switchs.html

音频编码

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  <dl> </dl>
  <dt><strong>* <kbd>-r</kbd><a name="r">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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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  <dl> </dl>
  <dt><strong>* <kbd>--replaygain-fast</kbd><a name="-replaygain-fast">&nbsp;&nbsp;&nbsp;&nbsp;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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  <dl> </dl>
  <dt><strong>* <kbd>--resample 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="-resample">&nbsp;&nbsp;&nbsp;&nbsp;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> 
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  <dl> </dl>
  <dt><strong>* <kbd>-s 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="s">&nbsp;&nbsp;&nbsp;&nbsp;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>
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  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-S / --silent / --quiet</kbd><a name="-silent">&nbsp;&nbsp;&nbsp;&nbsp;silent 
    operation</a></strong> </dt>
</dl>
<dl> 
  <dd> Don't print progress report. 
  <dt><br>
    <br>
  </dt>
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  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--scale n</kbd><a name="-scale">&nbsp;&nbsp;&nbsp;&nbsp;scales 
    input by n</a></strong> </dt>
  <dt><strong>* <kbd>--scale-l n</kbd><a name="-scale-l">&nbsp;&nbsp;&nbsp;&nbsp;scales 
    input channel 0 (left) by n</a></strong> </dt>
  <dt><strong>* <kbd>--scale-r n</kbd><a name="-scale-r">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;use 
    short blocks</a></strong> </dt>
</dl>
<dl> 
  <dd>Let LAME use short blocks when appropriate. It is the default setting. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
    <br>
  </dt>
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  <dl> </dl>
  <dt><strong>* <kbd>--strictly-enforce-ISO</kbd><a name="-strictly-enforce-ISO">&nbsp;&nbsp;&nbsp;&nbsp;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>&nbsp; 
  <dt><br>
    <br>
  </dt>
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  <dl> </dl>
  <dt><strong>* <kbd>-t</kbd><a name="t">&nbsp;&nbsp;&nbsp;&nbsp;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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  <dl> </dl>
  <dt><strong>* <kbd>-V 0...9</kbd><a name="V">&nbsp;&nbsp;&nbsp;&nbsp;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>
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  <dl> </dl>
  <dt><strong>* <kbd>--vbr-new</kbd><a name="-vbr-new">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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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  <dl> </dl>
  <dt><strong>* <kbd>--verbose</kbd><a name="-verbose">&nbsp;&nbsp;&nbsp;&nbsp;verbosity</a></strong></dt>
</dl>
<dl> 
  <dd> Print a lot of information on screen. 
  <dt><br>
    <br>
  </dt>
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  <dl> </dl>
  <dt><strong>* <kbd>-x</kbd><a name="x">&nbsp;&nbsp;&nbsp;&nbsp;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>
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  <dl> </dl>
  <dt><strong>* <kbd>-X 0...7</kbd><a name="Xquant">&nbsp;&nbsp;&nbsp;&nbsp;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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