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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>
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