a
           DC bias is clearly undesirable, the Dynamic Audio Normalizer provides optional DC bias
           correction.  With DC bias correction enabled, the Dynamic Audio Normalizer will
           determine the mean value, or "DC correction" offset, of each input frame and subtract
           that value from all of the frame's sample values which ensures those samples are
           centered around 0.0 again. Also, in order to avoid "gaps" at the frame boundaries, the
           DC correction offset values will be interpolated smoothly between neighbouring frames.

       b   Enable alternative boundary mode. By default is disabled.  The Dynamic Audio
           Normalizer takes into account a certain neighbourhood around each frame. This includes
           the preceding frames as well as the subsequent frames. However, for the "boundary"
           frames, located at the very beginning and at the very end of the audio file, not all
           neighbouring frames are available. In particular, for the first few frames in the
           audio file, the preceding frames are not known. And, similarly, for the last few
           frames in the audio file, the subsequent frames are not known. Thus, the question
           arises which gain factors should be assumed for the missing frames in the "boundary"
           region. The Dynamic Audio Normalizer implements two modes to deal with this situation.
           The default boundary mode assumes a gain factor of exactly 1.0 for the missing frames,
           resulting in a smooth "fade in" and "fade out" at the beginning and at the end of the
           input, respectively.

       s   Set the compress factor. In range from 0.0 to 30.0. Default is 0.0.  By default, the
           Dynamic Audio Normalizer does not apply "traditional" compression. This means that
           signal peaks will not be pruned and thus the full dynamic range will be retained
           within each local neighbourhood. However, in some cases it may be desirable to combine
           the Dynamic Audio Normalizer's normalization algorithm with a more "traditional"
           compression.  For this purpose, the Dynamic Audio Normalizer provides an optional
           compression (thresholding) function. If (and only if) the compression feature is
           enabled, all input frames will be processed by a soft knee thresholding function prior
           to the actual normalization process. Put simply, the thresholding function is going to
           prune all samples whose magnitude exceeds a certain threshold value.  However, the
           Dynamic Audio Normalizer does not simply apply a fixed threshold value. Instead, the
           threshold value will be adjusted for each individual frame.  In general, smaller
           parameters result in stronger compression, and vice versa.  Values below 3.0 are not
           recommended, because audible distortion may appear.

   earwax
       Make audio easier to listen to on headphones.

       This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when
       listened to on headphones the stereo image is moved from inside your head (standard for
       headphones) to outside and in front of the listener (standard for speakers).

       Ported from SoX.

   equalizer
       Apply a two-pole peaking equalisation (EQ) filter. With this filter, the signal-level at
       and around a selected frequency can be increased or decreased, whilst (unlike bandpass and
       bandreject filters) that at all other frequencies is unchanged.

       In order to produce complex equalisation curves, this filter can be given several times,
       each with a different central frequency.

       The filter accepts the following options:

       frequency, f
           Set the filter's central frequency in Hz.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.

       gain, g
           Set the required gain or attenuation in dB.  Beware of clipping when using a positive
           gain.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Examples

       ·   Attenuate 10 dB at 1000 Hz, with a bandwidth of 200 Hz:

                   equalizer=f=1000:t=h:width=200:g=-10

       ·   Apply 2 dB gain at 1000 Hz with Q 1 and attenuate 5 dB at 100 Hz with Q 2:

                   equalizer=f=1000:t=q:w=1:g=2,equalizer=f=100:t=q:w=2:g=-5

       Commands

       This filter supports the following commands:

       frequency, f
           Change equalizer frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change equalizer width_type.  Syntax for the command is : "width_type"

       width, w
           Change equalizer width.  Syntax for the command is : "width"

       gain, g
           Change equalizer gain.  Syntax for the command is : "gain"

   extrastereo
       Linearly increases the difference between left and right channels which adds some sort of
       "live" effect to playback.

       The filter accepts the following options:

       m   Sets the difference coefficient (default: 2.5). 0.0 means mono sound (average of both
           channels), with 1.0 sound will be unchanged, with -1.0 left and right channels will be
           swapped.

       c   Enable clipping. By default is enabled.

   firequalizer
       Apply FIR Equalization using arbitrary frequency response.

       The filter accepts the following option:

       gain
           Set gain curve equation (in dB). The expression can contain variables:

           f   the evaluated frequency

           sr  sample rate

           ch  channel number, set to 0 when multichannels evaluation is disabled

           chid
               channel id, see libavutil/channel_layout.h, set to the first channel id when
               multichannels evaluation is disabled

           chs number of channels

           chlayout
               channel_layout, see libavutil/channel_layout.h

           and functions:

           gain_interpolate(f)
               interpolate gain on frequency f based on gain_entry

           cubic_interpolate(f)
               same as gain_interpolate, but smoother

           This option is also available as command. Default is gain_interpolate(f).

       gain_entry
           Set gain entry for gain_interpolate function. The expression can contain functions:

           entry(f, g)
               store gain entry at frequency f with value g

           This option is also available as command.

       delay
           Set filter delay in seconds. Higher value means more accurate.  Default is 0.01.

       accuracy
           Set filter accuracy in Hz. Lower value means more accurate.  Default is 5.

       wfunc
           Set window function. Acceptable values are:

           rectangular
               rectangular window, useful when gain curve is already smooth

           hann
               hann window (default)

           hamming
               hamming window

           blackman
               blackman window

           nuttall3
               3-terms continuous 1st derivative nuttall window

           mnuttall3
               minimum 3-terms discontinuous nuttall window

           nuttall
               4-terms continuous 1st derivative nuttall window

           bnuttall
               minimum 4-terms discontinuous nuttall (blackman-nuttall) window

           bharris
               blackman-harris window

           tukey
               tukey window

       fixed
           If enabled, use fixed number of audio samples. This improves speed when filtering with
           large delay. Default is disabled.

       multi
           Enable multichannels evaluation on gain. Default is disabled.

       zero_phase
           Enable zero phase mode by subtracting timestamp to compensate delay.  Default is
           disabled.

       scale
           Set scale used by gain. Acceptable values are:

           linlin
               linear frequency, linear gain

           linlog
               linear frequency, logarithmic (in dB) gain (default)

           loglin
               logarithmic (in octave scale where 20 Hz is 0) frequency, linear gain

           loglog
               logarithmic frequency, logarithmic gain

       dumpfile
           Set file for dumping, suitable for gnuplot.

       dumpscale
           Set scale for dumpfile. Acceptable values are same with scale option.  Default is
           linlog.

       fft2
           Enable 2-channel convolution using complex FFT. This improves speed significantly.
           Default is disabled.

       min_phase
           Enable minimum phase impulse response. Default is disabled.

       Examples

       ·   lowpass at 1000 Hz:

                   firequalizer=gain='if(lt(f,1000), 0, -INF)'

       ·   lowpass at 1000 Hz with gain_entry:

                   firequalizer=gain_entry='entry(1000,0); entry(1001, -INF)'

       ·   custom equalization:

                   firequalizer=gain_entry='entry(100,0); entry(400, -4); entry(1000, -6); entry(2000, 0)'

       ·   higher delay with zero phase to compensate delay:

                   firequalizer=delay=0.1:fixed=on:zero_phase=on

       ·   lowpass on left channel, highpass on right channel:

                   firequalizer=gain='if(eq(chid,1), gain_interpolate(f), if(eq(chid,2), gain_interpolate(1e6+f), 0))'
                   :gain_entry='entry(1000, 0); entry(1001,-INF); entry(1e6+1000,0)':multi=on

   flanger
       Apply a flanging effect to the audio.

       The filter accepts the following options:

       delay
           Set base delay in milliseconds. Range from 0 to 30. Default value is 0.

       depth
           Set added sweep delay in milliseconds. Range from 0 to 10. Default value is 2.

       regen
           Set percentage regeneration (delayed signal feedback). Range from -95 to 95.  Default
           value is 0.

       width
           Set percentage of delayed signal mixed with original. Range from 0 to 100.  Default
           value is 71.

       speed
           Set sweeps per second (Hz). Range from 0.1 to 10. Default value is 0.5.

       shape
           Set swept wave shape, can be triangular or sinusoidal.  Default value is sinusoidal.

       phase
           Set swept wave percentage-shift for multi channel. Range from 0 to 100.  Default value
           is 25.

       interp
           Set delay-line interpolation, linear or quadratic.  Default is linear.

   haas
       Apply Haas effect to audio.

       Note that this makes most sense to apply on mono signals.  With this filter applied to
       mono signals it give some directionality and stretches its stereo image.

       The filter accepts the following options:

       level_in
           Set input level. By default is 1, or 0dB

       level_out
           Set output level. By default is 1, or 0dB.

       side_gain
           Set gain applied to side part of signal. By default is 1.

       middle_source
           Set kind of middle source. Can be one of the following:

           left
               Pick left channel.

           right
               Pick right channel.

           mid Pick middle part signal of stereo image.

           side
               Pick side part signal of stereo image.

       middle_phase
           Change middle phase. By default is disabled.

       left_delay
           Set left channel delay. By default is 2.05 milliseconds.

       left_balance
           Set left channel balance. By default is -1.

       left_gain
           Set left channel gain. By default is 1.

       left_phase
           Change left phase. By default is disabled.

       right_delay
           Set right channel delay. By defaults is 2.12 milliseconds.

       right_balance
           Set right channel balance. By default is 1.

       right_gain
           Set right channel gain. By default is 1.

       right_phase
           Change right phase. By default is enabled.

   hdcd
       Decodes High Definition Compatible Digital (HDCD) data. A 16-bit PCM stream with embedded
       HDCD codes is expanded into a 20-bit PCM stream.

       The filter supports the Peak Extend and Low-level Gain Adjustment features of HDCD, and
       detects the Transient Filter flag.

               ffmpeg -i HDCD16.flac -af hdcd OUT24.flac

       When using the filter with wav, note the default encoding for wav is 16-bit, so the
       resulting 20-bit stream will be truncated back to 16-bit. Use something like -acodec
       pcm_s24le after the filter to get 24-bit PCM output.

               ffmpeg -i HDCD16.wav -af hdcd OUT16.wav
               ffmpeg -i HDCD16.wav -af hdcd -c:a pcm_s24le OUT24.wav

       The filter accepts the following options:

       disable_autoconvert
           Disable any automatic format conversion or resampling in the filter graph.

       process_stereo
           Process the stereo channels together. If target_gain does not match between channels,
           consider it invalid and use the last valid target_gain.

       cdt_ms
           Set the code detect timer period in ms.

       force_pe
           Always extend peaks above -3dBFS even if PE isn't signaled.

       analyze_mode
           Replace audio with a solid tone and adjust the amplitude to signal some specific
           aspect of the decoding process. The output file can be loaded in an audio editor
           alongside the original to aid analysis.

           "analyze_mode=pe:force_pe=true" can be used to see all samples above the PE level.

           Modes are:

           0, off
               Disabled

           1, lle
               Gain adjustment level at each sample

           2, pe
               Samples where peak extend occurs

           3, cdt
               Samples where the code detect timer is active

           4, tgm
               Samples where the target gain does not match between channels

   headphone
       Apply head-related transfer functions (HRTFs) to create virtual loudspeakers around the
       user for binaural listening via headphones.  The HRIRs are provided via additional
       streams, for each channel one stereo input stream is needed.

       The filter accepts the following options:

       map Set mapping of input streams for convolution.  The argument is a '|'-separated list of
           channel names in order as they are given as additional stream inputs for filter.  This
           also specify number of input streams. Number of input streams must be not less than
           number of channels in first stream plus one.

       gain
           Set gain applied to audio. Value is in dB. Default is 0.

       type
           Set processing type. Can be time or freq. time is processing audio in time domain
           which is slow.  freq is processing audio in frequency domain which is fast.  Default
           is freq.

       lfe Set custom gain for LFE channels. Value is in dB. Default is 0.

       size
           Set size of frame in number of samples which will be processed at once.  Default value
           is 1024. Allowed range is from 1024 to 96000.

       hrir
           Set format of hrir stream.  Default value is stereo. Alternative value is multich.  If
           value is set to stereo, number of additional streams should be greater or equal to
           number of input channels in first input stream.  Also each additional stream should
           have stereo number of channels.  If value is set to multich, number of additional
           streams should be exactly one. Also number of input channels of additional stream
           should be equal or greater than twice number of channels of first input stream.

       Examples

       ·   Full example using wav files as coefficients with amovie filters for 7.1 downmix, each
           amovie filter use stereo file with IR coefficients as input.  The files give
           coefficients for each position of virtual loudspeaker:

                   ffmpeg -i input.wav -lavfi-complex "amovie=azi_270_ele_0_DFC.wav[sr],amovie=azi_90_ele_0_DFC.wav[sl],amovie=azi_225_ele_0_DFC.wav[br],amovie=azi_135_ele_0_DFC.wav[bl],amovie=azi_0_ele_0_DFC.wav,asplit[fc][lfe],amovie=azi_35_ele_0_DFC.wav[fl],amovie=azi_325_ele_0_DFC.wav[fr],[a:0][fl][fr][fc][lfe][bl][br][sl][sr]headphone=FL|FR|FC|LFE|BL|BR|SL|SR"
                   output.wav

       ·   Full example using wav files as coefficients with amovie filters for 7.1 downmix, but
           now in multich hrir format.

                   ffmpeg -i input.wav -lavfi-complex "amovie=minp.wav[hrirs],[a:0][hrirs]headphone=map=FL|FR|FC|LFE|BL|BR|SL|SR:hrir=multich"
                   output.wav

   highpass
       Apply a high-pass filter with 3dB point frequency.  The filter can be either single-pole,
       or double-pole (the default).  The filter roll off at 6dB per pole per octave (20dB per
       pole per decade).

       The filter accepts the following options:

       frequency, f
           Set frequency in Hz. Default is 3000.

       poles, p
           Set number of poles. Default is 2.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.  Applies only to double-pole
           filter.  The default is 0.707q and gives a Butterworth response.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change highpass frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change highpass width_type.  Syntax for the command is : "width_type"

       width, w
           Change highpass width.  Syntax for the command is : "width"

   join
       Join multiple input streams into one multi-channel stream.

       It accepts the following parameters:

       inputs
           The number of input streams. It defaults to 2.

       channel_layout
           The desired output channel layout. It defaults to stereo.

       map Map channels from inputs to output. The argument is a '|'-separated list of mappings,
           each in the "input_idx.in_channel-out_channel" form. input_idx is the 0-based index of
           the input stream. in_channel can be either the name of the input channel (e.g. FL for
           front left) or its index in the specified input stream. out_channel is the name of the
           output channel.

       The filter will attempt to guess the mappings when they are not specified explicitly. It
       does so by first trying to find an unused matching input channel and if that fails it
       picks the first unused input channel.

       Join 3 inputs (with properly set channel layouts):

               ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

       Build a 5.1 output from 6 single-channel streams:

               ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex
               'join=inputs=6:channel_layout=5.1:map=0.0-FL|1.0-FR|2.0-FC|3.0-SL|4.0-SR|5.0-LFE'
               out

   ladspa
       Load a LADSPA (Linux Audio Developer's Simple Plugin API) plugin.

       To enable compilation of this filter you need to configure FFmpeg with "--enable-ladspa".

       file, f
           Specifies the name of LADSPA plugin library to load. If the environment variable
           LADSPA_PATH is defined, the LADSPA plugin is searched in each one of the directories
           specified by the colon separated list in LADSPA_PATH, otherwise in the standard LADSPA
           paths, which are in this order: HOME/.ladspa/lib/, /usr/local/lib/ladspa/,
           /usr/lib/ladspa/.

       plugin, p
           Specifies the plugin within the library. Some libraries contain only one plugin, but
           others contain many of them. If this is not set filter will list all available plugins
           within the specified library.

       controls, c
           Set the '|' separated list of controls which are zero or more floating point values
           that determine the behavior of the loaded plugin (for example delay, threshold or
           gain).  Controls need to be defined using the following syntax:
           c0=value0|c1=value1|c2=value2|..., where valuei is the value set on the i-th control.
           Alternatively they can be also defined using the following syntax:
           value0|value1|value2|..., where valuei is the value set on the i-th control.  If
           controls is set to "help", all available controls and their valid ranges are printed.

       sample_rate, s
           Specify the sample rate, default to 44100. Only used if plugin have zero inputs.

       nb_samples, n
           Set the number of samples per channel per each output frame, default is 1024. Only
           used if plugin have zero inputs.

       duration, d
           Set the minimum duration of the sourced audio. See the Time duration section in the
           ffmpeg-utils(1) manual for the accepted syntax.  Note that the resulting duration may
           be greater than the specified duration, as the generated audio is always cut at the
           end of a complete frame.  If not specified, or the expressed duration is negative, the
           audio is supposed to be generated forever.  Only used if plugin have zero inputs.

       Examples

       ·   List all available plugins within amp (LADSPA example plugin) library:

                   ladspa=file=amp

       ·   List all available controls and their valid ranges for "vcf_notch" plugin from "VCF"
           library:

                   ladspa=f=vcf:p=vcf_notch:c=help

       ·   Simulate low quality audio equipment using "Computer Music Toolkit" (CMT) plugin
           library:

                   ladspa=file=cmt:plugin=lofi:controls=c0=22|c1=12|c2=12

       ·   Add reverberation to the audio using TAP-plugins (Tom's Audio Processing plugins):

                   ladspa=file=tap_reverb:tap_reverb

       ·   Generate white noise, with 0.2 amplitude:

                   ladspa=file=cmt:noise_source_white:c=c0=.2

       ·   Generate 20 bpm clicks using plugin "C* Click - Metronome" from the "C* Audio Plugin
           Suite" (CAPS) library:

                   ladspa=file=caps:Click:c=c1=20'

       ·   Apply "C* Eq10X2 - Stereo 10-band equaliser" effect:

                   ladspa=caps:Eq10X2:c=c0=-48|c9=-24|c3=12|c4=2

       ·   Increase volume by 20dB using fast lookahead limiter from Steve Harris "SWH Plugins"
           collection:

                   ladspa=fast_lookahead_limiter_1913:fastLookaheadLimiter:20|0|2

       ·   Attenuate low frequencies using Multiband EQ from Steve Harris "SWH Plugins"
           collection:

                   ladspa=mbeq_1197:mbeq:-24|-24|-24|0|0|0|0|0|0|0|0|0|0|0|0

       ·   Reduce stereo image using "Narrower" from the "C* Audio Plugin Suite" (CAPS) library:

                   ladspa=caps:Narrower

       ·   Another white noise, now using "C* Audio Plugin Suite" (CAPS) library:

                   ladspa=caps:White:.2

       ·   Some fractal noise, using "C* Audio Plugin Suite" (CAPS) library:

                   ladspa=caps:Fractal:c=c1=1

       ·   Dynamic volume normalization using "VLevel" plugin:

                   ladspa=vlevel-ladspa:vlevel_mono

       Commands

       This filter supports the following commands:

       cN  Modify the N-th control value.

           If the specified value is not valid, it is ignored and prior one is kept.

   loudnorm
       EBU R128 loudness normalization. Includes both dynamic and linear normalization modes.
       Support for both single pass (livestreams, files) and double pass (files) modes.  This
       algorithm can target IL, LRA, and maximum true peak. To accurately detect true peaks, the
       audio stream will be upsampled to 192 kHz unless the normalization mode is linear.  Use
       the "-ar" option or "aresample" filter to explicitly set an output sample rate.

       The filter accepts the following options:

       I, i
           Set integrated loudness target.  Range is -70.0 - -5.0. Default value is -24.0.

       LRA, lra
           Set loudness range target.  Range is 1.0 - 20.0. Default value is 7.0.

       TP, tp
           Set maximum true peak.  Range is -9.0 - +0.0. Default value is -2.0.

       measured_I, measured_i
           Measured IL of input file.  Range is -99.0 - +0.0.

       measured_LRA, measured_lra
           Measured LRA of input file.  Range is  0.0 - 99.0.

       measured_TP, measured_tp
           Measured true peak of input file.  Range is  -99.0 - +99.0.

       measured_thresh
           Measured threshold of input file.  Range is -99.0 - +0.0.

       offset
           Set offset gain. Gain is applied before the true-peak limiter.  Range is  -99.0 -
           +99.0. Default is +0.0.

       linear
           Normalize linearly if possible.  measured_I, measured_LRA, measured_TP, and
           measured_thresh must also to be specified in order to use this mode.  Options are true
           or false. Default is true.

       dual_mono
           Treat mono input files as "dual-mono". If a mono file is intended for playback on a
           stereo system, its EBU R128 measurement will be perceptually incorrect.  If set to
           "true", this option will compensate for this effect.  Multi-channel input files are
           not affected by this option.  Options are true or false. Default is false.

       print_format
           Set print format for stats. Options are summary, json, or none.  Default value is
           none.

   lowpass
       Apply a low-pass filter with 3dB point frequency.  The filter can be either single-pole or
       double-pole (the default).  The filter roll off at 6dB per pole per octave (20dB per pole
       per decade).

       The filter accepts the following options:

       frequency, f
           Set frequency in Hz. Default is 500.

       poles, p
           Set number of poles. Default is 2.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Specify the band-width of a filter in width_type units.  Applies only to double-pole
           filter.  The default is 0.707q and gives a Butterworth response.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Examples

       ·   Lowpass only LFE channel, it LFE is not present it does nothing:

                   lowpass=c=LFE

       Commands

       This filter supports the following commands:

       frequency, f
           Change lowpass frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change lowpass width_type.  Syntax for the command is : "width_type"

       width, w
           Change lowpass width.  Syntax for the command is : "width"

   lv2
       Load a LV2 (LADSPA Version 2) plugin.

       To enable compilation of this filter you need to configure FFmpeg with "--enable-lv2".

       plugin, p
           Specifies the plugin URI. You may need to escape ':'.

       controls, c
           Set the '|' separated list of controls which are zero or more floating point values
           that determine the behavior of the loaded plugin (for example delay, threshold or
           gain).  If controls is set to "help", all available controls and their valid ranges
           are printed.

       sample_rate, s
           Specify the sample rate, default to 44100. Only used if plugin have zero inputs.

       nb_samples, n
           Set the number of samples per channel per each output frame, default is 1024. Only
           used if plugin have zero inputs.

       duration, d
           Set the minimum duration of the sourced audio. See the Time duration section in the
           ffmpeg-utils(1) manual for the accepted syntax.  Note that the resulting duration may
           be greater than the specified duration, as the generated audio is always cut at the
           end of a complete frame.  If not specified, or the expressed duration is negative, the
           audio is supposed to be generated forever.  Only used if plugin have zero inputs.

       Examples

       ·   Apply bass enhancer plugin from Calf:

                   lv2=p=http\\\\://calf.sourceforge.net/plugins/BassEnhancer:c=amount=2

       ·   Apply vinyl plugin from Calf:

                   lv2=p=http\\\\://calf.sourceforge.net/plugins/Vinyl:c=drone=0.2|aging=0.5

       ·   Apply bit crusher plugin from ArtyFX:

                   lv2=p=http\\\\://www.openavproductions.com/artyfx#bitta:c=crush=0.3

   mcompand
       Multiband Compress or expand the audio's dynamic range.

       The input audio is divided into bands using 4th order Linkwitz-Riley IIRs.  This is akin
       to the crossover of a loudspeaker, and results in flat frequency response when absent
       compander action.

       It accepts the following parameters:

       args
           This option syntax is: attack,decay,[attack,decay..] soft-knee points
           crossover_frequency [delay [initial_volume [gain]]] | attack,decay ...  For
           explanation of each item refer to compand filter documentation.

   pan
       Mix channels with specific gain levels. The filter accepts the output channel layout
       followed by a set of channels definitions.

       This filter is also designed to efficiently remap the channels of an audio stream.

       The filter accepts parameters of the form: "l|outdef|outdef|..."

       l   output channel layout or number of channels

       outdef
           output channel specification, of the form:
           "out_name=[gain*]in_name[(+-)[gain*]in_name...]"

       out_name
           output channel to define, either a channel name (FL, FR, etc.) or a channel number
           (c0, c1, etc.)

       gain
           multiplicative coefficient for the channel, 1 leaving the volume unchanged

       in_name
           input channel to use, see out_name for details; it is not possible to mix named and
           numbered input channels

       If the `=' in a channel specification is replaced by `<', then the gains for that
       specification will be renormalized so that the total is 1, thus avoiding clipping noise.

       Mixing examples

       For example, if you want to down-mix from stereo to mono, but with a bigger factor for the
       left channel:

               pan=1c|c0=0.9*c0+0.1*c1

       A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels
       surround:

               pan=stereo| FL < FL + 0.5*FC + 0.6*BL + 0.6*SL | FR < FR + 0.5*FC + 0.6*BR + 0.6*SR

       Note that ffmpeg integrates a default down-mix (and up-mix) system that should be
       preferred (see "-ac" option) unless you have very specific needs.

       Remapping examples

       The channel remapping will be effective if, and only if:

       *<gain coefficients are zeroes or ones,>
       *<only one input per channel output,>

       If all these conditions are satisfied, the filter will notify the user ("Pure channel
       mapping detected"), and use an optimized and lossless method to do the remapping.

       For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra
       channels:

               pan="stereo| c0=FL | c1=FR"

       Given the same source, you can also switch front left and front right channels and keep
       the input channel layout:

               pan="5.1| c0=c1 | c1=c0 | c2=c2 | c3=c3 | c4=c4 | c5=c5"

       If the input is a stereo audio stream, you can mute the front left channel (and still keep
       the stereo channel layout) with:

               pan="stereo|c1=c1"

       Still with a stereo audio stream input, you can copy the right channel in both front left
       and right:

               pan="stereo| c0=FR | c1=FR"

   replaygain
       ReplayGain scanner filter. This filter takes an audio stream as an input and outputs it
       unchanged.  At end of filtering it displays "track_gain" and "track_peak".

   resample
       Convert the audio sample format, sample rate and channel layout. It is not meant to be
       used directly.

   rubberband
       Apply time-stretching and pitch-shifting with librubberband.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-librubberband".

       The filter accepts the following options:

       tempo
           Set tempo scale factor.

       pitch
           Set pitch scale factor.

       transients
           Set transients detector.  Possible values are:

           crisp
           mixed
           smooth
       detector
           Set detector.  Possible values are:

           compound
           percussive
           soft
       phase
           Set phase.  Possible values are:

           laminar
           independent
       window
           Set processing window size.  Possible values are:

           standard
           short
           long
       smoothing
           Set smoothing.  Possible values are:

           off
           on
       formant
           Enable formant preservation when shift pitching.  Possible values are:

           shifted
           preserved
       pitchq
           Set pitch quality.  Possible values are:

           quality
           speed
           consistency
       channels
           Set channels.  Possible values are:

           apart
           together

   sidechaincompress
       This filter acts like normal compressor but has the ability to compress detected signal
       using second input signal.  It needs two input streams and returns one output stream.
       First input stream will be processed depending on second stream signal.  The filtered
       signal then can be filtered with other filters in later stages of processing. See pan and
       amerge filter.

       The filter accepts the following options:

       level_in
           Set input gain. Default is 1. Range is between 0.015625 and 64.

       threshold
           If a signal of second stream raises above this level it will affect the gain reduction
           of first stream.  By default is 0.125. Range is between 0.00097563 and 1.

       ratio
           Set a ratio about which the signal is reduced. 1:2 means that if the level raised 4dB
           above the threshold, it will be only 2dB above after the reduction.  Default is 2.
           Range is between 1 and 20.

       attack
           Amount of milliseconds the signal has to rise above the threshold before gain
           reduction starts. Default is 20. Range is between 0.01 and 2000.

       release
           Amount of milliseconds the signal has to fall below the threshold before reduction is
           decreased again. Default is 250. Range is between 0.01 and 9000.

       makeup
           Set the amount by how much signal will be amplified after processing.  Default is 1.
           Range is from 1 to 64.

       knee
           Curve the sharp knee around the threshold to enter gain reduction more softly.
           Default is 2.82843. Range is between 1 and 8.

       link
           Choose if the "average" level between all channels of side-chain stream or the
           louder("maximum") channel of side-chain stream affects the reduction. Default is
           "average".

       detection
           Should the exact signal be taken in case of "peak" or an RMS one in case of "rms".
           Default is "rms" which is mainly smoother.

       level_sc
           Set sidechain gain. Default is 1. Range is between 0.015625 and 64.

       mix How much to use compressed signal in output. Default is 1.  Range is between 0 and 1.

       Examples

       ·   Full ffmpeg example taking 2 audio inputs, 1st input to be compressed depending on the
           signal of 2nd input and later compressed signal to be merged with 2nd input:

                   ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0:a][sc]sidechaincompress[compr];[compr][mix]amerge"

   sidechaingate
       A sidechain gate acts like a normal (wideband) gate but has the ability to filter the
       detected signal before sending it to the gain reduction stage.  Normally a gate uses the
       full range signal to detect a level above the threshold.  For example: If you cut all
       lower frequencies from your sidechain signal the gate will decrease the volume of your
       track only if not enough highs appear. With this technique you are able to reduce the
       resonation of a natural drum or remove "rumbling" of muted strokes from a heavily
       distorted guitar.  It needs two input streams and returns one output stream.  First input
       stream will be processed depending on second stream signal.

       The filter accepts the following options:

       level_in
           Set input level before filtering.  Default is 1. Allowed range is from 0.015625 to 64.

       range
           Set the level of gain reduction when the signal is below the threshold.  Default is
           0.06125. Allowed range is from 0 to 1.

       threshold
           If a signal rises above this level the gain reduction is released.  Default is 0.125.
           Allowed range is from 0 to 1.

       ratio
           Set a ratio about which the signal is reduced.  Default is 2. Allowed range is from 1
           to 9000.

       attack
           Amount of milliseconds the signal has to rise above the threshold before gain
           reduction stops.  Default is 20 milliseconds. Allowed range is from 0.01 to 9000.

       release
           Amount of milliseconds the signal has to fall below the threshold before the reduction
           is increased again. Default is 250 milliseconds.  Allowed range is from 0.01 to 9000.

       makeup
           Set amount of amplification of signal after processing.  Default is 1. Allowed range
           is from 1 to 64.

       knee
           Curve the sharp knee around the threshold to enter gain reduction more softly.
           Default is 2.828427125. Allowed range is from 1 to 8.

       detection
           Choose if exact signal should be taken for detection or an RMS like one.  Default is
           rms. Can be peak or rms.

       link
           Choose if the average level between all channels or the louder channel affects the
           reduction.  Default is average. Can be average or maximum.

       level_sc
           Set sidechain gain. Default is 1. Range is from 0.015625 to 64.

   silencedetect
       Detect silence in an audio stream.

       This filter logs a message when it detects that the input audio volume is less or equal to
       a noise tolerance value for a duration greater or equal to the minimum detected noise
       duration.

       The printed times and duration are expressed in seconds.

       The filter accepts the following options:

       noise, n
           Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified
           value) or amplitude ratio. Default is -60dB, or 0.001.

       duration, d
           Set silence duration until notification (default is 2 seconds).

       mono, m
           Process each channel separately, instead of combined. By default is disabled.

       Examples

       ·   Detect 5 seconds of silence with -50dB noise tolerance:

                   silencedetect=n=-50dB:d=5

       ·   Complete example with ffmpeg to detect silence with 0.0001 noise tolerance in
           silence.mp3:

                   ffmpeg -i silence.mp3 -af silencedetect=noise=0.0001 -f null -

   silenceremove
       Remove silence from the beginning, middle or end of the audio.

       The filter accepts the following options:

       start_periods
           This value is used to indicate if audio should be trimmed at beginning of the audio. A
           value of zero indicates no silence should be trimmed from the beginning. When
           specifying a non-zero value, it trims audio up until it finds non-silence. Normally,
           when trimming silence from beginning of audio the start_periods will be 1 but it can
           be increased to higher values to trim all audio up to specific count of non-silence
           periods.  Default value is 0.

       start_duration
           Specify the amount of time that non-silence must be detected before it stops trimming
           audio. By increasing the duration, bursts of noises can be treated as silence and
           trimmed off. Default value is 0.

       start_threshold
           This indicates what sample value should be treated as silence. For digital audio, a
           value of 0 may be fine but for audio recorded from analog, you may wish to increase
           the value to account for background noise.  Can be specified in dB (in case "dB" is
           appended to the specified value) or amplitude ratio. Default value is 0.

       start_silence
           Specify max duration of silence at beginning that will be kept after trimming. Default
           is 0, which is equal to trimming all samples detected as silence.

       start_mode
           Specify mode of detection of silence end in start of multi-channel audio.  Can be any
           or all. Default is any.  With any, any sample that is detected as non-silence will
           cause stopped trimming of silence.  With all, only if all channels are detected as
           non-silence will cause stopped trimming of silence.

       stop_periods
           Set the count for trimming silence from the end of audio.  To remove silence from the
           middle of a file, specify a stop_periods that is negative. This value is then treated
           as a positive value and is used to indicate the effect should restart processing as
           specified by start_periods, making it suitable for removing periods of silence in the
           middle of the audio.  Default value is 0.

       stop_duration
           Specify a duration of silence that must exist before audio is not copied any more. By
           specifying a higher duration, silence that is wanted can be left in the audio.
           Default value is 0.

       stop_threshold
           This is the same as start_threshold but for trimming silence from the end of audio.
           Can be specified in dB (in case "dB" is appended to the specified value) or amplitude
           ratio. Default value is 0.

       stop_silence
           Specify max duration of silence at end that will be kept after trimming. Default is 0,
           which is equal to trimming all samples detected as silence.

       stop_mode
           Specify mode of detection of silence start in end of multi-channel audio.  Can be any
           or all. Default is any.  With any, any sample that is detected as non-silence will
           cause stopped trimming of silence.  With all, only if all channels are detected as
           non-silence will cause stopped trimming of silence.

       detection
           Set how is silence detected. Can be "rms" or "peak". Second is faster and works better
           with digital silence which is exactly 0.  Default value is "rms".

       window
           Set duration in number of seconds used to calculate size of window in number of
           samples for detecting silence.  Default value is 0.02. Allowed range is from 0 to 10.

       Examples

       ·   The following example shows how this filter can be used to start a recording that does
           not contain the delay at the start which usually occurs between pressing the record
           button and the start of the performance:

                   silenceremove=start_periods=1:start_duration=5:start_threshold=0.02

       ·   Trim all silence encountered from beginning to end where there is more than 1 second
           of silence in audio:

                   silenceremove=stop_periods=-1:stop_duration=1:stop_threshold=-90dB

   sofalizer
       SOFAlizer uses head-related transfer functions (HRTFs) to create virtual loudspeakers
       around the user for binaural listening via headphones (audio formats up to 9 channels
       supported).  The HRTFs are stored in SOFA files (see <http://www.sofacoustics.org/> for a
       database).  SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
       Austrian Academy of Sciences.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libmysofa".

       The filter accepts the following options:

       sofa
           Set the SOFA file used for rendering.

       gain
           Set gain applied to audio. Value is in dB. Default is 0.

       rotation
           Set rotation of virtual loudspeakers in deg. Default is 0.

       elevation
           Set elevation of virtual speakers in deg. Default is 0.

       radius
           Set distance in meters between loudspeakers and the listener with near-field HRTFs.
           Default is 1.

       type
           Set processing type. Can be time or freq. time is processing audio in time domain
           which is slow.  freq is processing audio in frequency domain which is fast.  Default
           is freq.

       speakers
           Set custom positions of virtual loudspeakers. Syntax for this option is: <CH> <AZIM>
           <ELEV>[|<CH> <AZIM> <ELEV>|...].  Each virtual loudspeaker is described with short
           channel name following with azimuth and elevation in degrees.  Each virtual
           loudspeaker description is separated by '|'.  For example to override front left and
           front right channel positions use: 'speakers=FL 45 15|FR 345 15'.  Descriptions with
           unrecognised channel names are ignored.

       lfegain
           Set custom gain for LFE channels. Value is in dB. Default is 0.

       Examples

       ·   Using ClubFritz6 sofa file:

                   sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=1

       ·   Using ClubFritz12 sofa file and bigger radius with small rotation:

                   sofalizer=sofa=/path/to/ClubFritz12.sofa:type=freq:radius=2:rotation=5

       ·   Similar as above but with custom speaker positions for front left, front right, back
           left and back right and also with custom gain:

                   "sofalizer=sofa=/path/to/ClubFritz6.sofa:type=freq:radius=2:speakers=FL 45|FR 315|BL 135|BR 225:gain=28"

   stereotools
       This filter has some handy utilities to manage stereo signals, for converting M/S stereo
       recordings to L/R signal while having control over the parameters or spreading the stereo
       image of master track.

       The filter accepts the following options:

       level_in
           Set input level before filtering for both channels. Defaults is 1.  Allowed range is
           from 0.015625 to 64.

       level_out
           Set output level after filtering for both channels. Defaults is 1.  Allowed range is
           from 0.015625 to 64.

       balance_in
           Set input balance between both channels. Default is 0.  Allowed range is from -1 to 1.

       balance_out
           Set output balance between both channels. Default is 0.  Allowed range is from -1 to
           1.

       softclip
           Enable softclipping. Results in analog distortion instead of harsh digital 0dB
           clipping. Disabled by default.

       mutel
           Mute the left channel. Disabled by default.

       muter
           Mute the right channel. Disabled by default.

       phasel
           Change the phase of the left channel. Disabled by default.

       phaser
           Change the phase of the right channel. Disabled by default.

       mode
           Set stereo mode. Available values are:

           lr>lr
               Left/Right to Left/Right, this is default.

           lr>ms
               Left/Right to Mid/Side.

           ms>lr
               Mid/Side to Left/Right.

           lr>ll
               Left/Right to Left/Left.

           lr>rr
               Left/Right to Right/Right.

           lr>l+r
               Left/Right to Left + Right.

           lr>rl
               Left/Right to Right/Left.

           ms>ll
               Mid/Side to Left/Left.

           ms>rr
               Mid/Side to Right/Right.

       slev
           Set level of side signal. Default is 1.  Allowed range is from 0.015625 to 64.

       sbal
           Set balance of side signal. Default is 0.  Allowed range is from -1 to 1.

       mlev
           Set level of the middle signal. Default is 1.  Allowed range is from 0.015625 to 64.

       mpan
           Set middle signal pan. Default is 0. Allowed range is from -1 to 1.

       base
           Set stereo base between mono and inversed channels. Default is 0.  Allowed range is
           from -1 to 1.

       delay
           Set delay in milliseconds how much to delay left from right channel and vice versa.
           Default is 0. Allowed range is from -20 to 20.

       sclevel
           Set S/C level. Default is 1. Allowed range is from 1 to 100.

       phase
           Set the stereo phase in degrees. Default is 0. Allowed range is from 0 to 360.

       bmode_in, bmode_out
           Set balance mode for balance_in/balance_out option.

           Can be one of the following:

           balance
               Classic balance mode. Attenuate one channel at time.  Gain is raised up to 1.

           amplitude
               Similar as classic mode above but gain is raised up to 2.

           power
               Equal power distribution, from -6dB to +6dB range.

       Examples

       ·   Apply karaoke like effect:

                   stereotools=mlev=0.015625

       ·   Convert M/S signal to L/R:

                   "stereotools=mode=ms>lr"

   stereowiden
       This filter enhance the stereo effect by suppressing signal common to both channels and by
       delaying the signal of left into right and vice versa, thereby widening the stereo effect.

       The filter accepts the following options:

       delay
           Time in milliseconds of the delay of left signal into right and vice versa.  Default
           is 20 milliseconds.

       feedback
           Amount of gain in delayed signal into right and vice versa. Gives a delay effect of
           left signal in right output and vice versa which gives widening effect. Default is
           0.3.

       crossfeed
           Cross feed of left into right with inverted phase. This helps in suppressing the mono.
           If the value is 1 it will cancel all the signal common to both channels. Default is
           0.3.

       drymix
           Set level of input signal of original channel. Default is 0.8.

   superequalizer
       Apply 18 band equalizer.

       The filter accepts the following options:

       1b  Set 65Hz band gain.

       2b  Set 92Hz band gain.

       3b  Set 131Hz band gain.

       4b  Set 185Hz band gain.

       5b  Set 262Hz band gain.

       6b  Set 370Hz band gain.

       7b  Set 523Hz band gain.

       8b  Set 740Hz band gain.

       9b  Set 1047Hz band gain.

       10b Set 1480Hz band gain.

       11b Set 2093Hz band gain.

       12b Set 2960Hz band gain.

       13b Set 4186Hz band gain.

       14b Set 5920Hz band gain.

       15b Set 8372Hz band gain.

       16b Set 11840Hz band gain.

       17b Set 16744Hz band gain.

       18b Set 20000Hz band gain.

   surround
       Apply audio surround upmix filter.

       This filter allows to produce multichannel output from audio stream.

       The filter accepts the following options:

       chl_out
           Set output channel layout. By default, this is 5.1.

           See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

       chl_in
           Set input channel layout. By default, this is stereo.

           See the Channel Layout section in the ffmpeg-utils(1) manual for the required syntax.

       level_in
           Set input volume level. By default, this is 1.

       level_out
           Set output volume level. By default, this is 1.

       lfe Enable LFE channel output if output channel layout has it. By default, this is
           enabled.

       lfe_low
           Set LFE low cut off frequency. By default, this is 128 Hz.

       lfe_high
           Set LFE high cut off frequency. By default, this is 256 Hz.

       fc_in
           Set front center input volume. By default, this is 1.

       fc_out
           Set front center output volume. By default, this is 1.

       lfe_in
           Set LFE input volume. By default, this is 1.

       lfe_out
           Set LFE output volume. By default, this is 1.

   treble, highshelf
       Boost or cut treble (upper) frequencies of the audio using a two-pole shelving filter with
       a response similar to that of a standard hi-fi's tone-controls. This is also known as
       shelving equalisation (EQ).

       The filter accepts the following options:

       gain, g
           Give the gain at whichever is the lower of ~22 kHz and the Nyquist frequency. Its
           useful range is about -20 (for a large cut) to +20 (for a large boost). Beware of
           clipping when using a positive gain.

       frequency, f
           Set the filter's central frequency and so can be used to extend or reduce the
           frequency range to be boosted or cut.  The default value is 3000 Hz.

       width_type, t
           Set method to specify band-width of filter.

           h   Hz

           q   Q-Factor

           o   octave

           s   slope

           k   kHz

       width, w
           Determine how steep is the filter's shelf transition.

       channels, c
           Specify which channels to filter, by default all available are filtered.

       Commands

       This filter supports the following commands:

       frequency, f
           Change treble frequency.  Syntax for the command is : "frequency"

       width_type, t
           Change treble width_type.  Syntax for the command is : "width_type"

       width, w
           Change treble width.  Syntax for the command is : "width"

       gain, g
           Change treble gain.  Syntax for the command is : "gain"

   tremolo
       Sinusoidal amplitude modulation.

       The filter accepts the following options:

       f   Modulation frequency in Hertz. Modulation frequencies in the subharmonic range (20 Hz
           or lower) will result in a tremolo effect.  This filter may also be used as a ring
           modulator by specifying a modulation frequency higher than 20 Hz.  Range is 0.1 -
           20000.0. Default value is 5.0 Hz.

       d   Depth of modulation as a percentage. Range is 0.0 - 1.0.  Default value is 0.5.

   vibrato
       Sinusoidal phase modulation.

       The filter accepts the following options:

       f   Modulation frequency in Hertz.  Range is 0.1 - 20000.0. Default value is 5.0 Hz.

       d   Depth of modulation as a percentage. Range is 0.0 - 1.0.  Default value is 0.5.

   volume
       Adjust the input audio volume.

       It accepts the following parameters:

       volume
           Set audio volume expression.

           Output values are clipped to the maximum value.

           The output audio volume is given by the relation:

                   <output_volume> = <volume> * <input_volume>

           The default value for volume is "1.0".

       precision
           This parameter represents the mathematical precision.

           It determines which input sample formats will be allowed, which affects the precision
           of the volume scaling.

           fixed
               8-bit fixed-point; this limits input sample format to U8, S16, and S32.

           float
               32-bit floating-point; this limits input sample format to FLT. (default)

           double
               64-bit floating-point; this limits input sample format to DBL.

       replaygain
           Choose the behaviour on encountering ReplayGain side data in input frames.

           drop
               Remove ReplayGain side data, ignoring its contents (the default).

           ignore
               Ignore ReplayGain side data, but leave it in the frame.

           track
               Prefer the track gain, if present.

           album
               Prefer the album gain, if present.

       replaygain_preamp
           Pre-amplification gain in dB to apply to the selected replaygain gain.

           Default value for replaygain_preamp is 0.0.

       eval
           Set when the volume expression is evaluated.

           It accepts the following values:

           once
               only evaluate expression once during the filter initialization, or when the volume
               command is sent

           frame
               evaluate expression for each incoming frame

           Default value is once.

       The volume expression can contain the following parameters.

       n   frame number (starting at zero)

       nb_channels
           number of channels

       nb_consumed_samples
           number of samples consumed by the filter

       nb_samples
           number of samples in the current frame

       pos original frame position in the file

       pts frame PTS

       sample_rate
           sample rate

       startpts
           PTS at start of stream

       startt
           time at start of stream

       t   frame time

       tb  timestamp timebase

       volume
           last set volume value

       Note that when eval is set to once only the sample_rate and tb variables are available,
       all other variables will evaluate to NAN.

       Commands

       This filter supports the following commands:

       volume
           Modify the volume expression.  The command accepts the same syntax of the
           corresponding option.

           If the specified expression is not valid, it is kept at its current value.

       replaygain_noclip
           Prevent clipping by limiting the gain applied.

           Default value for replaygain_noclip is 1.

       Examples

       ·   Halve the input audio volume:

                   volume=volume=0.5
                   volume=volume=1/2
                   volume=volume=-6.0206dB

           In all the above example the named key for volume can be omitted, for example like in:

                   volume=0.5

       ·   Increase input audio power by 6 decibels using fixed-point precision:

                   volume=volume=6dB:precision=fixed

       ·   Fade volume after time 10 with an annihilation period of 5 seconds:

                   volume='if(lt(t,10),1,max(1-(t-10)/5,0))':eval=frame

   volumedetect
       Detect the volume of the input video.

       The filter has no parameters. The input is not modified. Statistics about the volume will
       be printed in the log when the input stream end is reached.

       In particular it will show the mean volume (root mean square), maximum volume (on a per-
       sample basis), and the beginning of a histogram of the registered volume values (from the
       maximum value to a cumulated 1/1000 of the samples).

       All volumes are in decibels relative to the maximum PCM value.

       Examples

       Here is an excerpt of the output:

               [Parsed_volumedetect_0  0xa23120] mean_volume: -27 dB
               [Parsed_volumedetect_0  0xa23120] max_volume: -4 dB
               [Parsed_volumedetect_0  0xa23120] histogram_4db: 6
               [Parsed_volumedetect_0  0xa23120] histogram_5db: 62
               [Parsed_volumedetect_0  0xa23120] histogram_6db: 286
               [Parsed_volumedetect_0  0xa23120] histogram_7db: 1042
               [Parsed_volumedetect_0  0xa23120] histogram_8db: 2551
               [Parsed_volumedetect_0  0xa23120] histogram_9db: 4609
               [Parsed_volumedetect_0  0xa23120] histogram_10db: 8409

       It means that:

       ·   The mean square energy is approximately -27 dB, or 10^-2.7.

       ·   The largest sample is at -4 dB, or more precisely between -4 dB and -5 dB.

       ·   There are 6 samples at -4 dB, 62 at -5 dB, 286 at -6 dB, etc.

       In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5
       dB causes clipping for 6 samples, etc.

AUDIO SOURCES

       Below is a description of the currently available audio sources.

   abuffer
       Buffer audio frames, and make them available to the filter chain.

       This source is mainly intended for a programmatic use, in particular through the interface
       defined in libavfilter/asrc_abuffer.h.

       It accepts the following parameters:

       time_base
           The timebase which will be used for timestamps of submitted frames. It must be either
           a floating-point number or in numerator/denominator form.

       sample_rate
           The sample rate of the incoming audio buffers.

       sample_fmt
           The sample format of the incoming audio buffers.  Either a sample format name or its
           corresponding integer representation from the enum AVSampleFormat in
           libavutil/samplefmt.h

       channel_layout
           The channel layout of the incoming audio buffers.  Either a channel layout name from
           channel_layout_map in libavutil/channel_layout.c or its corresponding integer
           representation from the AV_CH_LAYOUT_* macros in libavutil/channel_layout.h

       channels
           The number of channels of the incoming audio buffers.  If both channels and
           channel_layout are specified, then they must be consistent.

       Examples

               abuffer=sample_rate=44100:sample_fmt=s16p:channel_layout=stereo

       will instruct the source to accept planar 16bit signed stereo at 44100Hz.  Since the
       sample format with name "s16p" corresponds to the number 6 and the "stereo" channel layout
       corresponds to the value 0x3, this is equivalent to:

               abuffer=sample_rate=44100:sample_fmt=6:channel_layout=0x3

   aevalsrc
       Generate an audio signal specified by an expression.

       This source accepts in input one or more expressions (one for each channel), which are
       evaluated and used to generate a corresponding audio signal.

       This source accepts the following options:

       exprs
           Set the '|'-separated expressions list for each separate channel. In case the
           channel_layout option is not specified, the selected channel layout depends on the
           number of provided expressions. Otherwise the last specified expression is applied to
           the remaining output channels.

       channel_layout, c
           Set the channel layout. The number of channels in the specified layout must be equal
           to the number of specified expressions.

       duration, d
           Set the minimum duration of the sourced audio. See the Time duration section in the
           ffmpeg-utils(1) manual for the accepted syntax.  Note that the resulting duration may
           be greater than the specified duration, as the generated audio is always cut at the
           end of a complete frame.

           If not specified, or the expressed duration is negative, the audio is supposed to be
           generated forever.

       nb_samples, n
           Set the number of samples per channel per each output frame, default to 1024.

       sample_rate, s
           Specify the sample rate, default to 44100.

       Each expression in exprs can contain the following constants:

       n   number of the evaluated sample, starting from 0

       t   time of the evaluated sample expressed in seconds, starting from 0

       s   sample rate

       Examples

       ·   Generate silence:

                   aevalsrc=0

       ·   Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

                   aevalsrc="sin(440*2*PI*t):s=8000"

       ·   Generate a two channels signal, specify the channel layout (Front Center + Back
           Center) explicitly:

                   aevalsrc="sin(420*2*PI*t)|cos(430*2*PI*t):c=FC|BC"

       ·   Generate white noise:

                   aevalsrc="-2+random(0)"

       ·   Generate an amplitude modulated signal:

                   aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

       ·   Generate 2.5 Hz binaural beats on a 360 Hz carrier:

                   aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)"

   anullsrc
       The null audio source, return unprocessed audio frames. It is mainly useful as a template
       and to be employed in analysis / debugging tools, or as the source for filters which
       ignore the input data (for example the sox synth filter).

       This source accepts the following options:

       channel_layout, cl
           Specifies the channel layout, and can be either an integer or a string representing a
           channel layout. The default value of channel_layout is "stereo".

           Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping
           between strings and channel layout values.

       sample_rate, r
           Specifies the sample rate, and defaults to 44100.

       nb_samples, n
           Set the number of samples per requested frames.

       Examples

       ·   Set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO.

                   anullsrc=r=48000:cl=4

       ·   Do the same operation with a more obvious syntax:

                   anullsrc=r=48000:cl=mono

       All the parameters need to be explicitly defined.

   flite
       Synthesize a voice utterance using the libflite library.

       To enable compilation of this filter you need to configure FFmpeg with
       "--enable-libflite".

       Note that versions of the flite library prior to 2.0 are not thread-safe.

       The filter accepts the following options:

       list_voices
           If set to 1, list the names of the available voices and exit immediately. Default
           value is 0.

       nb_samples, n
           Set the maximum number of samples per frame. Default value is 512.

       textfile
           Set the filename containing the text to speak.

       text
           Set the text to speak.

       voice, v
           Set the voice to use for the speech synthesis. Default value is "kal". See also the
           list_voices option.

       Examples

       ·   Read from file speech.txt, and synthesize the text using the standard flite voice:

                   flite=textfile=speech.txt

       ·   Read the specified text selecting the "slt" voice:

                   flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Input text to ffmpeg:

                   ffmpeg -f lavfi -i flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt

       ·   Make ffplay speak the specified text, using "flite" and the "lavfi" device:

                   ffplay -f lavfi flite=text='No more be grieved for which that thou hast done.'

       For more information about libflite, check: <http://www.festvox.org/flite/>

   anoisesrc
       Generate a noise audio signal.

       The filter accepts the following options:

       sample_rate, r
           Specify the sample rate. Default value is 48000 Hz.

       amplitude, a
           Specify the amplitude (0.0 - 1.0) of the generated audio stream. Default value is 1.0.

       duration, d
           Specify the duration of the generated audio stream. Not specifying this option results
           in noise with an infinite length.

       color, colour, c
           Specify the color of noise. Available noise colors are white, pink, brown, blue and
           violet. Default color is white.

       seed, s
           Specify a value used to seed the PRNG.

       nb_samples, n
           Set the number of samples per each output frame, default is 1024.

       Examples

       ·   Generate 60 seconds of pink noise, with a 44.1 kHz sampling rate and an amplitude of
           0.5:

                   anoisesrc=d=60:c=pink:r=44100:a=0.5

   hilbert
       Generate odd-tap Hilbert transform FIR coefficients.

       The resulting stream can be used with afir filter for phase-shifting the signal by 90
       degrees.

       This is used in many matrix coding schemes and for analytic signal generation.  The
       process is often written as a multiplication by i (or j), the imaginary unit.

       The filter accepts the following options:

       sample_rate, s
           Set sample rate, default is 44100.

       taps, t
           Set length of FIR filter, default is 22051.

       nb_samples, n
           Set number of samples per each frame.

       win_func, w
           Set window function to be used when generating FIR coefficients.

   sinc
       Generate a sinc kaiser-windowed low-pass, high-pass, band-pass, or band-reject FIR
       coefficients.

       The resulting stream can be used with afir filter for filtering the audio signal.

       The filter accepts the following options:

       sample_rate, r
           Set sample rate, default is 44100.

       nb_samples, n
           Set number of samples per each frame. Default is 1024.

       hp  Set high-pass frequency. Default is 0.

       lp  Set low-pass frequency. Default is 0.  If high-pass frequency is lower than low-pass
           frequency and low-pass frequency is higher than 0 then filter will create band-pass
           filter coefficients, otherwise band-reject filter coefficients.

       phase
           Set filter phase response. Default is 50. Allowed range is from 0 to 100.

       beta
           Set Kaiser window beta.

       att Set stop-band attenuation. Default is 120dB, allowed range is from 40 to 180 dB.

       round
           Enable rounding, by default is disabled.

       hptaps
           Set number of taps for high-pass filter.

       lptaps
           Set number of taps for low-pass filter.

   sine
       Generate an audio signal made of a sine wave with amplitude 1/8.

       The audio signal is bit-exact.

       The filter accepts the following options:

       frequency, f
           Set the carrier frequency. Default is 440 Hz.

       beep_factor, b
           Enable a periodic beep every second with frequency beep_factor times the carrier
           frequency. Default is 0, meaning the beep is disabled.

       sample_rate, r
           Specify the sample rate, default is 44100.

       duration, d
           Specify the duration of the generated audio stream.

       samples_per_frame
           Set the number of samples per output frame.

           The expression can contain the following constants:

           n   The (sequential) number of the output audio frame, starting from 0.

           pts The PTS (Presentation TimeStamp) of the output audio frame, expressed in TB units.

           t   The PTS of the output audio frame, expressed in seconds.

           TB  The timebase of the output audio frames.

           Default is 1024.

       Examples

       ·   Generate a simple 440 Hz sine wave:

                   sine

       ·   Generate a 220 Hz sine wave with a 880 Hz beep each second, for 5 seconds:

                   sine=220:4:d=5
                   sine=f=220:b=4:d=5
                   sine=frequency=220:beep_factor=4:duration=5

       ·   Generate a 1 kHz sine wave following "1602,1601,1602,1601,1602" NTSC pattern:

                   sine=1000:samples_per_frame='st(0,mod(n,5)); 1602-not(not(eq(ld(0),1)+eq(ld(0),3)))'

AUDIO SINKS

       Below is a description of the currently available audio sinks.

   abuffersink
       Buffer audio frames, and make them available to the end of filter chain.

       This sink is mainly intended for programmatic use, in particular through the interface
       defined in libavfilter/buffersink.h or the options system.

       It accepts a pointer to an AVABufferSinkContext structure, which defines the incoming
       buffers' formats, to be passed as the opaque parameter to "avfilter_init_filter" for
       initialization.

   anullsink
       Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a
       template and for use in analysis / debugging tools.

VIDEO FILTERS

       When you configure your FFmpeg build, you can disable any of the existing filters using
       "--disable-filters".  The configure output will show the video filters included in your
       build.

       Below is a description of the currently available video filters.

   alphaextract
       Extract the alpha component from the input as a grayscale video. This is especially useful
       with the alphamerge filter.

   alphamerge
       Add or replace the alpha component of the primary input with the grayscale value of a
       second input. This is intended for use with alphaextract to allow the transmission or
       storage of frame sequences that have alpha in a format that doesn't support an alpha
       channel.

       For example, to reconstruct full frames from a normal YUV-encoded video and a separate
       video created with alphaextract, you might use:

               movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out]

       Since this filter is designed for reconstruction, it operates on frame sequences without
       considering timestamps, and terminates when either input reaches end of stream. This will
       cause problems if your encoding pipeline drops frames. If you're trying to apply an image
       as an overlay to a video stream, consider the overlay filter instead.

   amplify
       Amplify differences between current pixel and pixels of adjacent frames in same pixel
       location.

       This filter accepts the following options:

       radius
           Set frame radius. Default is 2. Allowed range is from 1 to 63.  For example radius of
           3 will instruct filter to calculate average of 7 frames.

       factor
           Set factor to amplify difference. Default is 2. Allowed range is from 0 to 65535.

       threshold
           Set threshold for difference amplification. Any differrence greater or equal to this
           value will not alter source pixel. Default is 10.  Allowed range is from 0 to 65535.

       low Set lower limit for changing source pixel. Default is 65535. Allowed range is from 0
           to 65535.  This option controls maximum possible value that will decrease source pixel
           value.

       high
           Set high limit for changing source pixel. Default is 65535. Allowed range is from 0 to
           65535.  This option controls maximum possible value that will increase source pixel
           value.

       planes
           Set which planes to filter. Default is all. Allowed range is from 0 to 15.

   ass
       Same as the subtitles filter, except that it doesn't require libavcodec and libavformat to
       work. On the other hand, it is limited to ASS (Advanced Substation Alpha) subtitles files.

       This filter accepts the following option in addition to the common options from the
       subtitles filter:

       shaping
           Set the shaping engine

           Available values are:

           auto
               The default libass shaping engine, which is the best available.

           simple
               Fast, font-agnostic shaper that can do only substitutions

           complex
               Slower shaper using OpenType for substitutions and positioning

           The default is "auto".

   atadenoise
       Apply an Adaptive Temporal Averaging Denoiser to the video input.

       The filter accepts the following options:

       0a  Set threshold A for 1st plane. Default is 0.02.  Valid range is 0 to 0.3.

       0b  Set threshold B for 1st plane. Default is 0.04.  Valid range is 0 to 5.

       1a  Set threshold A for 2nd plane. Default is 0.02.  Valid range is 0 to 0.3.

       1b  Set threshold B for 2nd plane. Default is 0.04.  Valid range is 0 to 5.

       2a  Set threshold A for 3rd plane. Default is 0.02.  Valid range is 0 to 0.3.

       2b  Set threshold B for 3rd plane. Default is 0.04.  Valid range is 0 to 5.

           Threshold A is designed to react on abrupt changes in the input signal and threshold B
           is designed to react on continuous changes in the input signal.

       s   Set number of frames filter will use for averaging. Default is 9. Must be odd number
           in range [5, 129].

       p   Set what planes of frame filter will use for averaging. Default is all.

   avgblur
       Apply average blur filter.

       The filter accepts the following options:

       sizeX
           Set horizontal radius size.

       planes
           Set which planes to filter. By default all planes are filtered.

       sizeY
           Set vertical radius size, if zero it will be same as "sizeX".  Default is 0.

   bbox
       Compute the bounding box for the non-black pixels in the input frame luminance plane.

       This filter computes the bounding box containing all the pixels with a luminance value
       greater than the minimum allowed value.  The parameters describing the bounding box are
       printed on the filter log.

       The filter accepts the following option:

       min_val
           Set the minimal luminance value. Default is 16.

   bitplanenoise
       Show and measure bit plane noise.

       The filter accepts the following options:

       bitplane
           Set which plane to analyze. Default is 1.

       filter
           Filter out noisy pixels from "bitplane" set above.  Default is disabled.

   blackdetect
       Detect video intervals that are (almost) completely black. Can be useful to detect chapter
       transitions, commercials, or invalid recordings. Output lines contains the time for the
       start, end and duration of the detected black interval expressed in seconds.

       In order to display the output lines, you need to set the loglevel at least to the
       AV_LOG_INFO value.

       The filter accepts the following options:

       black_min_duration, d
           Set the minimum detected black duration expressed in seconds. It must be a non-
           negative floating point number.

           Default value is 2.0.

       picture_black_ratio_th, pic_th
           Set the threshold for considering a picture "black".  Express the minimum value for
           the ratio:

                   <nb_black_pixels> / <nb_pixels>

           for which a picture is considered black.  Default value is 0.98.

       pixel_black_th, pix_th
           Set the threshold for considering a pixel "black".

           The threshold expresses the maximum pixel luminance value for which a pixel is
           considered "black". The provided value is scaled according to the following equation:

                   <absolute_threshold> = <luminance_minimum_value> + <pixel_black_th> * <luminance_range_size>

           luminance_range_size and luminance_minimum_value depend on the input video format, the
           range is [0-255] for YUV full-range formats and [16-235] for YUV non full-range
           formats.

           Default value is 0.10.

       The following example sets the maximum pixel threshold to the minimum value, and detects
       only black intervals of 2 or more seconds:

               blackdetect=d=2:pix_th=0.00

   blackframe
       Detect frames that are (almost) completely black. Can be useful to detect chapter
       transitions or commercials. Output lines consist of the frame number of the detected
       frame, the percentage of blackness, the position in the file if known or -1 and the
       timestamp in seconds.

       In order to display the output lines, you need to set the loglevel at least to the
       AV_LOG_INFO value.

       This filter exports frame metadata "lavfi.blackframe.pblack".  The value represents the
       percentage of pixels in the picture that are below the threshold value.

       It accepts the following parameters:

       amount
           The percentage of the pixels that have to be below the threshold; it defaults to 98.

       threshold, thresh
           The threshold below which a pixel value is considered black; it defaults to 32.

   blend, tblend
       Blend two video frames into each other.

       The "blend" filter takes two input streams and outputs one stream, the first input is the
       "top" layer and second input is "bottom" layer.  By default, the output terminates when
       the longest input terminates.

       The "tblend" (time blend) filter takes two consecutive frames from one single stream, and
       outputs the result obtained by blending the new frame on top of the old frame.

       A description of the accepted options follows.

       c0_mode
       c1_mode
       c2_mode
       c3_mode
       all_mode
           Set blend mode for specific pixel component or all pixel components in case of
           all_mode. Default value is "normal".

           Available values for component modes are:

           addition
           grainmerge
           and
           average
           burn
           darken
           difference
           grainextract
           divide
           dodge
           freeze
           exclusion
           extremity
           glow
           hardlight
           hardmix
           heat
           lighten
           linearlight
           multiply
           multiply128
           negation
           normal
           or
           overlay
           phoenix
           pinlight
           reflect
           screen
           softlight
           subtract
           vividlight
           xor
       c0_opacity
       c1_opacity
       c2_opacity
       c3_opacity
       all_opacity
           Set blend opacity for specific pixel component or all pixel components in case of
           all_opacity. Only used in combination with pixel component blend modes.

       c0_expr
       c1_expr
       c2_expr
       c3_expr
       all_expr
           Set blend expression for specific pixel component or all pixel components in case of
           all_expr. Note that related mode options will be ignored if those are set.

           The expressions can use the following variables:

           N   The sequential number of the filtered frame, starting from 0.

           X
           Y   the coordinates of the current sample

           W
           H   the width and height of currently filtered plane

           SW
           SH  Width and height scale for the plane being filtered. It is the ratio between the
               dimensions of the current plane to the luma plane, e.g. for a "yuv420p" frame, the
               values are "1,1" for the luma plane and "0.5,0.5" for the chroma planes.

           T   Time of the current frame, expressed in seconds.

           TOP, A
               Value of pixel component at current location for first video frame (top layer).

           BOTTOM, B
               Value of pixel component at current location for second video frame (bottom
               layer).

       The "blend" filter also supports the framesync options.

       Examples

       ·   Apply transition from bottom layer to top layer in first 10 seconds:

                   blend=all_expr='A*(if(gte(T,10),1,T/10))+B*(1-(if(gte(T,10),1,T/10)))'

       ·   Apply linear horizontal transition from top layer to bottom layer:

                   blend=all_expr='A*(X/W)+B*(1-X/W)'

       ·   Apply 1x1 checkerboard effect:

                   blend=all_expr='if(eq(mod(X,2),mod(Y,2)),A,B)'

       ·   Apply uncover left effect:

                   blend=all_expr='if(gte(N*SW+X,W),A,B)'

       ·   Apply uncover down effect:

                   blend=all_expr='if(gte(Y-N*SH,0),A,B)'

       ·   Apply uncover up-left effect:

                   blend=all_expr='if(gte(T*SH*40+Y,H)*gte((T*40*SW+X)*W/H,W),A,B)'

       ·   Split diagonally video and shows top and bottom layer on each side:

                   blend=all_expr='if(gt(X,Y*(W/H)),A,B)'

       ·   Display differences between the current and the previous frame:

                   tblend=all_mode=grainextract

   bm3d
       Denoise frames using Block-Matching 3D algorithm.

       The filter accepts the following options.

       sigma
           Set denoising strength. Default value is 1.  Allowed range is from 0 to 999.9.  The
           denoising algorith is very sensitive to sigma, so adjust it according to the source.

       block
           Set local patch size. This sets dimensions in 2D.

       bstep
           Set sliding step for processing blocks. Default value is 4.  Allowed range is from 1
           to 64.  Smaller values allows processing more reference blocks and is slower.

       group
           Set maximal number of similar blocks for 3rd dimension. Default value is 1.  When set
           to 1, no block matching is done. Larger values allows more blocks in single group.
           Allowed range is from 1 to 256.

       range
           Set radius for search block matching. Default is 9.  Allowed range is from 1 to
           INT32_MAX.

       mstep
           Set step between two search locations for block matching. Default is 1.  Allowed range
           is from 1 to 64. Smaller is slower.

       thmse
           Set threshold of mean square error for block matching. Valid range is 0 to INT32_MAX.

       hdthr
           Set thresholding parameter for hard thresholding in 3D transformed domain.  Larger
           values results in stronger hard-thresholding filtering in frequency domain.

       estim
           Set filtering estimation mode. Can be "basic" or "final".  Default is "basic".

       ref If enabled, filter will use 2nd stream for block matching.  Default is disabled for
           "basic" value of estim option, and always enabled if value of estim is "final".

       planes
           Set planes to filter. Default is all available except alpha.

       Examples

       ·   Basic filtering with bm3d:

                   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic

       ·   Same as above, but filtering only luma:

                   bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic:planes=1

       ·   Same as above, but with both estimation modes:

                   split[a][b],[a]bm3d=sigma=3:block=4:bstep=2:group=1:estim=basic[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

       ·   Same as above, but prefilter with nlmeans filter instead:

                   split[a][b],[a]nlmeans=s=3:r=7:p=3[a],[b][a]bm3d=sigma=3:block=4:bstep=2:group=16:estim=final:ref=1

   boxblur
       Apply a boxblur algorithm to the input video.

       It accepts the following parameters:

       luma_radius, lr
       luma_power, lp
       chroma_radius, cr
       chroma_power, cp
       alpha_radius, ar
       alpha_power, ap

       A description of the accepted options follows.

       luma_radius, lr
       chroma_radius, cr
       alpha_radius, ar
           Set an expression for the box radius in pixels used for blurring the corresponding
           input plane.

           The radius value must be a non-negative number, and must not be greater than the value
           of the expression "min(w,h)/2" for the luma and alpha planes, and of "min(cw,ch)/2"
           for the chroma planes.

           Default value for luma_radius is "2". If not specified, chroma_radius and alpha_radius
           default to the corresponding value set for luma_radius.

           The expressions can contain the following constants:

           w
           h   The input width and height in pixels.

           cw
           ch  The input chroma image width and height in pixels.

           hsub
           vsub
               The horizontal and vertical chroma subsample values. For example, for the pixel
               format "yuv422p", hsub is 2 and vsub is 1.

       luma_power, lp
       chroma_power, cp
       alpha_power, ap
           Specify how many times the boxblur filter is applied to the corresponding plane.

           Default value for luma_power is 2. If not specified, chroma_power and alpha_power
           default to the corresponding value set for luma_power.

           A value of 0 will disable the effect.

       Examples

       ·   Apply a boxblur filter with the luma, chroma, and alpha radii set to 2:

                   boxblur=luma_radius=2:luma_power=1
                   boxblur=2:1

       ·   Set the luma radius to 2, and alpha and chroma radius to 0:

                   boxblur=2:1:cr=0:ar=0

       ·   Set the luma and chroma radii to a fraction of the video dimension:

                   boxblur=luma_radius=min(h\,w)/10:luma_power=1:chroma_radius=min(cw\,ch)/10:chroma_power=1

   bwdif
       Deinterlace the input video ("bwdif" stands for "Bob Weaver Deinterlacing Filter").

       Motion adaptive deinterlacing based on yadif with the use of w3fdif and cubic
       interpolation algorithms.  It accepts the following parameters:

       mode
           The interlacing mode to adopt. It accepts one of the following values:

           0, send_frame
               Output one frame for each frame.

           1, send_field
               Output one frame for each field.

           The default value is "send_field".

       parity
           The picture field parity assumed for the input interlaced video. It accepts one of the
           following values:

           0, tff
               Assume the top field is first.

           1, bff
               Assume the bottom field is first.

           -1, auto
               Enable automatic detection of field parity.

           The default value is "auto".  If the interlacing is unknown or the decoder does not
           export this information, top field first will be assumed.

       deint
           Specify which frames to deinterlace. Accept one of the following values:

           0, all
               Deinterlace all frames.

           1, interlaced
               Only deinterlace frames marked as interlaced.

           The default value is "all".

   chromahold
       Remove all color information for all colors except for certain one.

       The filter accepts the following options:

       color
           The color which will not be replaced with neutral chroma.

       similarity
           Similarity percentage with the above color.  0.01 matches only the exact key color,
           while 1.0 matches everything.

       yuv Signals that the color passed is already in YUV instead of RGB.

           Literal colors like "green" or "red" don't make sense with this enabled anymore.  This
           can be used to pass exact YUV values as hexadecimal numbers.

   chromakey
       YUV colorspace color/chroma keying.

       The filter accepts the following options:

       color
           The color which will be replaced with transparency.

       similarity
           Similarity percentage with the key color.

           0.01 matches only the exact key color, while 1.0 matches everything.

       blend
           Blend percentage.

           0.0 makes pixels either fully transparent, or not transparent at all.

           Higher values result in semi-transparent pixels, with a higher transparency the more
           similar the pixels color is to the key color.

       yuv Signals that the color passed is already in YUV instead of RGB.

           Literal colors like "green" or "red" don't make sense with this enabled anymore.  This
           can be used to pass exact YUV values as hexadecimal numbers.

       Examples

       ·   Make every green pixel in the input image transparent:

                   ffmpeg -i input.png -vf chromakey=green out.png

       ·   Overlay a greenscreen-video on top of a static black background.

                   ffmpeg -f lavfi -i color=c=black:s=1280x720 -i video.mp4 -shortest -filter_complex "[1:v]chromakey=0x70de77:0.1:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.mkv

   ciescope
       Display CIE color diagram with pixels overlaid onto it.

       The filter accepts the following options:

       system
           Set color system.

           ntsc, 470m
           ebu, 470bg
           smpte
           240m
           apple
           widergb
           cie1931
           rec709, hdtv
           uhdtv, rec2020
       cie Set CIE system.

           xyy
           ucs
           luv
       gamuts
           Set what gamuts to draw.

           See "system" option for available values.

       size, s
           Set ciescope size, by default set to 512.

       intensity, i
           Set intensity used to map input pixel values to CIE diagram.

       contrast
           Set contrast used to draw tongue colors that are out of active color system gamut.

       corrgamma
           Correct gamma displayed on scope, by default enabled.

       showwhite
           Show white point on CIE diagram, by default disabled.

       gamma
           Set input gamma. Used only with XYZ input color space.

   codecview
       Visualize information exported by some codecs.

       Some codecs can export information through frames using side-data or other means. For
       example, some MPEG based codecs export motion vectors through the export_mvs flag in the
       codec flags2 option.

       The filter accepts the following option:

       mv  Set motion vectors to visualize.

           Available flags for mv are:

           pf  forward predicted MVs of P-frames

           bf  forward predicted MVs of B-frames

           bb  backward predicted MVs of B-frames

       qp  Display quantization parameters using the chroma planes.

       mv_type, mvt
           Set motion vectors type to visualize. Includes MVs from all frames unless specified by
           frame_type option.

           Available flags for mv_type are:

           fp  forward predicted MVs

           bp  backward predicted MVs

       frame_type, ft
           Set frame type to visualize motion vectors of.

           Available flags for frame_type are:

           if  intra-coded frames (I-frames)

           pf  predicted frames (P-frames)

           bf  bi-directionally predicted frames (B-frames)

       Examples

       ·   Visualize forward predicted MVs of all frames using ffplay:

                   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv_type=fp

       ·   Visualize multi-directionals MVs of P and B-Frames using ffplay:

                   ffplay -flags2 +export_mvs input.mp4 -vf codecview=mv=pf+bf+bb

   colorbalance
       Modify intensity of primary colors (red, green and blue) of input frames.

       The filter allows an input frame to be adjusted in the shadows, midtones or highlights
       regions for the red-cyan, green-magenta or blue-yellow balance.

       A positive adjustment value shifts the balance towards the primary color, a negative value
       towards the complementary color.

       The filter accepts the following options:

       rs
       gs
       bs  Adjust red, green and blue shadows (darkest pixels).

       rm
       gm
       bm  Adjust red, green and blue midtones (medium pixels).

       rh
       gh
       bh  Adjust red, green and blue highlights (brightest pixels).

           Allowed ranges for options are "[-1.0, 1.0]". Defaults are 0.

       Examples

       ·   Add red color cast to shadows:

                   colorbalance=rs=.3

   colorkey
       RGB colorspace color keying.

       The filter accepts the following options:

       color
           The color which will be replaced with transparency.

       similarity
           Similarity percentage with the key color.

           0.01 matches only the exact key color, while 1.0 matches everything.

       blend
           Blend percentage.

           0.0 makes pixels either fully transparent, or not transparent at all.

           Higher values result in semi-transparent pixels, with a higher transparency the more
           similar the pixels color is to the key color.

       Examples

       ·   Make every green pixel in the input image transparent:

                   ffmpeg -i input.png -vf colorkey=green out.png

       ·   Overlay a greenscreen-video on top of a static background image.

                   ffmpeg -i background.png -i video.mp4 -filter_complex "[1:v]colorkey=0x3BBD1E:0.3:0.2[ckout];[0:v][ckout]overlay[out]" -map "[out]" output.flv

   colorlevels
       Adjust video input frames using levels.

       The filter accepts the following options:

       rimin
       gimin
       bimin
       aimin
           Adjust red, green, blue and alpha input black point.  Allowed ranges for options are
           "[-1.0, 1.0]". Defaults are 0.

       rimax
       gimax
       bimax
       aimax
           Adjust red, green, blue and alpha input white point.  Allowed ranges for options are
           "[-1.0, 1.0]". Defaults are 1.

           Input levels are used to lighten highlights (bright tones), darken shadows (dark
           tones), change the balance of bright and dark tones.

       romin
       gomin
       bomin
       aomin
           Adjust red, green, blue and alpha output black point.  Allowed ranges for options are
           "[0, 1.0]". Defaults are 0.

       romax
       gomax
       bomax
       aomax
           Adjust red, green, blue and alpha output white point.  Allowed ranges for options are
           "[0, 1.0]". Defaults are 1.

           Output levels allows manual selection of a constrained output level range.

       Examples

       ·   Make video output darker:

                   colorlevels=rimin=0.058:gimin=0.058:bimin=0.058

       ·   Increase contrast:

                   colorlevels=rimin=0.039:gimin=0.039:bimin=0.039:rimax=0.96:gimax=0.96:bimax=0.96

       ·   Make video output lighter:

                   colorlevels=rimax=0.902:gimax=0.902:bimax=0.902

       ·   Increase brightness:

                   colorlevels=romin=0.5:gomin=0.5:bomin=0.5

   colorchannelmixer
       Adjust video input frames by re-mixing color channels.

       This filter modifies a color channel by adding the values associated to the other channels
       of the same pixels. For example if the value to modify is red, the output value will be:

               <red>=<red>*<rr> + <blue>*<rb> + <green>*<rg> + <alpha>*<ra>

       The filter accepts the following options:

       rr
       rg
       rb
       ra  Adjust contribution of input red, green, blue and alpha channels for output red
           channel.  Default is 1 for rr, and 0 for rg, rb and ra.

       gr
       gg
       gb
       ga  Adjust contribution of input red, green, blue and alpha channels for output green
           channel.  Default is 1 for gg, and 0 for gr, gb and ga.

       br
       bg
       bb
       ba  Adjust contribution of input red, green, blue and alpha channels for output blue
           channel.  Default is 1 for bb, and 0 for br, bg and ba.

       ar
       ag
       ab
       aa  Adjust contribution of input red, green, blue and alpha channels for output alpha
           channel.  Default is 1 for aa, and 0 for ar, ag and ab.

           Allowed ranges for options are "[-2.0, 2.0]".

       Examples

       ·   Convert source to grayscale:

                   colorchannelmixer=.3:.4:.3:0:.3:.4:.3:0:.3:.4:.3

       ·   Simulate sepia tones:

                   colorchannelmixer=.393:.769:.189:0:.349:.686:.168:0:.272:.534:.131

   colormatrix
       Convert color matrix.

       The filter accepts the following options:

       src
       dst Specify the source and destination color matrix. Both values must be specified.

           The accepted values are:

           bt709
               BT.709

           fcc FCC

           bt601
               BT.601

           bt470
               BT.470

           bt470bg
               BT.470BG

           smpte170m
               SMPTE-170M

           smpte240m
               SMPTE-240M

           bt2020
               BT.2020

       For example to convert from BT.601 to SMPTE-240M, use the command:

               colormatrix=bt601:smpte240m

   colorspace
       Convert colorspace, transfer characteristics or color primaries.  Input video needs to
       have an even size.

       The filter accepts the following options:

       all Specify all color properties at once.

           The accepted values are:

           bt470m
               BT.470M

           bt470bg
               BT.470BG

           bt601-6-525
               BT.601-6 525

           bt601-6-625
               BT.601-6 625

           bt709
               BT.709

           smpte170m
               SMPTE-170M

           smpte240m
               SMPTE-240M

           bt2020
               BT.2020

       space
           Specify output colorspace.

           The accepted values are:

           bt709
               BT.709

           fcc FCC

           bt470bg
               BT.470BG or BT.601-6 625

           smpte170m
               SMPTE-170M or BT.601-6 525

           smpte240m
               SMPTE-240M

           ycgco
               YCgCo

           bt2020ncl
               BT.2020 with non-constant luminance

       trc Specify output transfer characteristics.

           The accepted values are:

           bt709
               BT.709

           bt470m
               BT.470M

           bt470bg
               BT.470BG

           gamma22
               Constant gamma of 2.2

           gamma28
               Constant gamma of 2.8

           smpte170m
               SMPTE-170M, BT.601-6 625 or BT.601-6 525

           smpte240m
               SMPTE-240M

           srgb
               SRGB

           iec61966-2-1
               iec61966-2-1

           iec61966-2-4
               iec61966-2-4

           xvycc
               xvycc

           bt2020-10
               BT.2020 for 10-bits content

           bt2020-12
               BT.2020 for 12-bits content

       primaries
           Specify output color primaries.

           The accepted values are:

           bt709
               BT.709

           bt470m
               BT.470M

           bt470bg
               BT.470BG or BT.601-6 625

           smpte170m
               SMPTE-170M or BT.601-6 525

           smpte240m
               SMPTE-240M

           film
               film

           smpte431
               SMPTE-431

           smpte432
               SMPTE-432

           bt2020
               BT.2020

           jedec-p22
               JEDEC P22 phosphors

       range
           Specify output color range.

           The accepted values are:

           tv  TV (restricted) range

           mpeg
               MPEG (restricted) range

           pc  PC (full) range

           jpeg
               JPEG (full) range

       format
           Specify output color format.

           The accepted values are:

           yuv420p
               YUV 4:2:0 planar 8-bits

           yuv420p10
               YUV 4:2:0 planar 10-bits

           yuv420p12
               YUV 4:2:0 planar 12-bits

           yuv422p
               YUV 4:2:2 planar 8-bits

           yuv422p10
               YUV 4:2:2 planar 10-bits

           yuv422p12
               YUV 4:2:2 planar 12-bits

           yuv444p
               YUV 4:4:4 planar 8-bits

           yuv444p10
               YUV 4:4:4 planar 10-bits

           yuv444p12
               YUV 4:4:4 planar 12-bits

       fast
           Do a fast conversion, which skips gamma/primary correction. This will take
           significantly less CPU, but will be mathematically incorrect. To get output compatible
           with that produced by the colormatrix filter, use fast=1.

       dither
           Specify dithering mode.

           The accepted values are:

           none
               No dithering

           fsb Floyd-Steinberg dithering

       wpadapt
           Whitepoint adaptation mode.

           The accepted values are:

           bradford
               Bradford whitepoint adaptation

           vonkries
               von Kries whitepoint adaptation

           identity
               identity whitepoint adaptation (i.e. no whitepoint adaptation)

       iall
           Override all input properties at once. Same accepted values as all.

       ispace
           Override input colorspace. Same accepted values as space.

       iprimaries
           Override input color primaries. Same accepted values as primaries.

       itrc
           Override input transfer characteristics. Same accepted values as trc.

       irange
           Override input color range. Same accepted values as range.

       The filter converts the transfer characteristics, color space and color primaries to the
       specified user values. The output value, if not specified, is set to a default value based
       on the "all" property. If that property is also not specified, the filter will log an
       error. The output color range and format default to the same value as the input color
       range and format. The input transfer characteristics, color space, color primaries and
       color range should be set on the input data. If any of these are missing, the filter will
       log an error and no conversion will take place.

       For example to convert the input to SMPTE-240M, use the command:

               colorspace=smpte240m

   convolution
       Apply convolution of 3x3, 5x5, 7x7 or horizontal/vertical up to 49 elements.

       The filter accepts the following options:

       0m
       1m
       2m
       3m  Set matrix for each plane.  Matrix is sequence of 9, 25 or 49 signed integers in
           square mode, and from 1 to 49 odd number of signed integers in row mode.

       0rdiv
       1rdiv
       2rdiv
       3rdiv
           Set multiplier for calculated value for each plane.  If unset or 0, it will be sum of
           all matrix elements.

       0bias
       1bias
       2bias
       3bias
           Set bias for each plane. This value is added to the result of the multiplication.
           Useful for making the overall image brighter or darker. Default is 0.0.

       0mode
       1mode
       2mode
       3mode
           Set matrix mode for each plane. Can be square, row or column.  Default is square.

       Examples

       ·   Apply sharpen:

                   convolution="0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0:0 -1 0 -1 5 -1 0 -1 0"

       ·   Apply blur:

                   convolution="1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1 1 1 1 1 1 1 1 1:1/9:1/9:1/9:1/9"

       ·   Apply edge enhance:

                   convolution="0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:0 0 0 -1 1 0 0 0 0:5:1:1:1:0:128:128:128"

       ·   Apply edge detect:

                   convolution="0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:0 1 0 1 -4 1 0 1 0:5:5:5:1:0:128:128:128"

       ·   Apply laplacian edge detector which includes diagonals:

                   convolution="1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:1 1 1 1 -8 1 1 1 1:5:5:5:1:0:128:128:0"

       ·   Apply emboss:

                   convolution="-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2:-2 -1 0 -1 1 1 0 1 2"

   convolve
       Apply 2D convolution of video stream in frequency domain using second stream as impulse.

       The filter accepts the following options:

       planes
           Set which planes to process.

       impulse
           Set which impulse video frames will be processed, can be first or all. Default is all.

       The "convolve" filter also supports the framesync options.

   copy
       Copy the input video source unchanged to the output. This is mainly useful for testing
       purposes.

   coreimage
       Video filtering on GPU using Apple's CoreImage API on OSX.

       Hardware acceleration is based on an OpenGL context. Usually, this means it is processed
       by video hardware. However, software-based OpenGL implementations exist which means there
       is no guarantee for hardware processing. It depends on the respective OSX.

       There are many filters and image generators provided by Apple that come with a large
       variety of options. The filter has to be referenced by its name along with its options.

       The coreimage filter accepts the following options:

       list_filters
           List all available filters and generators along with all their respective options as
           well as possible minimum and maximum values along with the default values.

                   list_filters=true

       filter
           Specify all filters by their respective name and options.  Use list_filters to
           determine all valid filter names and options.  Numerical options are specified by a
           float value and are automatically clamped to their respective value range.  Vector and
           color options have to be specified by a list of space separated float values.
           Character escaping has to be done.  A special option name "default" is available to
           use default options for a filter.

           It is required to specify either "default" or at least one of the filter options.  All
           omitted options are used with their default values.  The syntax of the filter string
           is as follows:

                   filter=<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...][#<NAME>@<OPTION>=<VALUE>[@<OPTION>=<VALUE>][@...]][#...]

       output_rect
           Specify a rectangle where the output of the filter chain is copied into the input
           image. It is given by a list of space separated float values:

                   output_rect=x\ y\ width\ height

           If not given, the output rectangle equals the dimensions of the input image.  The
           output rectangle is automatically cropped at the borders of the input image. Negative
           values are valid for each component.

                   output_rect=25\ 25\ 100\ 100

       Several filters can be chained for successive processing without GPU-HOST transfers
       allowing for fast processing of complex filter chains.  Currently, only filters with zero
       (generators) or exactly one (filters) input image and one output image are supported.
       Also, transition filters are not yet usable as intended.

       Some filters generate output images with additional padding depending on the respective
       filter kernel. The padding is automatically removed to ensure the filter output has the
       same size as the input image.

       For image generators, the size of the output image is determined by the previous output
       image of the filter chain or the input image of the whole filterchain, respectively. The
       generators do not use the pixel information of this image to generate their output.
       However, the generated output is blended onto this image, resulting in partial or complete
       coverage of the output image.

       The coreimagesrc video source can be used for generating input images which are directly
       fed into the filter chain. By using it, providing input images by another video source or
       an input video is not required.

       Examples

       ·   List all filters available:

                   coreimage=list_filters=true

       ·   Use the CIBoxBlur filter with default options to blur an image:

                   coreimage=filter=CIBoxBlur@default

       ·   Use a filter chain with CISepiaTone at default values and CIVignetteEffect with its
           center at 100x100 and a radius of 50 pixels:

                   coreimage=filter=CIBoxBlur@default#CIVignetteEffect@inputCenter=100\ 100@inputRadius=50

       ·   Use nullsrc and CIQRCodeGenerator to create a QR code for the FFmpeg homepage, given
           as complete and escaped command-line for Apple's standard bash shell:

                   ffmpeg -f lavfi -i nullsrc=s=100x100,coreimage=filter=CIQRCodeGenerator@inputMessage=https\\\\\://FFmpeg.org/@inputCorrectionLevel=H -frames:v 1 QRCode.png

   crop
       Crop the input video to given dimensions.

       It accepts the following parameters:

       w, out_w
           The width of the output video. It defaults to "iw".  This expression is evaluated only
           once during the filter configuration, or when the w or out_w command is sent.

       h, out_h
           The height of the output video. It defaults to "ih".  This expression is evaluated
           only once during the filter configuration, or when the h or out_h command is sent.

       x   The horizontal position, in the input video, of the left edge of the output video. It
           defaults to "(in_w-out_w)/2".  This expression is evaluated per-frame.

       y   The vertical position, in the input video, of the top edge of the output video.  It
           defaults to "(in_h-out_h)/2".  This expression is evaluated per-frame.

       keep_aspect
           If set to 1 will force the output display aspect ratio to be the same of the input, by
           changing the output sample aspect ratio. It defaults to 0.

       exact
           Enable exact cropping. If enabled, subsampled videos will be cropped at exact
           width/height/x/y as specified and will not be rounded to nearest smaller value.  It
           defaults to 0.

       The out_w, out_h, x, y parameters are expressions containing the following constants:

       x
       y   The computed values for x and y. They are evaluated for each new frame.

       in_w
       in_h
           The input width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output (cropped) width and height.

       ow
       oh  These are the same as out_w and out_h.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       n   The number of the input frame, starting from 0.

       pos the position in the file of the input frame, NAN if unknown

       t   The timestamp expressed in seconds. It's NAN if the input timestamp is unknown.

       The expression for out_w may depend on the value of out_h, and the expression for out_h
       may depend on out_w, but they cannot depend on x and y, as x and y are evaluated after
       out_w and out_h.

       The x and y parameters specify the expressions for the position of the top-left corner of
       the output (non-cropped) area. They are evaluated for each frame. If the evaluated value
       is not valid, it is approximated to the nearest valid value.

       The expression for x may depend on y, and the expression for y may depend on x.

       Examples

       ·   Crop area with size 100x100 at position (12,34).

                   crop=100:100:12:34

           Using named options, the example above becomes:

                   crop=w=100:h=100:x=12:y=34

       ·   Crop the central input area with size 100x100:

                   crop=100:100

       ·   Crop the central input area with size 2/3 of the input video:

                   crop=2/3*in_w:2/3*in_h

       ·   Crop the input video central square:

                   crop=out_w=in_h
                   crop=in_h

       ·   Delimit the rectangle with the top-left corner placed at position 100:100 and the
           right-bottom corner corresponding to the right-bottom corner of the input image.

                   crop=in_w-100:in_h-100:100:100

       ·   Crop 10 pixels from the left and right borders, and 20 pixels from the top and bottom
           borders

                   crop=in_w-2*10:in_h-2*20

       ·   Keep only the bottom right quarter of the input image:

                   crop=in_w/2:in_h/2:in_w/2:in_h/2

       ·   Crop height for getting Greek harmony:

                   crop=in_w:1/PHI*in_w

       ·   Apply trembling effect:

                   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)

       ·   Apply erratic camera effect depending on timestamp:

                   crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)"

       ·   Set x depending on the value of y:

                   crop=in_w/2:in_h/2:y:10+10*sin(n/10)

       Commands

       This filter supports the following commands:

       w, out_w
       h, out_h
       x
       y   Set width/height of the output video and the horizontal/vertical position in the input
           video.  The command accepts the same syntax of the corresponding option.

           If the specified expression is not valid, it is kept at its current value.

   cropdetect
       Auto-detect the crop size.

       It calculates the necessary cropping parameters and prints the recommended parameters via
       the logging system. The detected dimensions correspond to the non-black area of the input
       video.

       It accepts the following parameters:

       limit
           Set higher black value threshold, which can be optionally specified from nothing (0)
           to everything (255 for 8-bit based formats). An intensity value greater to the set
           value is considered non-black. It defaults to 24.  You can also specify a value
           between 0.0 and 1.0 which will be scaled depending on the bitdepth of the pixel
           format.

       round
           The value which the width/height should be divisible by. It defaults to 16. The offset
           is automatically adjusted to center the video. Use 2 to get only even dimensions
           (needed for 4:2:2 video). 16 is best when encoding to most video codecs.

       reset_count, reset
           Set the counter that determines after how many frames cropdetect will reset the
           previously detected largest video area and start over to detect the current optimal
           crop area. Default value is 0.

           This can be useful when channel logos distort the video area. 0 indicates 'never
           reset', and returns the largest area encountered during playback.

   cue
       Delay video filtering until a given wallclock timestamp. The filter first passes on
       preroll amount of frames, then it buffers at most buffer amount of frames and waits for
       the cue. After reaching the cue it forwards the buffered frames and also any subsequent
       frames coming in its input.

       The filter can be used synchronize the output of multiple ffmpeg processes for realtime
       output devices like decklink. By putting the delay in the filtering chain and pre-
       buffering frames the process can pass on data to output almost immediately after the
       target wallclock timestamp is reached.

       Perfect frame accuracy cannot be guaranteed, but the result is good enough for some use
       cases.

       cue The cue timestamp expressed in a UNIX timestamp in microseconds. Default is 0.

       preroll
           The duration of content to pass on as preroll expressed in seconds. Default is 0.

       buffer
           The maximum duration of content to buffer before waiting for the cue expressed in
           seconds. Default is 0.

   curves
       Apply color adjustments using curves.

       This filter is similar to the Adobe Photoshop and GIMP curves tools. Each component (red,
       green and blue) has its values defined by N key points tied from each other using a smooth
       curve. The x-axis represents the pixel values from the input frame, and the y-axis the new
       pixel values to be set for the output frame.

       By default, a component curve is defined by the two points (0;0) and (1;1). This creates a
       straight line where each original pixel value is "adjusted" to its own value, which means
       no change to the image.

       The filter allows you to redefine these two points and add some more. A new curve (using a
       natural cubic spline interpolation) will be define to pass smoothly through all these new
       coordinates. The new defined points needs to be strictly increasing over the x-axis, and
       their x and y values must be in the [0;1] interval.  If the computed curves happened to go
       outside the vector spaces, the values will be clipped accordingly.

       The filter accepts the following options:

       preset
           Select one of the available color presets. This option can be used in addition to the
           r, g, b parameters; in this case, the later options takes priority on the preset
           values.  Available presets are:

           none
           color_negative
           cross_process
           darker
           increase_contrast
           lighter
           linear_contrast
           medium_contrast
           negative
           strong_contrast
           vintage

           Default is "none".

       master, m
           Set the master key points. These points will define a second pass mapping. It is
           sometimes called a "luminance" or "value" mapping. It can be used with r, g, b or all
           since it acts like a post-processing LUT.

       red, r
           Set the key points for the red component.

       green, g
           Set the key points for the green component.

       blue, b
           Set the key points for the blue component.

       all Set the key points for all components (not including master).  Can be used in addition
           to the other key points component options. In this case, the unset component(s) will
           fallback on this all setting.

       psfile
           Specify a Photoshop curves file (".acv") to import the settings from.

       plot
           Save Gnuplot script of the curves in specified file.

       To avoid some filtergraph syntax conflicts, each key points list need to be defined using
       the following syntax: "x0/y0 x1/y1 x2/y2 ...".

       Examples

       ·   Increase slightly the middle level of blue:

                   curves=blue='0/0 0.5/0.58 1/1'

       ·   Vintage effect:

                   curves=r='0/0.11 .42/.51 1/0.95':g='0/0 0.50/0.48 1/1':b='0/0.22 .49/.44 1/0.8'

           Here we obtain the following coordinates for each components:

           red "(0;0.11) (0.42;0.51) (1;0.95)"

           green
               "(0;0) (0.50;0.48) (1;1)"

           blue
               "(0;0.22) (0.49;0.44) (1;0.80)"

       ·   The previous example can also be achieved with the associated built-in preset:

                   curves=preset=vintage

       ·   Or simply:

                   curves=vintage

       ·   Use a Photoshop preset and redefine the points of the green component:

                   curves=psfile='MyCurvesPresets/purple.acv':green='0/0 0.45/0.53 1/1'

       ·   Check out the curves of the "cross_process" profile using ffmpeg and gnuplot:

                   ffmpeg -f lavfi -i color -vf curves=cross_process:plot=/tmp/curves.plt -frames:v 1 -f null -
                   gnuplot -p /tmp/curves.plt

   datascope
       Video data analysis filter.

       This filter shows hexadecimal pixel values of part of video.

       The filter accepts the following options:

       size, s
           Set output video size.

       x   Set x offset from where to pick pixels.

       y   Set y offset from where to pick pixels.

       mode
           Set scope mode, can be one of the following:

           mono
               Draw hexadecimal pixel values with white color on black background.

           color
               Draw hexadecimal pixel values with input video pixel color on black background.

           color2
               Draw hexadecimal pixel values on color background picked from input video, the
               text color is picked in such way so its always visible.

       axis
           Draw rows and columns numbers on left and top of video.

       opacity
           Set background opacity.

   dctdnoiz
       Denoise frames using 2D DCT (frequency domain filtering).

       This filter is not designed for real time.

       The filter accepts the following options:

       sigma, s
           Set the noise sigma constant.

           This sigma defines a hard threshold of "3 * sigma"; every DCT coefficient (absolute
           value) below this threshold with be dropped.

           If you need a more advanced filtering, see expr.

           Default is 0.

       overlap
           Set number overlapping pixels for each block. Since the filter can be slow, you may
           want to reduce this value, at the cost of a less effective filter and the risk of
           various artefacts.

           If the overlapping value doesn't permit processing the whole input width or height, a
           warning will be displayed and according borders won't be denoised.

           Default value is blocksize-1, which is the best possible setting.

       expr, e
           Set the coefficient factor expression.

           For each coefficient of a DCT block, this expression will be evaluated as a multiplier
           value for the coefficient.

           If this is option is set, the sigma option will be ignored.

           The absolute value of the coefficient can be accessed through the c variable.

       n   Set the blocksize using the number of bits. "1<<n" defines the blocksize, which is the
           width and height of the processed blocks.

           The default value is 3 (8x8) and can be raised to 4 for a blocksize of 16x16. Note
           that changing this setting has huge consequences on the speed processing. Also, a
           larger block size does not necessarily means a better de-noising.

       Examples

       Apply a denoise with a sigma of 4.5:

               dctdnoiz=4.5

       The same operation can be achieved using the expression system:

               dctdnoiz=e='gte(c, 4.5*3)'

       Violent denoise using a block size of "16x16":

               dctdnoiz=15:n=4

   deband
       Remove banding artifacts from input video.  It works by replacing banded pixels with
       average value of referenced pixels.

       The filter accepts the following options:

       1thr
       2thr
       3thr
       4thr
           Set banding detection threshold for each plane. Default is 0.02.  Valid range is
           0.00003 to 0.5.  If difference between current pixel and reference pixel is less than
           threshold, it will be considered as banded.

       range, r
           Banding detection range in pixels. Default is 16. If positive, random number in range
           0 to set value will be used. If negative, exact absolute value will be used.  The
           range defines square of four pixels around current pixel.

       direction, d
           Set direction in radians from which four pixel will be compared. If positive, random
           direction from 0 to set direction will be picked. If negative, exact of absolute value
           will be picked. For example direction 0, -PI or -2*PI radians will pick only pixels on
           same row and -PI/2 will pick only pixels on same column.

       blur, b
           If enabled, current pixel is compared with average value of all four surrounding
           pixels. The default is enabled. If disabled current pixel is compared with all four
           surrounding pixels. The pixel is considered banded if only all four differences with
           surrounding pixels are less than threshold.

       coupling, c
           If enabled, current pixel is changed if and only if all pixel components are banded,
           e.g. banding detection threshold is triggered for all color components.  The default
           is disabled.

   deblock
       Remove blocking artifacts from input video.

       The filter accepts the following options:

       filter
           Set filter type, can be weak or strong. Default is strong.  This controls what kind of
           deblocking is applied.

       block
           Set size of block, allowed range is from 4 to 512. Default is 8.

       alpha
       beta
       gamma
       delta
           Set blocking detection thresholds. Allowed range is 0 to 1.  Defaults are: 0.098 for
           alpha and 0.05 for the rest.  Using higher threshold gives more deblocking strength.
           Setting alpha controls threshold detection at exact edge of block.  Remaining options
           controls threshold detection near the edge. Each one for below/above or left/right.
           Setting any of those to 0 disables deblocking.

       planes
           Set planes to filter. Default is to filter all available planes.

       Examples

       ·   Deblock using weak filter and block size of 4 pixels.

                   deblock=filter=weak:block=4

       ·   Deblock using strong filter, block size of 4 pixels and custom thresholds for
           deblocking more edges.

                   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05

       ·   Similar as above, but filter only first plane.

                   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=1

       ·   Similar as above, but filter only second and third plane.

                   deblock=filter=strong:block=4:alpha=0.12:beta=0.07:gamma=0.06:delta=0.05:planes=6

   decimate
       Drop duplicated frames at regular intervals.

       The filter accepts the following options:

       cycle
           Set the number of frames from which one will be dropped. Setting this to N means one
           frame in every batch of N frames will be dropped.  Default is 5.

       dupthresh
           Set the threshold for duplicate detection. If the difference metric for a frame is
           less than or equal to this value, then it is declared as duplicate. Default is 1.1

       scthresh
           Set scene change threshold. Default is 15.

       blockx
       blocky
           Set the size of the x and y-axis blocks used during metric calculations.  Larger
           blocks give better noise suppression, but also give worse detection of small
           movements. Must be a power of two. Default is 32.

       ppsrc
           Mark main input as a pre-processed input and activate clean source input stream. This
           allows the input to be pre-processed with various filters to help the metrics
           calculation while keeping the frame selection lossless. When set to 1, the first
           stream is for the pre-processed input, and the second stream is the clean source from
           where the kept frames are chosen. Default is 0.

       chroma
           Set whether or not chroma is considered in the metric calculations. Default is 1.

   deconvolve
       Apply 2D deconvolution of video stream in frequency domain using second stream as impulse.

       The filter accepts the following options:

       planes
           Set which planes to process.

       impulse
           Set which impulse video frames will be processed, can be first or all. Default is all.

       noise
           Set noise when doing divisions. Default is 0.0000001. Useful when width and height are
           not same and not power of 2 or if stream prior to convolving had noise.

       The "deconvolve" filter also supports the framesync options.

   deflate
       Apply deflate effect to the video.

       This filter replaces the pixel by the local(3x3) average by taking into account only
       values lower than the pixel.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
           Limit the maximum change for each plane, default is 65535.  If 0, plane will remain
           unchanged.

   deflicker
       Remove temporal frame luminance variations.

       It accepts the following options:

       size, s
           Set moving-average filter size in frames. Default is 5. Allowed range is 2 - 129.

       mode, m
           Set averaging mode to smooth temporal luminance variations.

           Available values are:

           am  Arithmetic mean

           gm  Geometric mean

           hm  Harmonic mean

           qm  Quadratic mean

           cm  Cubic mean

           pm  Power mean

           median
               Median

       bypass
           Do not actually modify frame. Useful when one only wants metadata.

   dejudder
       Remove judder produced by partially interlaced telecined content.

       Judder can be introduced, for instance, by pullup filter. If the original source was
       partially telecined content then the output of "pullup,dejudder" will have a variable
       frame rate. May change the recorded frame rate of the container. Aside from that change,
       this filter will not affect constant frame rate video.

       The option available in this filter is:

       cycle
           Specify the length of the window over which the judder repeats.

           Accepts any integer greater than 1. Useful values are:

           4   If the original was telecined from 24 to 30 fps (Film to NTSC).

           5   If the original was telecined from 25 to 30 fps (PAL to NTSC).

           20  If a mixture of the two.

           The default is 4.

   delogo
       Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a
       rectangle covering the logo and watch it disappear (and sometimes something even uglier
       appear - your mileage may vary).

       It accepts the following parameters:

       x
       y   Specify the top left corner coordinates of the logo. They must be specified.

       w
       h   Specify the width and height of the logo to clear. They must be specified.

       band, t
           Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The
           default value is 1. This option is deprecated, setting higher values should no longer
           be necessary and is not recommended.

       show
           When set to 1, a green rectangle is drawn on the screen to simplify finding the right
           x, y, w, and h parameters.  The default value is 0.

           The rectangle is drawn on the outermost pixels which will be (partly) replaced with
           interpolated values. The values of the next pixels immediately outside this rectangle
           in each direction will be used to compute the interpolated pixel values inside the
           rectangle.

       Examples

       ·   Set a rectangle covering the area with top left corner coordinates 0,0 and size
           100x77, and a band of size 10:

                   delogo=x=0:y=0:w=100:h=77:band=10

   deshake
       Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove
       camera shake from hand-holding a camera, bumping a tripod, moving on a vehicle, etc.

       The filter accepts the following options:

       x
       y
       w
       h   Specify a rectangular area where to limit the search for motion vectors.  If desired
           the search for motion vectors can be limited to a rectangular area of the frame
           defined by its top left corner, width and height. These parameters have the same
           meaning as the drawbox filter which can be used to visualise the position of the
           bounding box.

           This is useful when simultaneous movement of subjects within the frame might be
           confused for camera motion by the motion vector search.

           If any or all of x, y, w and h are set to -1 then the full frame is used. This allows
           later options to be set without specifying the bounding box for the motion vector
           search.

           Default - search the whole frame.

       rx
       ry  Specify the maximum extent of movement in x and y directions in the range 0-64 pixels.
           Default 16.

       edge
           Specify how to generate pixels to fill blanks at the edge of the frame. Available
           values are:

           blank, 0
               Fill zeroes at blank locations

           original, 1
               Original image at blank locations

           clamp, 2
               Extruded edge value at blank locations

           mirror, 3
               Mirrored edge at blank locations

           Default value is mirror.

       blocksize
           Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.

       contrast
           Specify the contrast threshold for blocks. Only blocks with more than the specified
           contrast (difference between darkest and lightest pixels) will be considered. Range
           1-255, default 125.

       search
           Specify the search strategy. Available values are:

           exhaustive, 0
               Set exhaustive search

           less, 1
               Set less exhaustive search.

           Default value is exhaustive.

       filename
           If set then a detailed log of the motion search is written to the specified file.

   despill
       Remove unwanted contamination of foreground colors, caused by reflected color of
       greenscreen or bluescreen.

       This filter accepts the following options:

       type
           Set what type of despill to use.

       mix Set how spillmap will be generated.

       expand
           Set how much to get rid of still remaining spill.

       red Controls amount of red in spill area.

       green
           Controls amount of green in spill area.  Should be -1 for greenscreen.

       blue
           Controls amount of blue in spill area.  Should be -1 for bluescreen.

       brightness
           Controls brightness of spill area, preserving colors.

       alpha
           Modify alpha from generated spillmap.

   detelecine
       Apply an exact inverse of the telecine operation. It requires a predefined pattern
       specified using the pattern option which must be the same as that passed to the telecine
       filter.

       This filter accepts the following options:

       first_field
           top, t
               top field first

           bottom, b
               bottom field first The default value is "top".

       pattern
           A string of numbers representing the pulldown pattern you wish to apply.  The default
           value is 23.

       start_frame
           A number representing position of the first frame with respect to the telecine
           pattern. This is to be used if the stream is cut. The default value is 0.

   dilation
       Apply dilation effect to the video.

       This filter replaces the pixel by the local(3x3) maximum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
           Limit the maximum change for each plane, default is 65535.  If 0, plane will remain
           unchanged.

       coordinates
           Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are
           used.

           Flags to local 3x3 coordinates maps like this:

               1 2 3
               4   5
               6 7 8

   displace
       Displace pixels as indicated by second and third input stream.

       It takes three input streams and outputs one stream, the first input is the source, and
       second and third input are displacement maps.

       The second input specifies how much to displace pixels along the x-axis, while the third
       input specifies how much to displace pixels along the y-axis.  If one of displacement map
       streams terminates, last frame from that displacement map will be used.

       Note that once generated, displacements maps can be reused over and over again.

       A description of the accepted options follows.

       edge
           Set displace behavior for pixels that are out of range.

           Available values are:

           blank
               Missing pixels are replaced by black pixels.

           smear
               Adjacent pixels will spread out to replace missing pixels.

           wrap
               Out of range pixels are wrapped so they point to pixels of other side.

           mirror
               Out of range pixels will be replaced with mirrored pixels.

           Default is smear.

       Examples

       ·   Add ripple effect to rgb input of video size hd720:

                   ffmpeg -i INPUT -f lavfi -i nullsrc=s=hd720,lutrgb=128:128:128 -f lavfi -i nullsrc=s=hd720,geq='r=128+30*sin(2*PI*X/400+T):g=128+30*sin(2*PI*X/400+T):b=128+30*sin(2*PI*X/400+T)' -lavfi '[0][1][2]displace' OUTPUT

       ·   Add wave effect to rgb input of video size hd720:

                   ffmpeg -i INPUT -f lavfi -i nullsrc=hd720,geq='r=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):g=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T)):b=128+80*(sin(sqrt((X-W/2)*(X-W/2)+(Y-H/2)*(Y-H/2))/220*2*PI+T))' -lavfi '[1]split[x][y],[0][x][y]displace' OUTPUT

   drawbox
       Draw a colored box on the input image.

       It accepts the following parameters:

       x
       y   The expressions which specify the top left corner coordinates of the box. It defaults
           to 0.

       width, w
       height, h
           The expressions which specify the width and height of the box; if 0 they are
           interpreted as the input width and height. It defaults to 0.

       color, c
           Specify the color of the box to write. For the general syntax of this option, check
           the "Color" section in the ffmpeg-utils manual. If the special value "invert" is used,
           the box edge color is the same as the video with inverted luma.

       thickness, t
           The expression which sets the thickness of the box edge.  A value of "fill" will
           create a filled box. Default value is 3.

           See below for the list of accepted constants.

       replace
           Applicable if the input has alpha. With value 1, the pixels of the painted box will
           overwrite the video's color and alpha pixels.  Default is 0, which composites the box
           onto the input, leaving the video's alpha intact.

       The parameters for x, y, w and h and t are expressions containing the following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
           The input width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y offset coordinates where the box is drawn.

       w
       h   The width and height of the drawn box.

       t   The thickness of the drawn box.

           These constants allow the x, y, w, h and t expressions to refer to each other, so you
           may for example specify "y=x/dar" or "h=w/dar".

       Examples

       ·   Draw a black box around the edge of the input image:

                   drawbox

       ·   Draw a box with color red and an opacity of 50%:

                   drawbox=10:20:200:60:red@0.5

           The previous example can be specified as:

                   drawbox=x=10:y=20:w=200:h=60:color=red@0.5

       ·   Fill the box with pink color:

                   drawbox=x=10:y=10:w=100:h=100:color=pink@0.5:t=fill

       ·   Draw a 2-pixel red 2.40:1 mask:

                   drawbox=x=-t:y=0.5*(ih-iw/2.4)-t:w=iw+t*2:h=iw/2.4+t*2:t=2:c=red

   drawgrid
       Draw a grid on the input image.

       It accepts the following parameters:

       x
       y   The expressions which specify the coordinates of some point of grid intersection
           (meant to configure offset). Both default to 0.

       width, w
       height, h
           The expressions which specify the width and height of the grid cell, if 0 they are
           interpreted as the input width and height, respectively, minus "thickness", so image
           gets framed. Default to 0.

       color, c
           Specify the color of the grid. For the general syntax of this option, check the
           "Color" section in the ffmpeg-utils manual. If the special value "invert" is used, the
           grid color is the same as the video with inverted luma.

       thickness, t
           The expression which sets the thickness of the grid line. Default value is 1.

           See below for the list of accepted constants.

       replace
           Applicable if the input has alpha. With 1 the pixels of the painted grid will
           overwrite the video's color and alpha pixels.  Default is 0, which composites the grid
           onto the input, leaving the video's alpha intact.

       The parameters for x, y, w and h and t are expressions containing the following constants:

       dar The input display aspect ratio, it is the same as (w / h) * sar.

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       in_h, ih
       in_w, iw
           The input grid cell width and height.

       sar The input sample aspect ratio.

       x
       y   The x and y coordinates of some point of grid intersection (meant to configure
           offset).

       w
       h   The width and height of the drawn cell.

       t   The thickness of the drawn cell.

           These constants allow the x, y, w, h and t expressions to refer to each other, so you
           may for example specify "y=x/dar" or "h=w/dar".

       Examples

       ·   Draw a grid with cell 100x100 pixels, thickness 2 pixels, with color red and an
           opacity of 50%:

                   drawgrid=width=100:height=100:thickness=2:color=red@0.5

       ·   Draw a white 3x3 grid with an opacity of 50%:

                   drawgrid=w=iw/3:h=ih/3:t=2:c=white@0.5

   drawtext
       Draw a text string or text from a specified file on top of a video, using the libfreetype
       library.

       To enable compilation of this filter, you need to configure FFmpeg with
       "--enable-libfreetype".  To enable default font fallback and the font option you need to
       configure FFmpeg with "--enable-libfontconfig".  To enable the text_shaping option, you
       need to configure FFmpeg with "--enable-libfribidi".

       Syntax

       It accepts the following parameters:

       box Used to draw a box around text using the background color.  The value must be either 1
           (enable) or 0 (disable).  The default value of box is 0.

       boxborderw
           Set the width of the border to be drawn around the box using boxcolor.  The default
           value of boxborderw is 0.

       boxcolor
           The color to be used for drawing box around text. For the syntax of this option, check
           the "Color" section in the ffmpeg-utils manual.

           The default value of boxcolor is "white".

       line_spacing
           Set the line spacing in pixels of the border to be drawn around the box using box.
           The default value of line_spacing is 0.

       borderw
           Set the width of the border to be drawn around the text using bordercolor.  The
           default value of borderw is 0.

       bordercolor
           Set the color to be used for drawing border around text. For the syntax of this
           option, check the "Color" section in the ffmpeg-utils manual.

           The default value of bordercolor is "black".

       expansion
           Select how the text is expanded. Can be either "none", "strftime" (deprecated) or
           "normal" (default). See the drawtext_expansion, Text expansion section below for
           details.

       basetime
           Set a start time for the count. Value is in microseconds. Only applied in the
           deprecated strftime expansion mode. To emulate in normal expansion mode use the "pts"
           function, supplying the start time (in seconds) as the second argument.

       fix_bounds
           If true, check and fix text coords to avoid clipping.

       fontcolor
           The color to be used for drawing fonts. For the syntax of this option, check the
           "Color" section in the ffmpeg-utils manual.

           The default value of fontcolor is "black".

       fontcolor_expr
           String which is expanded the same way as text to obtain dynamic fontcolor value. By
           default this option has empty value and is not processed. When this option is set, it
           overrides fontcolor option.

       font
           The font family to be used for drawing text. By default Sans.

       fontfile
           The font file to be used for drawing text. The path must be included.  This parameter
           is mandatory if the fontconfig support is disabled.

       alpha
           Draw the text applying alpha blending. The value can be a number between 0.0 and 1.0.
           The expression accepts the same variables x, y as well.  The default value is 1.
           Please see fontcolor_expr.

       fontsize
           The font size to be used for drawing text.  The default value of fontsize is 16.

       text_shaping
           If set to 1, attempt to shape the text (for example, reverse the order of right-to-
           left text and join Arabic characters) before drawing it.  Otherwise, just draw the
           text exactly as given.  By default 1 (if supported).

       ft_load_flags
           The flags to be used for loading the fonts.

           The flags map the corresponding flags supported by libfreetype, and are a combination
           of the following values:

           default
           no_scale
           no_hinting
           render
           no_bitmap
           vertical_layout
           force_autohint
           crop_bitmap
           pedantic
           ignore_global_advance_width
           no_recurse
           ignore_transform
           monochrome
           linear_design
           no_autohint

           Default value is "default".

           For more information consult the documentation for the FT_LOAD_* libfreetype flags.

       shadowcolor
           The color to be used for drawing a shadow behind the drawn text. For the syntax of
           this option, check the "Color" section in the ffmpeg-utils manual.

           The default value of shadowcolor is "black".

       shadowx
       shadowy
           The x and y offsets for the text shadow position with respect to the position of the
           text. They can be either positive or negative values. The default value for both is
           "0".

       start_number
           The starting frame number for the n/frame_num variable. The default value is "0".

       tabsize
           The size in number of spaces to use for rendering the tab.  Default value is 4.

       timecode
           Set the initial timecode representation in "hh:mm:ss[:;.]ff" format. It can be used
           with or without text parameter. timecode_rate option must be specified.

       timecode_rate, rate, r
           Set the timecode frame rate (timecode only). Value will be rounded to nearest integer.
           Minimum value is "1".  Drop-frame timecode is supported for frame rates 30 & 60.

       tc24hmax
           If set to 1, the output of the timecode option will wrap around at 24 hours.  Default
           is 0 (disabled).

       text
           The text string to be drawn. The text must be a sequence of UTF-8 encoded characters.
           This parameter is mandatory if no file is specified with the parameter textfile.

       textfile
           A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded
           characters.

           This parameter is mandatory if no text string is specified with the parameter text.

           If both text and textfile are specified, an error is thrown.

       reload
           If set to 1, the textfile will be reloaded before each frame.  Be sure to update it
           atomically, or it may be read partially, or even fail.

       x
       y   The expressions which specify the offsets where text will be drawn within the video
           frame. They are relative to the top/left border of the output image.

           The default value of x and y is "0".

           See below for the list of accepted constants and functions.

       The parameters for x and y are expressions containing the following constants and
       functions:

       dar input display aspect ratio, it is the same as (w / h) * sar

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       line_h, lh
           the height of each text line

       main_h, h, H
           the input height

       main_w, w, W
           the input width

       max_glyph_a, ascent
           the maximum distance from the baseline to the highest/upper grid coordinate used to
           place a glyph outline point, for all the rendered glyphs.  It is a positive value, due
           to the grid's orientation with the Y axis upwards.

       max_glyph_d, descent
           the maximum distance from the baseline to the lowest grid coordinate used to place a
           glyph outline point, for all the rendered glyphs.  This is a negative value, due to
           the grid's orientation, with the Y axis upwards.

       max_glyph_h
           maximum glyph height, that is the maximum height for all the glyphs contained in the
           rendered text, it is equivalent to ascent - descent.

       max_glyph_w
           maximum glyph width, that is the maximum width for all the glyphs contained in the
           rendered text

       n   the number of input frame, starting from 0

       rand(min, max)
           return a random number included between min and max

       sar The input sample aspect ratio.

       t   timestamp expressed in seconds, NAN if the input timestamp is unknown

       text_h, th
           the height of the rendered text

       text_w, tw
           the width of the rendered text

       x
       y   the x and y offset coordinates where the text is drawn.

           These parameters allow the x and y expressions to refer each other, so you can for
           example specify "y=x/dar".

       Text expansion

       If expansion is set to "strftime", the filter recognizes strftime() sequences in the
       provided text and expands them accordingly. Check the documentation of strftime(). This
       feature is deprecated.

       If expansion is set to "none", the text is printed verbatim.

       If expansion is set to "normal" (which is the default), the following expansion mechanism
       is used.

       The backslash character \, followed by any character, always expands to the second
       character.

       Sequences of the form "%{...}" are expanded. The text between the braces is a function
       name, possibly followed by arguments separated by ':'.  If the arguments contain special
       characters or delimiters (':' or '}'), they should be escaped.

       Note that they probably must also be escaped as the value for the text option in the
       filter argument string and as the filter argument in the filtergraph description, and
       possibly also for the shell, that makes up to four levels of escaping; using a text file
       avoids these problems.

       The following functions are available:

       expr, e
           The expression evaluation result.

           It must take one argument specifying the expression to be evaluated, which accepts the
           same constants and functions as the x and y values. Note that not all constants should
           be used, for example the text size is not known when evaluating the expression, so the
           constants text_w and text_h will have an undefined value.

       expr_int_format, eif
           Evaluate the expression's value and output as formatted integer.

           The first argument is the expression to be evaluated, just as for the expr function.
           The second argument specifies the output format. Allowed values are x, X, d and u.
           They are treated exactly as in the "printf" function.  The third parameter is optional
           and sets the number of positions taken by the output.  It can be used to add padding
           with zeros from the left.

       gmtime
           The time at which the filter is running, expressed in UTC.  It can accept an argument:
           a strftime() format string.

       localtime
           The time at which the filter is running, expressed in the local time zone.  It can
           accept an argument: a strftime() format string.

       metadata
           Frame metadata. Takes one or two arguments.

           The first argument is mandatory and specifies the metadata key.

           The second argument is optional and specifies a default value, used when the metadata
           key is not found or empty.

       n, frame_num
           The frame number, starting from 0.

       pict_type
           A 1 character description of the current picture type.

       pts The timestamp of the current frame.  It can take up to three arguments.

           The first argument is the format of the timestamp; it defaults to "flt" for seconds as
           a decimal number with microsecond accuracy; "hms" stands for a formatted
           [-]HH:MM:SS.mmm timestamp with millisecond accuracy.  "gmtime" stands for the
           timestamp of the frame formatted as UTC time; "localtime" stands for the timestamp of
           the frame formatted as local time zone time.

           The second argument is an offset added to the timestamp.

           If the format is set to "hms", a third argument "24HH" may be supplied to present the
           hour part of the formatted timestamp in 24h format (00-23).

           If the format is set to "localtime" or "gmtime", a third argument may be supplied: a
           strftime() format string.  By default, YYYY-MM-DD HH:MM:SS format will be used.

       Examples

       ·   Draw "Test Text" with font FreeSerif, using the default values for the optional
           parameters.

                   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

       ·   Draw 'Test Text' with font FreeSerif of size 24 at position x=100 and y=50 (counting
           from the top-left corner of the screen), text is yellow with a red box around it. Both
           the text and the box have an opacity of 20%.

                   drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
                             x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"

           Note that the double quotes are not necessary if spaces are not used within the
           parameter list.

       ·   Show the text at the center of the video frame:

                   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h)/2"

       ·   Show the text at a random position, switching to a new position every 30 seconds:

                   drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=if(eq(mod(t\,30)\,0)\,rand(0\,(w-text_w))\,x):y=if(eq(mod(t\,30)\,0)\,rand(0\,(h-text_h))\,y)"

       ·   Show a text line sliding from right to left in the last row of the video frame. The
           file LONG_LINE is assumed to contain a single line with no newlines.

                   drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t"

       ·   Show the content of file CREDITS off the bottom of the frame and scroll up.

                   drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t"

       ·   Draw a single green letter "g", at the center of the input video.  The glyph baseline
           is placed at half screen height.

                   drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent"

       ·   Show text for 1 second every 3 seconds:

                   drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:enable=lt(mod(t\,3)\,1):text='blink'"

       ·   Use fontconfig to set the font. Note that the colons need to be escaped.

                   drawtext='fontfile=Linux Libertine O-40\:style=Semibold:text=FFmpeg'

       ·   Print the date of a real-time encoding (see strftime(3)):

                   drawtext='fontfile=FreeSans.ttf:text=%{localtime\:%a %b %d %Y}'

       ·   Show text fading in and out (appearing/disappearing):

                   #!/bin/sh
                   DS=1.0 # display start
                   DE=10.0 # display end
                   FID=1.5 # fade in duration
                   FOD=5 # fade out duration
                   ffplay -f lavfi "color,drawtext=text=TEST:fontsize=50:fontfile=FreeSerif.ttf:fontcolor_expr=ff0000%{eif\\\\: clip(255*(1*between(t\\, $DS + $FID\\, $DE - $FOD) + ((t - $DS)/$FID)*between(t\\, $DS\\, $DS + $FID) + (-(t - $DE)/$FOD)*between(t\\, $DE - $FOD\\, $DE) )\\, 0\\, 255) \\\\: x\\\\: 2 }"

       ·   Horizontally align multiple separate texts. Note that max_glyph_a and the fontsize
           value are included in the y offset.

                   drawtext=fontfile=FreeSans.ttf:text=DOG:fontsize=24:x=10:y=20+24-max_glyph_a,
                   drawtext=fontfile=FreeSans.ttf:text=cow:fontsize=24:x=80:y=20+24-max_glyph_a

       For more information about libfreetype, check: <http://www.freetype.org/>.

       For more information about fontconfig, check:
       <http://freedesktop.org/software/fontconfig/fontconfig-user.html>.

       For more information about libfribidi, check: <http://fribidi.org/>.

   edgedetect
       Detect and draw edges. The filter uses the Canny Edge Detection algorithm.

       The filter accepts the following options:

       low
       high
           Set low and high threshold values used by the Canny thresholding algorithm.

           The high threshold selects the "strong" edge pixels, which are then connected through
           8-connectivity with the "weak" edge pixels selected by the low threshold.

           low and high threshold values must be chosen in the range [0,1], and low should be
           lesser or equal to high.

           Default value for low is "20/255", and default value for high is "50/255".

       mode
           Define the drawing mode.

           wires
               Draw white/gray wires on black background.

           colormix
               Mix the colors to create a paint/cartoon effect.

           canny
               Apply Canny edge detector on all selected planes.

           Default value is wires.

       planes
           Select planes for filtering. By default all available planes are filtered.

       Examples

       ·   Standard edge detection with custom values for the hysteresis thresholding:

                   edgedetect=low=0.1:high=0.4

       ·   Painting effect without thresholding:

                   edgedetect=mode=colormix:high=0

   eq
       Set brightness, contrast, saturation and approximate gamma adjustment.

       The filter accepts the following options:

       contrast
           Set the contrast expression. The value must be a float value in range "-2.0" to 2.0.
           The default value is "1".

       brightness
           Set the brightness expression. The value must be a float value in range "-1.0" to 1.0.
           The default value is "0".

       saturation
           Set the saturation expression. The value must be a float in range 0.0 to 3.0. The
           default value is "1".

       gamma
           Set the gamma expression. The value must be a float in range 0.1 to 10.0.  The default
           value is "1".

       gamma_r
           Set the gamma expression for red. The value must be a float in range 0.1 to 10.0. The
           default value is "1".

       gamma_g
           Set the gamma expression for green. The value must be a float in range 0.1 to 10.0.
           The default value is "1".

       gamma_b
           Set the gamma expression for blue. The value must be a float in range 0.1 to 10.0. The
           default value is "1".

       gamma_weight
           Set the gamma weight expression. It can be used to reduce the effect of a high gamma
           value on bright image areas, e.g. keep them from getting overamplified and just plain
           white. The value must be a float in range 0.0 to 1.0. A value of 0.0 turns the gamma
           correction all the way down while 1.0 leaves it at its full strength. Default is "1".

       eval
           Set when the expressions for brightness, contrast, saturation and gamma expressions
           are evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization or when a command
               is processed

           frame
               evaluate expressions for each incoming frame

           Default value is init.

       The expressions accept the following parameters:

       n   frame count of the input frame starting from 0

       pos byte position of the corresponding packet in the input file, NAN if unspecified

       r   frame rate of the input video, NAN if the input frame rate is unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is unknown

       Commands

       The filter supports the following commands:

       contrast
           Set the contrast expression.

       brightness
           Set the brightness expression.

       saturation
           Set the saturation expression.

       gamma
           Set the gamma expression.

       gamma_r
           Set the gamma_r expression.

       gamma_g
           Set gamma_g expression.

       gamma_b
           Set gamma_b expression.

       gamma_weight
           Set gamma_weight expression.

           The command accepts the same syntax of the corresponding option.

           If the specified expression is not valid, it is kept at its current value.

   erosion
       Apply erosion effect to the video.

       This filter replaces the pixel by the local(3x3) minimum.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
           Limit the maximum change for each plane, default is 65535.  If 0, plane will remain
           unchanged.

       coordinates
           Flag which specifies the pixel to refer to. Default is 255 i.e. all eight pixels are
           used.

           Flags to local 3x3 coordinates maps like this:

               1 2 3
               4   5
               6 7 8

   extractplanes
       Extract color channel components from input video stream into separate grayscale video
       streams.

       The filter accepts the following option:

       planes
           Set plane(s) to extract.

           Available values for planes are:

           y
           u
           v
           a
           r
           g
           b

           Choosing planes not available in the input will result in an error.  That means you
           cannot select "r", "g", "b" planes with "y", "u", "v" planes at same time.

       Examples

       ·   Extract luma, u and v color channel component from input video frame into 3 grayscale
           outputs:

                   ffmpeg -i video.avi -filter_complex 'extractplanes=y+u+v[y][u][v]' -map '[y]' y.avi -map '[u]' u.avi -map '[v]' v.avi

   elbg
       Apply a posterize effect using the ELBG (Enhanced LBG) algorithm.

       For each input image, the filter will compute the optimal mapping from the input to the
       output given the codebook length, that is the number of distinct output colors.

       This filter accepts the following options.

       codebook_length, l
           Set codebook length. The value must be a positive integer, and represents the number
           of distinct output colors. Default value is 256.

       nb_steps, n
           Set the maximum number of iterations to apply for computing the optimal mapping. The
           higher the value the better the result and the higher the computation time. Default
           value is 1.

       seed, s
           Set a random seed, must be an integer included between 0 and UINT32_MAX. If not
           specified, or if explicitly set to -1, the filter will try to use a good random seed
           on a best effort basis.

       pal8
           Set pal8 output pixel format. This option does not work with codebook length greater
           than 256.

   entropy
       Measure graylevel entropy in histogram of color channels of video frames.

       It accepts the following parameters:

       mode
           Can be either normal or diff. Default is normal.

           diff mode measures entropy of histogram delta values, absolute differences between
           neighbour histogram values.

   fade
       Apply a fade-in/out effect to the input video.

       It accepts the following parameters:

       type, t
           The effect type can be either "in" for a fade-in, or "out" for a fade-out effect.
           Default is "in".

       start_frame, s
           Specify the number of the frame to start applying the fade effect at. Default is 0.

       nb_frames, n
           The number of frames that the fade effect lasts. At the end of the fade-in effect, the
           output video will have the same intensity as the input video.  At the end of the fade-
           out transition, the output video will be filled with the selected color.  Default is
           25.

       alpha
           If set to 1, fade only alpha channel, if one exists on the input.  Default value is 0.

       start_time, st
           Specify the timestamp (in seconds) of the frame to start to apply the fade effect. If
           both start_frame and start_time are specified, the fade will start at whichever comes
           last.  Default is 0.

       duration, d
           The number of seconds for which the fade effect has to last. At the end of the fade-in
           effect the output video will have the same intensity as the input video, at the end of
           the fade-out transition the output video will be filled with the selected color.  If
           both duration and nb_frames are specified, duration is used. Default is 0 (nb_frames
           is used by default).

       color, c
           Specify the color of the fade. Default is "black".

       Examples

       ·   Fade in the first 30 frames of video:

                   fade=in:0:30

           The command above is equivalent to:

                   fade=t=in:s=0:n=30

       ·   Fade out the last 45 frames of a 200-frame video:

                   fade=out:155:45
                   fade=type=out:start_frame=155:nb_frames=45

       ·   Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video:

                   fade=in:0:25, fade=out:975:25

       ·   Make the first 5 frames yellow, then fade in from frame 5-24:

                   fade=in:5:20:color=yellow

       ·   Fade in alpha over first 25 frames of video:

                   fade=in:0:25:alpha=1

       ·   Make the first 5.5 seconds black, then fade in for 0.5 seconds:

                   fade=t=in:st=5.5:d=0.5

   fftfilt
       Apply arbitrary expressions to samples in frequency domain

       dc_Y
           Adjust the dc value (gain) of the luma plane of the image. The filter accepts an
           integer value in range 0 to 1000. The default value is set to 0.

       dc_U
           Adjust the dc value (gain) of the 1st chroma plane of the image. The filter accepts an
           integer value in range 0 to 1000. The default value is set to 0.

       dc_V
           Adjust the dc value (gain) of the 2nd chroma plane of the image. The filter accepts an
           integer value in range 0 to 1000. The default value is set to 0.

       weight_Y
           Set the frequency domain weight expression for the luma plane.

       weight_U
           Set the frequency domain weight expression for the 1st chroma plane.

       weight_V
           Set the frequency domain weight expression for the 2nd chroma plane.

       eval
           Set when the expressions are evaluated.

           It accepts the following values:

           init
               Only evaluate expressions once during the filter initialization.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

           The filter accepts the following variables:

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       N   The number of input frame, starting from 0.

       Examples

       ·   High-pass:

                   fftfilt=dc_Y=128:weight_Y='squish(1-(Y+X)/100)'

       ·   Low-pass:

                   fftfilt=dc_Y=0:weight_Y='squish((Y+X)/100-1)'

       ·   Sharpen:

                   fftfilt=dc_Y=0:weight_Y='1+squish(1-(Y+X)/100)'

       ·   Blur:

                   fftfilt=dc_Y=0:weight_Y='exp(-4 * ((Y+X)/(W+H)))'

   fftdnoiz
       Denoise frames using 3D FFT (frequency domain filtering).

       The filter accepts the following options:

       sigma
           Set the noise sigma constant. This sets denoising strength.  Default value is 1.
           Allowed range is from 0 to 30.  Using very high sigma with low overlap may give
           blocking artifacts.

       amount
           Set amount of denoising. By default all detected noise is reduced.  Default value is
           1. Allowed range is from 0 to 1.

       block
           Set size of block, Default is 4, can be 3, 4, 5 or 6.  Actual size of block in pixels
           is 2 to power of block, so by default block size in pixels is 2^4 which is 16.

       overlap
           Set block overlap. Default is 0.5. Allowed range is from 0.2 to 0.8.

       prev
           Set number of previous frames to use for denoising. By default is set to 0.

       next
           Set number of next frames to to use for denoising. By default is set to 0.

       planes
           Set planes which will be filtered, by default are all available filtered except alpha.

   field
       Extract a single field from an interlaced image using stride arithmetic to avoid wasting
       CPU time. The output frames are marked as non-interlaced.

       The filter accepts the following options:

       type
           Specify whether to extract the top (if the value is 0 or "top") or the bottom field
           (if the value is 1 or "bottom").

   fieldhint
       Create new frames by copying the top and bottom fields from surrounding frames supplied as
       numbers by the hint file.

       hint
           Set file containing hints: absolute/relative frame numbers.

           There must be one line for each frame in a clip. Each line must contain two numbers
           separated by the comma, optionally followed by "-" or "+".  Numbers supplied on each
           line of file can not be out of [N-1,N+1] where N is current frame number for
           "absolute" mode or out of [-1, 1] range for "relative" mode. First number tells from
           which frame to pick up top field and second number tells from which frame to pick up
           bottom field.

           If optionally followed by "+" output frame will be marked as interlaced, else if
           followed by "-" output frame will be marked as progressive, else it will be marked
           same as input frame.  If line starts with "#" or ";" that line is skipped.

       mode
           Can be item "absolute" or "relative". Default is "absolute".

       Example of first several lines of "hint" file for "relative" mode:

               0,0 - # first frame
               1,0 - # second frame, use third's frame top field and second's frame bottom field
               1,0 - # third frame, use fourth's frame top field and third's frame bottom field
               1,0 -
               0,0 -
               0,0 -
               1,0 -
               1,0 -
               1,0 -
               0,0 -
               0,0 -
               1,0 -
               1,0 -
               1,0 -
               0,0 -

   fieldmatch
       Field matching filter for inverse telecine. It is meant to reconstruct the progressive
       frames from a telecined stream. The filter does not drop duplicated frames, so to achieve
       a complete inverse telecine "fieldmatch" needs to be followed by a decimation filter such
       as decimate in the filtergraph.

       The separation of the field matching and the decimation is notably motivated by the
       possibility of inserting a de-interlacing filter fallback between the two.  If the source
       has mixed telecined and real interlaced content, "fieldmatch" will not be able to match
       fields for the interlaced parts.  But these remaining combed frames will be marked as
       interlaced, and thus can be de-interlaced by a later filter such as yadif before
       decimation.

       In addition to the various configuration options, "fieldmatch" can take an optional second
       stream, activated through the ppsrc option. If enabled, the frames reconstruction will be
       based on the fields and frames from this second stream. This allows the first input to be
       pre-processed in order to help the various algorithms of the filter, while keeping the
       output lossless (assuming the fields are matched properly). Typically, a field-aware
       denoiser, or brightness/contrast adjustments can help.

       Note that this filter uses the same algorithms as TIVTC/TFM (AviSynth project) and
       VIVTC/VFM (VapourSynth project). The later is a light clone of TFM from which "fieldmatch"
       is based on. While the semantic and usage are very close, some behaviour and options names
       can differ.

       The decimate filter currently only works for constant frame rate input.  If your input has
       mixed telecined (30fps) and progressive content with a lower framerate like 24fps use the
       following filterchain to produce the necessary cfr stream:
       "dejudder,fps=30000/1001,fieldmatch,decimate".

       The filter accepts the following options:

       order
           Specify the assumed field order of the input stream. Available values are:

           auto
               Auto detect parity (use FFmpeg's internal parity value).

           bff Assume bottom field first.

           tff Assume top field first.

           Note that it is sometimes recommended not to trust the parity announced by the stream.

           Default value is auto.

       mode
           Set the matching mode or strategy to use. pc mode is the safest in the sense that it
           won't risk creating jerkiness due to duplicate frames when possible, but if there are
           bad edits or blended fields it will end up outputting combed frames when a good match
           might actually exist. On the other hand, pcn_ub mode is the most risky in terms of
           creating jerkiness, but will almost always find a good frame if there is one. The
           other values are all somewhere in between pc and pcn_ub in terms of risking jerkiness
           and creating duplicate frames versus finding good matches in sections with bad edits,
           orphaned fields, blended fields, etc.

           More details about p/c/n/u/b are available in p/c/n/u/b meaning section.

           Available values are:

           pc  2-way matching (p/c)

           pc_n
               2-way matching, and trying 3rd match if still combed (p/c + n)

           pc_u
               2-way matching, and trying 3rd match (same order) if still combed (p/c + u)

           pc_n_ub
               2-way matching, trying 3rd match if still combed, and trying 4th/5th matches if
               still combed (p/c + n + u/b)

           pcn 3-way matching (p/c/n)

           pcn_ub
               3-way matching, and trying 4th/5th matches if all 3 of the original matches are
               detected as combed (p/c/n + u/b)

           The parenthesis at the end indicate the matches that would be used for that mode
           assuming order=tff (and field on auto or top).

           In terms of speed pc mode is by far the fastest and pcn_ub is the slowest.

           Default value is pc_n.

       ppsrc
           Mark the main input stream as a pre-processed input, and enable the secondary input
           stream as the clean source to pick the fields from. See the filter introduction for
           more details. It is similar to the clip2 feature from VFM/TFM.

           Default value is 0 (disabled).

       field
           Set the field to match from. It is recommended to set this to the same value as order
           unless you experience matching failures with that setting. In certain circumstances
           changing the field that is used to match from can have a large impact on matching
           performance. Available values are:

           auto
               Automatic (same value as order).

           bottom
               Match from the bottom field.

           top Match from the top field.

           Default value is auto.

       mchroma
           Set whether or not chroma is included during the match comparisons. In most cases it
           is recommended to leave this enabled. You should set this to 0 only if your clip has
           bad chroma problems such as heavy rainbowing or other artifacts. Setting this to 0
           could also be used to speed things up at the cost of some accuracy.

           Default value is 1.

       y0
       y1  These define an exclusion band which excludes the lines between y0 and y1 from being
           included in the field matching decision. An exclusion band can be used to ignore
           subtitles, a logo, or other things that may interfere with the matching. y0 sets the
           starting scan line and y1 sets the ending line; all lines in between y0 and y1
           (including y0 and y1) will be ignored. Setting y0 and y1 to the same value will
           disable the feature.  y0 and y1 defaults to 0.

       scthresh
           Set the scene change detection threshold as a percentage of maximum change on the luma
           plane. Good values are in the "[8.0, 14.0]" range. Scene change detection is only
           relevant in case combmatch=sc.  The range for scthresh is "[0.0, 100.0]".

           Default value is 12.0.

       combmatch
           When combatch is not none, "fieldmatch" will take into account the combed scores of
           matches when deciding what match to use as the final match. Available values are:

           none
               No final matching based on combed scores.

           sc  Combed scores are only used when a scene change is detected.

           full
               Use combed scores all the time.

           Default is sc.

       combdbg
           Force "fieldmatch" to calculate the combed metrics for certain matches and print them.
           This setting is known as micout in TFM/VFM vocabulary.  Available values are:

           none
               No forced calculation.

           pcn Force p/c/n calculations.

           pcnub
               Force p/c/n/u/b calculations.

           Default value is none.

       cthresh
           This is the area combing threshold used for combed frame detection. This essentially
           controls how "strong" or "visible" combing must be to be detected.  Larger values mean
           combing must be more visible and smaller values mean combing can be less visible or
           strong and still be detected. Valid settings are from "-1" (every pixel will be
           detected as combed) to 255 (no pixel will be detected as combed). This is basically a
           pixel difference value. A good range is "[8, 12]".

           Default value is 9.

       chroma
           Sets whether or not chroma is considered in the combed frame decision.  Only disable
           this if your source has chroma problems (rainbowing, etc.) that are causing problems
           for the combed frame detection with chroma enabled. Actually, using chroma=0 is
           usually more reliable, except for the case where there is chroma only combing in the
           source.

           Default value is 0.

       blockx
       blocky
           Respectively set the x-axis and y-axis size of the window used during combed frame
           detection. This has to do with the size of the area in which combpel pixels are
           required to be detected as combed for a frame to be declared combed. See the combpel
           parameter description for more info.  Possible values are any number that is a power
           of 2 starting at 4 and going up to 512.

           Default value is 16.

       combpel
           The number of combed pixels inside any of the blocky by blockx size blocks on the
           frame for the frame to be detected as combed. While cthresh controls how "visible" the
           combing must be, this setting controls "how much" combing there must be in any
           localized area (a window defined by the blockx and blocky settings) on the frame.
           Minimum value is 0 and maximum is "blocky x blockx" (at which point no frames will
           ever be detected as combed). This setting is known as MI in TFM/VFM vocabulary.

           Default value is 80.

       p/c/n/u/b meaning

       p/c/n

       We assume the following telecined stream:

               Top fields:     1 2 2 3 4
               Bottom fields:  1 2 3 4 4

       The numbers correspond to the progressive frame the fields relate to. Here, the first two
       frames are progressive, the 3rd and 4th are combed, and so on.

       When "fieldmatch" is configured to run a matching from bottom (field=bottom) this is how
       this input stream get transformed:

               Input stream:
                               T     1 2 2 3 4
                               B     1 2 3 4 4   <-- matching reference

               Matches:              c c n n c

               Output stream:
                               T     1 2 3 4 4
                               B     1 2 3 4 4

       As a result of the field matching, we can see that some frames get duplicated.  To perform
       a complete inverse telecine, you need to rely on a decimation filter after this operation.
       See for instance the decimate filter.

       The same operation now matching from top fields (field=top) looks like this:

               Input stream:
                               T     1 2 2 3 4   <-- matching reference
                               B     1 2 3 4 4

               Matches:              c c p p c

               Output stream:
                               T     1 2 2 3 4
                               B     1 2 2 3 4

       In these examples, we can see what p, c and n mean; basically, they refer to the frame and
       field of the opposite parity:

       *<p matches the field of the opposite parity in the previous frame>
       *<c matches the field of the opposite parity in the current frame>
       *<n matches the field of the opposite parity in the next frame>

       u/b

       The u and b matching are a bit special in the sense that they match from the opposite
       parity flag. In the following examples, we assume that we are currently matching the 2nd
       frame (Top:2, bottom:2). According to the match, a 'x' is placed above and below each
       matched fields.

       With bottom matching (field=bottom):

               Match:           c         p           n          b          u

                                x       x               x        x          x
                 Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
                 Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                                x         x           x        x              x

               Output frames:
                                2          1          2          2          2
                                2          2          2          1          3

       With top matching (field=top):

               Match:           c         p           n          b          u

                                x         x           x        x              x
                 Top          1 2 2     1 2 2       1 2 2      1 2 2      1 2 2
                 Bottom       1 2 3     1 2 3       1 2 3      1 2 3      1 2 3
                                x       x               x        x          x

               Output frames:
                                2          2          2          1          2
                                2          1          3          2          2

       Examples

       Simple IVTC of a top field first telecined stream:

               fieldmatch=order=tff:combmatch=none, decimate

       Advanced IVTC, with fallback on yadif for still combed frames:

               fieldmatch=order=tff:combmatch=full, yadif=deint=interlaced, decimate

   fieldorder
       Transform the field order of the input video.

       It accepts the following parameters:

       order
           The output field order. Valid values are tff for top field first or bff for bottom
           field first.

       The default value is tff.

       The transformation is done by shifting the picture content up or down by one line, and
       filling the remaining line with appropriate picture content.  This method is consistent
       with most broadcast field order converters.

       If the input video is not flagged as being interlaced, or it is already flagged as being
       of the required output field order, then this filter does not alter the incoming video.

       It is very useful when converting to or from PAL DV material, which is bottom field first.

       For example:

               ffmpeg -i in.vob -vf "fieldorder=bff" out.dv

   fifo, afifo
       Buffer input images and send them when they are requested.

       It is mainly useful when auto-inserted by the libavfilter framework.

       It does not take parameters.

   fillborders
       Fill borders of the input video, without changing video stream dimensions.  Sometimes
       video can have garbage at the four edges and you may not want to crop video input to keep
       size multiple of some number.

       This filter accepts the following options:

       left
           Number of pixels to fill from left border.

       right
           Number of pixels to fill from right border.

       top Number of pixels to fill from top border.

       bottom
           Number of pixels to fill from bottom border.

       mode
           Set fill mode.

           It accepts the following values:

           smear
               fill pixels using outermost pixels

           mirror
               fill pixels using mirroring

           fixed
               fill pixels with constant value

           Default is smear.

       color
           Set color for pixels in fixed mode. Default is black.

   find_rect
       Find a rectangular object

       It accepts the following options:

       object
           Filepath of the object image, needs to be in gray8.

       threshold
           Detection threshold, default is 0.5.

       mipmaps
           Number of mipmaps, default is 3.

       xmin, ymin, xmax, ymax
           Specifies the rectangle in which to search.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

                   ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

   cover_rect
       Cover a rectangular object

       It accepts the following options:

       cover
           Filepath of the optional cover image, needs to be in yuv420.

       mode
           Set covering mode.

           It accepts the following values:

           cover
               cover it by the supplied image

           blur
               cover it by interpolating the surrounding pixels

           Default value is blur.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

                   ffmpeg -i file.ts -vf find_rect=newref.pgm,cover_rect=cover.jpg:mode=cover new.mkv

   floodfill
       Flood area with values of same pixel components with another values.

       It accepts the following options:

       x   Set pixel x coordinate.

       y   Set pixel y coordinate.

       s0  Set source #0 component value.

       s1  Set source #1 component value.

       s2  Set source #2 component value.

       s3  Set source #3 component value.

       d0  Set destination #0 component value.

       d1  Set destination #1 component value.

       d2  Set destination #2 component value.

       d3  Set destination #3 component value.

   format
       Convert the input video to one of the specified pixel formats.  Libavfilter will try to
       pick one that is suitable as input to the next filter.

       It accepts the following parameters:

       pix_fmts
           A '|'-separated list of pixel format names, such as "pix_fmts=yuv420p|monow|rgb24".

       Examples

       ·   Convert the input video to the yuv420p format

                   format=pix_fmts=yuv420p

           Convert the input video to any of the formats in the list

                   format=pix_fmts=yuv420p|yuv444p|yuv410p

   fps
       Convert the video to specified constant frame rate by duplicating or dropping frames as
       necessary.

       It accepts the following parameters:

       fps The desired output frame rate. The default is 25.

       start_time
           Assume the first PTS should be the given value, in seconds. This allows for
           padding/trimming at the start of stream. By default, no assumption is made about the
           first frame's expected PTS, so no padding or trimming is done.  For example, this
           could be set to 0 to pad the beginning with duplicates of the first frame if a video
           stream starts after the audio stream or to trim any frames with a negative PTS.

       round
           Timestamp (PTS) rounding method.

           Possible values are:

           zero
               round towards 0

           inf round away from 0

           down
               round towards -infinity

           up  round towards +infinity

           near
               round to nearest

           The default is "near".

       eof_action
           Action performed when reading the last frame.

           Possible values are:

           round
               Use same timestamp rounding method as used for other frames.

           pass
               Pass through last frame if input duration has not been reached yet.

           The default is "round".

       Alternatively, the options can be specified as a flat string: fps[:start_time[:round]].

       See also the setpts filter.

       Examples

       ·   A typical usage in order to set the fps to 25:

                   fps=fps=25

       ·   Sets the fps to 24, using abbreviation and rounding method to round to nearest:

                   fps=fps=film:round=near

   framepack
       Pack two different video streams into a stereoscopic video, setting proper metadata on
       supported codecs. The two views should have the same size and framerate and processing
       will stop when the shorter video ends. Please note that you may conveniently adjust view
       properties with the scale and fps filters.

       It accepts the following parameters:

       format
           The desired packing format. Supported values are:

           sbs The views are next to each other (default).

           tab The views are on top of each other.

           lines
               The views are packed by line.

           columns
               The views are packed by column.

           frameseq
               The views are temporally interleaved.

       Some examples:

               # Convert left and right views into a frame-sequential video
               ffmpeg -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT

               # Convert views into a side-by-side video with the same output resolution as the input
               ffmpeg -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

   framerate
       Change the frame rate by interpolating new video output frames from the source frames.

       This filter is not designed to function correctly with interlaced media. If you wish to
       change the frame rate of interlaced media then you are required to deinterlace before this
       filter and re-interlace after this filter.

       A description of the accepted options follows.

       fps Specify the output frames per second. This option can also be specified as a value
           alone. The default is 50.

       interp_start
           Specify the start of a range where the output frame will be created as a linear
           interpolation of two frames. The range is [0-255], the default is 15.

       interp_end
           Specify the end of a range where the output frame will be created as a linear
           interpolation of two frames. The range is [0-255], the default is 240.

       scene
           Specify the level at which a scene change is detected as a value between 0 and 100 to
           indicate a new scene; a low value reflects a low probability for the current frame to
           introduce a new scene, while a higher value means the current frame is more likely to
           be one.  The default is 8.2.

       flags
           Specify flags influencing the filter process.

           Available value for flags is:

           scene_change_detect, scd
               Enable scene change detection using the value of the option scene.  This flag is
               enabled by default.

   framestep
       Select one frame every N-th frame.

       This filter accepts the following option:

       step
           Select frame after every "step" frames.  Allowed values are positive integers higher
           than 0. Default value is 1.

   frei0r
       Apply a frei0r effect to the input video.

       To enable the compilation of this filter, you need to install the frei0r header and
       configure FFmpeg with "--enable-frei0r".

       It accepts the following parameters:

       filter_name
           The name of the frei0r effect to load. If the environment variable FREI0R_PATH is
           defined, the frei0r effect is searched for in each of the directories specified by the
           colon-separated list in FREI0R_PATH.  Otherwise, the standard frei0r paths are
           searched, in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/,
           /usr/lib/frei0r-1/.

       filter_params
           A '|'-separated list of parameters to pass to the frei0r effect.

       A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a
       color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and
       1.0, inclusive) or a color description as specified in the "Color" section in the ffmpeg-
       utils manual, a position (specified as X/Y, where X and Y are floating point numbers)
       and/or a string.

       The number and types of parameters depend on the loaded effect. If an effect parameter is
       not specified, the default value is set.

       Examples

       ·   Apply the distort0r effect, setting the first two double parameters:

                   frei0r=filter_name=distort0r:filter_params=0.5|0.01

       ·   Apply the colordistance effect, taking a color as the first parameter:

                   frei0r=colordistance:0.2/0.3/0.4
                   frei0r=colordistance:violet
                   frei0r=colordistance:0x112233

       ·   Apply the perspective effect, specifying the top left and top right image positions:

                   frei0r=perspective:0.2/0.2|0.8/0.2

       For more information, see <http://frei0r.dyne.org>

   fspp
       Apply fast and simple postprocessing. It is a faster version of spp.

       It splits (I)DCT into horizontal/vertical passes. Unlike the simple post- processing
       filter, one of them is performed once per block, not per pixel.  This allows for much
       higher speed.

       The filter accepts the following options:

       quality
           Set quality. This option defines the number of levels for averaging. It accepts an
           integer in the range 4-5. Default value is 4.

       qp  Force a constant quantization parameter. It accepts an integer in range 0-63.  If not
           set, the filter will use the QP from the video stream (if available).

       strength
           Set filter strength. It accepts an integer in range -15 to 32. Lower values mean more
           details but also more artifacts, while higher values make the image smoother but also
           blurrier. Default value is 0 X PSNR optimal.

       use_bframe_qp
           Enable the use of the QP from the B-Frames if set to 1. Using this option may cause
           flicker since the B-Frames have often larger QP. Default is 0 (not enabled).

   gblur
       Apply Gaussian blur filter.

       The filter accepts the following options:

       sigma
           Set horizontal sigma, standard deviation of Gaussian blur. Default is 0.5.

       steps
           Set number of steps for Gaussian approximation. Defauls is 1.

       planes
           Set which planes to filter. By default all planes are filtered.

       sigmaV
           Set vertical sigma, if negative it will be same as "sigma".  Default is "-1".

   geq
       Apply generic equation to each pixel.

       The filter accepts the following options:

       lum_expr, lum
           Set the luminance expression.

       cb_expr, cb
           Set the chrominance blue expression.

       cr_expr, cr
           Set the chrominance red expression.

       alpha_expr, a
           Set the alpha expression.

       red_expr, r
           Set the red expression.

       green_expr, g
           Set the green expression.

       blue_expr, b
           Set the blue expression.

       The colorspace is selected according to the specified options. If one of the lum_expr,
       cb_expr, or cr_expr options is specified, the filter will automatically select a YCbCr
       colorspace. If one of the red_expr, green_expr, or blue_expr options is specified, it will
       select an RGB colorspace.

       If one of the chrominance expression is not defined, it falls back on the other one. If no
       alpha expression is specified it will evaluate to opaque value.  If none of chrominance
       expressions are specified, they will evaluate to the luminance expression.

       The expressions can use the following variables and functions:

       N   The sequential number of the filtered frame, starting from 0.

       X
       Y   The coordinates of the current sample.

       W
       H   The width and height of the image.

       SW
       SH  Width and height scale depending on the currently filtered plane. It is the ratio
           between the corresponding luma plane number of pixels and the current plane ones. E.g.
           for YUV4:2:0 the values are "1,1" for the luma plane, and "0.5,0.5" for chroma planes.

       T   Time of the current frame, expressed in seconds.

       p(x, y)
           Return the value of the pixel at location (x,y) of the current plane.

       lum(x, y)
           Return the value of the pixel at location (x,y) of the luminance plane.

       cb(x, y)
           Return the value of the pixel at location (x,y) of the blue-difference chroma plane.
           Return 0 if there is no such plane.

       cr(x, y)
           Return the value of the pixel at location (x,y) of the red-difference chroma plane.
           Return 0 if there is no such plane.

       r(x, y)
       g(x, y)
       b(x, y)
           Return the value of the pixel at location (x,y) of the red/green/blue component.
           Return 0 if there is no such component.

       alpha(x, y)
           Return the value of the pixel at location (x,y) of the alpha plane. Return 0 if there
           is no such plane.

       For functions, if x and y are outside the area, the value will be automatically clipped to
       the closer edge.

       Examples

       ·   Flip the image horizontally:

                   geq=p(W-X\,Y)

       ·   Generate a bidimensional sine wave, with angle "PI/3" and a wavelength of 100 pixels:

                   geq=128 + 100*sin(2*(PI/100)*(cos(PI/3)*(X-50*T) + sin(PI/3)*Y)):128:128

       ·   Generate a fancy enigmatic moving light:

                   nullsrc=s=256x256,geq=random(1)/hypot(X-cos(N*0.07)*W/2-W/2\,Y-sin(N*0.09)*H/2-H/2)^2*1000000*sin(N*0.02):128:128

       ·   Generate a quick emboss effect:

                   format=gray,geq=lum_expr='(p(X,Y)+(256-p(X-4,Y-4)))/2'

       ·   Modify RGB components depending on pixel position:

                   geq=r='X/W*r(X,Y)':g='(1-X/W)*g(X,Y)':b='(H-Y)/H*b(X,Y)'

       ·   Create a radial gradient that is the same size as the input (also see the vignette
           filter):

                   geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

   gradfun
       Fix the banding artifacts that are sometimes introduced into nearly flat regions by
       truncation to 8-bit color depth.  Interpolate the gradients that should go where the bands
       are, and dither them.

       It is designed for playback only.  Do not use it prior to lossy compression, because
       compression tends to lose the dither and bring back the bands.

       It accepts the following parameters:

       strength
           The maximum amount by which the filter will change any one pixel. This is also the
           threshold for detecting nearly flat regions. Acceptable values range from .51 to 64;
           the default value is 1.2. Out-of-range values will be clipped to the valid range.

       radius
           The neighborhood to fit the gradient to. A larger radius makes for smoother gradients,
           but also prevents the filter from modifying the pixels near detailed regions.
           Acceptable values are 8-32; the default value is 16. Out-of-range values will be
           clipped to the valid range.

       Alternatively, the options can be specified as a flat string: strength[:radius]

       Examples

       ·   Apply the filter with a 3.5 strength and radius of 8:

                   gradfun=3.5:8

       ·   Specify radius, omitting the strength (which will fall-back to the default value):

                   gradfun=radius=8

   graphmonitor, agraphmonitor
       Show various filtergraph stats.

       With this filter one can debug complete filtergraph.  Especially issues with links filling
       with queued frames.

       The filter accepts the following options:

       size, s
           Set video output size. Default is hd720.

       opacity, o
           Set video opacity. Default is 0.9. Allowed range is from 0 to 1.

       mode, m
           Set output mode, can be fulll or compact.  In compact mode only filters with some
           queued frames have displayed stats.

       flags, f
           Set flags which enable which stats are shown in video.

           Available values for flags are:

           queue
               Display number of queued frames in each link.

           frame_count_in
               Display number of frames taken from filter.

           frame_count_out
               Display number of frames given out from filter.

           pts Display current filtered frame pts.

           time
               Display current filtered frame time.

           timebase
               Display time base for filter link.

           format
               Display used format for filter link.

           size
               Display video size or number of audio channels in case of audio used by filter
               link.

           rate
               Display video frame rate or sample rate in case of audio used by filter link.

       rate, r
           Set upper limit for video rate of output stream, Default value is 25.  This guarantee
           that output video frame rate will not be higher than this value.

   greyedge
       A color constancy variation filter which estimates scene illumination via grey edge
       algorithm and corrects the scene colors accordingly.

       See: <https://staff.science.uva.nl/th.gevers/pub/GeversTIP07.pdf>

       The filter accepts the following options:

       difford
           The order of differentiation to be applied on the scene. Must be chosen in the range
           [0,2] and default value is 1.

       minknorm
           The Minkowski parameter to be used for calculating the Minkowski distance. Must be
           chosen in the range [0,20] and default value is 1. Set to 0 for getting max value
           instead of calculating Minkowski distance.

       sigma
           The standard deviation of Gaussian blur to be applied on the scene. Must be chosen in
           the range [0,1024.0] and default value = 1. floor( sigma * break_off_sigma(3) ) can't
           be euqal to 0 if difford is greater than 0.

       Examples

       ·   Grey Edge:

                   greyedge=difford=1:minknorm=5:sigma=2

       ·   Max Edge:

                   greyedge=difford=1:minknorm=0:sigma=2

   haldclut
       Apply a Hald CLUT to a video stream.

       First input is the video stream to process, and second one is the Hald CLUT.  The Hald
       CLUT input can be a simple picture or a complete video stream.

       The filter accepts the following options:

       shortest
           Force termination when the shortest input terminates. Default is 0.

       repeatlast
           Continue applying the last CLUT after the end of the stream. A value of 0 disable the
           filter after the last frame of the CLUT is reached.  Default is 1.

       "haldclut" also has the same interpolation options as lut3d (both filters share the same
       internals).

       More information about the Hald CLUT can be found on Eskil Steenberg's website (Hald CLUT
       author) at <http://www.quelsolaar.com/technology/clut.html>.

       Workflow examples

       Hald CLUT video stream

       Generate an identity Hald CLUT stream altered with various effects:

               ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "hue=H=2*PI*t:s=sin(2*PI*t)+1, curves=cross_process" -t 10 -c:v ffv1 clut.nut

       Note: make sure you use a lossless codec.

       Then use it with "haldclut" to apply it on some random stream:

               ffmpeg -f lavfi -i mandelbrot -i clut.nut -filter_complex '[0][1] haldclut' -t 20 mandelclut.mkv

       The Hald CLUT will be applied to the 10 first seconds (duration of clut.nut), then the
       latest picture of that CLUT stream will be applied to the remaining frames of the
       "mandelbrot" stream.

       Hald CLUT with preview

       A Hald CLUT is supposed to be a squared image of "Level*Level*Level" by
       "Level*Level*Level" pixels. For a given Hald CLUT, FFmpeg will select the biggest possible
       square starting at the top left of the picture. The remaining padding pixels (bottom or
       right) will be ignored. This area can be used to add a preview of the Hald CLUT.

       Typically, the following generated Hald CLUT will be supported by the "haldclut" filter:

               ffmpeg -f lavfi -i B<haldclutsrc>=8 -vf "
                  pad=iw+320 [padded_clut];
                  smptebars=s=320x256, split [a][b];
                  [padded_clut][a] overlay=W-320:h, curves=color_negative [main];
                  [main][b] overlay=W-320" -frames:v 1 clut.png

       It contains the original and a preview of the effect of the CLUT: SMPTE color bars are
       displayed on the right-top, and below the same color bars processed by the color changes.

       Then, the effect of this Hald CLUT can be visualized with:

               ffplay input.mkv -vf "movie=clut.png, [in] haldclut"

   hflip
       Flip the input video horizontally.

       For example, to horizontally flip the input video with ffmpeg:

               ffmpeg -i in.avi -vf "hflip" out.avi

   histeq
       This filter applies a global color histogram equalization on a per-frame basis.

       It can be used to correct video that has a compressed range of pixel intensities.  The
       filter redistributes the pixel intensities to equalize their distribution across the
       intensity range. It may be viewed as an "automatically adjusting contrast filter". This
       filter is useful only for correcting degraded or poorly captured source video.

       The filter accepts the following options:

       strength
           Determine the amount of equalization to be applied.  As the strength is reduced, the
           distribution of pixel intensities more-and-more approaches that of the input frame.
           The value must be a float number in the range [0,1] and defaults to 0.200.

       intensity
           Set the maximum intensity that can generated and scale the output values
           appropriately.  The strength should be set as desired and then the intensity can be
           limited if needed to avoid washing-out. The value must be a float number in the range
           [0,1] and defaults to 0.210.

       antibanding
           Set the antibanding level. If enabled the filter will randomly vary the luminance of
           output pixels by a small amount to avoid banding of the histogram. Possible values are
           "none", "weak" or "strong". It defaults to "none".

   histogram
       Compute and draw a color distribution histogram for the input video.

       The computed histogram is a representation of the color component distribution in an
       image.

       Standard histogram displays the color components distribution in an image.  Displays color
       graph for each color component. Shows distribution of the Y, U, V, A or R, G, B
       components, depending on input format, in the current frame. Below each graph a color
       component scale meter is shown.

       The filter accepts the following options:

       level_height
           Set height of level. Default value is 200.  Allowed range is [50, 2048].

       scale_height
           Set height of color scale. Default value is 12.  Allowed range is [0, 40].

       display_mode
           Set display mode.  It accepts the following values:

           stack
               Per color component graphs are placed below each other.

           parade
               Per color component graphs are placed side by side.

           overlay
               Presents information identical to that in the "parade", except that the graphs
               representing color components are superimposed directly over one another.

           Default is "stack".

       levels_mode
           Set mode. Can be either "linear", or "logarithmic".  Default is "linear".

       components
           Set what color components to display.  Default is 7.

       fgopacity
           Set foreground opacity. Default is 0.7.

       bgopacity
           Set background opacity. Default is 0.5.

       Examples

       ·   Calculate and draw histogram:

                   ffplay -i input -vf histogram

   hqdn3d
       This is a high precision/quality 3d denoise filter. It aims to reduce image noise,
       producing smooth images and making still images really still. It should enhance
       compressibility.

       It accepts the following optional parameters:

       luma_spatial
           A non-negative floating point number which specifies spatial luma strength.  It
           defaults to 4.0.

       chroma_spatial
           A non-negative floating point number which specifies spatial chroma strength.  It
           defaults to 3.0*luma_spatial/4.0.

       luma_tmp
           A floating point number which specifies luma temporal strength. It defaults to
           6.0*luma_spatial/4.0.

       chroma_tmp
           A floating point number which specifies chroma temporal strength. It defaults to
           luma_tmp*chroma_spatial/luma_spatial.

   hwdownload
       Download hardware frames to system memory.

       The input must be in hardware frames, and the output a non-hardware format.  Not all
       formats will be supported on the output - it may be necessary to insert an additional
       format filter immediately following in the graph to get the output in a supported format.

   hwmap
       Map hardware frames to system memory or to another device.

       This filter has several different modes of operation; which one is used depends on the
       input and output formats:

       ·   Hardware frame input, normal frame output

           Map the input frames to system memory and pass them to the output.  If the original
           hardware frame is later required (for example, after overlaying something else on part
           of it), the hwmap filter can be used again in the next mode to retrieve it.

       ·   Normal frame input, hardware frame output

           If the input is actually a software-mapped hardware frame, then unmap it - that is,
           return the original hardware frame.

           Otherwise, a device must be provided.  Create new hardware surfaces on that device for
           the output, then map them back to the software format at the input and give those
           frames to the preceding filter.  This will then act like the hwupload filter, but may
           be able to avoid an additional copy when the input is already in a compatible format.

       ·   Hardware frame input and output

           A device must be supplied for the output, either directly or with the derive_device
           option.  The input and output devices must be of different types and compatible - the
           exact meaning of this is system-dependent, but typically it means that they must refer
           to the same underlying hardware context (for example, refer to the same graphics
           card).

           If the input frames were originally created on the output device, then unmap to
           retrieve the original frames.

           Otherwise, map the frames to the output device - create new hardware frames on the
           output corresponding to the frames on the input.

       The following additional parameters are accepted:

       mode
           Set the frame mapping mode.  Some combination of:

           read
               The mapped frame should be readable.

           write
               The mapped frame should be writeable.

           overwrite
               The mapping will always overwrite the entire frame.

               This may improve performance in some cases, as the original contents of the frame
               need not be loaded.

           direct
               The mapping must not involve any copying.

               Indirect mappings to copies of frames are created in some cases where either
               direct mapping is not possible or it would have unexpected properties.  Setting
               this flag ensures that the mapping is direct and will fail if that is not
               possible.

           Defaults to read+write if not specified.

       derive_device type
           Rather than using the device supplied at initialisation, instead derive a new device
           of type type from the device the input frames exist on.

       reverse
           In a hardware to hardware mapping, map in reverse - create frames in the sink and map
           them back to the source.  This may be necessary in some cases where a mapping in one
           direction is required but only the opposite direction is supported by the devices
           being used.

           This option is dangerous - it may break the preceding filter in undefined ways if
           there are any additional constraints on that filter's output.  Do not use it without
           fully understanding the implications of its use.

   hwupload
       Upload system memory frames to hardware surfaces.

       The device to upload to must be supplied when the filter is initialised.  If using ffmpeg,
       select the appropriate device with the -filter_hw_device option.

   hwupload_cuda
       Upload system memory frames to a CUDA device.

       It accepts the following optional parameters:

       device
           The number of the CUDA device to use

   hqx
       Apply a high-quality magnification filter designed for pixel art. This filter was
       originally created by Maxim Stepin.

       It accepts the following option:

       n   Set the scaling dimension: 2 for "hq2x", 3 for "hq3x" and 4 for "hq4x".  Default is 3.

   hstack
       Stack input videos horizontally.

       All streams must be of same pixel format and of same height.

       Note that this filter is faster than using overlay and pad filter to create same output.

       The filter accept the following option:

       inputs
           Set number of input streams. Default is 2.

       shortest
           If set to 1, force the output to terminate when the shortest input terminates. Default
           value is 0.

   hue
       Modify the hue and/or the saturation of the input.

       It accepts the following parameters:

       h   Specify the hue angle as a number of degrees. It accepts an expression, and defaults
           to "0".

       s   Specify the saturation in the [-10,10] range. It accepts an expression and defaults to
           "1".

       H   Specify the hue angle as a number of radians. It accepts an expression, and defaults
           to "0".

       b   Specify the brightness in the [-10,10] range. It accepts an expression and defaults to
           "0".

       h and H are mutually exclusive, and can't be specified at the same time.

       The b, h, H and s option values are expressions containing the following constants:

       n   frame count of the input frame starting from 0

       pts presentation timestamp of the input frame expressed in time base units

       r   frame rate of the input video, NAN if the input frame rate is unknown

       t   timestamp expressed in seconds, NAN if the input timestamp is unknown

       tb  time base of the input video

       Examples

       ·   Set the hue to 90 degrees and the saturation to 1.0:

                   hue=h=90:s=1

       ·   Same command but expressing the hue in radians:

                   hue=H=PI/2:s=1

       ·   Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:

                   hue="H=2*PI*t: s=sin(2*PI*t)+1"

       ·   Apply a 3 seconds saturation fade-in effect starting at 0:

                   hue="s=min(t/3\,1)"

           The general fade-in expression can be written as:

                   hue="s=min(0\, max((t-START)/DURATION\, 1))"

       ·   Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

                   hue="s=max(0\, min(1\, (8-t)/3))"

           The general fade-out expression can be written as:

                   hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))"

       Commands

       This filter supports the following commands:

       b
       s
       h
       H   Modify the hue and/or the saturation and/or brightness of the input video.  The
           command accepts the same syntax of the corresponding option.

           If the specified expression is not valid, it is kept at its current value.

   hysteresis
       Grow first stream into second stream by connecting components.  This makes it possible to
       build more robust edge masks.

       This filter accepts the following options:

       planes
           Set which planes will be processed as bitmap, unprocessed planes will be copied from
           first stream.  By default value 0xf, all planes will be processed.

       threshold
           Set threshold which is used in filtering. If pixel component value is higher than this
           value filter algorithm for connecting components is activated.  By default value is 0.

   idet
       Detect video interlacing type.

       This filter tries to detect if the input frames are interlaced, progressive, top or bottom
       field first. It will also try to detect fields that are repeated between adjacent frames
       (a sign of telecine).

       Single frame detection considers only immediately adjacent frames when classifying each
       frame.  Multiple frame detection incorporates the classification history of previous
       frames.

       The filter will log these metadata values:

       single.current_frame
           Detected type of current frame using single-frame detection. One of: ``tff'' (top
           field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''

       single.tff
           Cumulative number of frames detected as top field first using single-frame detection.

       multiple.tff
           Cumulative number of frames detected as top field first using multiple-frame
           detection.

       single.bff
           Cumulative number of frames detected as bottom field first using single-frame
           detection.

       multiple.current_frame
           Detected type of current frame using multiple-frame detection. One of: ``tff'' (top
           field first), ``bff'' (bottom field first), ``progressive'', or ``undetermined''

       multiple.bff
           Cumulative number of frames detected as bottom field first using multiple-frame
           detection.

       single.progressive
           Cumulative number of frames detected as progressive using single-frame detection.

       multiple.progressive
           Cumulative number of frames detected as progressive using multiple-frame detection.

       single.undetermined
           Cumulative number of frames that could not be classified using single-frame detection.

       multiple.undetermined
           Cumulative number of frames that could not be classified using multiple-frame
           detection.

       repeated.current_frame
           Which field in the current frame is repeated from the last. One of ``neither'',
           ``top'', or ``bottom''.

       repeated.neither
           Cumulative number of frames with no repeated field.

       repeated.top
           Cumulative number of frames with the top field repeated from the previous frame's top
           field.

       repeated.bottom
           Cumulative number of frames with the bottom field repeated from the previous frame's
           bottom field.

       The filter accepts the following options:

       intl_thres
           Set interlacing threshold.

       prog_thres
           Set progressive threshold.

       rep_thres
           Threshold for repeated field detection.

       half_life
           Number of frames after which a given frame's contribution to the statistics is halved
           (i.e., it contributes only 0.5 to its classification). The default of 0 means that all
           frames seen are given full weight of 1.0 forever.

       analyze_interlaced_flag
           When this is not 0 then idet will use the specified number of frames to determine if
           the interlaced flag is accurate, it will not count undetermined frames.  If the flag
           is found to be accurate it will be used without any further computations, if it is
           found to be inaccurate it will be cleared without any further computations. This
           allows inserting the idet filter as a low computational method to clean up the
           interlaced flag

   il
       Deinterleave or interleave fields.

       This filter allows one to process interlaced images fields without deinterlacing them.
       Deinterleaving splits the input frame into 2 fields (so called half pictures). Odd lines
       are moved to the top half of the output image, even lines to the bottom half.  You can
       process (filter) them independently and then re-interleave them.

       The filter accepts the following options:

       luma_mode, l
       chroma_mode, c
       alpha_mode, a
           Available values for luma_mode, chroma_mode and alpha_mode are:

           none
               Do nothing.

           deinterleave, d
               Deinterleave fields, placing one above the other.

           interleave, i
               Interleave fields. Reverse the effect of deinterleaving.

           Default value is "none".

       luma_swap, ls
       chroma_swap, cs
       alpha_swap, as
           Swap luma/chroma/alpha fields. Exchange even & odd lines. Default value is 0.

   inflate
       Apply inflate effect to the video.

       This filter replaces the pixel by the local(3x3) average by taking into account only
       values higher than the pixel.

       It accepts the following options:

       threshold0
       threshold1
       threshold2
       threshold3
           Limit the maximum change for each plane, default is 65535.  If 0, plane will remain
           unchanged.

   interlace
       Simple interlacing filter from progressive contents. This interleaves upper (or lower)
       lines from odd frames with lower (or upper) lines from even frames, halving the frame rate
       and preserving image height.

                  Original        Original             New Frame
                  Frame 'j'      Frame 'j+1'             (tff)
                 ==========      ===========       ==================
                   Line 0  -------------------->    Frame 'j' Line 0
                   Line 1          Line 1  ---->   Frame 'j+1' Line 1
                   Line 2 --------------------->    Frame 'j' Line 2
                   Line 3          Line 3  ---->   Frame 'j+1' Line 3
                    ...             ...                   ...
               New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

       It accepts the following optional parameters:

       scan
           This determines whether the interlaced frame is taken from the even (tff - default) or
           odd (bff) lines of the progressive frame.

       lowpass
           Vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.

           0, off
               Disable vertical lowpass filter

           1, linear
               Enable linear filter (default)

           2, complex
               Enable complex filter. This will slightly less reduce twitter and moire but better
               retain detail and subjective sharpness impression.

   kerndeint
       Deinterlace input video by applying Donald Graft's adaptive kernel deinterling. Work on
       interlaced parts of a video to produce progressive frames.

       The description of the accepted parameters follows.

       thresh
           Set the threshold which affects the filter's tolerance when determining if a pixel
           line must be processed. It must be an integer in the range [0,255] and defaults to 10.
           A value of 0 will result in applying the process on every pixels.

       map Paint pixels exceeding the threshold value to white if set to 1.  Default is 0.

       order
           Set the fields order. Swap fields if set to 1, leave fields alone if 0. Default is 0.

       sharp
           Enable additional sharpening if set to 1. Default is 0.

       twoway
           Enable twoway sharpening if set to 1. Default is 0.

       Examples

       ·   Apply default values:

                   kerndeint=thresh=10:map=0:order=0:sharp=0:twoway=0

       ·   Enable additional sharpening:

                   kerndeint=sharp=1

       ·   Paint processed pixels in white:

                   kerndeint=map=1

   lenscorrection
       Correct radial lens distortion

       This filter can be used to correct for radial distortion as can result from the use of
       wide angle lenses, and thereby re-rectify the image. To find the right parameters one can
       use tools available for example as part of opencv or simply trial-and-error.  To use
       opencv use the calibration sample (under samples/cpp) from the opencv sources and extract
       the k1 and k2 coefficients from the resulting matrix.

       Note that effectively the same filter is available in the open-source tools Krita and
       Digikam from the KDE project.

       In contrast to the vignette filter, which can also be used to compensate lens errors, this
       filter corrects the distortion of the image, whereas vignette corrects the brightness
       distribution, so you may want to use both filters together in certain cases, though you
       will have to take care of ordering, i.e. whether vignetting should be applied before or
       after lens correction.

       Options

       The filter accepts the following options:

       cx  Relative x-coordinate of the focal point of the image, and thereby the center of the
           distortion. This value has a range [0,1] and is expressed as fractions of the image
           width. Default is 0.5.

       cy  Relative y-coordinate of the focal point of the image, and thereby the center of the
           distortion. This value has a range [0,1] and is expressed as fractions of the image
           height. Default is 0.5.

       k1  Coefficient of the quadratic correction term. This value has a range [-1,1]. 0 means
           no correction. Default is 0.

       k2  Coefficient of the double quadratic correction term. This value has a range [-1,1].  0
           means no correction. Default is 0.

       The formula that generates the correction is:

       r_src = r_tgt * (1 + k1 * (r_tgt / r_0)^2 + k2 * (r_tgt / r_0)^4)

       where r_0 is halve of the image diagonal and r_src and r_tgt are the distances from the
       focal point in the source and target images, respectively.

   lensfun
       Apply lens correction via the lensfun library (<http://lensfun.sourceforge.net/>).

       The "lensfun" filter requires the camera make, camera model, and lens model to apply the
       lens correction. The filter will load the lensfun database and query it to find the
       corresponding camera and lens entries in the database. As long as these entries can be
       found with the given options, the filter can perform corrections on frames. Note that
       incomplete strings will result in the filter choosing the best match with the given
       options, and the filter will output the chosen camera and lens models (logged with level
       "info"). You must provide the make, camera model, and lens model as they are required.

       The filter accepts the following options:

       make
           The make of the camera (for example, "Canon"). This option is required.

       model
           The model of the camera (for example, "Canon EOS 100D"). This option is required.

       lens_model
           The model of the lens (for example, "Canon EF-S 18-55mm f/3.5-5.6 IS STM"). This
           option is required.

       mode
           The type of correction to apply. The following values are valid options:

           vignetting
               Enables fixing lens vignetting.

           geometry
               Enables fixing lens geometry. This is the default.

           subpixel
               Enables fixing chromatic aberrations.

           vig_geo
               Enables fixing lens vignetting and lens geometry.

           vig_subpixel
               Enables fixing lens vignetting and chromatic aberrations.

           distortion
               Enables fixing both lens geometry and chromatic aberrations.

           all Enables all possible corrections.

       focal_length
           The focal length of the image/video (zoom; expected constant for video). For example,
           a 18--55mm lens has focal length range of [18--55], so a value in that range should be
           chosen when using that lens. Default 18.

       aperture
           The aperture of the image/video (expected constant for video). Note that aperture is
           only used for vignetting correction. Default 3.5.

       focus_distance
           The focus distance of the image/video (expected constant for video). Note that focus
           distance is only used for vignetting and only slightly affects the vignetting
           correction process. If unknown, leave it at the default value (which is 1000).

       target_geometry
           The target geometry of the output image/video. The following values are valid options:

           rectilinear (default)
           fisheye
           panoramic
           equirectangular
           fisheye_orthographic
           fisheye_stereographic
           fisheye_equisolid
           fisheye_thoby
       reverse
           Apply the reverse of image correction (instead of correcting distortion, apply it).

       interpolation
           The type of interpolation used when correcting distortion. The following values are
           valid options:

           nearest
           linear (default)
           lanczos

       Examples

       ·   Apply lens correction with make "Canon", camera model "Canon EOS 100D", and lens model
           "Canon EF-S 18-55mm f/3.5-5.6 IS STM" with focal length of "18" and aperture of "8.0".

                   ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8 -c:v h264 -b:v 8000k output.mov

       ·   Apply the same as before, but only for the first 5 seconds of video.

                   ffmpeg -i input.mov -vf lensfun=make=Canon:model="Canon EOS 100D":lens_model="Canon EF-S 18-55mm f/3.5-5.6 IS STM":focal_length=18:aperture=8:enable='lte(t\,5)' -c:v h264 -b:v 8000k output.mov

   libvmaf
       Obtain the VMAF (Video Multi-Method Assessment Fusion) score between two input videos.

       The obtained VMAF score is printed through the logging system.

       It requires Netflix's vmaf library (libvmaf) as a pre-requisite.  After installing the
       library it can be enabled using: "./configure --enable-libvmaf --enable-version3".  If no
       model path is specified it uses the default model: "vmaf_v0.6.1.pkl".

       The filter has following options:

       model_path
           Set the model path which is to be used for SVM.  Default value: "vmaf_v0.6.1.pkl"

       log_path
           Set the file path to be used to store logs.

       log_fmt
           Set the format of the log file (xml or json).

       enable_transform
           Enables transform for computing vmaf.

       phone_model
           Invokes the phone model which will generate VMAF scores higher than in the regular
           model, which is more suitable for laptop, TV, etc. viewing conditions.

       psnr
           Enables computing psnr along with vmaf.

       ssim
           Enables computing ssim along with vmaf.

       ms_ssim
           Enables computing ms_ssim along with vmaf.

       pool
           Set the pool method (mean, min or harmonic mean) to be used for computing vmaf.

       n_threads
           Set number of threads to be used when computing vmaf.

       n_subsample
           Set interval for frame subsampling used when computing vmaf.

       enable_conf_interval
           Enables confidence interval.

       This filter also supports the framesync options.

       On the below examples the input file main.mpg being processed is compared with the
       reference file ref.mpg.

               ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf -f null -

       Example with options:

               ffmpeg -i main.mpg -i ref.mpg -lavfi libvmaf="psnr=1:enable-transform=1" -f null -

   limiter
       Limits the pixel components values to the specified range [min, max].

       The filter accepts the following options:

       min Lower bound. Defaults to the lowest allowed value for the input.

       max Upper bound. Defaults to the highest allowed value for the input.

       planes
           Specify which planes will be processed. Defaults to all available.

   loop
       Loop video frames.

       The filter accepts the following options:

       loop
           Set the number of loops. Setting this value to -1 will result in infinite loops.
           Default is 0.

       size
           Set maximal size in number of frames. Default is 0.

       start
           Set first frame of loop. Default is 0.

       Examples

       ·   Loop single first frame infinitely:

                   loop=loop=-1:size=1:start=0

       ·   Loop single first frame 10 times:

                   loop=loop=10:size=1:start=0

       ·   Loop 10 first frames 5 times:

                   loop=loop=5:size=10:start=0

   lut1d
       Apply a 1D LUT to an input video.

       The filter accepts the following options:

       file
           Set the 1D LUT file name.

           Currently supported formats:

           cube
               Iridas

       interp
           Select interpolation mode.

           Available values are:

           nearest
               Use values from the nearest defined point.

           linear
               Interpolate values using the linear interpolation.

           cubic
               Interpolate values using the cubic interpolation.

   lut3d
       Apply a 3D LUT to an input video.

       The filter accepts the following options:

       file
           Set the 3D LUT file name.

           Currently supported formats:

           3dl AfterEffects

           cube
               Iridas

           dat DaVinci

           m3d Pandora

       interp
           Select interpolation mode.

           Available values are:

           nearest
               Use values from the nearest defined point.

           trilinear
               Interpolate values using the 8 points defining a cube.

           tetrahedral
               Interpolate values using a tetrahedron.

       This filter also supports the framesync options.

   lumakey
       Turn certain luma values into transparency.

       The filter accepts the following options:

       threshold
           Set the luma which will be used as base for transparency.  Default value is 0.

       tolerance
           Set the range of luma values to be keyed out.  Default value is 0.

       softness
           Set the range of softness. Default value is 0.  Use this to control gradual transition
           from zero to full transparency.

   lut, lutrgb, lutyuv
       Compute a look-up table for binding each pixel component input value to an output value,
       and apply it to the input video.

       lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.

       These filters accept the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha component

       r   set red component expression

       g   set green component expression

       b   set blue component expression

       a   alpha component expression

       y   set Y/luminance component expression

       u   set U/Cb component expression

       v   set V/Cr component expression

       Each of them specifies the expression to use for computing the lookup table for the
       corresponding pixel component values.

       The exact component associated to each of the c* options depends on the format in input.

       The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB
       pixel formats in input, and lutyuv requires YUV.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       clipval
           The input value, clipped to the minval-maxval range.

       maxval
           The maximum value for the pixel component.

       minval
           The minimum value for the pixel component.

       negval
           The negated value for the pixel component value, clipped to the minval-maxval range;
           it corresponds to the expression "maxval-clipval+minval".

       clip(val)
           The computed value in val, clipped to the minval-maxval range.

       gammaval(gamma)
           The computed gamma correction value of the pixel component value, clipped to the
           minval-maxval range. It corresponds to the expression
           "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"

       All expressions default to "val".

       Examples

       ·   Negate input video:

                   lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val"
                   lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val"

           The above is the same as:

                   lutrgb="r=negval:g=negval:b=negval"
                   lutyuv="y=negval:u=negval:v=negval"

       ·   Negate luminance:

                   lutyuv=y=negval

       ·   Remove chroma components, turning the video into a graytone image:

                   lutyuv="u=128:v=128"

       ·   Apply a luma burning effect:

                   lutyuv="y=2*val"

       ·   Remove green and blue components:

                   lutrgb="g=0:b=0"

       ·   Set a constant alpha channel value on input:

                   format=rgba,lutrgb=a="maxval-minval/2"

       ·   Correct luminance gamma by a factor of 0.5:

                   lutyuv=y=gammaval(0.5)

       ·   Discard least significant bits of luma:

                   lutyuv=y='bitand(val, 128+64+32)'

       ·   Technicolor like effect:

                   lutyuv=u='(val-maxval/2)*2+maxval/2':v='(val-maxval/2)*2+maxval/2'

   lut2, tlut2
       The "lut2" filter takes two input streams and outputs one stream.

       The "tlut2" (time lut2) filter takes two consecutive frames from one single stream.

       This filter accepts the following parameters:

       c0  set first pixel component expression

       c1  set second pixel component expression

       c2  set third pixel component expression

       c3  set fourth pixel component expression, corresponds to the alpha component

       Each of them specifies the expression to use for computing the lookup table for the
       corresponding pixel component values.

       The exact component associated to each of the c* options depends on the format in inputs.

       The expressions can contain the following constants:

       w
       h   The input width and height.

       x   The first input value for the pixel component.

       y   The second input value for the pixel component.

       bdx The first input video bit depth.

       bdy The second input video bit depth.

       All expressions default to "x".

       Examples

       ·   Highlight differences between two RGB video streams:

                   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,0,pow(2,bdx)-1)'

       ·   Highlight differences between two YUV video streams:

                   lut2='ifnot(x-y,0,pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1):ifnot(x-y,pow(2,bdx-1),pow(2,bdx)-1)'

       ·   Show max difference between two video streams:

                   lut2='if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1))):if(lt(x,y),0,if(gt(x,y),pow(2,bdx)-1,pow(2,bdx-1)))'

   maskedclamp
       Clamp the first input stream with the second input and third input stream.

       Returns the value of first stream to be between second input stream - "undershoot" and
       third input stream + "overshoot".

       This filter accepts the following options:

       undershoot
           Default value is 0.

       overshoot
           Default value is 0.

       planes
           Set which planes will be processed as bitmap, unprocessed planes will be copied from
           first stream.  By default value 0xf, all planes will be processed.

   maskedmerge
       Merge the first input stream with the second input stream using per pixel weights in the
       third input stream.

       A value of 0 in the third stream pixel component means that pixel component from first
       stream is returned unchanged, while maximum value (eg. 255 for 8-bit videos) means that
       pixel component from second stream is returned unchanged. Intermediate values define the
       amount of merging between both input stream's pixel components.

       This filter accepts the following options:

       planes
           Set which planes will be processed as bitmap, unprocessed planes will be copied from
           first stream.  By default value 0xf, all planes will be processed.

   mcdeint
       Apply motion-compensation deinterlacing.

       It needs one field per frame as input and must thus be used together with yadif=1/3 or
       equivalent.

       This filter accepts the following options:

       mode
           Set the deinterlacing mode.

           It accepts one of the following values:

           fast
           medium
           slow
               use iterative motion estimation

           extra_slow
               like slow, but use multiple reference frames.

           Default value is fast.

       parity
           Set the picture field parity assumed for the input video. It must be one of the
           following values:

           0, tff
               assume top field first

           1, bff
               assume bottom field first

           Default value is bff.

       qp  Set per-block quantization parameter (QP) used by the internal encoder.

           Higher values should result in a smoother motion vector field but less optimal
           individual vectors. Default value is 1.

   mergeplanes
       Merge color channel components from several video streams.

       The filter accepts up to 4 input streams, and merge selected input planes to the output
       video.

       This filter accepts the following options:

       mapping
           Set input to output plane mapping. Default is 0.

           The mappings is specified as a bitmap. It should be specified as a hexadecimal number
           in the form 0xAa[Bb[Cc[Dd]]]. 'Aa' describes the mapping for the first plane of the
           output stream. 'A' sets the number of the input stream to use (from 0 to 3), and 'a'
           the plane number of the corresponding input to use (from 0 to 3). The rest of the
           mappings is similar, 'Bb' describes the mapping for the output stream second plane,
           'Cc' describes the mapping for the output stream third plane and 'Dd' describes the
           mapping for the output stream fourth plane.

       format
           Set output pixel format. Default is "yuva444p".

       Examples

       ·   Merge three gray video streams of same width and height into single video stream:

                   [a0][a1][a2]mergeplanes=0x001020:yuv444p

       ·   Merge 1st yuv444p stream and 2nd gray video stream into yuva444p video stream:

                   [a0][a1]mergeplanes=0x00010210:yuva444p

       ·   Swap Y and A plane in yuva444p stream:

                   format=yuva444p,mergeplanes=0x03010200:yuva444p

       ·   Swap U and V plane in yuv420p stream:

                   format=yuv420p,mergeplanes=0x000201:yuv420p

       ·   Cast a rgb24 clip to yuv444p:

                   format=rgb24,mergeplanes=0x000102:yuv444p

   mestimate
       Estimate and export motion vectors using block matching algorithms.  Motion vectors are
       stored in frame side data to be used by other filters.

       This filter accepts the following options:

       method
           Specify the motion estimation method. Accepts one of the following values:

           esa Exhaustive search algorithm.

           tss Three step search algorithm.

           tdls
               Two dimensional logarithmic search algorithm.

           ntss
               New three step search algorithm.

           fss Four step search algorithm.

           ds  Diamond search algorithm.

           hexbs
               Hexagon-based search algorithm.

           epzs
               Enhanced predictive zonal search algorithm.

           umh Uneven multi-hexagon search algorithm.

           Default value is esa.

       mb_size
           Macroblock size. Default 16.

       search_param
           Search parameter. Default 7.

   midequalizer
       Apply Midway Image Equalization effect using two video streams.

       Midway Image Equalization adjusts a pair of images to have the same histogram, while
       maintaining their dynamics as much as possible. It's useful for e.g. matching exposures
       from a pair of stereo cameras.

       This filter has two inputs and one output, which must be of same pixel format, but may be
       of different sizes. The output of filter is first input adjusted with midway histogram of
       both inputs.

       This filter accepts the following option:

       planes
           Set which planes to process. Default is 15, which is all available planes.

   minterpolate
       Convert the video to specified frame rate using motion interpolation.

       This filter accepts the following options:

       fps Specify the output frame rate. This can be rational e.g. "60000/1001". Frames are
           dropped if fps is lower than source fps. Default 60.

       mi_mode
           Motion interpolation mode. Following values are accepted:

           dup Duplicate previous or next frame for interpolating new ones.

           blend
               Blend source frames. Interpolated frame is mean of previous and next frames.

           mci Motion compensated interpolation. Following options are effective when this mode
               is selected:

               mc_mode
                   Motion compensation mode. Following values are accepted:

                   obmc
                       Overlapped block motion compensation.

                   aobmc
                       Adaptive overlapped block motion compensation. Window weighting
                       coefficients are controlled adaptively according to the reliabilities of
                       the neighboring motion vectors to reduce oversmoothing.

                   Default mode is obmc.

               me_mode
                   Motion estimation mode. Following values are accepted:

                   bidir
                       Bidirectional motion estimation. Motion vectors are estimated for each
                       source frame in both forward and backward directions.

                   bilat
                       Bilateral motion estimation. Motion vectors are estimated directly for
                       interpolated frame.

                   Default mode is bilat.

               me  The algorithm to be used for motion estimation. Following values are accepted:

                   esa Exhaustive search algorithm.

                   tss Three step search algorithm.

                   tdls
                       Two dimensional logarithmic search algorithm.

                   ntss
                       New three step search algorithm.

                   fss Four step search algorithm.

                   ds  Diamond search algorithm.

                   hexbs
                       Hexagon-based search algorithm.

                   epzs
                       Enhanced predictive zonal search algorithm.

                   umh Uneven multi-hexagon search algorithm.

                   Default algorithm is epzs.

               mb_size
                   Macroblock size. Default 16.

               search_param
                   Motion estimation search parameter. Default 32.

               vsbmc
                   Enable variable-size block motion compensation. Motion estimation is applied
                   with smaller block sizes at object boundaries in order to make the them less
                   blur. Default is 0 (disabled).

       scd Scene change detection method. Scene change leads motion vectors to be in random
           direction. Scene change detection replace interpolated frames by duplicate ones. May
           not be needed for other modes. Following values are accepted:

           none
               Disable scene change detection.

           fdiff
               Frame difference. Corresponding pixel values are compared and if it satisfies
               scd_threshold scene change is detected.

           Default method is fdiff.

       scd_threshold
           Scene change detection threshold. Default is 5.0.

   mix
       Mix several video input streams into one video stream.

       A description of the accepted options follows.

       nb_inputs
           The number of inputs. If unspecified, it defaults to 2.

       weights
           Specify weight of each input video stream as sequence.  Each weight is separated by
           space. If number of weights is smaller than number of frames last specified weight
           will be used for all remaining unset weights.

       scale
           Specify scale, if it is set it will be multiplied with sum of each weight multiplied
           with pixel values to give final destination pixel value. By default scale is auto
           scaled to sum of weights.

       duration
           Specify how end of stream is determined.

           longest
               The duration of the longest input. (default)

           shortest
               The duration of the shortest input.

           first
               The duration of the first input.

   mpdecimate
       Drop frames that do not differ greatly from the previous frame in order to reduce frame
       rate.

       The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup
       modem), but it could in theory be used for fixing movies that were inverse-telecined
       incorrectly.

       A description of the accepted options follows.

       max Set the maximum number of consecutive frames which can be dropped (if positive), or
           the minimum interval between dropped frames (if negative). If the value is 0, the
           frame is dropped disregarding the number of previous sequentially dropped frames.

           Default value is 0.

       hi
       lo
       frac
           Set the dropping threshold values.

           Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value
           differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel,
           or the same spread out differently over the block.

           A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold
           of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than
           a threshold of lo.

           Default value for hi is 64*12, default value for lo is 64*5, and default value for
           frac is 0.33.

   negate
       Negate (invert) the input video.

       It accepts the following option:

       negate_alpha
           With value 1, it negates the alpha component, if present. Default value is 0.

   nlmeans
       Denoise frames using Non-Local Means algorithm.

       Each pixel is adjusted by looking for other pixels with similar contexts. This context
       similarity is defined by comparing their surrounding patches of size pxp. Patches are
       searched in an area of rxr around the pixel.

       Note that the research area defines centers for patches, which means some patches will be
       made of pixels outside that research area.

       The filter accepts the following options.

       s   Set denoising strength.

       p   Set patch size.

       pc  Same as p but for chroma planes.

           The default value is 0 and means automatic.

       r   Set research size.

       rc  Same as r but for chroma planes.

           The default value is 0 and means automatic.

   nnedi
       Deinterlace video using neural network edge directed interpolation.

       This filter accepts the following options:

       weights
           Mandatory option, without binary file filter can not work.  Currently file can be
           found here:
           https://github.com/dubhater/vapoursynth-nnedi3/blob/master/src/nnedi3_weights.bin

       deint
           Set which frames to deinterlace, by default it is "all".  Can be "all" or
           "interlaced".

       field
           Set mode of operation.

           Can be one of the following:

           af  Use frame flags, both fields.

           a   Use frame flags, single field.

           t   Use top field only.

           b   Use bottom field only.

           tf  Use both fields, top first.

           bf  Use both fields, bottom first.

       planes
           Set which planes to process, by default filter process all frames.

       nsize
           Set size of local neighborhood around each pixel, used by the predictor neural
           network.

           Can be one of the following:

           s8x6
           s16x6
           s32x6
           s48x6
           s8x4
           s16x4
           s32x4
       nns Set the number of neurons in predictor neural network.  Can be one of the following:

           n16
           n32
           n64
           n128
           n256
       qual
           Controls the number of different neural network predictions that are blended together
           to compute the final output value. Can be "fast", default or "slow".

       etype
           Set which set of weights to use in the predictor.  Can be one of the following:

           a   weights trained to minimize absolute error

           s   weights trained to minimize squared error

       pscrn
           Controls whether or not the prescreener neural network is used to decide which pixels
           should be processed by the predictor neural network and which can be handled by simple
           cubic interpolation.  The prescreener is trained to know whether cubic interpolation
           will be sufficient for a pixel or whether it should be predicted by the predictor nn.
           The computational complexity of the prescreener nn is much less than that of the
           predictor nn. Since most pixels can be handled by cubic interpolation, using the
           prescreener generally results in much faster processing.  The prescreener is pretty
           accurate, so the difference between using it and not using it is almost always
           unnoticeable.

           Can be one of the following:

           none
           original
           new

           Default is "new".

       fapprox
           Set various debugging flags.

   noformat
       Force libavfilter not to use any of the specified pixel formats for the input to the next
       filter.

       It accepts the following parameters:

       pix_fmts
           A '|'-separated list of pixel format names, such as pix_fmts=yuv420p|monow|rgb24".

       Examples

       ·   Force libavfilter to use a format different from yuv420p for the input to the vflip
           filter:

                   noformat=pix_fmts=yuv420p,vflip

       ·   Convert the input video to any of the formats not contained in the list:

                   noformat=yuv420p|yuv444p|yuv410p

   noise
       Add noise on video input frame.

       The filter accepts the following options:

       all_seed
       c0_seed
       c1_seed
       c2_seed
       c3_seed
           Set noise seed for specific pixel component or all pixel components in case of
           all_seed. Default value is 123457.

       all_strength, alls
       c0_strength, c0s
       c1_strength, c1s
       c2_strength, c2s
       c3_strength, c3s
           Set noise strength for specific pixel component or all pixel components in case
           all_strength. Default value is 0. Allowed range is [0, 100].

       all_flags, allf
       c0_flags, c0f
       c1_flags, c1f
       c2_flags, c2f
       c3_flags, c3f
           Set pixel component flags or set flags for all components if all_flags.  Available
           values for component flags are:

           a   averaged temporal noise (smoother)

           p   mix random noise with a (semi)regular pattern

           t   temporal noise (noise pattern changes between frames)

           u   uniform noise (gaussian otherwise)

       Examples

       Add temporal and uniform noise to input video:

               noise=alls=20:allf=t+u

   normalize
       Normalize RGB video (aka histogram stretching, contrast stretching).  See:
       https://en.wikipedia.org/wiki/Normalization_(image_processing)

       For each channel of each frame, the filter computes the input range and maps it linearly
       to the user-specified output range. The output range defaults to the full dynamic range
       from pure black to pure white.

       Temporal smoothing can be used on the input range to reduce flickering (rapid changes in
       brightness) caused when small dark or bright objects enter or leave the scene. This is
       similar to the auto-exposure (automatic gain control) on a video camera, and, like a video
       camera, it may cause a period of over- or under-exposure of the video.

       The R,G,B channels can be normalized independently, which may cause some color shifting,
       or linked together as a single channel, which prevents color shifting. Linked
       normalization preserves hue. Independent normalization does not, so it can be used to
       remove some color casts. Independent and linked normalization can be combined in any
       ratio.

       The normalize filter accepts the following options:

       blackpt
       whitept
           Colors which define the output range. The minimum input value is mapped to the
           blackpt. The maximum input value is mapped to the whitept.  The defaults are black and
           white respectively. Specifying white for blackpt and black for whitept will give
           color-inverted, normalized video. Shades of grey can be used to reduce the dynamic
           range (contrast). Specifying saturated colors here can create some interesting
           effects.

       smoothing
           The number of previous frames to use for temporal smoothing. The input range of each
           channel is smoothed using a rolling average over the current frame and the smoothing
           previous frames. The default is 0 (no temporal smoothing).

       independence
           Controls the ratio of independent (color shifting) channel normalization to linked
           (color preserving) normalization. 0.0 is fully linked, 1.0 is fully independent.
           Defaults to 1.0 (fully independent).

       strength
           Overall strength of the filter. 1.0 is full strength. 0.0 is a rather expensive no-op.
           Defaults to 1.0 (full strength).

       Examples

       Stretch video contrast to use the full dynamic range, with no temporal smoothing; may
       flicker depending on the source content:

               normalize=blackpt=black:whitept=white:smoothing=0

       As above, but with 50 frames of temporal smoothing; flicker should be reduced, depending
       on the source content:

               normalize=blackpt=black:whitept=white:smoothing=50

       As above, but with hue-preserving linked channel normalization:

               normalize=blackpt=black:whitept=white:smoothing=50:independence=0

       As above, but with half strength:

               normalize=blackpt=black:whitept=white:smoothing=50:independence=0:strength=0.5

       Map the darkest input color to red, the brightest input color to cyan:

               normalize=blackpt=red:whitept=cyan

   null
       Pass the video source unchanged to the output.

   ocr
       Optical Character Recognition

       This filter uses Tesseract for optical character recognition. To enable compilation of
       this filter, you need to configure FFmpeg with "--enable-libtesseract".

       It accepts the following options:

       datapath
           Set datapath to tesseract data. Default is to use whatever was set at installation.

       language
           Set language, default is "eng".

       whitelist
           Set character whitelist.

       blacklist
           Set character blacklist.

       The filter exports recognized text as the frame metadata "lavfi.ocr.text".

   ocv
       Apply a video transform using libopencv.

       To enable this filter, install the libopencv library and headers and configure FFmpeg with
       "--enable-libopencv".

       It accepts the following parameters:

       filter_name
           The name of the libopencv filter to apply.

       filter_params
           The parameters to pass to the libopencv filter. If not specified, the default values
           are assumed.

       Refer to the official libopencv documentation for more precise information:
       <http://docs.opencv.org/master/modules/imgproc/doc/filtering.html>

       Several libopencv filters are supported; see the following subsections.

       dilate

       Dilate an image by using a specific structuring element.  It corresponds to the libopencv
       function "cvDilate".

       It accepts the parameters: struct_el|nb_iterations.

       struct_el represents a structuring element, and has the syntax:
       colsxrows+anchor_xxanchor_y/shape

       cols and rows represent the number of columns and rows of the structuring element,
       anchor_x and anchor_y the anchor point, and shape the shape for the structuring element.
       shape must be "rect", "cross", "ellipse", or "custom".

       If the value for shape is "custom", it must be followed by a string of the form
       "=filename". The file with name filename is assumed to represent a binary image, with each
       printable character corresponding to a bright pixel. When a custom shape is used, cols and
       rows are ignored, the number or columns and rows of the read file are assumed instead.

       The default value for struct_el is "3x3+0x0/rect".

       nb_iterations specifies the number of times the transform is applied to the image, and
       defaults to 1.

       Some examples:

               # Use the default values
               ocv=dilate

               # Dilate using a structuring element with a 5x5 cross, iterating two times
               ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2

               # Read the shape from the file diamond.shape, iterating two times.
               # The file diamond.shape may contain a pattern of characters like this
               #   *
               #  ***
               # *****
               #  ***
               #   *
               # The specified columns and rows are ignored
               # but the anchor point coordinates are not
               ocv=dilate:0x0+2x2/custom=diamond.shape|2

       erode

       Erode an image by using a specific structuring element.  It corresponds to the libopencv
       function "cvErode".

       It accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as
       the dilate filter.

       smooth

       Smooth the input video.

       The filter takes the following parameters: type|param1|param2|param3|param4.

       type is the type of smooth filter to apply, and must be one of the following values:
       "blur", "blur_no_scale", "median", "gaussian", or "bilateral". The default value is
       "gaussian".

       The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and
       param2 accept integer positive values or 0. param3 and param4 accept floating point
       values.

       The default value for param1 is 3. The default value for the other parameters is 0.

       These parameters correspond to the parameters assigned to the libopencv function
       "cvSmooth".

   oscilloscope
       2D Video Oscilloscope.

       Useful to measure spatial impulse, step responses, chroma delays, etc.

       It accepts the following parameters:

       x   Set scope center x position.

       y   Set scope center y position.

       s   Set scope size, relative to frame diagonal.

       t   Set scope tilt/rotation.

       o   Set trace opacity.

       tx  Set trace center x position.

       ty  Set trace center y position.

       tw  Set trace width, relative to width of frame.

       th  Set trace height, relative to height of frame.

       c   Set which components to trace. By default it traces first three components.

       g   Draw trace grid. By default is enabled.

       st  Draw some statistics. By default is enabled.

       sc  Draw scope. By default is enabled.

       Examples

       ·   Inspect full first row of video frame.

                   oscilloscope=x=0.5:y=0:s=1

       ·   Inspect full last row of video frame.

                   oscilloscope=x=0.5:y=1:s=1

       ·   Inspect full 5th line of video frame of height 1080.

                   oscilloscope=x=0.5:y=5/1080:s=1

       ·   Inspect full last column of video frame.

                   oscilloscope=x=1:y=0.5:s=1:t=1

   overlay
       Overlay one video on top of another.

       It takes two inputs and has one output. The first input is the "main" video on which the
       second input is overlaid.

       It accepts the following parameters:

       A description of the accepted options follows.

       x
       y   Set the expression for the x and y coordinates of the overlaid video on the main
           video. Default value is "0" for both expressions. In case the expression is invalid,
           it is set to a huge value (meaning that the overlay will not be displayed within the
           output visible area).

       eof_action
           See framesync.

       eval
           Set when the expressions for x, and y are evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization or when a command
               is processed

           frame
               evaluate expressions for each incoming frame

           Default value is frame.

       shortest
           See framesync.

       format
           Set the format for the output video.

           It accepts the following values:

           yuv420
               force YUV420 output

           yuv422
               force YUV422 output

           yuv444
               force YUV444 output

           rgb force packed RGB output

           gbrp
               force planar RGB output

           auto
               automatically pick format

           Default value is yuv420.

       repeatlast
           See framesync.

       alpha
           Set format of alpha of the overlaid video, it can be straight or premultiplied.
           Default is straight.

       The x, and y expressions can contain the following parameters.

       main_w, W
       main_h, H
           The main input width and height.

       overlay_w, w
       overlay_h, h
           The overlay input width and height.

       x
       y   The computed values for x and y. They are evaluated for each new frame.

       hsub
       vsub
           horizontal and vertical chroma subsample values of the output format. For example for
           the pixel format "yuv422p" hsub is 2 and vsub is 1.

       n   the number of input frame, starting from 0

       pos the position in the file of the input frame, NAN if unknown

       t   The timestamp, expressed in seconds. It's NAN if the input timestamp is unknown.

       This filter also supports the framesync options.

       Note that the n, pos, t variables are available only when evaluation is done per frame,
       and will evaluate to NAN when eval is set to init.

       Be aware that frames are taken from each input video in timestamp order, hence, if their
       initial timestamps differ, it is a good idea to pass the two inputs through a
       setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as the example
       for the movie filter does.

       You can chain together more overlays but you should test the efficiency of such approach.

       Commands

       This filter supports the following commands:

       x
       y   Modify the x and y of the overlay input.  The command accepts the same syntax of the
           corresponding option.

           If the specified expression is not valid, it is kept at its current value.

       Examples

       ·   Draw the overlay at 10 pixels from the bottom right corner of the main video:

                   overlay=main_w-overlay_w-10:main_h-overlay_h-10

           Using named options the example above becomes:

                   overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10

       ·   Insert a transparent PNG logo in the bottom left corner of the input, using the ffmpeg
           tool with the "-filter_complex" option:

                   ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output

       ·   Insert 2 different transparent PNG logos (second logo on bottom right corner) using
           the ffmpeg tool:

                   ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=H-h-10' output

       ·   Add a transparent color layer on top of the main video; "WxH" must specify the size of
           the main input to the overlay filter:

                   color=color=red@.3:size=WxH [over]; [in][over] overlay [out]

       ·   Play an original video and a filtered version (here with the deshake filter) side by
           side using the ffplay tool:

                   ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w'

           The above command is the same as:

                   ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w'

       ·   Make a sliding overlay appearing from the left to the right top part of the screen
           starting since time 2:

                   overlay=x='if(gte(t,2), -w+(t-2)*20, NAN)':y=0

       ·   Compose output by putting two input videos side to side:

                   ffmpeg -i left.avi -i right.avi -filter_complex "
                   nullsrc=size=200x100 [background];
                   [0:v] setpts=PTS-STARTPTS, scale=100x100 [left];
                   [1:v] setpts=PTS-STARTPTS, scale=100x100 [right];
                   [background][left]       overlay=shortest=1       [background+left];
                   [background+left][right] overlay=shortest=1:x=100 [left+right]
                   "

       ·   Mask 10-20 seconds of a video by applying the delogo filter to a section

                   ffmpeg -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
                   -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
                   masked.avi

       ·   Chain several overlays in cascade:

                   nullsrc=s=200x200 [bg];
                   testsrc=s=100x100, split=4 [in0][in1][in2][in3];
                   [in0] lutrgb=r=0, [bg]   overlay=0:0     [mid0];
                   [in1] lutrgb=g=0, [mid0] overlay=100:0   [mid1];
                   [in2] lutrgb=b=0, [mid1] overlay=0:100   [mid2];
                   [in3] null,       [mid2] overlay=100:100 [out0]

   owdenoise
       Apply Overcomplete Wavelet denoiser.

       The filter accepts the following options:

       depth
           Set depth.

           Larger depth values will denoise lower frequency components more, but slow down
           filtering.

           Must be an int in the range 8-16, default is 8.

       luma_strength, ls
           Set luma strength.

           Must be a double value in the range 0-1000, default is 1.0.

       chroma_strength, cs
           Set chroma strength.

           Must be a double value in the range 0-1000, default is 1.0.

   pad
       Add paddings to the input image, and place the original input at the provided x, y
       coordinates.

       It accepts the following parameters:

       width, w
       height, h
           Specify an expression for the size of the output image with the paddings added. If the
           value for width or height is 0, the corresponding input size is used for the output.

           The width expression can reference the value set by the height expression, and vice
           versa.

           The default value of width and height is 0.

       x
       y   Specify the offsets to place the input image at within the padded area, with respect
           to the top/left border of the output image.

           The x expression can reference the value set by the y expression, and vice versa.

           The default value of x and y is 0.

           If x or y evaluate to a negative number, they'll be changed so the input image is
           centered on the padded area.

       color
           Specify the color of the padded area. For the syntax of this option, check the "Color"
           section in the ffmpeg-utils manual.

           The default value of color is "black".

       eval
           Specify when to evaluate  width, height, x and y expression.

           It accepts the following values:

           init
               Only evaluate expressions once during the filter initialization or when a command
               is processed.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

       aspect
           Pad to aspect instead to a resolution.

       The value for the width, height, x, and y options are expressions containing the following
       constants:

       in_w
       in_h
           The input video width and height.

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output width and height (the size of the padded area), as specified by the width
           and height expressions.

       ow
       oh  These are the same as out_w and out_h.

       x
       y   The x and y offsets as specified by the x and y expressions, or NAN if not yet
           specified.

       a   same as iw / ih

       sar input sample aspect ratio

       dar input display aspect ratio, it is the same as (iw / ih) * sar

       hsub
       vsub
           The horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   Add paddings with the color "violet" to the input video. The output video size is
           640x480, and the top-left corner of the input video is placed at column 0, row 40

                   pad=640:480:0:40:violet

           The example above is equivalent to the following command:

                   pad=width=640:height=480:x=0:y=40:color=violet

       ·   Pad the input to get an output with dimensions increased by 3/2, and put the input
           video at the center of the padded area:

                   pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2"

       ·   Pad the input to get a squared output with size equal to the maximum value between the
           input width and height, and put the input video at the center of the padded area:

                   pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2"

       ·   Pad the input to get a final w/h ratio of 16:9:

                   pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2"

       ·   In case of anamorphic video, in order to set the output display aspect correctly, it
           is necessary to use sar in the expression, according to the relation:

                   (ih * X / ih) * sar = output_dar
                   X = output_dar / sar

           Thus the previous example needs to be modified to:

                   pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2"

       ·   Double the output size and put the input video in the bottom-right corner of the
           output padded area:

                   pad="2*iw:2*ih:ow-iw:oh-ih"

   palettegen
       Generate one palette for a whole video stream.

       It accepts the following options:

       max_colors
           Set the maximum number of colors to quantize in the palette.  Note: the palette will
           still contain 256 colors; the unused palette entries will be black.

       reserve_transparent
           Create a palette of 255 colors maximum and reserve the last one for transparency.
           Reserving the transparency color is useful for GIF optimization.  If not set, the
           maximum of colors in the palette will be 256. You probably want to disable this option
           for a standalone image.  Set by default.

       transparency_color
           Set the color that will be used as background for transparency.

       stats_mode
           Set statistics mode.

           It accepts the following values:

           full
               Compute full frame histograms.

           diff
               Compute histograms only for the part that differs from previous frame. This might
               be relevant to give more importance to the moving part of your input if the
               background is static.

           single
               Compute new histogram for each frame.

           Default value is full.

       The filter also exports the frame metadata "lavfi.color_quant_ratio" ("nb_color_in /
       nb_color_out") which you can use to evaluate the degree of color quantization of the
       palette. This information is also visible at info logging level.

       Examples

       ·   Generate a representative palette of a given video using ffmpeg:

                   ffmpeg -i input.mkv -vf palettegen palette.png

   paletteuse
       Use a palette to downsample an input video stream.

       The filter takes two inputs: one video stream and a palette. The palette must be a 256
       pixels image.

       It accepts the following options:

       dither
           Select dithering mode. Available algorithms are:

           bayer
               Ordered 8x8 bayer dithering (deterministic)

           heckbert
               Dithering as defined by Paul Heckbert in 1982 (simple error diffusion).  Note:
               this dithering is sometimes considered "wrong" and is included as a reference.

           floyd_steinberg
               Floyd and Steingberg dithering (error diffusion)

           sierra2
               Frankie Sierra dithering v2 (error diffusion)

           sierra2_4a
               Frankie Sierra dithering v2 "Lite" (error diffusion)

           Default is sierra2_4a.

       bayer_scale
           When bayer dithering is selected, this option defines the scale of the pattern (how
           much the crosshatch pattern is visible). A low value means more visible pattern for
           less banding, and higher value means less visible pattern at the cost of more banding.

           The option must be an integer value in the range [0,5]. Default is 2.

       diff_mode
           If set, define the zone to process

           rectangle
               Only the changing rectangle will be reprocessed. This is similar to GIF
               cropping/offsetting compression mechanism. This option can be useful for speed if
               only a part of the image is changing, and has use cases such as limiting the scope
               of the error diffusal dither to the rectangle that bounds the moving scene (it
               leads to more deterministic output if the scene doesn't change much, and as a
               result less moving noise and better GIF compression).

           Default is none.

       new Take new palette for each output frame.

       alpha_threshold
           Sets the alpha threshold for transparency. Alpha values above this threshold will be
           treated as completely opaque, and values below this threshold will be treated as
           completely transparent.

           The option must be an integer value in the range [0,255]. Default is 128.

       Examples

       ·   Use a palette (generated for example with palettegen) to encode a GIF using ffmpeg:

                   ffmpeg -i input.mkv -i palette.png -lavfi paletteuse output.gif

   perspective
       Correct perspective of video not recorded perpendicular to the screen.

       A description of the accepted parameters follows.

       x0
       y0
       x1
       y1
       x2
       y2
       x3
       y3  Set coordinates expression for top left, top right, bottom left and bottom right
           corners.  Default values are "0:0:W:0:0:H:W:H" with which perspective will remain
           unchanged.  If the "sense" option is set to "source", then the specified points will
           be sent to the corners of the destination. If the "sense" option is set to
           "destination", then the corners of the source will be sent to the specified
           coordinates.

           The expressions can use the following variables:

           W
           H   the width and height of video frame.

           in  Input frame count.

           on  Output frame count.

       interpolation
           Set interpolation for perspective correction.

           It accepts the following values:

           linear
           cubic

           Default value is linear.

       sense
           Set interpretation of coordinate options.

           It accepts the following values:

           0, source
               Send point in the source specified by the given coordinates to the corners of the
               destination.

           1, destination
               Send the corners of the source to the point in the destination specified by the
               given coordinates.

               Default value is source.

       eval
           Set when the expressions for coordinates x0,y0,...x3,y3 are evaluated.

           It accepts the following values:

           init
               only evaluate expressions once during the filter initialization or when a command
               is processed

           frame
               evaluate expressions for each incoming frame

           Default value is init.

   phase
       Delay interlaced video by one field time so that the field order changes.

       The intended use is to fix PAL movies that have been captured with the opposite field
       order to the film-to-video transfer.

       A description of the accepted parameters follows.

       mode
           Set phase mode.

           It accepts the following values:

           t   Capture field order top-first, transfer bottom-first.  Filter will delay the
               bottom field.

           b   Capture field order bottom-first, transfer top-first.  Filter will delay the top
               field.

           p   Capture and transfer with the same field order. This mode only exists for the
               documentation of the other options to refer to, but if you actually select it, the
               filter will faithfully do nothing.

           a   Capture field order determined automatically by field flags, transfer opposite.
               Filter selects among t and b modes on a frame by frame basis using field flags. If
               no field information is available, then this works just like u.

           u   Capture unknown or varying, transfer opposite.  Filter selects among t and b on a
               frame by frame basis by analyzing the images and selecting the alternative that
               produces best match between the fields.

           T   Capture top-first, transfer unknown or varying.  Filter selects among t and p
               using image analysis.

           B   Capture bottom-first, transfer unknown or varying.  Filter selects among b and p
               using image analysis.

           A   Capture determined by field flags, transfer unknown or varying.  Filter selects
               among t, b and p using field flags and image analysis. If no field information is
               available, then this works just like U. This is the default mode.

           U   Both capture and transfer unknown or varying.  Filter selects among t, b and p
               using image analysis only.

   pixdesctest
       Pixel format descriptor test filter, mainly useful for internal testing. The output video
       should be equal to the input video.

       For example:

               format=monow, pixdesctest

       can be used to test the monowhite pixel format descriptor definition.

   pixscope
       Display sample values of color channels. Mainly useful for checking color and levels.
       Minimum supported resolution is 640x480.

       The filters accept the following options:

       x   Set scope X position, relative offset on X axis.

       y   Set scope Y position, relative offset on Y axis.

       w   Set scope width.

       h   Set scope height.

       o   Set window opacity. This window also holds statistics about pixel area.

       wx  Set window X position, relative offset on X axis.

       wy  Set window Y position, relative offset on Y axis.

   pp
       Enable the specified chain of postprocessing subfilters using libpostproc. This library
       should be automatically selected with a GPL build ("--enable-gpl").  Subfilters must be
       separated by '/' and can be disabled by prepending a '-'.  Each subfilter and some options
       have a short and a long name that can be used interchangeably, i.e. dr/dering are the
       same.

       The filters accept the following options:

       subfilters
           Set postprocessing subfilters string.

       All subfilters share common options to determine their scope:

       a/autoq
           Honor the quality commands for this subfilter.

       c/chrom
           Do chrominance filtering, too (default).

       y/nochrom
           Do luminance filtering only (no chrominance).

       n/noluma
           Do chrominance filtering only (no luminance).

       These options can be appended after the subfilter name, separated by a '|'.

       Available subfilters are:

       hb/hdeblock[|difference[|flatness]]
           Horizontal deblocking filter

           difference
               Difference factor where higher values mean more deblocking (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking (default: 39).

       vb/vdeblock[|difference[|flatness]]
           Vertical deblocking filter

           difference
               Difference factor where higher values mean more deblocking (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking (default: 39).

       ha/hadeblock[|difference[|flatness]]
           Accurate horizontal deblocking filter

           difference
               Difference factor where higher values mean more deblocking (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking (default: 39).

       va/vadeblock[|difference[|flatness]]
           Accurate vertical deblocking filter

           difference
               Difference factor where higher values mean more deblocking (default: 32).

           flatness
               Flatness threshold where lower values mean more deblocking (default: 39).

       The horizontal and vertical deblocking filters share the difference and flatness values so
       you cannot set different horizontal and vertical thresholds.

       h1/x1hdeblock
           Experimental horizontal deblocking filter

       v1/x1vdeblock
           Experimental vertical deblocking filter

       dr/dering
           Deringing filter

       tn/tmpnoise[|threshold1[|threshold2[|threshold3]]], temporal noise reducer
           threshold1
               larger -> stronger filtering

           threshold2
               larger -> stronger filtering

           threshold3
               larger -> stronger filtering

       al/autolevels[:f/fullyrange], automatic brightness / contrast correction
           f/fullyrange
               Stretch luminance to "0-255".

       lb/linblenddeint
           Linear blend deinterlacing filter that deinterlaces the given block by filtering all
           lines with a "(1 2 1)" filter.

       li/linipoldeint
           Linear interpolating deinterlacing filter that deinterlaces the given block by
           linearly interpolating every second line.

       ci/cubicipoldeint
           Cubic interpolating deinterlacing filter deinterlaces the given block by cubically
           interpolating every second line.

       md/mediandeint
           Median deinterlacing filter that deinterlaces the given block by applying a median
           filter to every second line.

       fd/ffmpegdeint
           FFmpeg deinterlacing filter that deinterlaces the given block by filtering every
           second line with a "(-1 4 2 4 -1)" filter.

       l5/lowpass5
           Vertically applied FIR lowpass deinterlacing filter that deinterlaces the given block
           by filtering all lines with a "(-1 2 6 2 -1)" filter.

       fq/forceQuant[|quantizer]
           Overrides the quantizer table from the input with the constant quantizer you specify.

           quantizer
               Quantizer to use

       de/default
           Default pp filter combination ("hb|a,vb|a,dr|a")

       fa/fast
           Fast pp filter combination ("h1|a,v1|a,dr|a")

       ac  High quality pp filter combination ("ha|a|128|7,va|a,dr|a")

       Examples

       ·   Apply horizontal and vertical deblocking, deringing and automatic brightness/contrast:

                   pp=hb/vb/dr/al

       ·   Apply default filters without brightness/contrast correction:

                   pp=de/-al

       ·   Apply default filters and temporal denoiser:

                   pp=default/tmpnoise|1|2|3

       ·   Apply deblocking on luminance only, and switch vertical deblocking on or off
           automatically depending on available CPU time:

                   pp=hb|y/vb|a

   pp7
       Apply Postprocessing filter 7. It is variant of the spp filter, similar to spp = 6 with 7
       point DCT, where only the center sample is used after IDCT.

       The filter accepts the following options:

       qp  Force a constant quantization parameter. It accepts an integer in range 0 to 63. If
           not set, the filter will use the QP from the video stream (if available).

       mode
           Set thresholding mode. Available modes are:

           hard
               Set hard thresholding.

           soft
               Set soft thresholding (better de-ringing effect, but likely blurrier).

           medium
               Set medium thresholding (good results, default).

   premultiply
       Apply alpha premultiply effect to input video stream using first plane of second stream as
       alpha.

       Both streams must have same dimensions and same pixel format.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be copied.  By default
           value 0xf, all planes will be processed.

       inplace
           Do not require 2nd input for processing, instead use alpha plane from input stream.

   prewitt
       Apply prewitt operator to input video stream.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be copied.  By default
           value 0xf, all planes will be processed.

       scale
           Set value which will be multiplied with filtered result.

       delta
           Set value which will be added to filtered result.

   program_opencl
       Filter video using an OpenCL program.

       source
           OpenCL program source file.

       kernel
           Kernel name in program.

       inputs
           Number of inputs to the filter.  Defaults to 1.

       size, s
           Size of output frames.  Defaults to the same as the first input.

       The program source file must contain a kernel function with the given name, which will be
       run once for each plane of the output.  Each run on a plane gets enqueued as a separate 2D
       global NDRange with one work-item for each pixel to be generated.  The global ID offset
       for each work-item is therefore the coordinates of a pixel in the destination image.

       The kernel function needs to take the following arguments:

       ·   Destination image, __write_only image2d_t.

           This image will become the output; the kernel should write all of it.

       ·   Frame index, unsigned int.

           This is a counter starting from zero and increasing by one for each frame.

       ·   Source images, __read_only image2d_t.

           These are the most recent images on each input.  The kernel may read from them to
           generate the output, but they can't be written to.

       Example programs:

       ·   Copy the input to the output (output must be the same size as the input).

                   __kernel void copy(__write_only image2d_t destination,
                                      unsigned int index,
                                      __read_only  image2d_t source)
                   {
                       const sampler_t sampler = CLK_NORMALIZED_COORDS_FALSE;

                       int2 location = (int2)(get_global_id(0), get_global_id(1));

                       float4 value = read_imagef(source, sampler, location);

                       write_imagef(destination, location, value);
                   }

       ·   Apply a simple transformation, rotating the input by an amount increasing with the
           index counter.  Pixel values are linearly interpolated by the sampler, and the output
           need not have the same dimensions as the input.

                   __kernel void rotate_image(__write_only image2d_t dst,
                                              unsigned int index,
                                              __read_only  image2d_t src)
                   {
                       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
                                                  CLK_FILTER_LINEAR);

                       float angle = (float)index / 100.0f;

                       float2 dst_dim = convert_float2(get_image_dim(dst));
                       float2 src_dim = convert_float2(get_image_dim(src));

                       float2 dst_cen = dst_dim / 2.0f;
                       float2 src_cen = src_dim / 2.0f;

                       int2   dst_loc = (int2)(get_global_id(0), get_global_id(1));

                       float2 dst_pos = convert_float2(dst_loc) - dst_cen;
                       float2 src_pos = {
                           cos(angle) * dst_pos.x - sin(angle) * dst_pos.y,
                           sin(angle) * dst_pos.x + cos(angle) * dst_pos.y
                       };
                       src_pos = src_pos * src_dim / dst_dim;

                       float2 src_loc = src_pos + src_cen;

                       if (src_loc.x < 0.0f      || src_loc.y < 0.0f ||
                           src_loc.x > src_dim.x || src_loc.y > src_dim.y)
                           write_imagef(dst, dst_loc, 0.5f);
                       else
                           write_imagef(dst, dst_loc, read_imagef(src, sampler, src_loc));
                   }

       ·   Blend two inputs together, with the amount of each input used varying with the index
           counter.

                   __kernel void blend_images(__write_only image2d_t dst,
                                              unsigned int index,
                                              __read_only  image2d_t src1,
                                              __read_only  image2d_t src2)
                   {
                       const sampler_t sampler = (CLK_NORMALIZED_COORDS_FALSE |
                                                  CLK_FILTER_LINEAR);

                       float blend = (cos((float)index / 50.0f) + 1.0f) / 2.0f;

                       int2  dst_loc = (int2)(get_global_id(0), get_global_id(1));
                       int2 src1_loc = dst_loc * get_image_dim(src1) / get_image_dim(dst);
                       int2 src2_loc = dst_loc * get_image_dim(src2) / get_image_dim(dst);

                       float4 val1 = read_imagef(src1, sampler, src1_loc);
                       float4 val2 = read_imagef(src2, sampler, src2_loc);

                       write_imagef(dst, dst_loc, val1 * blend + val2 * (1.0f - blend));
                   }

   pseudocolor
       Alter frame colors in video with pseudocolors.

       This filter accept the following options:

       c0  set pixel first component expression

       c1  set pixel second component expression

       c2  set pixel third component expression

       c3  set pixel fourth component expression, corresponds to the alpha component

       i   set component to use as base for altering colors

       Each of them specifies the expression to use for computing the lookup table for the
       corresponding pixel component values.

       The expressions can contain the following constants and functions:

       w
       h   The input width and height.

       val The input value for the pixel component.

       ymin, umin, vmin, amin
           The minimum allowed component value.

       ymax, umax, vmax, amax
           The maximum allowed component value.

       All expressions default to "val".

       Examples

       ·   Change too high luma values to gradient:

                   pseudocolor="'if(between(val,ymax,amax),lerp(ymin,ymax,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(umax,umin,(val-ymax)/(amax-ymax)),-1):if(between(val,ymax,amax),lerp(vmin,vmax,(val-ymax)/(amax-ymax)),-1):-1'"

   psnr
       Obtain the average, maximum and minimum PSNR (Peak Signal to Noise Ratio) between two
       input videos.

       This filter takes in input two input videos, the first input is considered the "main"
       source and is passed unchanged to the output. The second input is used as a "reference"
       video for computing the PSNR.

       Both video inputs must have the same resolution and pixel format for this filter to work
       correctly. Also it assumes that both inputs have the same number of frames, which are
       compared one by one.

       The obtained average PSNR is printed through the logging system.

       The filter stores the accumulated MSE (mean squared error) of each frame, and at the end
       of the processing it is averaged across all frames equally, and the following formula is
       applied to obtain the PSNR:

               PSNR = 10*log10(MAX^2/MSE)

       Where MAX is the average of the maximum values of each component of the image.

       The description of the accepted parameters follows.

       stats_file, f
           If specified the filter will use the named file to save the PSNR of each individual
           frame. When filename equals "-" the data is sent to standard output.

       stats_version
           Specifies which version of the stats file format to use. Details of each format are
           written below.  Default value is 1.

       stats_add_max
           Determines whether the max value is output to the stats log.  Default value is 0.
           Requires stats_version >= 2. If this is set and stats_version < 2, the filter will
           return an error.

       This filter also supports the framesync options.

       The file printed if stats_file is selected, contains a sequence of key/value pairs of the
       form key:value for each compared couple of frames.

       If a stats_version greater than 1 is specified, a header line precedes the list of per-
       frame-pair stats, with key value pairs following the frame format with the following
       parameters:

       psnr_log_version
           The version of the log file format. Will match stats_version.

       fields
           A comma separated list of the per-frame-pair parameters included in the log.

       A description of each shown per-frame-pair parameter follows:

       n   sequential number of the input frame, starting from 1

       mse_avg
           Mean Square Error pixel-by-pixel average difference of the compared frames, averaged
           over all the image components.

       mse_y, mse_u, mse_v, mse_r, mse_g, mse_b, mse_a
           Mean Square Error pixel-by-pixel average difference of the compared frames for the
           component specified by the suffix.

       psnr_y, psnr_u, psnr_v, psnr_r, psnr_g, psnr_b, psnr_a
           Peak Signal to Noise ratio of the compared frames for the component specified by the
           suffix.

       max_avg, max_y, max_u, max_v
           Maximum allowed value for each channel, and average over all channels.

       For example:

               movie=ref_movie.mpg, setpts=PTS-STARTPTS [main];
               [main][ref] psnr="stats_file=stats.log" [out]

       On this example the input file being processed is compared with the reference file
       ref_movie.mpg. The PSNR of each individual frame is stored in stats.log.

   pullup
       Pulldown reversal (inverse telecine) filter, capable of handling mixed hard-telecine,
       24000/1001 fps progressive, and 30000/1001 fps progressive content.

       The pullup filter is designed to take advantage of future context in making its decisions.
       This filter is stateless in the sense that it does not lock onto a pattern to follow, but
       it instead looks forward to the following fields in order to identify matches and rebuild
       progressive frames.

       To produce content with an even framerate, insert the fps filter after pullup, use
       "fps=24000/1001" if the input frame rate is 29.97fps, "fps=24" for 30fps and the (rare)
       telecined 25fps input.

       The filter accepts the following options:

       jl
       jr
       jt
       jb  These options set the amount of "junk" to ignore at the left, right, top, and bottom
           of the image, respectively. Left and right are in units of 8 pixels, while top and
           bottom are in units of 2 lines.  The default is 8 pixels on each side.

       sb  Set the strict breaks. Setting this option to 1 will reduce the chances of filter
           generating an occasional mismatched frame, but it may also cause an excessive number
           of frames to be dropped during high motion sequences.  Conversely, setting it to -1
           will make filter match fields more easily.  This may help processing of video where
           there is slight blurring between the fields, but may also cause there to be interlaced
           frames in the output.  Default value is 0.

       mp  Set the metric plane to use. It accepts the following values:

           l   Use luma plane.

           u   Use chroma blue plane.

           v   Use chroma red plane.

           This option may be set to use chroma plane instead of the default luma plane for doing
           filter's computations. This may improve accuracy on very clean source material, but
           more likely will decrease accuracy, especially if there is chroma noise (rainbow
           effect) or any grayscale video.  The main purpose of setting mp to a chroma plane is
           to reduce CPU load and make pullup usable in realtime on slow machines.

       For best results (without duplicated frames in the output file) it is necessary to change
       the output frame rate. For example, to inverse telecine NTSC input:

               ffmpeg -i input -vf pullup -r 24000/1001 ...

   qp
       Change video quantization parameters (QP).

       The filter accepts the following option:

       qp  Set expression for quantization parameter.

       The expression is evaluated through the eval API and can contain, among others, the
       following constants:

       known
           1 if index is not 129, 0 otherwise.

       qp  Sequential index starting from -129 to 128.

       Examples

       ·   Some equation like:

                   qp=2+2*sin(PI*qp)

   random
       Flush video frames from internal cache of frames into a random order.  No frame is
       discarded.  Inspired by frei0r nervous filter.

       frames
           Set size in number of frames of internal cache, in range from 2 to 512. Default is 30.

       seed
           Set seed for random number generator, must be an integer included between 0 and
           "UINT32_MAX". If not specified, or if explicitly set to less than 0, the filter will
           try to use a good random seed on a best effort basis.

   readeia608
       Read closed captioning (EIA-608) information from the top lines of a video frame.

       This filter adds frame metadata for "lavfi.readeia608.X.cc" and "lavfi.readeia608.X.line",
       where "X" is the number of the identified line with EIA-608 data (starting from 0). A
       description of each metadata value follows:

       lavfi.readeia608.X.cc
           The two bytes stored as EIA-608 data (printed in hexadecimal).

       lavfi.readeia608.X.line
           The number of the line on which the EIA-608 data was identified and read.

       This filter accepts the following options:

       scan_min
           Set the line to start scanning for EIA-608 data. Default is 0.

       scan_max
           Set the line to end scanning for EIA-608 data. Default is 29.

       mac Set minimal acceptable amplitude change for sync codes detection.  Default is 0.2.
           Allowed range is "[0.001 - 1]".

       spw Set the ratio of width reserved for sync code detection.  Default is 0.27. Allowed
           range is "[0.01 - 0.7]".

       mhd Set the max peaks height difference for sync code detection.  Default is 0.1. Allowed
           range is "[0.0 - 0.5]".

       mpd Set max peaks period difference for sync code detection.  Default is 0.1. Allowed
           range is "[0.0 - 0.5]".

       msd Set the first two max start code bits differences.  Default is 0.02. Allowed range is
           "[0.0 - 0.5]".

       bhd Set the minimum ratio of bits height compared to 3rd start code bit.  Default is 0.75.
           Allowed range is "[0.01 - 1]".

       th_w
           Set the white color threshold. Default is 0.35. Allowed range is "[0.1 - 1]".

       th_b
           Set the black color threshold. Default is 0.15. Allowed range is "[0.0 - 0.5]".

       chp Enable checking the parity bit. In the event of a parity error, the filter will output
           0x00 for that character. Default is false.

       Examples

       ·   Output a csv with presentation time and the first two lines of identified EIA-608
           captioning data.

                   ffprobe -f lavfi -i movie=captioned_video.mov,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.cc,lavfi.readeia608.1.cc -of csv

   readvitc
       Read vertical interval timecode (VITC) information from the top lines of a video frame.

       The filter adds frame metadata key "lavfi.readvitc.tc_str" with the timecode value, if a
       valid timecode has been detected. Further metadata key "lavfi.readvitc.found" is set to
       0/1 depending on whether timecode data has been found or not.

       This filter accepts the following options:

       scan_max
           Set the maximum number of lines to scan for VITC data. If the value is set to "-1" the
           full video frame is scanned. Default is 45.

       thr_b
           Set the luma threshold for black. Accepts float numbers in the range [0.0,1.0],
           default value is 0.2. The value must be equal or less than "thr_w".

       thr_w
           Set the luma threshold for white. Accepts float numbers in the range [0.0,1.0],
           default value is 0.6. The value must be equal or greater than "thr_b".

       Examples

       ·   Detect and draw VITC data onto the video frame; if no valid VITC is detected, draw
           "--:--:--:--" as a placeholder:

                   ffmpeg -i input.avi -filter:v 'readvitc,drawtext=fontfile=FreeMono.ttf:text=%{metadata\\:lavfi.readvitc.tc_str\\:--\\\\\\:--\\\\\\:--\\\\\\:--}:x=(w-tw)/2:y=400-ascent'

   remap
       Remap pixels using 2nd: Xmap and 3rd: Ymap input video stream.

       Destination pixel at position (X, Y) will be picked from source (x, y) position where x =
       Xmap(X, Y) and y = Ymap(X, Y). If mapping values are out of range, zero value for pixel
       will be used for destination pixel.

       Xmap and Ymap input video streams must be of same dimensions. Output video stream will
       have Xmap/Ymap video stream dimensions.  Xmap and Ymap input video streams are 16bit
       depth, single channel.

   removegrain
       The removegrain filter is a spatial denoiser for progressive video.

       m0  Set mode for the first plane.

       m1  Set mode for the second plane.

       m2  Set mode for the third plane.

       m3  Set mode for the fourth plane.

       Range of mode is from 0 to 24. Description of each mode follows:

       0   Leave input plane unchanged. Default.

       1   Clips the pixel with the minimum and maximum of the 8 neighbour pixels.

       2   Clips the pixel with the second minimum and maximum of the 8 neighbour pixels.

       3   Clips the pixel with the third minimum and maximum of the 8 neighbour pixels.

       4   Clips the pixel with the fourth minimum and maximum of the 8 neighbour pixels.  This
           is equivalent to a median filter.

       5   Line-sensitive clipping giving the minimal change.

       6   Line-sensitive clipping, intermediate.

       7   Line-sensitive clipping, intermediate.

       8   Line-sensitive clipping, intermediate.

       9   Line-sensitive clipping on a line where the neighbours pixels are the closest.

       10  Replaces the target pixel with the closest neighbour.

       11  [1 2 1] horizontal and vertical kernel blur.

       12  Same as mode 11.

       13  Bob mode, interpolates top field from the line where the neighbours pixels are the
           closest.

       14  Bob mode, interpolates bottom field from the line where the neighbours pixels are the
           closest.

       15  Bob mode, interpolates top field. Same as 13 but with a more complicated interpolation
           formula.

       16  Bob mode, interpolates bottom field. Same as 14 but with a more complicated
           interpolation formula.

       17  Clips the pixel with the minimum and maximum of respectively the maximum and minimum
           of each pair of opposite neighbour pixels.

       18  Line-sensitive clipping using opposite neighbours whose greatest distance from the
           current pixel is minimal.

       19  Replaces the pixel with the average of its 8 neighbours.

       20  Averages the 9 pixels ([1 1 1] horizontal and vertical blur).

       21  Clips pixels using the averages of opposite neighbour.

       22  Same as mode 21 but simpler and faster.

       23  Small edge and halo removal, but reputed useless.

       24  Similar as 23.

   removelogo
       Suppress a TV station logo, using an image file to determine which pixels comprise the
       logo. It works by filling in the pixels that comprise the logo with neighboring pixels.

       The filter accepts the following options:

       filename, f
           Set the filter bitmap file, which can be any image format supported by libavformat.
           The width and height of the image file must match those of the video stream being
           processed.

       Pixels in the provided bitmap image with a value of zero are not considered part of the
       logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo
       and black (0) for the rest, you will be safe. For making the filter bitmap, it is
       recommended to take a screen capture of a black frame with the logo visible, and then
       using a threshold filter followed by the erode filter once or twice.

       If needed, little splotches can be fixed manually. Remember that if logo pixels are not
       covered, the filter quality will be much reduced. Marking too many pixels as part of the
       logo does not hurt as much, but it will increase the amount of blurring needed to cover
       over the image and will destroy more information than necessary, and extra pixels will
       slow things down on a large logo.

   repeatfields
       This filter uses the repeat_field flag from the Video ES headers and hard repeats fields
       based on its value.

   reverse
       Reverse a video clip.

       Warning: This filter requires memory to buffer the entire clip, so trimming is suggested.

       Examples

       ·   Take the first 5 seconds of a clip, and reverse it.

                   trim=end=5,reverse

   roberts
       Apply roberts cross operator to input video stream.

       The filter accepts the following option:

       planes
           Set which planes will be processed, unprocessed planes will be copied.  By default
           value 0xf, all planes will be processed.

       scale
           Set value which will be multiplied with filtered result.

       delta
           Set value which will be added to filtered result.

   rotate
       Rotate video by an arbitrary angle expressed in radians.

       The filter accepts the following options:

       A description of the optional parameters follows.

       angle, a
           Set an expression for the angle by which to rotate the input video clockwise,
           expressed as a number of radians. A negative value will result in a counter-clockwise
           rotation. By default it is set to "0".

           This expression is evaluated for each frame.

       out_w, ow
           Set the output width expression, default value is "iw".  This expression is evaluated
           just once during configuration.

       out_h, oh
           Set the output height expression, default value is "ih".  This expression is evaluated
           just once during configuration.

       bilinear
           Enable bilinear interpolation if set to 1, a value of 0 disables it. Default value is
           1.

       fillcolor, c
           Set the color used to fill the output area not covered by the rotated image. For the
           general syntax of this option, check the "Color" section in the ffmpeg-utils manual.
           If the special value "none" is selected then no background is printed (useful for
           example if the background is never shown).

           Default value is "black".

       The expressions for the angle and the output size can contain the following constants and
       functions:

       n   sequential number of the input frame, starting from 0. It is always NAN before the
           first frame is filtered.

       t   time in seconds of the input frame, it is set to 0 when the filter is configured. It
           is always NAN before the first frame is filtered.

       hsub
       vsub
           horizontal and vertical chroma subsample values. For example for the pixel format
           "yuv422p" hsub is 2 and vsub is 1.

       in_w, iw
       in_h, ih
           the input video width and height

       out_w, ow
       out_h, oh
           the output width and height, that is the size of the padded area as specified by the
           width and height expressions

       rotw(a)
       roth(a)
           the minimal width/height required for completely containing the input video rotated by
           a radians.

           These are only available when computing the out_w and out_h expressions.

       Examples

       ·   Rotate the input by PI/6 radians clockwise:

                   rotate=PI/6

       ·   Rotate the input by PI/6 radians counter-clockwise:

                   rotate=-PI/6

       ·   Rotate the input by 45 degrees clockwise:

                   rotate=45*PI/180

       ·   Apply a constant rotation with period T, starting from an angle of PI/3:

                   rotate=PI/3+2*PI*t/T

       ·   Make the input video rotation oscillating with a period of T seconds and an amplitude
           of A radians:

                   rotate=A*sin(2*PI/T*t)

       ·   Rotate the video, output size is chosen so that the whole rotating input video is
           always completely contained in the output:

                   rotate='2*PI*t:ow=hypot(iw,ih):oh=ow'

       ·   Rotate the video, reduce the output size so that no background is ever shown:

                   rotate=2*PI*t:ow='min(iw,ih)/sqrt(2)':oh=ow:c=none

       Commands

       The filter supports the following commands:

       a, angle
           Set the angle expression.  The command accepts the same syntax of the corresponding
           option.

           If the specified expression is not valid, it is kept at its current value.

   sab
       Apply Shape Adaptive Blur.

       The filter accepts the following options:

       luma_radius, lr
           Set luma blur filter strength, must be a value in range 0.1-4.0, default value is 1.0.
           A greater value will result in a more blurred image, and in slower processing.

       luma_pre_filter_radius, lpfr
           Set luma pre-filter radius, must be a value in the 0.1-2.0 range, default value is
           1.0.

       luma_strength, ls
           Set luma maximum difference between pixels to still be considered, must be a value in
           the 0.1-100.0 range, default value is 1.0.

       chroma_radius, cr
           Set chroma blur filter strength, must be a value in range -0.9-4.0. A greater value
           will result in a more blurred image, and in slower processing.

       chroma_pre_filter_radius, cpfr
           Set chroma pre-filter radius, must be a value in the -0.9-2.0 range.

       chroma_strength, cs
           Set chroma maximum difference between pixels to still be considered, must be a value
           in the -0.9-100.0 range.

       Each chroma option value, if not explicitly specified, is set to the corresponding luma
       option value.

   scale
       Scale (resize) the input video, using the libswscale library.

       The scale filter forces the output display aspect ratio to be the same of the input, by
       changing the output sample aspect ratio.

       If the input image format is different from the format requested by the next filter, the
       scale filter will convert the input to the requested format.

       Options

       The filter accepts the following options, or any of the options supported by the
       libswscale scaler.

       See the ffmpeg-scaler manual for the complete list of scaler options.

       width, w
       height, h
           Set the output video dimension expression. Default value is the input dimension.

           If the width or w value is 0, the input width is used for the output. If the height or
           h value is 0, the input height is used for the output.

           If one and only one of the values is -n with n >= 1, the scale filter will use a value
           that maintains the aspect ratio of the input image, calculated from the other
           specified dimension. After that it will, however, make sure that the calculated
           dimension is divisible by n and adjust the value if necessary.

           If both values are -n with n >= 1, the behavior will be identical to both values being
           set to 0 as previously detailed.

           See below for the list of accepted constants for use in the dimension expression.

       eval
           Specify when to evaluate width and height expression. It accepts the following values:

           init
               Only evaluate expressions once during the filter initialization or when a command
               is processed.

           frame
               Evaluate expressions for each incoming frame.

           Default value is init.

       interl
           Set the interlacing mode. It accepts the following values:

           1   Force interlaced aware scaling.

           0   Do not apply interlaced scaling.

           -1  Select interlaced aware scaling depending on whether the source frames are flagged
               as interlaced or not.

           Default value is 0.

       flags
           Set libswscale scaling flags. See the ffmpeg-scaler manual for the complete list of
           values. If not explicitly specified the filter applies the default flags.

       param0, param1
           Set libswscale input parameters for scaling algorithms that need them. See the ffmpeg-
           scaler manual for the complete documentation. If not explicitly specified the filter
           applies empty parameters.

       size, s
           Set the video size. For the syntax of this option, check the "Video size" section in
           the ffmpeg-utils manual.

       in_color_matrix
       out_color_matrix
           Set in/output YCbCr color space type.

           This allows the autodetected value to be overridden as well as allows forcing a
           specific value used for the output and encoder.

           If not specified, the color space type depends on the pixel format.

           Possible values:

           auto
               Choose automatically.

           bt709
               Format conforming to International Telecommunication Union (ITU) Recommendation
               BT.709.

           fcc Set color space conforming to the United States Federal Communications Commission
               (FCC) Code of Federal Regulations (CFR) Title 47 (2003) 73.682 (a).

           bt601
               Set color space conforming to:

               ·   ITU Radiocommunication Sector (ITU-R) Recommendation BT.601

               ·   ITU-R Rec. BT.470-6 (1998) Systems B, B1, and G

               ·   Society of Motion Picture and Television Engineers (SMPTE) ST 170:2004

           smpte240m
               Set color space conforming to SMPTE ST 240:1999.

       in_range
       out_range
           Set in/output YCbCr sample range.

           This allows the autodetected value to be overridden as well as allows forcing a
           specific value used for the output and encoder. If not specified, the range depends on
           the pixel format. Possible values:

           auto/unknown
               Choose automatically.

           jpeg/full/pc
               Set full range (0-255 in case of 8-bit luma).

           mpeg/limited/tv
               Set "MPEG" range (16-235 in case of 8-bit luma).

       force_original_aspect_ratio
           Enable decreasing or increasing output video width or height if necessary to keep the
           original aspect ratio. Possible values:

           disable
               Scale the video as specified and disable this feature.

           decrease
               The output video dimensions will automatically be decreased if needed.

           increase
               The output video dimensions will automatically be increased if needed.

           One useful instance of this option is that when you know a specific device's maximum
           allowed resolution, you can use this to limit the output video to that, while
           retaining the aspect ratio. For example, device A allows 1280x720 playback, and your
           video is 1920x800. Using this option (set it to decrease) and specifying 1280x720 to
           the command line makes the output 1280x533.

           Please note that this is a different thing than specifying -1 for w or h, you still
           need to specify the output resolution for this option to work.

       The values of the w and h options are expressions containing the following constants:

       in_w
       in_h
           The input width and height

       iw
       ih  These are the same as in_w and in_h.

       out_w
       out_h
           The output (scaled) width and height

       ow
       oh  These are the same as out_w and out_h

       a   The same as iw / ih

       sar input sample aspect ratio

       dar The input display aspect ratio. Calculated from "(iw / ih) * sar".

       hsub
       vsub
           horizontal and vertical input chroma subsample values. For example for the pixel
           format "yuv422p" hsub is 2 and vsub is 1.

       ohsub
       ovsub
           horizontal and vertical output chroma subsample values. For example for the pixel
           format "yuv422p" hsub is 2 and vsub is 1.

       Examples

       ·   Scale the input video to a size of 200x100

                   scale=w=200:h=100

           This is equiva