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<H1>Full command line switch reference</H1>
<P> <font size="-1">note: Options which could exist without being documented
here are considered as experimental ones. Such experimental options should usually
not be used.</font>
<P>
<TABLE CELLPADDING=3 BORDER="1">
<TR VALIGN="TOP">
<TD ALIGN="LEFT" nowrap><b>switch</b></TD>
<TD ALIGN="LEFT" nowrap><b>parameter</b></TD>
</TR>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#a">-a</a></kbd></td>
<td align="LEFT" nowrap>downmix stereo file to mono</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-abr">--abr</a></kbd></td>
<td align="LEFT" nowrap>average bitrate encoding</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-allshort">--allshort</a></kbd></td>
<td align="LEFT" nowrap>use short blocks only</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-athlower">--athlower</a></kbd></td>
<td align="LEFT" nowrap>lower the ATH</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-athonly">--athonly</a></kbd></td>
<td align="LEFT" nowrap>ATH only</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-athshort">--athshort</a></kbd></td>
<td align="LEFT" nowrap>ATH only for short blocks</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-athtype">--athtype</a></kbd></td>
<td align="LEFT" nowrap>select ATH type</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#b">-b</a></kbd></td>
<td align="LEFT" nowrap>bitrate (8...320)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#Bmax">-B</a></kbd></td>
<td align="LEFT" nowrap>max VBR/ABR bitrate (8...320)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-bitwidth">--bitwidth</a></kbd></td>
<td align="LEFT" nowrap>input bit width</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#c">-c</a></kbd></td>
<td align="LEFT" nowrap>copyright</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-cbr">--cbr</a></kbd></td>
<td align="LEFT" nowrap>enforce use of constant bitrate</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-clipdetect">--clipdetect</a></kbd></td>
<td align="LEFT" nowrap>clipping detection</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-comp">--comp</a></kbd></td>
<td align="LEFT" nowrap>choose compression ratio</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-cwlimit">--cwlimit</a></kbd></td>
<td align="LEFT" nowrap>tonality limit</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#d">-d</a></kbd></td>
<td align="LEFT" nowrap>block type control</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-decode">--decode</a></kbd></td>
<td align="LEFT" nowrap>decoding only</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-disptime">--disptime</a></kbd></td>
<td align="LEFT" nowrap>time between display updates</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#e">-e</a></kbd></td>
<td align="LEFT" nowrap>de-emphasis (<b>n</b>, 5, c)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#f">-f</a></kbd></td>
<td align="LEFT" nowrap> fast mode</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#FF">-F</a></kbd></td>
<td align="LEFT" nowrap> strictly enforce the -b option</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-freeformat">--freeformat</a></kbd></td>
<td align="LEFT" nowrap> free format bitstream</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#h">-h</a></kbd></td>
<td align="LEFT" nowrap>high quality</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-help">--help</a></kbd></td>
<td align="LEFT" nowrap> help</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass</a></kbd></td>
<td align="LEFT" nowrap> highpass filtering frequency in kHz</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass-width</a></kbd></td>
<td align="LEFT" nowrap> width of highpass filtering in kHz</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#k">-k</a></kbd></td>
<td align="LEFT" nowrap> full bandwidth</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-lowpass">--lowpass</a></kbd></td>
<td align="LEFT" nowrap> lowpass filtering frequency in kHz</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-lowpass-width">--lowpass-width</a></kbd></td>
<td align="LEFT" nowrap> width of lowpass filtering in kHz</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#m">-m</a></kbd></td>
<td align="LEFT" nowrap>stereo mode (s, <b>j</b>, f, m)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-mp1input">--mp1input</a></kbd></td>
<td align="LEFT" nowrap>MPEG Layer I input file</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-mp2input">--mp2input</a></kbd></td>
<td align="LEFT" nowrap>MPEG Layer II input file</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-mp3input">--mp3input</a></kbd></td>
<td align="LEFT" nowrap>MPEG Layer III input file</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-noath">--noath</a></kbd></td>
<td align="LEFT" nowrap>disable ATH</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-noasm">--noasm</a></kbd></td>
<td align="LEFT" nowrap>disable assembly optimizations (mmx/3dnow/sse)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-nohist">--nohist</a></kbd></td>
<td align="LEFT" nowrap>disable histogram display</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-noreplaygain">--noreplaygain</a></kbd></td>
<td align="LEFT" nowrap>disable ReplayGain analysis</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-nores">--nores</a></kbd></td>
<td align="LEFT" nowrap>disable bit reservoir</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-noshort">--noshort</a></kbd></td>
<td align="LEFT" nowrap>disable short blocks frames</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-notemp">--notemp</a></kbd></td>
<td align="LEFT" nowrap>disable temporal masking</td>
</tr>
<TR VALIGN="TOP">
<TD ALIGN="LEFT" nowrap><kbd><a href="#o">-o</a></kbd></TD>
<TD ALIGN="LEFT" nowrap>non-original</TD>
</TR>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#p">-p</a></kbd></td>
<td align="LEFT" nowrap>error protection</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-preset">--preset</a></kbd></td>
<td align="LEFT" nowrap>use built-in preset</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-priority">--priority</a></kbd></td>
<td align="LEFT" nowrap>OS/2 process priority control</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#q">-q</a></kbd></td>
<td align="LEFT" nowrap>algorithm quality selection</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-silent">--quiet</a></kbd></td>
<td align="LEFT" nowrap>silent operation</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#r">-r</a></kbd></td>
<td align="LEFT" nowrap>input file is raw PCM</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-replaygain-accurate">--replaygain-accurate</a></kbd></td>
<td align="LEFT" nowrap>compute ReplayGain more accurately and find the peak sample</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-replaygain-fast">--replaygain-fast</a></kbd></td>
<td align="LEFT" nowrap>compute ReplayGain fast but slightly inaccurately (default)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-resample">--resample</a></kbd></td>
<td align="LEFT" nowrap>output sampling frequency in kHz (encoding only)</td>
</tr>
<TR VALIGN="TOP">
<TD ALIGN="LEFT" nowrap><kbd><a href="#s">-s</a></kbd></TD>
<TD ALIGN="LEFT" nowrap>sampling frequency in kHz</TD>
</TR>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-silent">-S</a></kbd></td>
<td align="LEFT" nowrap>silent operation</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-scale">--scale</a></kbd></td>
<td align="LEFT" nowrap>scale input</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-scale-l">--scale-l</a></kbd></td>
<td align="LEFT" nowrap>scale input channel 0 (left)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-scale-r">--scale-r</a></kbd></td>
<td align="LEFT" nowrap>scale input channel 1 (right)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-short">--short</a></kbd></td>
<td align="LEFT" nowrap>use short blocks</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-silent">--silent</a></kbd></td>
<td align="LEFT" nowrap>silent operation</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-strictly-enforce-ISO">--strictly-enforce-ISO</a></kbd></td>
<td align="LEFT" nowrap>strict ISO compliance</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#t">-t</a></kbd></td>
<td align="LEFT" nowrap>disable INFO/WAV header</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#V">-V</a></kbd></td>
<td align="LEFT" nowrap>VBR quality setting (0...9)</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-vbr-new">--vbr-new</a></kbd></td>
<td align="LEFT" nowrap>new VBR mode</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-vbr-old">--vbr-old</a></kbd></td>
<td align="LEFT" nowrap>older VBR mode</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#-verbose">--verbose</a></kbd></td>
<td align="LEFT" nowrap>verbosity</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#x">-x</a></kbd></td>
<td align="LEFT" nowrap>swapbytes</td>
</tr>
<tr valign="TOP">
<td align="LEFT" nowrap><kbd><a href="#Xquant">-X</a></kbd></td>
<td align="LEFT" nowrap>change quality measure</td>
</tr>
</TABLE>
<BR>
<dl>
<dt><strong>* <kbd>-a</kbd><a name="a"> downmix </a></strong>
<dd>Mix the stereo input file to mono and encode as mono.<br>
The downmix is calculated as the sum of the left and right channel, attenuated
by 6 dB. <br>
<br>
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).<br>
To encode a stereo PCM input file as mono, use "lame -m s -a".<br>
<br>
For WAV and AIFF input files, using "-m m" will always produce a mono .mp3
file from both mono and stereo input.
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--abr n</kbd><a name="-abr"> average
bitrate encoding</a></strong> </dt>
</dl>
<dl>
<dd>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.<br>
<br>
It can be combined with the -b and -B switches like:<br>
lame --abr 123 -b 64 -B 192 a.wav a.mp3<br>
which would limit the allowed frame sizes between 64 and 192 kbits. <br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--allshort</kbd><a name="-allshort"> use
short blocks only</a></strong> </dt>
</dl>
<dl>
<dd>Use only short blocks, no long ones.
</dl>
<dl>
<dd>
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--athlower n</kbd><a name="-athlower"> lower
the ATH</a></strong> </dt>
</dl>
<dl>
<dd>Lower the ATH (absolute threshold of hearing) by n dB.<br>
Normally, humans are unable to hear any sound below this threshold, but for
music recorded at very low level this option might be useful.
</dl>
<dl>
<dd>
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--athonly</kbd><a name="-athonly"> ATH
only</a></strong> </dt>
</dl>
<dl>
<dd>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.
</dl>
<dl>
<dd>
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--athshort</kbd><a name="-athshort"> ATH
only for short blocks</a></strong> </dt>
</dl>
<dl>
<dd>Ignore psychoacoustic model for short blocks, use ATH only.
</dl>
<dl>
<dd>
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>--athtype 0/1/2</kbd><a name="-athtype"> select
ATH type</a></strong> </dt>
</dl>
<dl>
<dd>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. <br>
<br>
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.
</dl>
<dl>
<dd>
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>-b n</kbd><a name="b"> bitrate</a></strong>
</dt>
</dl>
<dl>
<dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
<br>
For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
<br>
Default is 128 kbps for MPEG1 and 64 kbps for MPEG2. <br>
<br>
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.
<dt><br>
</dt>
<hr width="50%" noshade align="center">
<br>
</dl>
<dl>
<dt><strong>* <kbd>-B n</kbd><a name="Bmax"> maximum
VBR/ABR bitrate </a></strong> </dt>
</dl>
<dl>
<dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
<br>
For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
<br>
Specifies the maximum allowed bitrate when using VBR/ABR <br>
<br>
The use of -B is NOT RECOMMENDED. A 128kbps CBR bitstream, because of the bit reservoir,
can actually have frames which use as many bits as a 320kbps frame. VBR modes
minimize the use of the bit reservoir, and thus need to allow 320kbps frames
to get the same flexibility as CBR streams.<br>
<br>
<i>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.</i> <br>
</dl>
<dl>
<dt><strong>* <kbd>--bitwidth 8/16/24/32</kbd><a name="-bitwidth"> input
bit width </a></strong> </dt>
</dl>
<dl>
<dd> Required only for raw PCM input files. Otherwise it will be determined
from the header of the input file. <br>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--clipdetect</kbd><a name="-clipdetect"> clipping detection</a></strong>
</dt>
</dl>
<dl>
<dd>
Enable --replaygain-accurate and print a message whether clipping
occurs and how far in dB the waveform is from full scale.<br>
<br>
This option is not usable if the MP3 decoder was <b>explicitly</b>
disabled in the build of LAME.<br>
<br>
See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
enforce use of constant bitrate</a></strong>
</dt>
</dl>
<dl>
<dd>This switch enforces the use of constant bitrate encoding.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
enforce use of constant bitrate</a></strong>
</dt>
</dl>
<dl>
<dd>This switch enforces the use of constant bitrate encoding.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>--comp</kbd><a name="-comp"> choose
compression ratio</a></strong> </dt>
</dl>
<dl>
<dd>Instead of choosing bitrate, using this option, user can choose compression
ratio to achieve.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>--cwlimit n</kbd><a name="-cwlimit"> tonality
limit</a></strong> </dt>
</dl>
<dl>
<dd>Compute tonality up to freq (in kHz). Default setting is 8.8717.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>-d</kbd><a name="d"> block type control</a></strong>
</dt>
</dl>
<dl>
<dd>Allows the left and right channels to use different block size types.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dt><strong>* <kbd>--decode</kbd><a name="-decode"> decoding
only</a></strong> </dt>
</dl>
<dl>
<dd>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.<br>
<br>
If -t is used (disable WAV header), Lame will output raw PCM in native endian
format. You can use -x to swap bytes order. <br>
<br>
This option is not usable if the MP3 decoder was <b>explicitly</b>
disabled in the build of LAME.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--disptime n</kbd><a name="-disptime"> time
between display updates</a></strong> </dt>
</dl>
<dl>
<dd>Set the delay in seconds between two display updates.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-e n/5/c</kbd><a name="e"> de-emphasis</a></strong>
</dt>
</dl>
<dl>
<dd> <br>
n = (none, default)<br>
5 = 0/15 microseconds<br>
c = citt j.17<br>
<br>
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.<br>
<br>
A better solution would be to apply the de-emphasis with a standalone utility
before encoding, and then encode without -e.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-f</kbd><a name="f"> fast mode</a></strong>
</dt>
</dl>
<dl>
<dd> 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.<br>
<br>
Noise shaping will be disabled, but psycho acoustics will still be computed
for bit allocation and pre-echo detection.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-F</kbd><a name="FF"> strictly enforce the
-b option</a></strong> </dt>
</dl>
<dl>
<dd> This is mainly for use with hardware players that do not support low bitrate
mp3.<br>
<br>
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).
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--freeformat</kbd><a name="-freeformat"> free
format bitstream</a></strong> </dt>
</dl>
<dl>
<dd> Produces a free format bitstream. With this option, you can use -b with
any bitrate higher than 8 kbps.<br>
<br>
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.<br>
<br>
Tests have shown that the following decoders support free format:<br>
<br>
FreeAmp up to 440 kbps<br>
in_mpg123 up to 560 kbps<br>
l3dec up to 310 kbps<br>
LAME up to 560 kbps<br>
MAD up to 640 kbps<br>
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-h</kbd><a name="h"> high quality</a></strong>
</dt>
</dl>
<dl>
<dd> 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.<br>
This switch is always enabled when using VBR.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--help</kbd><a name="-help"> help</a></strong>
</dt>
</dl>
<dl>
<dd> Display a list of all available options.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--highpass</kbd><a name="-highpass"> highpass
filtering frequency in kHz</a></strong> </dt>
</dl>
<dl>
<dd> Set an highpass filtering frequency. Frequencies below the specified one
will be cutoff.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--highpass-width</kbd><a name="-highpass-width"> width
of highpass filtering in kHz</a></strong> </dt>
</dl>
<dl>
<dd> Set the width of the highpass filter. The default value is 15% of the highpass
frequency.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-k</kbd><a name="k"> full bandwidth</a></strong>
</dt>
</dl>
<dl>
<dd> Tells the encoder to use full bandwidth and to disable all filters. By
default, the encoder uses some lowpass filtering at lower bitrates, in order
to keep a good quality by giving more bits to more important frequencies.<br>
Increasing the bandwidth from the default setting might produce ringing artefacts
at low bitrates. Use with care!
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--lowpass</kbd><a name="-lowpass"> lowpass
filtering frequency in kHz</a></strong></dt>
</dl>
<dl>
<dd> Set a lowpass filtering frequency. Frequencies above the specified one
will be cutoff.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--lowpass-width</kbd><a name="-lowpass-width"> width
of lowpass filtering in kHz</a></strong></dt>
</dl>
<dl>
<dd> Set the width of the lowpass filter. The default value is 15% of the lowpass
frequency.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-m s/<b>j/</b>f/d/m</kbd><a name="m"> stereo
mode</a></strong> </dt>
</dl>
<dl>
<dd> Joint-stereo is the default mode for input files featuring two channels..
<b><i><br>
<br>
stereo</i></b> <br>
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.<br>
<br>
<i><b>joint stereo</b></i><br>
In this mode, the encoder will make use of 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.<br>
This will effectively increase the bandwidth if the signal does not have too
much stereo separation, thus giving a significant gain in encoding quality.
In joint stereo, the encoder can select between Left/Right and Mid/Side representation
on a frame basis.<br>
<br>
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.<br>
<br>
<b><i>forced joint stereo </i></b><br>
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.<br>
<br>
<b><i>dual channels </i></b><br>
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 (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.<br>
<br>
<b><i>mono</i></b><br>
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.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--mp1input</kbd><a name="-mp1input"> MPEG
Layer I input file</a></strong> </dt>
</dl>
<dl>
<dd> Assume the input file is a MPEG Layer I file.<br>
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.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--mp2input</kbd><a name="-mp2input"> MPEG
Layer II input file</a></strong> </dt>
</dl>
<dl>
<dd> Assume the input file is a MPEG Layer II (ie MP2) file.<br>
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.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--mp3input</kbd><a name="-mp3input"> MPEG
Layer III input file</a></strong> </dt>
</dl>
<dl>
<dd> 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.<br>
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.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--noath</kbd><a name="-noath"> disable
ATH</a></strong> </dt>
</dl>
<dl>
<dd> Disable any use of the ATH (absolute threshold of hearing) for masking.
Normally, humans are unable to hear any sound below this threshold.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--noasm mmx/3dnow/sse</kbd><a name="-noasm">
disable assembly optimizations</a></strong> </dt>
</dl>
<dl>
<dd>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.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--nohist</kbd><a name="-nohist"> disable
histogram display</a></strong> </dt>
</dl>
<dl>
<dd> By default, LAME will display a bitrate histogram while producing VBR mp3
files. This will disable that feature.<br>
Histogram display might not be available on your release.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--noreplaygain</kbd><a name="-noreplaygain"> disable
ReplayGain analysis</a></strong></dt>
</dl>
<dl>
<dd> By default ReplayGain analysis is enabled. This switch disables it.<br>
<br>
See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>,
<a href="#-replaygain-fast">--replaygain-fast</a>
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--nores</kbd><a name="-nores"> disable
bit reservoir</a></strong></dt>
</dl>
<dl>
<dd> Disable the bit reservoir. Each frame will then become independent from
previous ones, but the quality will be lower.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--noshort</kbd><a name="-noshort"> disable
short blocks frames</a></strong></dt>
</dl>
<dl>
<dd> Encode all frames using long blocks only. This could increase quality when
encoding at very low bitrates, but can produce serious pre-echo artefacts.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--notemp</kbd><a name="-notemp"> disable
temporal masking</a></strong></dt>
</dl>
<dl>
<dd>Don't make use of the temporal masking effect.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-o</kbd><a name="o"> non-original</a></strong>
</dt>
</dl>
<dl>
<dd> Mark the encoded file as being a copy.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-p</kbd><a name="p"> error protection</a></strong></dt>
</dl>
<dl>
<dd> Turn on CRC error protection.<br>
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.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--preset presetName</kbd> <a name="-preset"> use
built-in preset</a></strong></dt>
</dl>
<dd> Use one of the built-in presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
<br>
<dd> "--preset help" gives more information about the usage possibilities for these presets.
<dt><br>
<br>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--priority 0...4</kbd><a name="-priority"> OS/2
process priority control</a></strong> </dt>
<dl>
<dd> With this option, LAME will run with a different process priority under
IBM OS/2.<br>
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.<br>
<br>
<dd><b>0 (Low priority)</b><br>
Priority 0 assumes "IDLE" class, with delta 0.<br>
LAME will have the lowest priority possible, and the encoding may be suspended
very frequently by user interaction.<br>
<br>
<dd><b>1 (Medium priority)</b><br>
Priority 1 assumes "IDLE" class, with delta +31.<br>
LAME won't interfere at all with what you're doing.<br>
Recommended if you have a slower machine. <br>
<br>
<dd><b>2 (Regular priority)</b><br>
Priority 2 assumes "REGULAR" class, with delta -31.<br>
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. <br>
<br>
<dd><b>3 (High priority)</b><br>
Priority 3 assumes "REGULAR" class, with delta 0.<br>
LAME will run with a priority a bit higher than a normal process. <br>
Good if you're just running LAME by itself or with moderate user interaction.<br>
<br>
<dd><b>4 (Maximum priority)</b><br>
Priority 4 assumes "REGULAR" class, with delta +31.<br>
LAME will run with a very high priority, and may interfere with the machine
response.<br>
Recommended if you only intend to run LAME by itself, or if you have a fast
processor. <br>
<br>
<br>
Priority 1 or 2 is recommended for most users.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-q 0..9</kbd><a name="q"> algorithm
quality selection</a></strong></dt>
</dl>
<dl>
<dd> 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).<br>
<br>
-q 0: use slowest & best possible version of all algorithms. -q 0 and -q 1
are slow and may not produce significantly higher quality.<br>
<br>
-q 2: recommended. Same as -h.<br>
<br>
-q 5: default value. Good speed, reasonable quality.<br>
<br>
-q 7: same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo
& M/S, but no noise shaping is done.<br>
<br>
-q 9: disables almost all algorithms including psy-model. poor quality.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-r</kbd><a name="r"> input file is
raw PCM</a></strong></dt>
</dl>
<dl>
<dd> 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.<br>
Might not be available on your release.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--replaygain-accurate</kbd><a name="-replaygain-accurate"> compute
ReplayGain more accurately and find the peak sample</a></strong></dt>
</dl>
<dl>
<dd>
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.<br>
<br>
ReplayGain analysis does <i>not</i> 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
<a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
<br>
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:
<ul>
<li> 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
</li>
<li> decoding on the fly significantly slows down the encoding process
</li>
</ul>
The apparent advantage is that:
<ul>
<li> 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.
</li>
</ul>
<br>
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.<br>
<br>
This option is not usable if the MP3 decoder was <b>explicitly</b>
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).<br>
<br>
See also: <a href="#-replaygain-fast">--replaygain-fast</a>,
<a href="#-noreplaygain">--noreplaygain</a>, <a href="#-clipdetect">--clipdetect</a>
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--replaygain-fast</kbd><a name="-replaygain-fast"> compute
ReplayGain fast but slightly inaccurately (default)</a></strong></dt>
</dl>
<dl>
<dd>
Compute "Radio" ReplayGain on the input data stream after user-specified
volume scaling and/or resampling.<br>
<br>
ReplayGain analysis does <i>not</i> 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
<a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
<br>
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.<br>
<br>
This switch is enabled by default.<br>
<br>
See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>,
<a href="#-noreplaygain">--noreplaygain</a>
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--resample 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="-resample"> output
sampling frequency in kHz</a></strong></dt>
</dl>
<dl>
<dd> Select output sampling frequency (for encoding only). <br>
If not specified, LAME will automatically resample the input when using high
compression ratios.
<dt><br>
</dt>
</dl>
<dl>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-s 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="s"> sampling
frequency</a></strong> </dt>
</dl>
<dl>
<dd> Required only for raw PCM input files. Otherwise it will be determined
from the header of the input file.<br>
<br>
LAME will automatically resample the input file to one of the supported MP3
samplerates if necessary.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-S / --silent / --quiet</kbd><a name="-silent"> silent
operation</a></strong> </dt>
</dl>
<dl>
<dd> Don't print progress report.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--scale n</kbd><a name="-scale"> scales
input by n</a></strong> </dt>
<dt><strong>* <kbd>--scale-l n</kbd><a name="-scale-l"> scales
input channel 0 (left) by n</a></strong> </dt>
<dt><strong>* <kbd>--scale-r n</kbd><a name="-scale-r"> scales
input channel 1 (right) by n</a></strong> </dt>
</dl>
<dl>
<dd>Scales input by n. This just multiplies the PCM data (after it has been
converted to floating point) by n. <br>
<br>
n > 1: increase volume<br>
n = 1: no effect<br>
n < 1: reduce volume<br>
<br>
Use with care, since most MP3 decoders will truncate data which decodes to
values greater than 32768.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--short</kbd><a name="-short"> use
short blocks</a></strong> </dt>
</dl>
<dl>
<dd>Let LAME use short blocks when appropriate. It is the default setting.
</dl>
<dl>
<dd>
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--strictly-enforce-ISO</kbd><a name="-strictly-enforce-ISO"> strict
ISO compliance</a></strong> </dt>
</dl>
<dl>
<dd> With this option, LAME will enforce the 7680 bit limitation on total frame
size.<br>
This results in many wasted bits for high bitrate encodings but will ensure
strict ISO compatibility. This compatibility might be important for hardware
players.
</dl>
<dl>
<dd>
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-t</kbd><a name="t"> disable INFO/WAV
header </a></strong></dt>
</dl>
<dl>
<dd> Disable writing of the INFO Tag on encoding.<br>
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.<br>
<br>
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.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-V 0...9</kbd><a name="V"> VBR quality
setting</a></strong></dt>
</dl>
<dl>
<dd> Enable VBR (Variable BitRate) and specifies the value of VBR quality.<br>
default=4<br>
0=highest quality.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--vbr-new</kbd><a name="-vbr-new"> new
VBR mode</a></strong></dt>
</dl>
<dl>
<dd> 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. <br>
It has the added advantage of being very fast (over twice as fast as --vbr-old).
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--vbr-old</kbd><a name="-vbr-old"> older
VBR mode</a></strong></dt>
</dl>
<dl>
<dd> 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.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>--verbose</kbd><a name="-verbose"> verbosity</a></strong></dt>
</dl>
<dl>
<dd> Print a lot of information on screen.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-x</kbd><a name="x"> swapbytes</a></strong>
</dt>
</dl>
<dl>
<dd> Swap bytes in the input file or output file when using --decode.<br>
For sorting out little endian/big endian type problems. If your encodings
sounds like static, try this first.
<dt><br>
<br>
</dt>
<hr width="50%" noshade align="center">
<br>
<dl> </dl>
<dt><strong>* <kbd>-X 0...7</kbd><a name="Xquant"> change
quality measure</a></strong> </dt>
</dl>
<dl>
<dd> 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:<br>
<br>
<b>-X0 </b><br>
The criterions are (in order of importance):<br>
* less distorted scalefactor bands<br>
* the sum of noise over the thresholds is lower<br>
* the total noise is lower<br>
<br>
<b>-X1</b><br>
The actual is better if the maximum noise over all scalefactor bands is less
than the best so far .<br>
<br>
<b>-X2</b><br>
The actual is better if the total sum of noise is lower than the best so far.<br>
<br>
<b>-X3</b><br>
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.<br>
<br>
<b>-X4</b> <br>
Not yet documented.<br>
<br>
<b>-X5</b><br>
The criterions are (in order of importance):<br>
* the sum of noise over the thresholds is lower <br>
* the total sum of noise is lower<br>
<br>
<b>-X6</b> <br>
The criterions are (in order of importance):<br>
* the sum of noise over the thresholds is lower<br>
* the maximum noise over all scalefactor bands is lower<br>
* the total sum of noise is lower<br>
<br>
<b>-X7</b> <br>
The criterions are:<br>
* less distorted scalefactor bands<br>
or<br>
* the sum of noise over the thresholds is lower
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