lame
create mp3 audio files
see also :
sox
Synopsis
lame [options]
<infile> <outfile>
add an example, a script, a trick and tips
examples
Fixed bit rate jstereo 128kbs encoding:
lame sample.wav sample.mp3
Fixed bit rate jstereo 128 kbps encoding, highest quality
(recommended):
lame -h sample.wav sample.mp3
Fixed bit rate jstereo 112 kbps encoding:
lame -b 112 sample.wav sample.mp3
To disable joint stereo encoding (slightly faster, but less
quality at bitrates <= 128 kbps):
lame -m s sample.wav sample.mp3
Fast encode, low quality (no psycho-acoustics):
lame -f sample.wav sample.mp3
Variable bitrate (use -V n to adjust quality/filesize):
lame -h -V 6 sample.wav sample.mp3
Streaming mono 22.05 kHz raw pcm, 24 kbps output:
cat inputfile | lame -r -m m
-b 24 -s 22.05 - - >
output
Streaming mono 44.1 kHz raw pcm, with downsampling to 22.05 kHz:
cat inputfile | lame -r -m m
-b 24 --resample 22.05 - -
> output
Encode with the standard preset:
lame --preset standard sample.wav sample.mp3
source
Most efficient way to convert a large ALAC library to MP3
In the examples below, ~/Music/ is assumed as the
source directory.
Create a convert.sh script:
$ cat > convert
#!/bin/bash
input=$1
output=${input#.*}.mp3
ffmpeg -i "$input" -ac 2 -f wav - | lame -V 2 - "$output"
[Ctrl-D]
$ chmod +x convert
-
If you want a different location to be used for the outputs,
add this before ffmpeg:
output=~/Converted/${output#~/Music/}
mkdir -p "${output%/*}"
Convert using parallel from moreutils:
$ find ~/Music/ -type f -name '*.mp4' -print0 | xargs -0 parallel ./convert --
Not to be confused with GNU parallel, which uses a
different syntax:
$ find ~/Music/ -type f -name '*.mp4' | parallel ./convert
source
Dealing with conflicts on linux installs
This show there is already same package installed with other
version.
Try rpm -Uvh packagename*. This should low down your trouble.
description
LAME is a
program which can be used to create compressed audio files.
(Lame ain’t an MP3 encoder). These audio files can be
played back by popular MP3 players such as mpg123 or
madplay. To read from stdin, use "-" for
<infile>. To write to stdout, use "-"
for <outfile>.
options
Input
options:
-r
Assume the input file is raw pcm. Sampling rate and
mono/stereo/jstereo must be specified on the command line.
For each stereo sample, LAME expects the input data to be
ordered left channel first, then right channel. By default,
LAME expects them to be signed integers with a bitwidth of
16. Without -r, LAME will perform several
fseek()’s on the input file looking for WAV and
AIFF headers.
Might not be available on your
release.
-x
Swap bytes in the input file or
output file when using --decode.
For sorting out little
endian/big endian type problems. If your encodings sounds
like static, try this first.
Without using -x, LAME will treat input file as
native endian.
-s
sfreq
sfreq =
8/11.025/12/16/22.05/24/32/44.1/48
Required only
for raw PCM input files. Otherwise it will be determined
from the header of the input file.
LAME will
automatically resample the input file to one of the
supported MP3 samplerates if necessary.
--bitwidth
n
Input bit width per sample.
n = 8, 16, 24, 32 (default 16)
Required only
for raw PCM input files. Otherwise it will be determined
from the header of the input file.
--signed
Instructs LAME that the samples
from the input are signed (the default for 16, 24 and 32
bits raw pcm data).
Required only
for raw PCM input files.
--unsigned
Instructs LAME that the samples
from the input are unsigned (the default for 8 bits raw pcm
data, where 0x80 is zero).
Required only
for raw PCM input files and only available at bitwidth
8.
--little-endian
Instructs LAME that the samples
from the input are in little-endian form.
Required only
for raw PCM input files.
--big-endian
Instructs LAME that the samples
from the input are in big-endian form.
Required only
for raw PCM input files.
--mp2input
Assume the input file is a MPEG
Layer II (ie MP2) file.
If the filename ends in ".mp2" LAME will assume it
is a MPEG Layer II file. For stdin or Layer II files which
do not end in .mp2 you need to use this switch.
--mp3input
Assume the input file is a MP3
file.
Useful for downsampling from one mp3 to another. As an
example, it can be useful for streaming through an IceCast
server.
If the filename ends in ".mp3" LAME will assume it
is an MP3. For stdin or MP3 files which do not end in .mp3
you need to use this switch.
--nogap
file1 file2 ...
gapless encoding for a set of
contiguous files
--nogapout
dir
output dir for gapless encoding
(must precede --nogap)
Operational
options:
-m mode
mode = s, j, f, d, m, l,
r
Joint-stereo is
the default mode for stereo files with VBR when
-V is more than 4 or fixed bitrates of 160kbs
or less. At higher fixed bitrates or higher VBR settings,
the default is stereo.
(s)imple
stereo
In this mode, the encoder makes no use of potentially
existing correlations between the two input channels. It
can, however, negotiate the bit demand between both channel,
i.e. give one channel more bits if the other contains
silence or needs less bits because of a lower
complexity.
(j)oint
stereo
In this mode, the encoder will make use of a correlation
between both channels. The signal will be matrixed into a
sum ("mid"), computed by L+R, and difference
("side") signal, computed by L-R, and more
bits are allocated to the mid channel. This will effectively
increase the bandwidth if the signal does not have too much
stereo separation, thus giving a significant gain in
encoding quality.
Using mid/side
stereo inappropriately can result in audible compression
artifacts. To much switching between mid/side and regular
stereo can also sound bad. To determine when to switch to
mid/side stereo, LAME uses a much more sophisticated
algorithm than that described in the ISO documentation, and
thus is safe to use in joint stereo mode.
(f)orced MS
stereo
This mode will force MS stereo on all frames. It is slightly
faster than joint stereo, but it should be used only if you
are sure that every frame of the input file has very little
stereo separation.
(d)ual
mono
In this mode, the 2 channels will be totally independently
encoded. Each channel will have exactly half of the bitrate.
This mode is designed for applications like dual languages
encoding (for example: English in one channel and French in
the other). Using this encoding mode for regular stereo
files will result in a lower quality encoding.
(m)ono
The input will be encoded as a mono signal. If it was a
stereo signal, it will be downsampled to mono. The downmix
is calculated as the sum of the left and right channel,
attenuated by 6 dB.
(l)eft
channel only
The input will be encoded as a mono signal. If it was a
stereo signal, the left channel will be encoded only.
(r)ight
channel only
The input will be encoded as a mono signal. If it was a
stereo signal, the right channel will be encoded only.
-a
Mix the stereo input file to mono and encode as
mono.
The downmix is calculated as
the sum of the left and right channel, attenuated by 6
dB.
This option is
only needed in the case of raw PCM stereo input (because
LAME cannot determine the number of channels in the input
file). To encode a stereo PCM input file as mono, use
lame -m s -a.
For WAV and
AIFF input files, using -m will always produce
a mono .mp3 file from both mono and stereo input.
-d
Allows the left and right
channels to use different block size types.
--freeformat
Produces a free format
bitstream. With this option, you can use -b
with any bitrate higher than 8 kbps.
However, even
if an mp3 decoder is required to support free bitrates at
least up to 320 kbps, many players are unable to deal with
it.
Tests have
shown that the following decoders support free format:
FreeAmp up to 440 kbps
in_mpg123 up to 560 kbps
l3dec up to 310 kbps
LAME up to 560 kbps
MAD up to 640 kbps
--decode
Uses LAME for decoding to a wav
file. The input file can be any input type supported by
encoding, including layer II files. LAME uses a bugfixed
version of mpglib for decoding.
If
-t is used (disable wav header), LAME will
output raw pcm in native endian format. You can use
-x to swap bytes order.
This option is
not usable if the MP3 decoder was explicitly disabled
in the build of LAME.
-t
Disable writing of the INFO Tag on encoding.
This tag in embedded in frame 0
of the MP3 file. It includes some information about the
encoding options of the file, and in VBR it lets VBR aware
players correctly seek and compute playing times of VBR
files.
When
--decode is specified (decode to WAV),
this flag will disable writing of the WAV header. The output
will be raw pcm, native endian format. Use -x
to swap bytes.
--comp
arg
Instead of choosing bitrate,
using this option, user can choose compression ratio to
achieve.
--scale
n
--scale-l n
--scale-r n
Scales input (every channel,
only left channel or only right channel) by n. This
just multiplies the PCM data (after it has been converted to
floating point) by n.
n >
1: increase volume
n = 1: no effect
n < 1: reduce volume
Use with care,
since most MP3 decoders will truncate data which decodes to
values greater than 32768.
--replaygain-fast
Compute ReplayGain fast but
slightly inaccurately.
This computes
"Radio" ReplayGain on the input data stream after
user-specified volume-scaling and/or resampling.
The ReplayGain
analysis does not affect the content of a compressed
data stream itself, it is a value stored in the header of a
sound file. Information on the purpose of ReplayGain and the
algorithms used is available from
http://www.replaygain.org/.
Only the
"RadioGain" Replaygain value is computed, it is
stored in the LAME tag. The analysis is performed with the
reference volume equal to 89dB. Note: the reference volume
has been changed from 83dB on transition from version 3.95
to 3.95.1.
This switch is
enabled by default.
See also:
--replaygain-accurate,
--noreplaygain
--replaygain-accurate
Compute ReplayGain more
accurately and find the peak sample.
This enables
decoding on the fly, computes "Radio" ReplayGain
on the decoded data stream, finds the peak sample of the
decoded data stream and stores it in the file.
The ReplayGain
analysis does not affect the content of a compressed
data stream itself, it is a value stored in the header of a
sound file. Information on the purpose of ReplayGain and the
algorithms used is available from
http://www.replaygain.org/.
By default,
LAME performs ReplayGain analysis on the input data (after
the user-specified volume scaling). This behavior might give
slightly inaccurate results because the data on the output
of a lossy compression/decompression sequence differs from
the initial input data. When
--replaygain-accurate is specified the
mp3 stream gets decoded on the fly and the analysis is
performed on the decoded data stream. Although theoretically
this method gives more accurate results, it has several
disadvantages:
*
tests have shown that the difference between the
ReplayGain values computed on the input data and decoded
data is usually not greater than 0.5dB, although the minimum
volume difference the human ear can perceive is about
1.0dB
*
decoding on the fly significantly slows down the
encoding process
The apparent
advantage is that:
*
with
--replaygain-accurate the real peak
sample is determined and stored in the file. The knowledge
of the peak sample can be useful to decoders (players) to
prevent a negative effect called ’clipping’ that
introduces distortion into the sound.
Only the
"RadioGain" ReplayGain value is computed, it is
stored in the LAME tag. The analysis is performed with the
reference volume equal to 89dB. Note: the reference volume
has been changed from 83dB on transition from version 3.95
to 3.95.1.
This option is
not usable if the MP3 decoder was explicitly disabled
in the build of LAME. (Note: if LAME is compiled without the
MP3 decoder, ReplayGain analysis is performed on the input
data after user-specified volume scaling).
See also:
--replaygain-fast, --noreplaygain
--clipdetect
--noreplaygain
Disable ReplayGain
analysis.
By default
ReplayGain analysis is enabled. This switch disables it.
See also:
--replaygain-fast,
--replaygain-accurate
--clipdetect
Clipping detection.
Enable
--replaygain-accurate and print a message
whether clipping occurs and how far in dB the waveform is
from full scale.
This option is
not usable if the MP3 decoder was explicitly disabled
in the build of LAME.
See also:
--replaygain-accurate
--preset type |
[cbr] kbps
Use one of the built-in
presets.
Have a look at
the PRESETS section below.
--preset
help gives more infos about the the used options in
these presets.
--preset type |
[cbr] kbps
Use one of the built-in
presets.
--noasm
type
Disable specific assembly
optimizations ( mmx / 3dnow / sse ).
Quality will not increase, only speed will be reduced. If
you have problems running Lame on a Cyrix/Via processor,
disabling mmx optimizations might solve your problem.
Verbosity:
--disptime n
Set the delay in seconds
between two display updates.
--nohist
By default, LAME will display a
bitrate histogram while producing VBR mp3 files. This will
disable that feature.
Histogram display might not be available on your
release.
--silent
--quiet
Do not print anything on the
screen.
--verbose
Print a lot of information on
the screen.
--help
Display a list of available options.
Noise shaping
& psycho acoustic algorithms:
-q qual
0 <= qual <= 9
Bitrate is of
course the main influence on quality. The higher the
bitrate, the higher the quality. But for a given bitrate, we
have a choice of algorithms to determine the best
scalefactors and Huffman encoding (noise shaping).
-q 0:
use slowest & best possible version of all algorithms.
-q 0 and -q 1 are slow and may not produce
significantly higher quality.
-q 2:
recommended. Same as -h.
-q 5:
default value. Good speed, reasonable quality.
-q 7:
same as -f. Very fast, ok quality. Psycho acoustics
are used for pre-echo & M/S, but no noise shaping is
done.
-q 9:
disables almost all algorithms including psy-model. Poor
quality.
-h
Use some quality improvements. Encoding will be slower,
but the result will be of higher quality. The behavior is
the same as the -q 2 switch.
This switch is always enabled
when using VBR.
-f
This switch forces the encoder
to use a faster encoding mode, but with a lower quality. The
behavior is the same as the -q 7 switch.
Noise shaping
will be disabled, but psycho acoustics will still be
computed for bit allocation and pre-echo detection.
CBR (constant
bitrate, the default) options:
-b n
For MPEG-1 (sampling frequencies
of 32, 44.1 and 48 kHz)
n = 32, 40, 48, 56, 64,
80, 96, 112, 128, 160, 192, 224, 256, 320
For MPEG-2
(sampling frequencies of 16, 22.05 and 24 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128,
144, 160
For MPEG-2.5
(sampling frequencies of 8, 11.025 and 12 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64
Default is 128
for MPEG1 and 64 for MPEG2.
--cbr
enforce use of constant
bitrate
ABR (average
bitrate) options:
--abr n
Turns on encoding with a
targeted average bitrate of n kbits, allowing to use frames
of different sizes. The allowed range of n is 8 -
310, you can use any integer value within that range.
It can be
combined with the -b and -B switches like:
lame --abr 123 -b 64
-B 192 a.wav a.mp3 which would limit the
allowed frame sizes between 64 and 192 kbits.
The use of
-B is NOT RECOMMENDED. A 128 kbps CBR bitstream,
because of the bit reservoir, can actually have frames which
use as many bits as a 320 kbps frame. VBR modes minimize the
use of the bit reservoir, and thus need to allow 320 kbps
frames to get the same flexibility as CBR streams.
VBR (variable
bitrate) options:
-v
use variable bitrate (--vbr-new)
--vbr-old
Invokes the oldest, most tested
VBR algorithm. It produces very good quality files, though
is not very fast. This has, up through v3.89, been
considered the "workhorse" VBR algorithm.
--vbr-new
Invokes the newest VBR
algorithm. During the development of version 3.90,
considerable tuning was done on this algorithm, and it is
now considered to be on par with the original
--vbr-old. It has the added advantage of
being very fast (over twice as fast as
--vbr-old).
Enable VBR (Variable BitRate)
and specifies the value of VBR quality (default = 4). 0 =
highest quality.
ABR and VBR
options:
-b bitrate
For MPEG-1 (sampling
frequencies of 32, 44.1 and 48 kHz)
n = 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224,
256, 320
For MPEG-2
(sampling frequencies of 16, 22.05 and 24 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128,
144, 160
For MPEG-2.5
(sampling frequencies of 8, 11.025 and 12 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64
Specifies the
minimum bitrate to be used. However, in order to avoid
wasted space, the smallest frame size available will be used
during silences.
-B bitrate
For MPEG-1 (sampling
frequencies of 32, 44.1 and 48 kHz)
n = 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224,
256, 320
For MPEG-2
(sampling frequencies of 16, 22.05 and 24 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128,
144, 160
For MPEG-2.5
(sampling frequencies of 8, 11.025 and 12 kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64
Specifies the
maximum allowed bitrate.
Note: If you
own an mp3 hardware player build upon a MAS 3503 chip, you
must set maximum bitrate to no more than 224 kpbs.
-F
Strictly enforce the -b option.
This is mainly for use with
hardware players that do not support low bitrate mp3.
Without this
option, the minimum bitrate will be ignored for passages of
analog silence, i.e. when the music level is below the
absolute threshold of human hearing (ATH).
Experimental
options:
When LAME
searches for a "good" quantization, it has to
compare the actual one with the best one found so far. The
comparison says which one is better, the best so far or the
actual. The -X parameter selects between different
approaches to make this decision, -X0 being the
default mode:
-X0
The criteria are (in order of importance):
* less distorted scalefactor bands
* the sum of noise over the thresholds is lower
* the total noise is lower
-X1
The actual is better if the maximum noise over all
scalefactor bands is less than the best so far.
-X2
The actual is better if the total sum of noise is lower than
the best so far.
-X3
The actual is better if the total sum of noise is lower than
the best so far and the maximum noise over all scalefactor
bands is less than the best so far plus 2dB.
-X4
Not yet documented.
-X5
The criteria are (in order of importance):
* the sum of noise over the thresholds is lower
* the total sum of noise is lower
-X6
The criteria are (in order of importance):
* the sum of noise over the thresholds is lower
* the maximum noise over all scalefactor bands is lower
* the total sum of noise is lower
-X7
The criteria are:
* less distorted scalefactor bands
or
* the sum of noise over the thresholds is lower
-Y
lets LAME ignore noise in sfb21,
like in CBR
MP3
header/stream options:
n = (none,
default)
5 = 0/15 microseconds
c = citt j.17
All this does
is set a flag in the bitstream. If you have a PCM input file
where one of the above types of (obsolete) emphasis has been
applied, you can set this flag in LAME. Then the mp3 decoder
should de-emphasize the output during playback, although
most decoders ignore this flag.
A better
solution would be to apply the de-emphasis with a standalone
utility before encoding, and then encode without
-e.
-c
Mark the encoded file as being
copyrighted.
-o
Mark the encoded file as being a copy.
-p
Turn on CRC error protection.
It will add a cyclic redundancy
check (CRC) code in each frame, allowing to detect
transmission errors that could occur on the MP3 stream.
However, it takes 16 bits per frame that would otherwise be
used for encoding, and then will slightly reduce the sound
quality.
--nores
Disable the bit reservoir. Each
frame will then become independent from previous ones, but
the quality will be lower.
--strictly-enforce-ISO
With this option, LAME will
enforce the 7680 bit limitation on total frame size.
This results in many wasted bits for high bitrate encodings
but will ensure strict ISO compatibility. This compatibility
might be important for hardware players.
Filter options:
--lowpass freq
Set a lowpass filtering
frequency in kHz. Frequencies above the specified one will
be cutoff.
--lowpass-width
freq
Set the width of the lowpass
filter. The default value is 15% of the lowpass
frequency.
--highpass
freq
Set an highpass filtering
frequency in kHz. Frequencies below the specified one will
be cutoff.
--highpass-width
freq
Set the width of the highpass
filter in kHz. The default value is 15% of the highpass
frequency.
--resample
sfreq
sfreq = 8, 11.025, 12,
16, 22.05, 24, 32, 44.1, 48
Select output sampling frequency (only supported for
encoding).
If not specified, LAME will automatically resample the input
when using high compression ratios.
ID3 tag
options:
--tt title
audio/song title (max 30 chars
for version 1 tag)
--ta
artist
audio/song artist (max 30 chars
for version 1 tag)
--tl
album
audio/song album (max 30 chars
for version 1 tag)
--ty
year
audio/song year of issue (1 to
9999)
--tc
comment
user-defined text (max 30 chars
for v1 tag, 28 for v1.1)
--tn
track[/total]
audio/song track number and
(optionally) the total number of tracks on the original
recording. (track and total each 1 to 255. Providing just
the track number creates v1.1 tag, providing a total forces
v2.0).
--tg
genre
audio/song genre (name or
number in list)
--add-id3v2
force addition of version 2
tag
--id3v1-only
add only a version 1 tag
--id3v2-only
add only a version 2 tag
--id3v2-latin1
add following options in
ISO-8859-1 text encoding.
--id3v2-utf16
add following options in
unicode text encoding.
--space-id3v1
pad version 1 tag with spaces
instead of nulls
--pad-id3v2
same as
--pad-id3v2-size 128
--pad-id3v2-size
num
adds version 2 tag, pad with
extra "num" bytes
--genre-list
print alphabetically sorted ID3
genre list and exit
--ignore-tag-errors
ignore errors in values passed
for tags, use defaults in case an error occurs
Analysis
options:
-g
run graphical analysis on <infile>. <infile>
can also be a .mp3 file. (This feature is a compile time
option. Your binary may for speed reasons be compiled
without this.)
encoding modes
LAME is able to encode your music using one of its 3 encoding
modes: constant bitrate (CBR), average bitrate (ABR) and variable
bitrate (VBR).
Constant Bitrate (CBR)
This is the default encoding mode, and also the most basic. In
this mode, the bitrate will be the same for the whole file. It
means that each part of your mp3 file will be using the same
number of bits. The musical passage being a difficult one to
encode or an easy one, the encoder will use the same bitrate, so
the quality of your mp3 is variable. Complex parts will be of a
lower quality than the easiest ones. The main advantage is that
the final files size won’t change and can be accurately
predicted.
Average Bitrate (ABR)
In this mode, you choose the encoder will maintain an average
bitrate while using higher bitrates for the parts of your music
that need more bits. The result will be of higher quality than
CBR encoding but the average file size will remain predictable,
so this mode is highly recommended over CBR. This encoding mode
is similar to what is referred as vbr in AAC or Liquid Audio (2
other compression technologies).
Variable bitrate (VBR)
In this mode, you choose the desired quality on a scale from 9
(lowest quality/biggest distortion) to 0 (highest quality/lowest
distortion). Then encoder tries to maintain the given quality in
the whole file by choosing the optimal number of bits to spend
for each part of your music. The main advantage is that you are
able to specify the quality level that you want to reach, but the
inconvenient is that the final file size is totally
unpredictable.
id3 tags
LAME is able to embed ID3 v1, v1.1 or v2 tags inside the encoded
MP3 file. This allows to have some useful information about the
music track included inside the file. Those data can be read by
most MP3 players.
Lame will smartly choose which tags to use. It will add ID3 v2
tags only if the input comments won’t fit in v1 or v1.1 tags,
i.e. if they are more than 30 characters. In this case, both v1
and v2 tags will be added, to ensure reading of tags by MP3
players which are unable to read ID3 v2 tags.
presets
The --preset switches are aliases over LAME settings.
To activate these presets:
For VBR modes (generally highest quality):
--preset medium
This preset should provide near transparency to most people on
most music.
--preset standard
This preset should generally be transparent to most people on
most music and is already quite high in quality.
--preset extreme
If you have extremely good hearing and similar equipment, this
preset will generally provide slightly higher quality than the
standard mode.
For CBR 320kbps (highest quality possible from the
--preset switches):
--preset insane
This preset will usually be overkill for most people and most
situations, but if you must have the absolute highest quality
with no regard to filesize, this is the way to go.
For ABR modes (high quality per given bitrate but not as high as
VBR):
--preset kbps
Using this preset will usually give you good quality at a
specified bitrate. Depending on the bitrate entered, this preset
will determine the optimal settings for that particular
situation. While this approach works, it is not nearly as
flexible as VBR, and usually will not attain the same level of
quality as VBR at higher bitrates.
The following options are also available for the corresponding
profiles:
standard|extreme
cbr kbps
cbr
If you use the ABR mode (read above) with a significant bitrate
such as 80, 96, 112, 128, 160, 192, 224, 256, 320, you can use
the cbr option to force CBR mode encoding instead of the
standard ABR mode. ABR does provide higher quality but CBR may be
useful in situations such as when streaming an MP3 over the
Internet may be important.
bugs
Probably there
are some.
see also
mpg123,
madplay, sox
authors
LAME originally
developed by Mike Cheng and now maintained by
Mark Taylor, and the LAME team.
GPSYCHO
psycho-acoustic model by Mark Taylor.
(See http://www.mp3dev.org/).
mpglib by
Michael Hipp
Manual page by
William Schelter, Nils Faerber, Alexander Leidinger,
and Rogério Brito.