📄 switchs.html
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<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 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.<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>Determine whether clipping occurs and display an appropriate message.
Store the MP3 peak sample in LAME Tag.
<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.
<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 highpass filtering at low 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 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. <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 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.<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.<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 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.<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. Usefull for downsampling from one
mp3 to another. As an example, it can be usefull 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 optimisations</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>
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