switchs.html

音频编码

    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>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athonly</kbd><a name="-athonly">&nbsp;&nbsp;&nbsp;&nbsp;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>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athshort</kbd><a name="-athshort">&nbsp;&nbsp;&nbsp;&nbsp;ATH 
    only for short blocks</a></strong> </dt>
</dl>
<dl> 
  <dd>Ignore psychoacoustic model for short blocks, use ATH only. 
</dl>
<dl> 
  <dd>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>--athtype 0/1/2</kbd><a name="-athtype">&nbsp;&nbsp;&nbsp;&nbsp;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>&nbsp; 
  <dt><br>
  </dt>
  <hr width="50%" noshade align="center">
  <br>
</dl>
<dl> 
  <dt><strong>* <kbd>-b n</kbd><a name="b">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;maximum 
    VBR/ABR bitrate&nbsp;</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">&nbsp;&nbsp;&nbsp;&nbsp;input 
    bit width&nbsp;</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">&nbsp;&nbsp;&nbsp;&nbsp;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">
    &nbsp;&nbsp;&nbsp;&nbsp;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">
    &nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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">&nbsp;&nbsp;&nbsp;&nbsp;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>