NAME

       wavpack - encode wav files to wavpack

SYNOPSIS

       wavpack [-options] INFILE... [-o OUTFILE]

DESCRIPTION

       wavpack encodes the specified source file(s) into WavPack files using the options provided. The source
       files may be uncompressed WAV files, existing WavPack files, or raw PCM data. The resulting filename will
       be source-name.wv unless overridden with the -o switch. Multiple input files may be specified resulting
       in multiple WavPack files, and in that case -o may be used to specify an alternate target directory.
       Stdin and stdout may be specified with “-”. In the case of transcoding from existing WavPack files, all
       tags are copied (and may be modified with additional args) and unless an alternate name or directory is
       specified, the source files are safely overwritten. To decode WavPack files back to wav or raw PCM use
       the wvunpack program.

OPTIONS

       -a
           Adobe Audition (CoolEdit) mode for 32-bit floats

       --allow-huge-tags
           allow tag data up to 16 MB (embedding > 1 MB is not recommended for portable devices and may not work
           with some programs including WavPack pre-4.70)

       -bn
           enable hybrid compression, n = 2.0 to 23.9 bits/sample, or n = 24-9600 kbits/second (kbps)

       --blocksize=n
           specify block size in samples (max = 131072 and min = 16 with --merge-blocks, otherwise 128)

       -c
           create correction file (.wvc) for hybrid mode (results in 2-file lossless compression)

       -cc
           maximum hybrid compression (hurts lossy quality & decode speed)

       --channel-order=list
           specify (comma separated) channel order if not Microsoft standard (which is
           FL,FR,FC,LFE,BL,BR,FLC,FRC,BC,SL,SR,TC,TFL,TFC,TFR,TBL,TBC,TBR); specify “...”  to indicate that
           channels are not assigned to specific speakers, or terminate list with “...”  to indicate that any
           channels beyond those specified are unassigned

       -d
           delete source file if successful (use with caution!)

       -f
           fast mode (fast, but some compromise in compression ratio)

       -h
           high quality (better compression ratio, but slower encode and decode than default mode)

       -hh
           very high quality (best compression, but slowest and NOT recommended for use on portable playback
           devices)

       --help
           display extended help

       -i
           ignore length in wav header (no pipe output allowed)

       -jn
           joint-stereo override (0 = left/right, 1 = mid/side)

       -m
           compute & store MD5 signature of raw audio data

       --merge-blocks
           merge consecutive blocks with equal redundancy (used with --blocksize option and is useful for files
           generated by the lossyWAV program or decoded HDCD files)

       -n
           calculate average and peak quantization noise (hybrid only, reference fullscale sine)

       --no-utf8-convert
           don't recode passed tags from local encoding to UTF-8, assume they are in UTF-8 already

       -o OUTFILE
           specify output filename (only if single source file) or target directory (must exist)

       --optimize-mono
           optimization for stereo files that are really mono (result may be incompatible with very old
           decoders)

       -p
           practical float storage (also 32-bit ints, no longer technically lossless)

       --pair-unassigned-chans
           encode unassigned channels into stereo pairs

       -q
           quiet (keep console output to a minimum)

       -r
           generate a new RIFF WAV header (any extra RIFF info in original file will be discarded)

       --raw-pcm
           intput data is raw pcm (44,100 Hz, 16-bit, 2-channels)

       --raw-pcm=sr,bits,chans
           intput data is raw pcm with specified sample-rate, bit-depth, and number of channels (specify 32f for
           32-bit floating point data)

       -sn
           override default hybrid mode noise shaping where n is a float value between -1.0 and 1.0; negative
           values move noise lower in freq, positive values move noise higher in freq, use 0 for no shaping
           (white noise)

       -t
           copy input file's time stamp to output file(s)

       --use-dns
           force use of dynamic noise shaping (hybrid mode only)

       -v
           verify output file integrity after write (not for piped output)

       --version
           write program version to stdout

       -w “Field=Value”
           write specified text metadata to APEv2 tag

       -w “Field=@file.ext”
           write specified text metadata from file to APEv2 tag, normally used for embedded cuesheets and logs
           (field names “Cuesheet” and “Log”)

       --write-binary-tag “Field=@file.ext”
           write the specified binary metadata file to APEv2 tag, normally used for cover art with the specified
           field name “Cover Art (Front)”

       -x[n]
           extra encode processing (optional n = 1 to 6, 1=default), -x1 to -x3 to choose best of predefined
           filters, -x4 to -x6 to generate custom filters (very slow!)

       -y
           yes to all warnings (use with caution!)

       -z[n]
           don't set (n = 0 or omitted) or set (n = 1) console title to indicate progress (leaves "WavPack
           Completed")

SEE ALSO

       wvunpack(1), wvgain(1)

       Please visit www.wavpack.com for more information

COPYRIGHT

       This manual page was written by Sebastian Dröge <slomo@debian.org> and David Bryant <david@wavpack.com>.
       Permission is granted to copy, distribute and/or modify this document under the terms of the BSD License.

AUTHORS

       Sebastian Dröge <slomo@debian.org>
           Original author

       David Bryant <david@wavpack.com>
           Updates

COPYRIGHT

       Copyright © 2005 Sebastian Dröge
       Copyright © 2015 David Bryant

WavPack 4.75.0                                     2015-05-10                                         WAVPACK(1)