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 * Microsoft RIFF, extension ".wav", includes BWF and RF64 varients * WavPack, extension ".wv", trancode operation, tags copied * Apple Core Audio, extension ".caf" * Sony Wave64, extension ".w64" * Philips DSDIFF, extension ".dff" * 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)
-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) Please visit www.wavpack.com for more information
COPYRIGHT
This manual page was written by Sebastian Drge <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 Drge <slomo@debian.org>
Original author
David Bryant <david@wavpack.com>
Updates
COPYRIGHT
Copyright 2005 Sebastian Drge Copyright 2016 David Bryant
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