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- <H1>Full command line switch reference</H1>
- <P> <font size="-1">note: Options which could exist without being documented
- here are considered as experimental ones. Such experimental options should usually
- not be used.</font>
- <P>
- <TABLE CELLPADDING=3 BORDER="1">
- <TR VALIGN="TOP">
- <TD ALIGN="LEFT" nowrap><b>switch</b></TD>
- <TD ALIGN="LEFT" nowrap><b>parameter</b></TD>
- </TR>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#a">-a</a></kbd></td>
- <td align="LEFT" nowrap>downmix stereo file to mono</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-abr">--abr</a></kbd></td>
- <td align="LEFT" nowrap>average bitrate encoding</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-allshort">--allshort</a></kbd></td>
- <td align="LEFT" nowrap>use short blocks only</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-athlower">--athlower</a></kbd></td>
- <td align="LEFT" nowrap>lower the ATH</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-athonly">--athonly</a></kbd></td>
- <td align="LEFT" nowrap>ATH only</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-athshort">--athshort</a></kbd></td>
- <td align="LEFT" nowrap>ATH only for short blocks</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-athtype">--athtype</a></kbd></td>
- <td align="LEFT" nowrap>select ATH type</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#b">-b</a></kbd></td>
- <td align="LEFT" nowrap>bitrate (8...320)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#Bmax">-B</a></kbd></td>
- <td align="LEFT" nowrap>max VBR/ABR bitrate (8...320)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-bitwidth">--bitwidth</a></kbd></td>
- <td align="LEFT" nowrap>input bit width</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#c">-c</a></kbd></td>
- <td align="LEFT" nowrap>copyright</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-cbr">--cbr</a></kbd></td>
- <td align="LEFT" nowrap>enforce use of constant bitrate</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-clipdetect">--clipdetect</a></kbd></td>
- <td align="LEFT" nowrap>clipping detection</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-comp">--comp</a></kbd></td>
- <td align="LEFT" nowrap>choose compression ratio</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-cwlimit">--cwlimit</a></kbd></td>
- <td align="LEFT" nowrap>tonality limit</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#d">-d</a></kbd></td>
- <td align="LEFT" nowrap>block type control</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-decode">--decode</a></kbd></td>
- <td align="LEFT" nowrap>decoding only</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-disptime">--disptime</a></kbd></td>
- <td align="LEFT" nowrap>time between display updates</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#e">-e</a></kbd></td>
- <td align="LEFT" nowrap>de-emphasis (<b>n</b>, 5, c)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#f">-f</a></kbd></td>
- <td align="LEFT" nowrap> fast mode</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#FF">-F</a></kbd></td>
- <td align="LEFT" nowrap> strictly enforce the -b option</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-freeformat">--freeformat</a></kbd></td>
- <td align="LEFT" nowrap> free format bitstream</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#h">-h</a></kbd></td>
- <td align="LEFT" nowrap>high quality</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-help">--help</a></kbd></td>
- <td align="LEFT" nowrap> help</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass</a></kbd></td>
- <td align="LEFT" nowrap> highpass filtering frequency in kHz</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass-width</a></kbd></td>
- <td align="LEFT" nowrap> width of highpass filtering in kHz</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#k">-k</a></kbd></td>
- <td align="LEFT" nowrap> full bandwidth</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-lowpass">--lowpass</a></kbd></td>
- <td align="LEFT" nowrap> lowpass filtering frequency in kHz</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-lowpass-width">--lowpass-width</a></kbd></td>
- <td align="LEFT" nowrap> width of lowpass filtering in kHz</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#m">-m</a></kbd></td>
- <td align="LEFT" nowrap>stereo mode (s, <b>j</b>, f, m)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-mp1input">--mp1input</a></kbd></td>
- <td align="LEFT" nowrap>MPEG Layer I input file</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-mp2input">--mp2input</a></kbd></td>
- <td align="LEFT" nowrap>MPEG Layer II input file</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-mp3input">--mp3input</a></kbd></td>
- <td align="LEFT" nowrap>MPEG Layer III input file</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-noath">--noath</a></kbd></td>
- <td align="LEFT" nowrap>disable ATH</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-noasm">--noasm</a></kbd></td>
- <td align="LEFT" nowrap>disable assembly optimizations (mmx/3dnow/sse)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-nohist">--nohist</a></kbd></td>
- <td align="LEFT" nowrap>disable histogram display</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-noreplaygain">--noreplaygain</a></kbd></td>
- <td align="LEFT" nowrap>disable ReplayGain analysis</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-nores">--nores</a></kbd></td>
- <td align="LEFT" nowrap>disable bit reservoir</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-noshort">--noshort</a></kbd></td>
- <td align="LEFT" nowrap>disable short blocks frames</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-notemp">--notemp</a></kbd></td>
- <td align="LEFT" nowrap>disable temporal masking</td>
- </tr>
- <TR VALIGN="TOP">
- <TD ALIGN="LEFT" nowrap><kbd><a href="#o">-o</a></kbd></TD>
- <TD ALIGN="LEFT" nowrap>non-original</TD>
- </TR>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#p">-p</a></kbd></td>
- <td align="LEFT" nowrap>error protection</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-preset">--preset</a></kbd></td>
- <td align="LEFT" nowrap>use built-in preset</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-priority">--priority</a></kbd></td>
- <td align="LEFT" nowrap>OS/2 process priority control</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#q">-q</a></kbd></td>
- <td align="LEFT" nowrap>algorithm quality selection</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-silent">--quiet</a></kbd></td>
- <td align="LEFT" nowrap>silent operation</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#r">-r</a></kbd></td>
- <td align="LEFT" nowrap>input file is raw PCM</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-replaygain-accurate">--replaygain-accurate</a></kbd></td>
- <td align="LEFT" nowrap>compute ReplayGain more accurately and find the peak sample</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-replaygain-fast">--replaygain-fast</a></kbd></td>
- <td align="LEFT" nowrap>compute ReplayGain fast but slightly inaccurately (default)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-resample">--resample</a></kbd></td>
- <td align="LEFT" nowrap>output sampling frequency in kHz (encoding only)</td>
- </tr>
- <TR VALIGN="TOP">
- <TD ALIGN="LEFT" nowrap><kbd><a href="#s">-s</a></kbd></TD>
- <TD ALIGN="LEFT" nowrap>sampling frequency in kHz</TD>
- </TR>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-silent">-S</a></kbd></td>
- <td align="LEFT" nowrap>silent operation</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-scale">--scale</a></kbd></td>
- <td align="LEFT" nowrap>scale input</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-scale-l">--scale-l</a></kbd></td>
- <td align="LEFT" nowrap>scale input channel 0 (left)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-scale-r">--scale-r</a></kbd></td>
- <td align="LEFT" nowrap>scale input channel 1 (right)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-short">--short</a></kbd></td>
- <td align="LEFT" nowrap>use short blocks</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-silent">--silent</a></kbd></td>
- <td align="LEFT" nowrap>silent operation</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-strictly-enforce-ISO">--strictly-enforce-ISO</a></kbd></td>
- <td align="LEFT" nowrap>strict ISO compliance</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#t">-t</a></kbd></td>
- <td align="LEFT" nowrap>disable INFO/WAV header</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#V">-V</a></kbd></td>
- <td align="LEFT" nowrap>VBR quality setting (0...9)</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-vbr-new">--vbr-new</a></kbd></td>
- <td align="LEFT" nowrap>new VBR mode</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-vbr-old">--vbr-old</a></kbd></td>
- <td align="LEFT" nowrap>older VBR mode</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#-verbose">--verbose</a></kbd></td>
- <td align="LEFT" nowrap>verbosity</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#x">-x</a></kbd></td>
- <td align="LEFT" nowrap>swapbytes</td>
- </tr>
- <tr valign="TOP">
- <td align="LEFT" nowrap><kbd><a href="#Xquant">-X</a></kbd></td>
- <td align="LEFT" nowrap>change quality measure</td>
- </tr>
- </TABLE>
- <BR>
- <dl>
- <dt><strong>* <kbd>-a</kbd><a name="a"> downmix </a></strong>
- <dd>Mix the stereo input file to mono and encode as mono.<br>
- The downmix is calculated as the sum of the left and right channel, attenuated
- by 6 dB. <br>
- <br>
- This option is only needed in the case of raw PCM stereo input (because LAME
- cannot determine the number of channels in the input file).<br>
- To encode a stereo PCM input file as mono, use "lame -m s -a".<br>
- <br>
- For WAV and AIFF input files, using "-m m" will always produce a mono .mp3
- file from both mono and stereo input.
- <dt><br>
- </dt>
- <hr width="50%" noshade align="center">
- <br>
- </dl>
- <dl>
- <dt><strong>* <kbd>--abr n</kbd><a name="-abr"> average
- bitrate encoding</a></strong> </dt>
- </dl>
- <dl>
- <dd>Turns on encoding with a targeted average bitrate of n kbits, allowing to
- use frames of different sizes. The allowed range of n is 8-310, you can use
- any integer value within that range.<br>
- <br>
- It can be combined with the -b and -B switches like:<br>
- lame --abr 123 -b 64 -B 192 a.wav a.mp3<br>
- which would limit the allowed frame sizes between 64 and 192 kbits. <br>
- <br>
- </dt>
- <hr width="50%" noshade align="center">
- <br>
- </dl>
- <dl>
- <dt><strong>* <kbd>--allshort</kbd><a name="-allshort"> use
- short blocks only</a></strong> </dt>
- </dl>
- <dl>
- <dd>Use only short blocks, no long ones.
- </dl>
- <dl>
- <dd>
- <dt><br>
- </dt>
- <hr width="50%" noshade align="center">
- <br>
- </dl>
- <dl>
- <dt><strong>* <kbd>--athlower n</kbd><a name="-athlower"> lower
- 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>
- </dt>
- <hr width="50%" noshade align="center">
- <br>
- <dl> </dl>
- <dt><strong>* <kbd>-k</kbd><a name="k"> 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"> 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"> 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"> 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>
- <hr width="50%" noshade align="center">
- <br>
- <dl> </dl>
- <dt><strong>* <kbd>--mp1input</kbd><a name="-mp1input"> 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 ".mpg" 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"> 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"> 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"> 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">
- 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"> 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"> 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"> 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"> 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"> 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"> 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"> 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"> 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"> 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"> 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 & 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
- & 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>
- <hr width="50%" noshade align="center">
- <br>
- <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>
- <hr width="50%" noshade align="center">
- <br>
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- <hr width="50%" noshade align="center">
- <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>
- </BODY>
- </HTML>