Full command line switch reference
note: Options which could exist without beeing documented
here are considered as experimental ones. Such experimental options should usually
not be used.
switch
parameter
#a
-a
downmix stereo file to mono
#-abr
--abr
average bitrate encoding
#-allshort
--allshort
use short blocks only
#-athlower
--athlower
lower the ATH
#-athonly
--athonly
ATH only
#-athshort
--athshort
ATH only for short blocks
#-athtype
--athtype
select ATH type
#b
-b
bitrate (8...320)
#Bmax
-B
max VBR/ABR bitrate (8...320)
#-bitwidth
--bitwidth
input bit width
#c
-c
copyright
#-cbr
--cbr
enforce use of constant bitrate
#-clipdetect
--clipdetect
clipping detection
#-comp
--comp
choose compression ratio
#-cwlimit
--cwlimit
tonality limit
#d
-d
block type control
#-decode
--decode
decoding only
#-disptime
--disptime
time between display updates
#e
-e
de-emphasis (
n
, 5, c)
#f
-f
fast mode
#FF
-F
strictly enforce the -b option
#-freeformat
--freeformat
free format bitstream
#h
-h
high quality
#-help
--help
help
#-highpass
--highpass
highpass filtering frequency in kHz
#-highpass
--highpass-width
width of highpass filtering in kHz
#k
-k
full bandwidth
#-lowpass
--lowpass
lowpass filtering frequency in kHz
#-lowpass-width
--lowpass-width
width of lowpass filtering in kHz
#m
-m
stereo mode (s,
j
, f, m)
#-mp1input
--mp1input
MPEG Layer I input file
#-mp2input
--mp2input
MPEG Layer II input file
#-mp3input
--mp3input
MPEG Layer III input file
#-noath
--noath
disable ATH
#-noasm
--noasm
disable assembly optimizations (mmx/3dnow/sse)
#-nohist
--nohist
disable histogram display
#-noreplaygain
--noreplaygain
disable ReplayGain analysis
#-nores
--nores
disable bit reservoir
#-noshort
--noshort
disable short blocks frames
#-notemp
--notemp
disable temporal masking
#o
-o
non-original
#p
-p
error protection
#-preset
--preset
use built-in preset
#-priority
--priority
OS/2 process priority control
#q
-q
algorithm quality selection
#-silent
--quiet
silent operation
#r
-r
input file is raw pcm
#-replaygain-accurate
--replaygain-accurate
compute ReplayGain more accurately and find the peak sample
#-replaygain-fast
--replaygain-fast
compute ReplayGain fast but slightly inaccurately (default)
#-resample
--resample
output sampling frequency in kHz (encoding only)
#s
-s
sampling frequency in kHz
#-silent
-S
silent operation
#-scale
--scale
scale input
#-scale-l
--scale-l
scale input channel 0 (left)
#-scale-r
--scale-r
scale input channel 1 (right)
#-short
--short
use short blocks
#-silent
--silent
silent operation
#-strictly-enforce-ISO
--strictly-enforce-ISO
strict ISO compliance
#t
-t
disable INFO/WAV header
#V
-V
VBR quality setting (0...9)
#-vbr-new
--vbr-new
new VBR mode
#-vbr-old
--vbr-old
older VBR mode
#-verbose
--verbose
verbosity
#x
-x
swapbytes
#Xquant
-X
change quality measure
*
-a
downmix
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 m" will always produce a mono .mp3
file from both mono and stereo input.
*
--abr n
average
bitrate encoding
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.
*
--allshort
use
short blocks only
Use only short blocks, no long ones.
*
--athlower n
lower
the ATH
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.
*
--athonly
ATH
only
This option causes LAME to ignore the output of the psy-model and only use
masking from the ATH (absolute threshold of hearing). Might be useful at very
high bitrates or for testing the ATH.
*
--athshort
ATH
only for short blocks
Ignore psychoacoustic model for short blocks, use ATH only.
*
--athtype 0/1/2
select
ATH type
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 sensitive, leading to very high bitrates.
Shape 2 formula was accurately modelized from real data in order to real optimal
quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape
2 by default.
In VBR mode, LAME is adapting its shape according to the
-V value, going gradually from the 0 shape at -V9 up to shape 2 at -V0.
*
-b n
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
Default is 128 kbs for MPEG1 and 64 kbs for MPEG2.
When used with variable bitrate encoding (VBR), -b 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 n
maximum
VBR/ABR 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 when using VBR/ABR
The use of -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the bit reservoir,
can actually have frames which use as many bits as a 320kbs frame. VBR modes
minimize the use of the bit reservoir, and thus need to allow 320kbs frames
to get the same flexibility as CBR streams.
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.
*
--bitwidth 8/16/24/32
input
bit width
Required only for raw PCM input files. Otherwise it will be determined
from the header of the input file.
*
--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
--replaygain-accurate
*
--cbr
enforce use of constant bitrate
This switch enforces the use of constant bitrate encoding.
*
--cbr
enforce use of constant bitrate
This switch enforces the use of constant bitrate encoding.
*
--comp
choose
compression ratio
Instead of choosing bitrate, using this option, user can choose compression
ratio to achieve.
*
--cwlimit n
tonality
limit
Compute tonality up to freq (in kHz). Default setting is 8.8717.
*
-d
block type control
Allows the left and right channels to use different block size types.
*
--decode
decoding
only
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 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.
*
--disptime n
time
between display updates
Set the delay in seconds between two display updates.
*
-e n/5/c
de-emphasis
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.
*
-f
fast mode
This switch forces the encoder to use a faster encoding mode, but with
a lower quality. The behaviour is the same as the -q7 switch.
Noise shaping will be disabled, but psycho acoustics will still be computed
for bit allocation and pre-echo detection.
*
-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, ie when the music level is below the absolute threshold of human
hearing (ATH).
*
--freeformat
free
format bitstream
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
*
-h
high quality
Use some quality improvements. Encoding will be slower, but the result
will be of higher quality. The behaviour is the same as the -q2 switch.
This switch is always enabled when using VBR.
*
--help
help
Display a list of all available options.
*
--highpass
highpass
filtering frequency in kHz
Set an highpass filtering frequency. Frequencies below the specified one
will be cutoff.
*
--highpass-width
width
of highpass filtering in kHz
Set the width of the highpass filter. The default value is 15% of the highpass
frequency.
*
-k
full bandwidth
Tells the encoder to use full bandwidth and to disable 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
lowpass
filtering frequency in kHz
Set a lowpass filtering frequency. Frequencies above the specified one
will be cutoff.
*
--lowpass-width
width
of lowpass filtering in kHz
Set the width of the lowpass filter. The default value is 15% of the lowpass
frequency.
*
-m s/
j/
f/d/m
stereo
mode
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.
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.
joint 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.
forced joint stereo
This mode will force MS joint stereo on all frames. It's 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.
dual channels
In this mode, the 2 channels will be totally indenpendently encoded. Each
channel will have exactly half of the bitrate. This mode is designed for applications
like dual languages encoding (ex: English in one channel and French in the
other). Using this encoding mode for regular stereo files will result in a
lower quality encoding.
mono
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.
*
--mp1input
MPEG
Layer I input file
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.
*
--mp2input
MPEG
Layer II input file
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
MPEG
Layer III input file
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 file. For stdin or
MP3 files which do not end in .mp3 you need to use this switch.
*
--noath
disable
ATH
Disable any use of the ATH (absolute threshold of hearing) for masking.
Normally, humans are unable to hear any sound below this threshold.
*
--noasm mmx/3dnow/sse
disable assembly optimisations
Disable specific assembly optimizations. 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.
*
--nohist
disable
histogram display
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.
*
--noreplaygain
disable
ReplayGain analysis
By default ReplayGain analysis is enabled. This switch disables it.
See also:
#-replaygain-accurate
--replaygain-accurate
,
#-replaygain-fast
--replaygain-fast
*
--nores
disable
bit reservoir
Disable the bit reservoir. Each frame will then become independent from
previous ones, but the quality will be lower.
*
--noshort
disable
short blocks frames
Encode all frames using long blocks only. This could increase quality when
encoding at very low bitrates, but can produce serious pre-echo artefacts.
*
--notemp
disable
temporal masking
Don't make use of the temporal masking effect.
*
-o
non-original
Mark the encoded file as being a copy.
*
-p
error protection
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.
*
--preset presetName
use
built-in preset
Use one of the built-in presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
"--preset help" gives more information about the usage possibilities for these presets.
*
--priority 0...4
OS/2
process priority control
With this option, LAME will run with a different process priority under
IBM OS/2.
This will greatly improve system responsiveness, since OS/2 will have more
free time to properly update the screen and poll the keyboard/mouse. It should
make quite a difference overall, especially on slower machines. LAME's performance
impact should be minimal.
0 (Low priority)
Priority 0 assumes "IDLE" class, with delta 0.
LAME will have the lowest priority possible, and the encoding may be suspended
very frequently by user interaction.
1 (Medium priority)
Priority 1 assumes "IDLE" class, with delta +31.
LAME won't interfere at all with what you're doing.
Recommended if you have a slower machine.
2 (Regular priority)
Priority 2 assumes "REGULAR" class, with delta -31.
LAME won't interfere with your activity. It'll run just like a regular process,
but will spare just a bit of idle time for the system. Recommended for most
users.
3 (High priority)
Priority 3 assumes "REGULAR" class, with delta 0.
LAME will run with a priority a bit higher than a normal process.
Good if you're just running LAME by itself or with moderate user interaction.
4 (Maximum priority)
Priority 4 assumes "REGULAR" class, with delta +31.
LAME will run with a very high priority, and may interfere with the machine
response.
Recommended if you only intend to run LAME by itself, or if you have a fast
processor.
Priority 1 or 2 is recommended for most users.
*
-q 0..9
algorithm
quality selection
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.
*
-r
input file is
raw pcm
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 several
fseek()'s on the input file looking for WAV and AIFF headers.
Might not be available on your release.
*
--replaygain-accurate
compute
ReplayGain more accurately and find the peak sample
Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded
data stream. Find the peak sample of the decoded data stream and store
it in the file.
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/
http://www.replaygain.org/
By default, LAME performs ReplayGain analysis on the input data
(after the user-specified volume scaling). This
behaviour 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 no 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 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 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
--replaygain-fast
,
#-noreplaygain
--noreplaygain
,
#-clipdetect
--clipdetect
*
--replaygain-fast
compute
ReplayGain fast but slightly inaccurately (default)
Compute "Radio" ReplayGain on the input data stream after user-specified
volume scaling and/or resampling.
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/
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
--replaygain-accurate
,
#-noreplaygain
--noreplaygain
*
--resample 8/11.025/12/16/22.05/24/32/44.1/48
output
sampling frequency in kHz
Select ouptut sampling frequency (for encoding only).
If not specified, LAME will automatically resample the input when using high
compression ratios.
*
-s 8/11.025/12/16/22.05/24/32/44.1/48
sampling
frequency
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.
*
-S / --silent / --quiet
silent
operation
Don't print progress report.
*
--scale n
scales
input by n
*
--scale-l n
scales
input channel 0 (left) by n
*
--scale-r n
scales
input channel 1 (right) by n
Scales input 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
Use with care, since most MP3 decoders will truncate data which decodes to
values greater than 32768.
*
--short
use
short blocks
Let LAME use short blocks when appropriate. It is the default setting.
*
--strictly-enforce-ISO
strict
ISO compliance
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.
*
-t
disable INFO/WAV
header
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.
*
-V 0...9
VBR quality
setting
Enable VBR (Variable BitRate) and specifies the value of VBR quality.
default=4
0=highest quality.
*
--vbr-new
new
VBR mode
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).
*
--vbr-old
older
VBR mode
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.
*
--verbose
verbosity
Print a lot of information on screen.
*
-x
swapbytes
Swap bytes in the input file or ouptut file when using --decode.
For sorting out little endian/big endian type problems. If your encodings
sounds like static, try this first.
*
-X 0...7
change
quality measure
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
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
