Appendix > Appendix A - Lame encoder command line

Lame encoder, used during the development of this product, is available for download from the following link.

Lame encoder latest build, with full source code, is available for download from its owner's website

Below you can find the full list of command line's options for Lame version 3.96.1:

LAME version 3.96.1 (http://lame.sourceforge.net/)

usage: lame [options] <infile> [outfile]

<infile> and/or <outfile> can be "-", which means stdin/stdout.

RECOMMENDED:

lame -h input.wav output.mp3

OPTIONS:

Input options:

-r input is raw pcm

-x force byte-swapping of input

-s sfreq sampling frequency of input file (kHz) - default 44.1 kHz

--bitwidth w input bit width is w (default 16)

--mp1input input file is a MPEG Layer I file

--mp2input input file is a MPEG Layer II file

--mp3input input file is a MPEG Layer III file

--nogap <file1> <file2> <...>

gapless encoding for a set of contiguous files

--nogapout <dir>

output dir for gapless encoding (must precede --nogap)

--nogaptags allow the use of VBR tags in gapless encoding

Operational options:

-m <mode> (s)tereo, (j)oint, (f)orce, (m)ono

default is (s) or (j) depending on bitrate

force = force ms_stereo on all frames.

-a downmix from stereo to mono file for mono encoding

--freeformat produce a free format bitstream

--decode input=mp3 file, output=wav

-t disable writing wav header when using --decode

--comp <arg> choose bitrate to achive a compression ratio of <arg>

--scale <arg> scale input (multiply PCM data) by <arg>

--scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>

--scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>

--replaygain-fast compute RG fast but slightly inaccurately (default)

--replaygain-accurate compute RG more accurately and find the peak sample

--noreplaygain disable ReplayGain analysis

--clipdetect enable --replaygain-accurate and print a message whether

clipping occurs and how far the waveform is from full scale

--preset type type must be "medium", "standard", "extreme", "insane",

or a value for an average desired bitrate and depending

on the value specified, appropriate quality settings will

be used.

"--preset help" gives more info on these

Verbosity:

--disptime <arg>print progress report every arg seconds

-S don't print progress report, VBR histograms

--nohist disable VBR histogram display

--silent don't print anything on screen

--quiet don't print anything on screen

--brief print more useful information

--verbose print a lot of useful information

Noise shaping & psycho acoustic algorithms:

-q <arg> <arg> = 0...9. Default -q 5

-q 0: Highest quality, very slow

-q 9: Poor quality, but fast

-h Same as -q 2. Recommended.

-f Same as -q 7. Fast, ok quality

CBR (constant bitrate, the default) options:

-b <bitrate> set the bitrate in kbps, default 128 kbps

--cbr enforce use of constant bitrate

ABR options:

--abr <bitrate> specify average bitrate desired (instead of quality)

VBR options:

-v use variable bitrate (VBR) (--vbr-old)

--vbr-old use old variable bitrate (VBR) routine

--vbr-new use new variable bitrate (VBR) routine

-V n quality setting for VBR. default n=4

0=high quality,bigger files. 9=smaller files

-b <bitrate> specify minimum allowed bitrate, default 32 kbps

-B <bitrate> specify maximum allowed bitrate, default 320 kbps

-F strictly enforce the -b option, for use with players that

do not support low bitrate mp3

-t disable writing LAME Tag

-T enable and force writing LAME Tag

ATH related:

--noath turns ATH down to a flat noise floor

--athshort ignore GPSYCHO for short blocks, use ATH only

--athonly ignore GPSYCHO completely, use ATH only

--athtype n selects between different ATH types [0-4]

--athlower x lowers ATH by x dB

--athaa-type n ATH auto adjust types 1-3, else no adjustment

--athaa-loudapprox n n=1 total energy or n=2 equal loudness curve

--athaa-sensitivity x activation offset in -/+ dB for ATH auto-adjustment

PSY related:

--short use short blocks when appropriate

--noshort do not use short blocks

--allshort use only short blocks

--cwlimit <freq> compute tonality up to freq (in kHz) default 8.8717

--notemp disable temporal masking effect

--nssafejoint M/S switching criterion

--nsmsfix <arg> M/S switching tuning [effective 0-3.5]

--interch x adjust inter-channel masking ratio

--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

--shortthreshold x,y short block switching threshold, x for L/R/M channel, y for S channel

Noise Shaping related:

--substep n use pseudo substep noise shaping method types 0-2

experimental switches:

-X n[,m] selects between different noise measurements

n for long block, m for short. if m is omitted, m = n

-Y lets LAME ignore noise in sfb21, like in CBR

-Z [n] toggles the scalefac-scale and subblock gain feature on

if n is set and minus, only scalefac-scale is enabled

MP3 header/stream options:

-e <emp> de-emphasis n/5/c (obsolete)

-c mark as copyright

-o mark as non-original

-p error protection. adds 16 bit checksum to every frame

(the checksum is computed correctly)

--nores disable the bit reservoir

--strictly-enforce-ISO comply as much as possible to ISO MPEG spec

Filter options:

-k keep ALL frequencies (disables all filters),

Can cause ringing and twinkling

--lowpass <freq> frequency(kHz), lowpass filter cutoff above freq

--lowpass-width <freq> frequency(kHz) - default 15% of lowpass freq

--highpass <freq> frequency(kHz), highpass filter cutoff below freq

--highpass-width <freq> frequency(kHz) - default 15% of highpass freq

--resample <sfreq> sampling frequency of output file(kHz)- default=automatic

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

--ignore-tag-errors ignore errors in values passed for tags

Note: A version 2 tag will NOT be added unless one of the input fields

won't fit in a version 1 tag (e.g. the title string is longer than 30

characters), or the '--add-id3v2' or '--id3v2-only' options are used,

or output is redirected to stdout.

MS-Windows-specific options:

--priority <type> sets the process priority:

0,1 = Low priority (IDLE_PRIORITY_CLASS)

2 = normal priority (NORMAL_PRIORITY_CLASS, default)

3,4 = High priority (HIGH_PRIORITY_CLASS))

Note: Calling '--priority' without a parameter will select priority 0.

MPEG-1 layer III sample frequencies (kHz): 32 48 44.1

bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320

MPEG-2 layer III sample frequencies (kHz): 16 24 22.05

bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160

MPEG-2.5 layer III sample frequencies (kHz): 8 12 11.025

bitrates (kbps): 8 16 24 32 40 48 56 64 80 96 112 128 144 160