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wavpack(1)                WavPack Executable Programs               wavpack(1)




NAME

       wavpack - encode audio 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 any of the file format
       types listed below, and the audio data may be either uncompressed PCM
       or DSD (depending on the format). Raw audio may also be specified (see
       --raw-pcm). The output 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 their
       original format (or raw audio) use the wvunpack program.

   INPUT FILE FORMATS
       o   Microsoft RIFF, extension ".wav", includes BWF and RF64 varients

       o   WavPack, extension ".wv", trancode operation, tags copied

       o   Apple Core Audio, extension ".caf"

       o   Sony Wave64, extension ".w64"

       o   Philips DSDIFF, extension ".dff"

       o   Sony DSD Stream, extension ".dsf"


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), not available with DSD audio

       --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

       --cross-decorr
           use cross-channel correlation in hybrid mode (on by default in
           lossless mode and with -cc option)

       -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)

       --import-id3
           import applicable tag items from ID3v2.3 tag present in DSF files
           into APEv2 tag (if there are > 1 MB cover images present add
           --allow-huge-tags to include them, and -r if you do not want large
           images appearing twice in the WavPack file, although this will
           remove the entire ID3 tag wrapper)

       -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)

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

       --pre-quantize=bits
           pre-quantize samples to bits depth BEFORE encoding and MD5
           calculation (common use would be --pre-quantize=20 for 24-bit or
           float material recorded with typical converters)

       -q
           quiet (keep console output to a minimum)

       -r
           remove file headers (file-appropriate headers will be regenerated
           during unpacking)

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

       --raw-pcm=sr,bits[f|s|u],chans,[le|be]
           input data is raw pcm with specified sample-rate, bit-depth
           (float,unsigned,signed), number of channels, and endianness
           (defaulted parameters may be omitted, specify bits=1 for DSD)

       --raw-pcm-skip=begin[,end]
           skip begin bytes before encoding raw PCM (header) and skip end
           bytes at the EOF (trailer)

       -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 Encoder
           write actual encoder metadata to APEv2 tag (e.g., "Encoder=WavPack
           5.0.0")

       -w Settings
           write actual user settings metadata to APEv2 tag (e.g.,
           "Settings=-hb384cx3")

       -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), wvtag(1)

       Please visit www.wavpack.com for more information


COPYRIGHT

       This manual page was written by Sebastian DrA9|ge <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 DrA9|ge <slomo@debian.org>
           Original author

       David Bryant <david@wavpack.com>
           Updates


COPYRIGHT

       Copyright (C) 2005 Sebastian DrA9|ge
       Copyright (C) 2017 David Bryant



WavPack 5.1.0                     2017-01-13                        wavpack(1)

wavpack 5.1.0 - Generated Fri Jan 27 15:49:40 CST 2017
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