switchs.html

音频编码

  </dt>
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  <br>
  <dl> </dl>
  <dt><strong>* <kbd>-k</kbd><a name="k">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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>
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  <br>
  <dl> </dl>
  <dt><strong>* <kbd>--mp1input</kbd><a name="-mp1input">&nbsp;&nbsp;&nbsp;&nbsp;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 &quot;.mpg&quot; 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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">
   &nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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 &amp; 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 
    &amp; 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>