NAME
lame - create mp3 audio files
SYNOPSIS
lame [options] <infile> <outfile>
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 a "-" 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. Without -r, LAME will perform sev-
eral 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 prob-
lems. If your encodings sounds like static, try
this first.
-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.
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.
--mp1input
Assume the input file is a MPEG Layer I file.
If the filename ends in ".mp1" or ".mpg" LAME will
assume it is a MPEG Layer I file. For stdin or
Layer I files which do not end in .mp1 or .mpg you
need to use this switch.
October 13, 2001 1
--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.
Usefull for downsampling from one mp3 to another.
As an example, it can be usefull 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
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)tereo
In this mode, the encoder makes no use of poten-
tially 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 corre-
lation 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 sig-
nal does not have too much stereo separation, thus
giving a significant gain in encoding quality.
Using mid/side stereo inappropriately can result in
October 13, 2001 2
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 algo-
rithm than that described in the ISO documentation,
and thus is safe to use in joint stereo mode.
(f)orced joint stereo
This mode will force MS joint 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 sep-
aration.
(d)ual channels
In this mode, the 2 channels will be totally inden-
pendently 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 encod-
ing.
(mo)no
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.
-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 -I 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 sup-
port free bitrates at least up to 320 kbps, many
October 13, 2001 3
players are unable to deal with it.
Tests have shown that the following decoders sup-
port 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 I,II,III (MP3) and OGG files. In
case of MPEG files, LAME uses a bugfixed version of
mpglib for decoding.
If -t is used (disable wav header), LAME will out-
put raw pcm in native endian format. You can use
-x to swap bytes order.
-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 trun-
cate data which decodes to values greater than
32768.
October 13, 2001 4
--preset [fast] type | [cbr] kbps
Use one of the built-in presets.
Have a look at the PRESETS section below.
Warning: with the current version fast presets
might result in too high bitrate compared to regu-
lar presets.
--preset help gives more infos about the the used
options in these presets.
--alt-preset [fast] type | [cbr] kbps
Use one of the built-in presets.
This option is deprecated and offers the same as
the --preset option above. Do not use it anymore,
it will go away in a later version.
--r3mix
Uses r3mix VBR preset.
See http://www.r3mix.net/ for more details.
--noasm type
Disable specific assembly optimizations ( mmx /
3dnow / sse ). Quality will not increase, only
speed will be reduced. If you have problems run-
ning 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.
-S
--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.
October 13, 2001 5
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 algo-
rithms to determine the best scalefactors and huff-
man encoding (noise shaping).
-q 0:
use slowest & best possible version of all algo-
rithms. -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 acous-
tics 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 behaviour 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
behaviour is the same as the -q 7 switch.
Noise shaping will be disabled, but psycho acous-
tics will still be computed for bit allocation and
pre-echo detection.
CBR (constant bitrate, the default) options:
-b n For MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24
kHz)
October 13, 2001 6
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112,
128, 144, 160
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 actu-
ally 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-old)
--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 con-
sidered 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 consid-
ered 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).
-V n 0 <= n <= 9
Enable VBR (Variable BitRate) and specifies the
value of VBR quality (default = 4). 0 = highest
quality.
ABR and VBR options:
October 13, 2001 7
-b bitrate
For MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24
kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112,
128, 144, 160
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 MPEG1 (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 MPEG2 (sampling frequencies of 16, 22.05 and 24
kHz)
n = 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112,
128, 144, 160
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).
ATH related:
--noath
Disable any use of the ATH (absolute threshold of
hearing) for masking. Normally, humans are unable
to hear any sound below this threshold.
--athshort
Ignore psychoacoustic model for short blocks, use
ATH only.
October 13, 2001 8
--athonly
This option causes LAME to ignore the output of the
psy-model and only use masking from the ATH (abso-
lute threshold of hearing). Might be useful at
very high bitrates or for testing the ATH.
--athtype shape
The Absolute Threshold of Hearing is the minimum
threshold under which humans are unable to hear any
sound.
In the past, LAME was using ATH shape 0 which is
the Painter & Spanias formula. Tests have shown
that this formula is innacurate for the 13 - 22 kHz
area, leading to audible artifacts in some cases.
Shape 1 was thus implemented, which is over sensi-
tive, leading to very high bitrates.
Shape 2 formula was accurately modelized from real
data in order to reach optimal quality while not
wasting bitrate. In CBR and ABR modes, LAME uses
ATH shape 2 by default, VBR selects one depending
on the specified parameter to the -V option.
--athlower n
Lower the ATH (absolute threshold of hearing) by n
dB.
Normally, humans are unable to hear any sound below
this threshold, but for music recorded at very low
level this option might be usefull.
--athaa-type n
ATH auto adjust types 1 - 3, else no adjustment
--athaa-sensitivity x
activation offset in -/+ dB for ATH auto-adjustment
PSY related:
--short
Let LAME use short blocks when appropriate. It is
the default setting.
--noshort
Encode all frames using long blocks only. This
could increase quality when encoding at very low
bitrates, but can produce serious pre-echo arte-
facts.
--allshort
Use only short blocks, no long ones.
--cwlimit freq
Compute tonality up to freq (in kHz). Default set-
ting is 8.8717.
October 13, 2001 9
--notemp
Do not make use of the temporal masking effect.
--nspsytune
Experimental PSY tunings by Naoki Shibata
--nssafejoint
M/S switching criterion
--nsmsfix arg
M/S switching tuning [effective 0-3.5]
--ns-bass x
Adjust masking for sfbs 0 - 6 (long) 0 - 5
(short)
--ns-alto x
Adjust masking for sfbs 7 - 13 (long) 6 - 10
(short)
--ns-treble x
Adjust masking for sfbs 14 - 21 (long) 11 - 12
(short)
--ns-sfb21 x
Change ns-treble by x dB for sfb21
Experimantal options:
-X n 0 <= n <= 7
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 beeing the default mode:
-X0
The criterions 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
October 13, 2001 10
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 criterions are (in order of importance):
* the sum of noise over the thresholds is lower
* the total sum of noise is lower
-X6
The criterions 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 criterions 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
-Z toggles the scalefac feature on
MP3 header/stream options:
-e emp emp = n, 5, c
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.
October 13, 2001 11
-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:
-k Tells the encoder to use full bandwidth and to dis-
able all filters. By default, the encoder uses
some highpass filtering at low bitrates, in order
to keep a good quality by giving more bits to more
important frequencies.
Increasing the bandwidth from the default setting
might produce ringing artefacts at low bitrates.
Use with care!
--lowpass freq
Set a lowpass filtering frequency in kHz. Frequen-
cies 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. Fre-
quencies 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 ouptut sampling frequency (only supported
for encoding).
If not specified, LAME will automatically resample
October 13, 2001 12
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
audio/song track number (1 to 255, creates v1.1
tag)
--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
--space-id3v1
pad version 1 tag with spaces instead of nulls
--pad-id3v2
pad version 2 tag with extra 128 bytes
--genre-list
print alphabetically sorted ID3 genre list and exit
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.)
October 13, 2001 13
ID3 TAGS
LAME is able to embed ID3 v1, v1.1 or v2 tags inside the
encoded MP3 file. This allows to have some usefull infor-
mation about the music track included inside the file.
Those data can be read by most MP3 players.
Lame will smartly choose wich 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 wich are unable to read ID3
v2 tags.
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 beeing a difficult one
to encode or an easy one, the encoder will use the
same bitrate, so the quality of your mp3 is vari-
able. 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 accu-
rately 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 predictible,
so this mode is highly recommended over CBR. This
encoding mode is similar to what is reffered as vbr
in AAC or Liquid Audio (2 other compression tech-
nologies).
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 qual-
ity level that you want to reach, but the inconve-
nient is that the final file size is totally unpre-
dictible.
October 13, 2001 14
PRESETS
The --preset switches are designed to provide the highest
possible quality.
They have for the most part been subject to and tuned via
rigorous double blind listening tests to verify and
achieve this objective.
These are continually updated to coincide with the latest
developments that occur and as a result should provide you
with nearly the best quality currently possible from LAME.
To activate these prests:
For VBR modes (generally highest quality):
--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 --pre-
set switches):
--preset insane
This preset will usually be overkill for most peo-
ple and most situations, but if you must have the
absolute highest quality with no regard to file-
size, 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 qual-
ity 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 corre-
sponding profiles:
fast standard|extreme|insane
cbr kbps
October 13, 2001 15
fast Enables the new fast VBR for a particular profile.
The disadvantage to the speed switch is that often
times the bitrate will be slightly higher than with
the normal mode and quality may be slightly lower
also.
cbr If you use the ABR mode (read above) with a signif-
icant 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 use-
ful in situations such as when streaming an MP3
over the internet may be important.
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
October 13, 2001 16
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 fast standard preset:
lame --preset fast standard sample.wav sample.mp3
BUGS
Probably there are some.
SEE ALSO
mpg123 (1) , madplay (1) , sox (1)
AUTHORS
LAME originally developed by Mike Cheng and now maintained by
Mark Taylor. GPSYCHO psycho-acoustic model by Mark Taylor.
( http://www.mp3dev.org/).
mpglib by Michael Hipp
Manual page by William Schelter, Nils Faerber, Alexander Leidinger
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Latest update of this package can be found at http://amiga.sourceforge.net/
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LhA Freeware Version 2.2
Copyright © 1991-94 by Stefan Boberg.
Copyright © 1998-2000 by Jim Cooper and David Tritscher.
Listing of archive 'lame-3.93.1.lha':
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5406 1761 67.4% 24-Oct-01 14:32:24 +basic.html
3591 1801 49.8% 15-Oct-02 09:32:00 +contributors.html
1842 767 58.3% 24-Oct-01 14:32:24 +examples.html
74810 24058 67.8% 01-Dec-02 16:09:22 +history.html
7211 2398 66.7% 13-Oct-01 11:44:50 +id3.html
2200 906 58.8% 01-Dec-02 13:15:32 +index.html
708 275 61.1% 04-Dec-00 14:32:36 +lame.css
707 423 40.1% 19-Dec-00 13:07:10 +LICENSE
10425 9453 9.3% 17-Nov-02 10:04:36 +testcase.mp3
100044 94371 5.6% 29-Nov-00 15:44:48 +testcase.wav
5351 2685 49.8% 06-Apr-02 07:22:58 +TODO
25632 8353 67.4% 22-Jan-02 12:20:40 +USAGE
2303 1093 52.5% 13-Oct-01 11:44:50 +modes.html
6835 2674 60.8% 24-Oct-01 14:32:24 +node6.html
3183 1423 55.2% 17-Nov-02 09:06:56 +presets.html
43403 10307 76.2% 03-Sep-02 11:08:24 +switchs.html
445912 202335 54.6% 24-Mar-03 05:22:10 +LAME
25905 9663 62.6% 17-Nov-02 09:20:42 +lame.1
-------- ------- ----- --------- --------
1233111 518519 57.9% Operation successful.
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.Readme created with: MRea \
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>»>»>»>»> Some additional info about this archive:
Source: http://prdownloads.sf.net/amiga/lame-3.93.1.lha?download
FileSize: 519552 Bytes
CRC: 8906A867
MD5: 1826BE2FE2E1A4700BF3601241447CF5
SHA: A2AA7C5DED0890A939868D6B0B021B051B820346
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