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flac(1)             Free Lossless Audio Codec conversion tool            flac(1)


       flac - Free Lossless Audio Codec


       flac [ OPTIONS ] [ infile.wav | infile.rf64 | infile.aiff | infile.raw |
       infile.flac | infile.oga | infile.ogg | - ... ]

       flac [ -d | --decode | -t | --test | -a | --analyze ] [ OPTIONS ] [
       infile.flac | infile.oga | infile.ogg | - ... ]


       flac is a command-line tool for encoding, decoding, testing and analyzing
       FLAC streams.


       flac supports as input RIFF WAVE, Wave64, RF64, AIFF, FLAC or Ogg FLAC
       format, or raw interleaved samples.  The decoder currently can output to
       RIFF WAVE, Wave64, RF64, or AIFF format, or raw interleaved samples.
       flac only supports linear PCM samples (in other words, no A-LAW, uLAW,
       etc.), and the input must be between 4 and 32 bits per sample.

       flac assumes that files ending in ".wav" or that have the RIFF WAVE
       header present are WAVE files, files ending in ".w64" or have the Wave64
       header present are Wave64 files, files ending in ".rf64" or have the RF64
       header present are RF64 files, files ending in ".aif" or ".aiff" or have
       the AIFF header present are AIFF files, files ending in ".flac" or have
       the FLAC header present are FLAC files and files ending in ".oga" or
       ".ogg" or have the Ogg FLAC header present are Ogg FLAC files.

       Other than this, flac makes no assumptions about file extensions, though
       the convention is that FLAC files have the extension ".flac" (or ".fla"
       on ancient "8.3" file systems like FAT-16).

       Before going into the full command-line description, a few other things
       help to sort it out: 1.  flac encodes by default, so you must use -d to
       decode 2.  the options -0 ..  -8 (or -fast and -best) that control the
       compression level actually are just synonyms for different groups of
       specific encoding options (described later) and you can get the same
       effect by using the same options.  When specific options are specified
       they take priority over the compression level no matter the order 3.
       flac behaves similarly to gzip in the way it handles input and output
       files 4.  the order in which options are specified is generally not

       Skip to the examples below for examples of some common tasks.

       flac will be invoked one of four ways, depending on whether you are
       encoding, decoding, testing, or analyzing.  Encoding is the default
       invocation, but can be switch to decoding with -d, analysis with -a or
       testing with -t.  Depending on which way is chosen, encoding, decoding,
       analysis or testing options can be used, see section OPTIONS for details.
       General options can be used for all.

       If only one inputfile is specified, it may be "-" for stdin.  When stdin
       is used as input, flac will write to stdout.  Otherwise flac will perform
       the desired operation on each input file to similarly named output files
       (meaning for encoding, the extension will be replaced with ".flac", or
       appended with ".flac" if the input file has no extension, and for
       decoding, the extension will be ".wav" for WAVE output and ".raw" for raw
       output).  The original file is not deleted unless -delete-input-file is

       If you are encoding/decoding from stdin to a file, you should use the -o
       option like so:

              flac [options] -o outputfile
              flac -d [options] -o outputfile

       which are better than:

              flac [options] > outputfile
              flac -d [options] > outputfile

       since the former allows flac to seek backwards to write the STREAMINFO or
       RIFF WAVE header contents when necessary.

       Also, you can force output data to go to stdout using -c.

       To encode or decode files that start with a dash, use - to signal the end
       of options, to keep the filenames themselves from being treated as

              flac -V -- -01-filename.wav

       The encoding options affect the compression ratio and encoding speed.
       The format options are used to tell flac the arrangement of samples if
       the input file (or output file when decoding) is a raw file.  If it is a
       RIFF WAVE, Wave64, RF64, or AIFF file the format options are not needed
       since they are read from the file's header.

       In test mode, flac acts just like in decode mode, except no output file
       is written.  Both decode and test modes detect errors in the stream, but
       they also detect when the MD5 signature of the decoded audio does not
       match the stored MD5 signature, even when the bitstream is valid.

       flac can also re-encode FLAC files.  In other words, you can specify a
       FLAC or Ogg FLAC file as an input to the encoder and it will decoder it
       and re-encode it according to the options you specify.  It will also
       preserve all the metadata unless you override it with other options (e.g.
       specifying new tags, seekpoints, cuesheet, padding, etc.).

       flac has been tuned so that the default settings yield a good speed vs.
       compression tradeoff for many kinds of input.  However, if you are
       looking to maximize the compression rate or speed, or want to use the
       full power of FLAC's metadata system, see the page titled `About the FLAC
       Format' on the FLAC website.


       Some common encoding tasks using flac:

       flac abc.wav
              Encode abc.wav to abc.flac using the default compression setting.
              abc.wav is not deleted.

       flac --delete-input-file abc.wav
              Like above, except abc.wav is deleted if there were no errors.

       flac --delete-input-file -w abc.wav
              Like above, except abc.wav is deleted if there were no errors or

       flac --best abc.wav
              Encode abc.wav to abc.flac using the highest compression setting.

       flac --verify abc.wav
              Encode abc.wav to abc.flac and internally decode abc.flac to make
              sure it matches abc.wav.

       flac -o my.flac abc.wav
              Encode abc.wav to my.flac.

       flac -T "TITLE=Bohemian Rhapsody" -T "ARTIST=Queen" abc.wav
              Encode abc.wav and add some tags at the same time to abc.flac.

       flac *.wav
              Encode all .wav files in the current directory.

       flac abc.aiff
              Encode abc.aiff to abc.flac.

       flac abc.rf64
              Encode abc.rf64 to abc.flac.

       flac abc.w64
              Encode abc.w64 to abc.flac.

       flac abc.flac --force
              This one's a little tricky: notice that flac is in encode mode by
              default (you have to specify -d to decode) so this command
              actually recompresses abc.flac back to abc.flac.  -force is needed
              to make sure you really want to overwrite abc.flac with a new
              version.  Why would you want to do this?  It allows you to
              recompress an existing FLAC file with (usually) higher compression
              options or a newer version of FLAC and preserve all the metadata
              like tags too.

       Some common decoding tasks using flac:

       flac -d abc.flac
              Decode abc.flac to abc.wav.  abc.flac is not deleted.  NOTE:
              Without -d it means re-encode abc.flac to abc.flac (see above).

       flac -d --force-aiff-format abc.flac
       flac -d -o abc.aiff abc.flac : Two different ways of decoding abc.flac to
       abc.aiff (AIFF format).  abc.flac is not deleted.

       flac -d --force-rf64-format abc.flac
       flac -d -o abc.rf64 abc.flac : Two different ways of decoding abc.flac to
       abc.rf64 (RF64 format).  abc.flac is not deleted.

       flac -d --force-wave64-format abc.flac
       flac -d -o abc.w64 abc.flac : Two different ways of decoding abc.flac to
       abc.w64 (Wave64 format).  abc.flac is not deleted.

       flac -d -F abc.flac
              Decode abc.flac to abc.wav and don't abort if errors are found
              (useful for recovering as much as possible from corrupted files).


       A summary of options is included below.  For a complete description, see
       the HTML documentation.

       -v, --version
              Show the flac version number

       -h, --help
              Show basic usage and a list of all options

       -H, --explain
              Show detailed explanation of usage and all options

       -d, --decode
              Decode (the default behavior is to encode)

       -t, --test
              Test a flac encoded file (same as -d except no decoded file is

       -a, --analyze
              Analyze a FLAC encoded file (same as -d except an analysis file is

       -c, --stdout
              Write output to stdout

       -s, --silent
              Silent mode (do not write runtime encode/decode statistics to

              Do not print anything of any kind, including warnings or errors.
              The exit code will be the only way to determine successful

              Do not convert tags from local charset to UTF-8.  This is useful
              for scripts, and setting tags in situations where the locale is
              wrong.  This option must appear before any tag options!

       -w, --warnings-as-errors
              Treat all warnings as errors (which cause flac to terminate with a
              non-zero exit code).

       -f, --force
              Force overwriting of output files.  By default, flac warns that
              the output file already exists and continues to the next file.

       -o filename, --output-name=filename
              Force the output file name (usually flac just changes the
              extension).  May only be used when encoding a single file.  May
              not be used in conjunction with --output-prefix.

              Prefix each output file name with the given string.  This can be
              useful for encoding or decoding files to a different directory.
              Make sure if your string is a path name that it ends with a
              trailing `/' (slash).

              Automatically delete the input file after a successful encode or
              decode.  If there was an error (including a verify error) the
              input file is left intact.

              Output files have their timestamps/permissions set to match those
              of their inputs (this is default).  Use --no-preserve-modtime to
              make output files have the current time and default permissions.

              If encoding, save WAVE, RF64, or AIFF non-audio chunks in FLAC
              metadata.  If decoding, restore any saved non-audio chunks from
              FLAC metadata when writing the decoded file.  Foreign metadata
              cannot be transcoded, e.g. WAVE chunks saved in a FLAC file cannot
              be restored when decoding to AIFF.  Input and output must be
              regular files (not stdin or stdout).

              Like --keep-foreign-metadata, but without throwing an error if
              foreign metadata cannot be found or restored, instead printing a

              Skip over the first number of samples of the input.  This works
              for both encoding and decoding, but not testing.  The alternative
              form can be used to specify minutes, seconds, and
              fractions of a second.

              Stop at the given sample number for each input file.  This works
              for both encoding and decoding, but not testing.  The given sample
              number is not included in the decoded output.  The alternative
              form can be used to specify minutes, seconds, and
              fractions of a second.  If a `+' (plus) sign is at the beginning,
              the --until point is relative to the --skip point.  If a `-'
              (minus) sign is at the beginning, the --until point is relative to
              end of the audio.

       --ogg  When encoding, generate Ogg FLAC output instead of native FLAC.
              Ogg FLAC streams are FLAC streams wrapped in an Ogg transport
              layer.  The resulting file should have an `.oga' extension and
              will still be decodable by flac.  When decoding, force the input
              to be treated as Ogg FLAC.  This is useful when piping input from
              stdin or when the filename does not end in `.oga' or `.ogg'.

              When used with --ogg, specifies the serial number to use for the
              first Ogg FLAC stream, which is then incremented for each
              additional stream.  When encoding and no serial number is given,
              flac uses a random number for the first stream, then increments it
              for each additional stream.  When decoding and no number is given,
              flac uses the serial number of the first page.

              Includes the residual signal in the analysis file.  This will make
              the file very big, much larger than even the decoded file.

              Generates a gnuplot file for every subframe; each file will
              contain the residual distribution of the subframe.  This will
              create a lot of files.

              Set the beginning and ending cuepoints to decode.  The optional
              first #.# is the track and index point at which decoding will
              start; the default is the beginning of the stream.  The optional
              second #.# is the track and index point at which decoding will
              end; the default is the end of the stream.  If the cuepoint does
              not exist, the closest one before it (for the start point) or
              after it (for the end point) will be used.  If those don't exist,
              the start of the stream (for the start point) or end of the stream
              (for the end point) will be used.  The cuepoints are merely
              translated into sample numbers then used as --skip and --until.  A
              CD track can always be cued by, for example, --cue=9.1-10.1 for
              track 9, even if the CD has no 10th track.

       -F, --decode-through-errors
              By default flac stops decoding with an error and removes the
              partially decoded file if it encounters a bitstream error.  With
              -F, errors are still printed but flac will continue decoding to
              completion.  Note that errors may cause the decoded audio to be
              missing some samples or have silent sections.

              Applies ReplayGain values while decoding.  WARNING: THIS IS NOT
              WITH THIS OPTION. This option is useful for example in transcoding
              media servers, where the client does not support ReplayGain.  For
              details on the use of this option, see the section ReplayGain
              application specification.

       -V, --verify
              Verify a correct encoding by decoding the output in parallel and
              comparing to the original

       --lax  Allow encoder to generate non-Subset files.  The resulting FLAC
              file may not be streamable or might have trouble being played in
              all players (especially hardware devices), so you should only use
              this option in combination with custom encoding options meant for

              Calculate ReplayGain values and store them as FLAC tags, similar
              to vorbisgain.  Title gains/peaks will be computed for each input
              file, and an album gain/peak will be computed for all files.  All
              input files must have the same resolution, sample rate, and number
              of channels.  Only mono and stereo files are allowed, and the
              sample rate must be one of 8, 11.025, 12, 16, 22.05, 24, 32, 44.1,
              or 48 kHz.  Also note that this option may leave a few extra bytes
              in a PADDING block as the exact size of the tags is not known
              until all files are processed.  Note that this option cannot be
              used when encoding to standard output (stdout).

              Import the given cuesheet file and store it in a CUESHEET metadata
              block.  This option may only be used when encoding a single file.
              A seekpoint will be added for each index point in the cuesheet to
              the SEEKTABLE unless --no-cued-seekpoints is specified.

              Import a picture and store it in a PICTURE metadata block.  More
              than one --picture option can be specified.  Either a filename for
              the picture file or a more complete specification form can be
              used.  The SPECIFICATION is a string whose parts are separated by
              | (pipe) characters.  Some parts may be left empty to invoke
              default values.  FILENAME is just shorthand for "||||FILENAME".
              For the format of SPECIFICATION, see the section picture

              Align encoding of multiple CD format files on sector boundaries.
              See the HTML documentation for more information.  This option is
              DEPRECATED and may not exist in future versions of flac.

              When encoding to flac, ignore the file size headers in WAV and
              AIFF files to attempt to work around problems with over-sized or
              malformed files.  WAV and AIFF files both have an unsigned 32 bit
              numbers in the file header which specifes the length of audio
              data.  Since this number is unsigned 32 bits, that limits the size
              of a valid file to being just over 4 Gigabytes.  Files larger than
              this are mal-formed, but should be read correctly using this

       -S {#|X|#x|#s}, --seekpoint={#|X|#x|#s}
              Include a point or points in a SEEKTABLE.  Using #, a seek point
              at that sample number is added.  Using X, a placeholder point is
              added at the end of a the table.  Using #x, # evenly spaced seek
              points will be added, the first being at sample 0.  Using #s, a
              seekpoint will be added every # seconds (# does not have to be a
              whole number; it can be, for example, 9.5, meaning a seekpoint
              every 9.5 seconds).  You may use many -S options; the resulting
              SEEKTABLE will be the unique-ified union of all such values.  With
              no -S options, flac defaults to `-S 10s'.  Use --no-seektable for
              no SEEKTABLE.  Note: `-S #x' and `-S #s' will not work if the
              encoder can't determine the input size before starting.  Note: if
              you use `-S #' and # is >= samples in the input, there will be
              either no seek point entered (if the input size is determinable
              before encoding starts) or a placeholder point (if input size is
              not determinable).

       -P #, --padding=#
              Tell the encoder to write a PADDING metadata block of the given
              length (in bytes) after the STREAMINFO block.  This is useful if
              you plan to tag the file later with an APPLICATION block; instead
              of having to rewrite the entire file later just to insert your
              block, you can write directly over the PADDING block.  Note that
              the total length of the PADDING block will be 4 bytes longer than
              the length given because of the 4 metadata block header bytes.
              You can force no PADDING block at all to be written with --no-
              padding.  The encoder writes a PADDING block of 8192 bytes by
              default (or 65536 bytes if the input audio stream is more that 20
              minutes long).

              Add a FLAC tag.  The comment must adhere to the Vorbis comment
              spec; i.e. the FIELD must contain only legal characters,
              terminated by an `equals' sign.  Make sure to quote the comment if
              necessary.  This option may appear more than once to add several
              comments.  NOTE: all tags will be added to all encoded files.

              Like --tag, except FILENAME is a file whose contents will be read
              verbatim to set the tag value.  The contents will be converted to
              UTF-8 from the local charset.  This can be used to store a
              cuesheet in a tag (e.g. --tag-from-file="CUESHEET=image.cue").  Do
              not try to store binary data in tag fields!  Use APPLICATION
              blocks for that.

       -b #, --blocksize=#
              Specify the blocksize in samples.  The default is 1152 for -l 0,
              else 4096.  For subset streams this must be <= 4608 if the
              samplerate <= 48kHz, for subset streams with higher samplerates it
              must be <= 16384.

       -m, --mid-side
              Try mid-side coding for each frame (stereo input only)

       -M, --adaptive-mid-side
              Adaptive mid-side coding for all frames (stereo input only)

       -0..-8, --compression-level-0..--compression-level-8
              Fastest compression..highest compression (default is -5).  These
              are synonyms for other options:

       -0, --compression-level-0
              Synonymous with -l 0 -b 1152 -r 3 --no-mid-side

       -1, --compression-level-1
              Synonymous with -l 0 -b 1152 -M -r 3

       -2, --compression-level-2
              Synonymous with -l 0 -b 1152 -m -r 3

       -3, --compression-level-3
              Synonymous with -l 6 -b 4096 -r 4 --no-mid-side

       -4, --compression-level-4
              Synonymous with -l 8 -b 4096 -M -r 4

       -5, --compression-level-5
              Synonymous with -l 8 -b 4096 -m -r 5

       -6, --compression-level-6
              Synonymous with -l 8 -b 4096 -m -r 6 -A subdivide_tukey(2)

       -7, --compression-level-7
              Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(2)

       -8, --compression-level-8
              Synonymous with -l 12 -b 4096 -m -r 6 -A subdivide_tukey(3)

       --fast Fastest compression.  Currently synonymous with -0.

       --best Highest compression.  Currently synonymous with -8.

       -e, --exhaustive-model-search
              Do exhaustive model search (expensive!)

       -A function, --apodization=function
              Window audio data with given the apodization function.  See
              section Apodization functions for details.

       -l #, --max-lpc-order=#
              Specifies the maximum LPC order.  This number must be <= 32.  For
              subset streams, it must be <=12 if the sample rate is <=48kHz.  If
              0, the encoder will not attempt generic linear prediction, and use
              only fixed predictors.  Using fixed predictors is faster but
              usually results in files being 5-10% larger.

       -p, --qlp-coeff-precision-search
              Do exhaustive search of LP coefficient quantization (expensive!).
              Overrides -q; does nothing if using -l 0

       -q #, --qlp-coeff-precision=#
              Precision of the quantized linear-predictor coefficients, 0 => let
              encoder decide (min is 5, default is 0)

       -r [#,]#, --rice-partition-order=[#,]#
              Set the [min,]max residual partition order (0..15).  min defaults
              to 0 if unspecified.  Default is -r 5.

              Set the byte order for samples

              Set number of channels.

              Set bits per sample.

              Set sample rate (in Hz).

              Set the sign of samples (the default is signed).

              Specify the size of the raw input in bytes.  If you are encoding
              raw samples from stdin, you must set this option in order to be
              able to use --skip, --until, --cuesheet, or other options that
              need to know the size of the input beforehand.  If the size given
              is greater than what is found in the input stream, the encoder
              will complain about an unexpected end-of-file.  If the size given
              is less, samples will be truncated.

              Force input (when encoding) or output (when decoding) to be
              treated as raw samples (even if filename ends in .wav).

              Force the decoder to output AIFF format.  This option is not
              needed if the output filename (as set by -o) ends with .aif or
              .aiff.  Also, this option has no effect when encoding since input
              AIFF is auto-detected.

              Force the decoder to output RF64 format.  This option is not
              needed if the output filename (as set by -o) ends with .rf64.
              Also, this option has no effect when encoding since input RF64 is

              Force the decoder to output Wave64 format.  This option is not
              needed if the output filename (as set by -o) ends with .w64.
              Also, this option has no effect when encoding since input Wave64
              is auto-detected.


       These flags can be used to invert the sense of the corresponding normal

   ReplayGain application specification
       The option --apply-replaygain-which-is-not-lossless[=<specification>]
       applies ReplayGain values while decoding.  WARNING: THIS IS NOT LOSSLESS.
       This option is useful for example in transcoding media servers, where the
       client does not support ReplayGain.

       The equals sign and <specification> is optional.  If omitted, the default
       specification is 0aLn1.

       The <specification> is a shorthand notation for describing how to apply
       ReplayGain.  All components are optional but order is important.  `[]'
       means `optional'.  `|' means `or'.  `{}' means required.  The format is:


       In which the following parameters are used:

       o preamp: A floating point number in dB.  This is added to the existing
         gain value.

       o a|t: Specify `a' to use the album gain, or `t' to use the track gain.
         If tags for the preferred kind (album/track) do not exist but tags for
         the other (track/album) do, those will be used instead.

       o l|L: Specify `l' to peak-limit the output, so that the ReplayGain peak
         value is full-scale.  Specify `L' to use a 6dB hard limiter that kicks
         in when the signal approaches full-scale.

       o n{0|1|2|3}: Specify the amount of noise shaping.  ReplayGain synthesis
         happens in floating point; the result is dithered before converting
         back to integer.  This quantization adds noise.  Noise shaping tries to
         move the noise where you won't hear it as much.  0 means no noise
         shaping, 1 means `low', 2 means `medium', 3 means `high'.

       For example, the default of 0aLn1 means 0dB preamp, use album gain, 6dB
       hard limit, low noise shaping.  --apply-replaygain-which-is-not-
       lossless=3 means 3dB preamp, use album gain, no limiting, no noise

       flac uses the ReplayGain tags for the calculation.  If a stream does not
       have the required tags or they can't be parsed, decoding will continue
       with a warning, and no ReplayGain is applied to that stream.

   Picture specification
       This described the specification used for the --picture option.

       TYPE is optional; it is a number from one of:

        0. Other

        1. 32x32 pixels `file icon' (PNG only)

        2. Other file icon

        3. Cover (front)

        4. Cover (back)

        5. Leaflet page

        6. Media (e.g. label side of CD)

        7. Lead artist/lead performer/soloist

        8. Artist/performer

        9. Conductor

       10. Band/Orchestra

       11. Composer

       12. Lyricist/text writer

       13. Recording Location

       14. During recording

       15. During performance

       16. Movie/video screen capture

       17. A bright coloured fish

       18. Illustration

       19. Band/artist logotype

       20. Publisher/Studio logotype

       The default is 3 (front cover).  There may only be one picture each of
       type 1 and 2 in a file.

       MIME-TYPE is optional; if left blank, it will be detected from the file.
       For best compatibility with players, use pictures with MIME type
       image/jpeg or image/png.  The MIME type can also be --> to mean that FILE
       is actually a URL to an image, though this use is discouraged.

       DESCRIPTION is optional; the default is an empty string.

       The next part specifies the resolution and color information.  If the
       MIME-TYPE is image/jpeg, image/png, or image/gif, you can usually leave
       this empty and they can be detected from the file.  Otherwise, you must
       specify the width in pixels, height in pixels, and color depth in bits-
       per-pixel.  If the image has indexed colors you should also specify the
       number of colors used.  When manually specified, it is not checked
       against the file for accuracy.

       FILE is the path to the picture file to be imported, or the URL if MIME
       type is -->

       For example, "|image/jpeg|||../cover.jpg" will embed the JPEG file at
       ../cover.jpg, defaulting to type 3 (front cover) and an empty
       description.  The resolution and color info will be retrieved from the
       file itself.

       The specification
       "4|-->|CD|320x300x24/173|http://blah.blah/backcover.tiff" will embed the
       given URL, with type 4 (back cover), description "CD", and a manually
       specified resolution of 320x300, 24 bits-per-pixel, and 173 colors.  The
       file at the URL will not be fetched; the URL itself is stored in the
       PICTURE metadata block.

   Apodization functions
       To improve LPC analysis, audio data is windowed .  The window can be
       selected with one or more -A options.  Possible functions are: bartlett,
       bartlett_hann, blackman, blackman_harris_4term_92db, connes, flattop,
       gauss(STDDEV), hamming, hann, kaiser_bessel, nuttall, rectangle,
       triangle, tukey(P), partial_tukey(n[/ov[/P]]),
       punchout_tukey(n[/ov[/P]]), subdivide_tukey(n[/P]) welch.

       o For gauss(STDDEV), STDDEV is the standard deviation (0<STDDEV<=0.5).

       o For tukey(P), P specifies the fraction of the window that is tapered
         (0<=P<=1; P=0 corresponds to "rectangle" and P=1 corresponds to

       o For partial_tukey(n) and punchout_tukey(n), n apodization functions are
         added that span different parts of each block.  Values of 2 to 6 seem
         to yield sane results.  If necessary, an overlap can be specified, as
         can be the taper parameter, for example partial_tukey(2/0.2) or
         partial_tukey(2/0.2/0.5).  ov should be smaller than 1 and can be
         negative.  The use of this is that different parts of a block are
         ignored as the might contain transients which are hard to predict
         anyway.  The encoder will try each different added apodization (each
         covering a different part of the block) to see which resulting
         predictor results in the smallest representation.

       o subdivide_tukey(n) is a more efficient reimplementation of
         partial_tukey and punchout_tukey taken together, recycling as much data
         as possible.  It combines all possible non-redundant partial_tukey(n)
         and punchout_tukey(n) up to the n specified.  Specifying
         subdivide_tukey(3) is equivalent to specifying tukey, partial_tukey(2),
         partial_tukey(3) and punchout_tukey(3), specifying subdivide_tukey(5)
         equivalently adds partial_tukey(4), punchout_tukey(4), partial_tukey(5)
         and punchout_tukey(5).  To be able to reuse data as much as possible,
         the tukey taper is taken equal for all windows, and the P specified is
         applied for the smallest used window.  In other words,
         subdivide_tukey(2/0.5) results in a taper equal to that of tukey(0.25)
         and subdivide_tukey(5) in a taper equal to that of tukey(0.1).  The
         default P for subdivide_tukey when none is specified is 0.5.

       Note that P, STDDEV and ov are locale specific, so a comma as decimal
       separator might be required instead of a dot.  Use scientific notation
       for a locale-independent specification, for example tukey(5e-1) instead
       of tukey(0.5) or tukey(0,5).

       More than one -A option (up to 32) may be used.  Any function that is
       specified erroneously is silently dropped.  The encoder chooses suitable
       defaults in the absence of any -A options; any -A option specified
       replaces the default(s).

       When more than one function is specified, then for every subframe the
       encoder will try each of them separately and choose the window that
       results in the smallest compressed subframe.  Multiple functions can
       greatly increase the encoding time.




       This manual page was initially written by Matt Zimmerman <>
       for the Debian GNU/Linux system (but may be used by others).  It has been
       kept up-to-date by the Foundation.

Version 1.4.2                                                            flac(1)

flac 1.4.2 - Generated Sat Jan 7 08:21:25 CST 2023
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