📄 switchs.html
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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>
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<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
</dl>
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