Joburgess
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lame-ffmpeg-wasm


			
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	<h2 class="hilight">Detailed command line switches</h2>
</div>

<h3>Options ordered by feature</h3>
<table class="settings" cellpadding="4">

<tr><th style="width:20%">Setting</th><th>Brief description</th></tr>
<tr><th colspan="2">Input options</th></tr>
<tr><td colspan="2">By default, LAME accepts a PCM audio sample inside a .WAV
container as the input file, in 8, 16, 24 and 32 bits integer and in IEEE FLOAT.
If it is compiled with <a href="http://www.mega-nerd.com/libsndfile/">libsndfile</a>,
then it also supports the extra formats that the library supports.<br/>
There is also support for RAW PCM data and and piped input</td></tr>
<tr><td>-</td><td>For piped/streamed Input, use "-" as the input and/or output
 file name.</td></tr>
<tr><td>--out-dir path</td><td>If no explicit output file is specified, a file will be written at given path. Ignored when using piped/streamed input</td></tr>
<tr><td><a href="#mp1input">--mp1input</a><a href="#note1"><sup>1</sup></a></td><td>Input file is an MPEG
 layer I file. Implicit if extension is .mp1</td></tr>
<tr><td><a href="#mp2input">--mp2input</a><a href="#note1"><sup>1</sup></a></td><td>Input file is an MPEG
 layer II file. Implicit if extension is .mp2</td></tr>
<tr><td><a href="#mp3input">--mp3input</a><a href="#note1"><sup>1</sup></a></td><td>Input file is an MPEG
 layer III file. Implicit if extension is .mp3</td></tr>
<tr><td><a href="#scale">--scale</a> number</td><td>Scale input (multiply PCM data) by number.</td></tr>
<tr><td><a href="#scale-l">--scale-l</a> number</td><td>Scale channel 0 (left) input (multiply PCM data
) by number</td></tr>
<tr><td><a href="#scale-r">--scale-r</a> number</td><td>Scale channel 1 (right) input (multiply PCM data)
 by number</td></tr>
<tr><td><a href="#gain">--gain</a> number</td><td>apply Gain adjustment in decibels, range -20.0 to +12.0.</td></tr>
<tr><td>--swap-channel</td><td>Swap Left and Right channels</td></tr>
<tr><td><a href="#nogap">--nogap</a> file1 file2 ...</td><td>Gapless encoding for a set of contiguous
 files</td></tr>
<tr><td><a href="#nogapout">--nogapout</a> dir</td><td>Output dir for gapless encoding (must precede
 --nogap)</td></tr>
<tr><td><a href="#nogaptags">--nogaptags</a></td><td>Allow the use of VBR tags in gapless encoding</td></tr>

<tr><th colspan="2">Input options for RAW PCM</th></tr>
<tr><td><a href="#r">-r</a></td><td>Assume input file is raw PCM.</td></tr>
<tr><td><a href="#s">-s</a> number</td><td>Input sampling frequency in kHz (Default is 44.1Khz)</td></tr>
<tr><td><a href="#signed">--signed</a> / <a href="#unsigned">--unsigned</a></td><td> Input is signed (default) or unsigned</td></tr>
<tr><td><a href="#bitwidth">--bitwidth</a> w</td><td> Input bit width is w (default 16)</td></tr>
<tr><td><a href="#x">-x</a></td><td> Force byte-swapping of input stream</td></tr>
<tr><td><a href="#little-endian">--little-endian</a> / <a href="#big-endian">--big-endian</a></td><td> Input is little-endian (default)
 or big-endian</td></tr>
<tr><td><a href="#a">-a</a></td><td>Downmix from stereo to mono file for mono encoding. Needed with RAW input for the -mm mode to do the downmix.</td></tr>

<tr><th colspan="2">Constant Bit Rate (CBR)</th></tr>
<tr><td><a href="#b">-b</a>  number</td><td>Set the bitrate in kbps, default 128 kbps</td></tr>
<tr><td><a href="#cbr">--cbr</a></td><td>Enforce use of constant bitrate</td></tr>
<tr><td><a href="#comp">--comp</a> ratio</td><td>Choose bitrate to obtain the specified compression
 ratio</td></tr>
<tr><td><a href="#freeformat">--freeformat</a></td><td>Produce a free format bitstream instead of the 
standard one. Supported bitrates between 8 and 640 kbps. Scarce support by
decoders.</td></tr>

<tr><th colspan="2">Variable Bit Rate (<a href="abr.html">ABR</a>)</th></tr>
<tr><td><a href="#abr">--abr</a> number</td><td>Specify average bitrate desired (instead of quality)
</td></tr>

<tr><th colspan="2">Variable Bit Rate (<a href="vbr.html">VBR</a>)</th></tr>
<tr><td><a href="#V">-V</a> number</td><td>Quality setting for VBR.  default number=4.<br/>
                    Decimal values can be specified, like: 4.51<br/>
                    0=highest quality, bigger files. 9.999=lowest quality,
                     smaller files</td></tr>
<tr><td><a href="#vbr-old">--vbr-old</a></td><td>Use old variable bitrate (VBR) routine</td></tr>
<tr><td><a href="#b">-b</a>  number</td><td>Specify a minimum allowed bitrate. default 32 kbps (See -F setting below)</td></tr>
<tr><td><a href="#B">-B</a>  number</td><td>Specify a maximum allowed bitrate. default 320 kbps</td></tr>
<tr><td><a href="#F">-F</a></td><td>Enforce -b option for digital silence frames too, rather than encoding them at 8 or 32 kbps.</td></tr>
<tr><td><a href="#t">-t</a></td><td>Disable VBR informational tag</td></tr>
<tr><td><a href="#T">-T</a></td><td>Enable and force writing LAME Tag</td></tr>
<tr><td><a href="#Y">-Y</a></td><td>Allows -V2, -V1 and -V0 to not to encode the highest frequencies accurately, if doing so causes disproportional increases in bitrate. This is the same that CBR and ABR modes do.</td></tr>

<tr><th colspan="2">Operational options</th></tr>
<tr><td><a href="#preset">--preset</a> type</td><td>Enables some preconfigured settings. Check below
for each of the valid values</td></tr>
<tr><td><a href="#priority">--priority</a> number</td><td>Sets the process priority:<br/>
                         0,1 = Low priority <br/>
                         2 = normal priority (default)<br/>
                         3,4 = High priority </td></tr>
<tr><td><a href="#flush">--flush</a></td><td>Flush output stream as soon as possible</td></tr>
<tr><td><a href="#decode">--decode</a></td><td>Input=mp3 file, output=wav</td></tr>
<tr><td><a href="#decode-mp3delay">--decode-mp3delay</a> samples</td><td>Set the encoder delay to use to decode 
the input .mp3 file</td></tr>
<tr><td><a href="#t_deco">-t</a></td><td>Disable writing wav header when using --decode</td></tr>
<tr><td><a href="#noasm">--noasm</a> type</td><td>Disable assembly optimizations for mmx/3dnow/sse</td></tr>

<tr><th colspan="2">Noise shaping &amp; psycho acoustic algorithms</th></tr>
<tr><td><a href="#q">-q</a> number</td><td>number = 0...9.  Default  -q 3<br/>
                    -q 0:  Highest quality, very slow<br/>
                    -q 9:  Poor quality, but fast<br/>
					Please, see notes in the detailed description</td></tr>
<tr><td><a href="#m">-m</a> mode</td><td><b>j</b>oint, <b>s</b>imple, <b>f</b>orce, <b>d</b>ual-mono,
 <b>m</b>ono, <b>l</b>eft, <b>r</b>ight. default is <b>j</b>.<br/>
                    joint  = joins the best possible of MS and LR stereo<br/>
                    simple = force LR stereo on all frames<br/>
                    force  = force MS stereo on all frames.</td></tr>

<tr><th colspan="2">Filter and resampling options:</th></tr>
<tr><td><a href="#resample">--resample</a> sfreq</td><td>Sampling frequency of output file(kHz)- 
default=automatic</td></tr>
<tr><td><a href="#lowpass">--lowpass</a> number</td><td>Frequency(kHz), lowpass filter cutoff above freq.
 Range [0.001..50]kHz or [50..50000]Hz</td></tr>
<tr><td><a href="#lowpass">--lowpass</a>-width number</td><td>Frequency(kHz), lowpass window width. Range [0.001..16]kHz or
 [16..50000]Hz - (See further restriction in the detailed explanation)</td></tr>
<tr><td><a href="#highpass">--highpass</a> number</td><td>Frequency(kHz), highpass filter cutoff below freq.  Range [0.001..16]kHz or
 [16..50000]Hz (See further restriction in the detailed explanation)</td></tr>
<tr><td><a href="#highpass">--highpass</a>-width number</td><td>Frequency(kHz), highpass window width - (See further restriction in the detailed explanation)</td></tr>

<tr><th colspan="2">ID3 tag Information</th></tr>
<tr><td><a href="#tt">--tt</a> title</td><td>Audio/song title (max 30 chars for version 1 tag)</td></tr>
<tr><td><a href="#ta">--ta</a> artist</td><td>Audio/song artist (max 30 chars for version 1 tag)</td></tr>
<tr><td><a href="#tl">--tl</a> album</td><td>Audio/song album (max 30 chars for version 1 tag)</td></tr>
<tr><td><a href="#ty">--ty</a> year</td><td>Audio/song year of issue (1 to 9999)</td></tr>
<tr><td><a href="#tc">--tc</a> comment</td><td>User-defined text (max 30 chars for v1 tag, 28 for v1.1)</td></tr>
<tr><td><a href="#tn">--tn</a> track[/total]</td><td>Audio/song track number and (optionally) the total
                           number of tracks on the original recording.<br/> (track
                           and total each 1 to 255. just the track number
                           creates v1.1 tag, providing a total forces v2.0).</td></tr>
<tr><td><a href="#tg">--tg</a> genre</td><td>Audio/song genre (name or number in list)</td></tr>
<tr><td><a href="#ti">--ti</a> file</td><td>Audio/song albumArt (jpeg/png/gif file, 128KB max, v2.3)</td></tr>
<tr><td><a href="#tv">--tv</a> id=value</td><td>Text or URL frame specified by id and value (v2.3 tag). User defined frame. Syntax: --tv "TXXX=description=content"</td></tr>
<tr><td><a href="#add-id3v2">--add-id3v2</a></td><td>Force addition of version 2 tag</td></tr>
<tr><td><a href="#id3v1-only">--id3v1-only</a></td><td>Add only a version 1 tag</td></tr>
<tr><td><a href="#id3v2-only">--id3v2-only</a></td><td>Add only a version 2 tag</td></tr>
<tr><td><a href="#id3v2-latin1">--id3v2-latin1</a></td><td>Add following options in ISO-8859-1 text encoding</td></tr>
<tr><td><a href="#id3v2-utf16">--id3v2-utf16</a></td><td>Add following options in unicode text encoding</td></tr>
<tr><td><a href="#space-id3v1">--space-id3v1</a></td><td>Pad version 1 tag with spaces instead of nulls</td></tr>
<tr><td><a href="#pad-id3v2-size">--pad-id3v2-size</a> n</td><td>Adds ID3v2 tag with n padding bytes</td></tr>
<tr><td><a href="#genre-list">--genre-list</a></td><td>Print alphabetically sorted ID3 genre list and exit</td></tr>
<tr><td><a href="#ignore-tag-errors">--ignore-tag-errors</a></td><td>Ignore errors in values passed for tags</td></tr>
<tr><td colspan="2">
    Note: A version 2 tag will NOT be added unless one of the input fields
    won't fit in a version 1 tag (e.g. the title string is longer than 30
    characters), or the '--add-id3v2' or '--id3v2-only' options are used,
    or output is redirected to stdout.</td></tr>

<tr><th colspan="2">MP3 header/stream options:</th></tr>
<tr><td><a href="#e">-e</a> mode</td><td>De-emphasis n/5/c  (obsolete)</td></tr>
<tr><td><a href="#c">-c</a></td><td>Mark as copyright</td></tr>
<tr><td><a href="#o">-o</a></td><td>Mark as non-original</td></tr>
<tr><td><a href="#p">-p</a></td><td>Error detection.  adds 16 bit checksum to every frame
                    (the checksum is computed correctly)</td></tr>
<tr><td><a href="#strictly-enforce-ISO">--strictly-enforce-ISO</a></td><td>Comply as much as possible to ISO MPEG spec</td></tr>
<tr><td><a href="#replaygain-fast">--replaygain-fast</a></td><td>Compute RG fast but slightly inaccurately (default)</td></tr>
<tr><td><a href="#replaygain-accurate">--replaygain-accurate</a><a href="#note1"><sup>1</sup></a></td><td>Compute RG
 more accurately and find the peak sample</td></tr>
<tr><td><a href="#noreplaygain">--noreplaygain</a></td><td>Disable ReplayGain analysis</td></tr>
<tr><td><a href="#clipdetect">--clipdetect</a><a href="#note1"><sup>1</sup></a></td><td>Enable
 --replaygain-accurate and print a message whether
 clipping occurs and how far the waveform is from full scale</td></tr>

<tr><th colspan="2">Verbosity:</th></tr>
<tr><td><a href="#disptime">--disptime</a> secs</td><td>Print progress report every secs seconds</td></tr>
<tr><td><a href="#nohist">--nohist</a></td><td>Disable VBR histogram display</td></tr>
<tr><td><a href="#silent">--silent</a> / <a href="#quiet">--quiet</a></td><td>Don't print anything on screen</td></tr>
<tr><td><a href="#verbose">--verbose</a></td><td>Print a lot of useful information</td></tr>
<tr><td><a href="#version">--version</a> / <a href="#license">--license</a></td><td>Print License information</td></tr>
<tr><td><a href="#help">--help</a> / <a href="#usage">--usage</a></td><td>Shows the common list of switches. Add <i>id3</i> or <i>dev</i> to get help for a specified topic</td></tr>
<tr><td><a href="#longhelp">--longhelp</a></td><td>Shows the complete list of switches</td></tr>

<tr><th colspan="2">Experimental switches for developers. Only enabled in alpha/debug 
versions. Can change from version to version</th></tr>
<tr><td>--nores</td><td>Disables the bit reservour. Each frame will become independent from previous ones, but the quality will be lower.</td></tr>
<tr><td>--buffer-constraint <i>x</i></td><td>available values for <i>x</i>: default, strict, maximum</td></tr>
<tr><td>--noath</td><td>turns ATH down to a flat noise floor</td></tr>
<tr><td>--athshort</td><td>ignore GPSYCHO for short blocks, use ATH only</td></tr>
<tr><td>--athonly</td><td>ignore GPSYCHO completely, use ATH only</td></tr>
<tr><td>--athtype x</td><td>selects between different ATH types [0-4]</td></tr>
<tr><td>--athlower x</td><td>Lower the ATH by x.x dB's</td></tr>
<tr><td>--athaa-type n</td><td>ATH auto adjust: 0 'no' else 'loudness based'</td></tr>
<tr><td>--athaa-sensitivity x</td><td>activation offset in -/+ dB for ATH auto-adjustment</td></tr>
<tr><td>--short</td><td>Allow the use of short blocs (default)</td></tr>
<tr><td>--noshort</td><td>Disable use of short blocks</td></tr>
<tr><td>--allshort</td><td>Use only short blocks</td></tr>
<tr><td>--shortthreshold x[,y]</td><td>short block switching threshold, x for L/R/M channel, y for S channel</td></tr>
<tr><td>--temporal-masking x</td><td>x=0 disables, x=1 enables temporal masking effect</td></tr>
<tr><td>--notemp</td><td>Alias of --temporal-masking 0</td></tr>
<tr><td>--nssafejoint</td><td>M/S switching criterion</td></tr>
<tr><td>--nsmsfix x</td><td>M/S switching tuning [effective 0-3.5]</td></tr>
<tr><td>--ns-bass x</td><td>Adjust masking for sfbs  0 -  6 (long)  0 -  5 (short)</td></tr>
<tr><td>--ns-alto x</td><td>Adjust masking for sfbs  7 - 13 (long)  6 - 10 (short)</td></tr>
<tr><td>--ns-treble x</td><td>Adjust masking for sfbs 14 - 21 (long) 11 - 12 (short)</td></tr>
<tr><td>--ns-sfb21 x</td><td>Change ns-treble by x dB for sfb21</td></tr>
<tr><td>-Z [n]</td><td>always do calculate short block maskings</td></tr>
<tr><td>--substep</td><td>use pseudo substep noise shaping method types 0-2</td></tr>
<tr><td>-X n[,m]</td><td>selects between different noise measurements n for long block, m for short. if m is omitted, m = n</td></tr>                    
<tr><td>--vbr-new</td><td>In LAME 3.98, the new VBR was made default, and this switch may be used for experimental tweaks.</td></tr>


<tr><th colspan="2">Aliases and removed settings</th></tr>
<tr><td>-f</td><td>Alias of <a href="#q">-q 7</a>.</td></tr>
<tr><td>-h</td><td>Alias of <a href="#q">-q 2</a>.</td></tr>
<tr><td>-v</td><td>Alias of <a href="#V">-V 4</a></td></tr>
<tr><td>-S</td><td>Alias of <a href="#silent">--silent</a></td></tr>
<tr><td>--alt-preset</td><td>Alias of <a href="#preset">--preset</a></td></tr>
<tr><td>--r3mix</td><td>Alias of <a href="#V">-V 3</a></td></tr>
<tr><td>--vbr-mtrh</td><td>Alias of LAME 3.98 default vbr mode</td></tr>
<tr><td>--pad-id3v2</td><td>Alias of <a href="#pad-id3v2-size">--pad-id3v2-size 128</a></td></tr>
<tr><td>-d</td><td>No longer supported</td></tr>
<tr><td>-k</td><td>No longer supported</td></tr>
<tr><td>--cwlimit</td><td>No longer supported</td></tr>
<tr><td>--ogginput</td><td>No longer supported. Compile with 
<a href="http://www.mega-nerd.com/libsndfile/">libsndfile</a> instead</td></tr>
<tr><td>--brief</td><td>Currently unused</td></tr>
<tr><td>--interch x</td><td>Currently unused</td></tr>
<tr><td style="border-bottom:0px;">--nspsytune</td><td style="border-bottom:0px;">Currently unused</td></tr>
</table>

<h3>Detailed description in alphabetical order</h3>

<p class="settingtitle">
<a name="a"><span class="hilight">-a</span></a> Downmix
</p>
<p>  
mix the stereo input file to mono and encode as mono.<br/>  
</p>
<p>
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).
To encode a stereo PCM input file as mono, use "lame -m m -a"<br/>
</p>
<p>
For WAV and AIFF input files, using "-m m" will always produce a
mono .mp3 file from both mono and stereo input.
</p>

<p class="settingtitle">
<a name="abr"><span class="hilight">--abr n</span></a> Average bitrate encoding (aka Safe VBR)
</p>
<p>
turns on encoding with a targeted average bitrate of n kbps, allowing
to use frames of different sizes.  The allowed range of n is 8...320 
kbps, you can use any integer value within that range.
</p>

<p class="settingtitle">
<a name="add-id3v2"><span class="hilight">--add-id3v2</span></a> Force addition of version 2 tag
</p>
<p>
Tells LAME to add the tag information as id3v2. This implies adding both, a version 1
and a version 2 tag, if the values fit on a version 1 tag. See --id3v1-only and
--id3v2-only if you want a more fine-grained control.
</p>

<p class="settingtitle">
<a name="b"><span class="hilight">-b n</span></a> Bitrate
</p>
<table cellpadding="5">
<tr><th>Codec</th>
<th>sample frequencies (kHz)</th>
<th>bitrates (kbps)</th></tr>
<tr><td>MPEG-1 layer III </td>  <td>32, 48, 44.1</td>
<td>32 40 48 56 64 80 96 112 128 160 192 224 256 320</td></tr>
<tr><td>MPEG-2 layer III</td><td> 16, 24, 22.05</td>
<td> 8 16 24 32 40 48 56 64 80 96 112 128 144 160</td></tr>
<tr><td style="border-bottom:0px;">MPEG-2.5 layer III</td>
<td style="border-bottom:0px;"> 8, 12, 11.025</td>
<td style="border-bottom:0px;">8 16 24 32 40 48 56 64</td></tr>
</table>
<p>
The bitrate to be used. Default is 128kbps in MPEG1 (64 for mono),
64kbps in MPEG2 (32 for mono) and 32kbps in MPEG2.5 (16 for mono).
</p>
<p>
When used with variable bitrate encodings (VBR), -b specifies the
minimum bitrate to use. This is useful only if you need to circumvent
a buggy hardware device with strange bitrate constrains. (You will need the -F setting too for digital silence).<br/>
Default for ABR/VBR is the minimum allowed bitrate for the MPEG version. (i.e. no limit)
</p>

<p class="settingtitle">
<a name="B"><span class="hilight">-B  n</span></a> Max bitrate
</p>
<p>
see also option "-b" for allowed bitrates.
</p>
<p>
Maximum allowed bitrate when using VBR/ABR.
</p>
<p>
Using -B is NOT RECOMMENDED.  A 128 kbps CBR bitstream, because of the
bit reservoir, can actually have frames which use as many bits as a
320 kbps frame.  ABR/VBR modes minimize the use of the bit reservoir, and
thus need to allow 320 kbps frames to get the same flexibility as CBR
streams.  This is useful only if you need to circumvent a buggy hardware
device with strange bitrate constrains.<br/>
Default value is the maximum allowed bitrate for the MPEG versio (i.e. no limit)
</p>

<p class="settingtitle">
<a name="big-endian"><span class="hilight">--big-endian</span></a> Set the byte order to big-endian.
</p>
<p>
This switch tells LAME that the RAW pcm input is encoded in big-endian instead
of little-endian.
</p>

<p class="settingtitle">
<a name="bitwidth"><span class="hilight">--bitwidth</span></a> Sets the bitwidth value
</p>
<p>
With RAW pcm input, this switch lets specify the bitwidth of the same (8 bits,
 16 bits...)
</p>

<p class="settingtitle">
<a name="c"><span class="hilight">-c</span></a> copyright flag   
</p>
<p>
mark the encoded file as copyrighted
</p>

<p class="settingtitle">
<a name="cbr"><span class="hilight">--cbr</span></a> Enforce constant bitrate encoding
</p>
<p>This switch disables the VBR encoding. Examples:</p>
<pre>lame -V 0 --cbr input.wav</pre>
Will encode at -b 128 (ignore completely -V and encode at default bitrate)
<pre>lame --preset 160 --cbr</pre>
Will encode at -b 160 (ignore ABR and use the specified bitrate)
<pre>lame --abr 200 --cbr</pre>
Will encode at -b 192 (ignore ABR and use the nearest bitrate to the one specified)

<p class="settingtitle">
<a name="clipdetect"><span class="hilight">--clipdetect</span></a> Clipping detection
</p>
<p>
Enable --replaygain-accurate and print a message whether clipping 
occurs and how far in dB the waveform is from full scale.
</p>
<p>
This option is not usable if the MP3 decoder was <b>explicitly</b> disabled 
in the build of LAME.
</p>
<p>
See also: --replaygain-accurate
</p>

<p class="settingtitle">
<a name="comp"><span class="hilight">--comp x</span></a> Indicate a compression ratio instead of a bitrate 
</p>
<p>
Use this in place of the -b setting if you prefer to indicate a compression ratio (integer value).
The ratio is the value of <sup>original-filesize</sup>/<sub>desired-filesize</sub>
</p>

<p class="settingtitle">
<a name="decode"><span class="hilight">--decode</span></a> MP3 decode capability
</p> 
<p>
This uses LAME's HIP<a href="#note2"><sup>2</sup></a> decoder to decode an MP3 (layers 1, 2 and 3) file to a wav file.
</p>
<p>
If -t is used (disable wav header), LAME will output
raw pcm in native endian format (use -x to swap bytes).
</p>
<p>
This option is not usable if the MP3 decoder was <b>explicitly</b> disabled
in the build of LAME.
</p>

<p class="settingtitle">
<a name="decode-mp3delay"><span class="hilight">--decode-mp3delay x</span></a> Indicate a different encoder delay
for decoding 
</p>
<p>
When decoding an mp3 file, LAME automatically corrects for the start delay that
the encoder had to put into it.<br/>
This setting lets you specify a different delay than LAME's own one, so that it
is possible to compensate for the delay of mp3's generated with other encoders.
</p>

<p class="settingtitle">
<a name="disptime"><span class="hilight">--disptime secs</span></a> Update the display each secs
</p>
<p>
Use this setting to change the frequency that LAME updates the display when
encoding a file. Allows decimal values (like 0.5)
</p>

<p class="settingtitle">
<a name="e"><span class="hilight">-e  n/5/c</span></a> Write the de-emphasis flag
</p>
<p>
  n = (none, default)<br/>
  5 = 50/15 microseconds<br/>
  c = ccitt j.17<br/>
</p>
<p>
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.
</p>
<p>
A better solution would be to apply the de-emphasis with a standalone
utility before encoding, and then encode without -e.  
</p>

<p class="settingtitle">
<a name="F"><span class="hilight">-F</span></a> Enforce the minimum bitrate
</p>
<p>
Enforces the minimum bitrate. Without this option, passages of analog
silence will be encoded at the minimum bitrate possible (32 or 8, depending on
MPEG version).
</p>

<p class="settingtitle">
<a name="flush"><span class="hilight">--flush</span></a> Flush the output stream as soon as possible
</p>
<p>
This setting forces a flush of the data written as soon as the operation has
finished. This is mostly useful in case of streams. With files, it may degrade
performance, since the OS has to write to disk in smaller chunks. 
</p>

<p class="settingtitle">
<a name="freeformat"><span class="hilight">--freeformat</span></a> Encode to freeformat stream
</p>
<p>
LAME will produce a fixed bitrate, free format bitstream.
User must specify the desired bitrate in kbps, which can
be any integer between 8 and 640.
</p>
<p>
Not supported by most decoders.  Compliant decoders (of which there
are few) are only required to support up to 320 kbps.  
</p>
<p>
Decoders knwon to handle free format:
</p>
<table cellpadding="4">
<tr><th></th><th>Supports up to</th></tr>
<tr><td>MAD</td><td>640 kbps</td></tr>
<tr><td>"lame --decode"</td><td>640 kbps</td></tr>
<tr><td style="border-bottom:0px;">l3dec</td><td style="border-bottom:0px;">310 kbps</td></tr>
</table>

<p class="settingtitle">
<a name="gain"><span class="hilight">--gain</span></a> Apply gain in decibels.
</p>
<p>
Apply Gain adjustment in decibels, range -20.0 to +12.0. 0dBFS means no amplification.
</p>

<p class="settingtitle">
<a name="genre-list"><span class="hilight">--genre-list</span></a> Print alphabetical ordered list of
known genres
</p>
<p>
Tell LAME to print the list of genres with their index to be used with the --tg
 setting.
</p>
<table cellpadding="4">
<tr><td style="border-bottom-width:0px;">123 Acappella<br />
 34 Acid<br />
 74 Acid Jazz<br />
 73 Acid Punk<br />
 99 Acoustic<br />
 20 Alternative<br />
 40 Alternative Rock<br />
 26 Ambient<br />
145 Anime<br />
 90 Avantgarde<br />
116 Ballad<br />
 41 Bass<br />
135 Beat<br />
 85 Bebob<br />
 96 Big Band<br />
138 Black Metal<br />
 89 Bluegrass<br />
  0 Blues<br />
107 Booty Bass<br />
132 BritPop<br />
 65 Cabaret<br />
 88 Celtic<br />
104 Chamber Music<br />
102 Chanson<br />
 97 Chorus<br />
136 Christian Gangsta<br />
 61 Christian Rap<br />
141 Christian Rock<br />
 32 Classical<br />
  1 Classic Rock<br />
112 Club<br />
128 Club-House<br />
 57 Comedy<br />
140 Contemporary Christian<br />
  2 Country<br />
139 Crossover<br />
 58 Cult</td><td style="border-bottom-width:0px;">
  3 Dance<br />
125 Dance Hall<br />
 50 Darkwave<br />
 22 Death Metal<br />
  4 Disco<br />
 55 Dream<br />
127 Drum &amp; Bass<br />
122 Drum Solo<br />
120 Duet<br />
 98 Easy Listening<br />
 52 Electronic<br />
 48 Ethnic<br />
 54 Eurodance<br />
124 Euro-House<br />
 25 Euro-Techno<br />
 84 Fast Fusion<br />
 80 Folk<br />
115 Folklore<br />
 81 Folk-Rock<br />
119 Freestyle<br />
  5 Funk<br />
 30 Fusion<br />
 36 Game<br />
 59 Gangsta<br />
126 Goa<br />
 38 Gospel<br />
 49 Gothic<br />
 91 Gothic Rock<br />
  6 Grunge<br />
129 Hardcore<br />
 79 Hard Rock<br />
137 Heavy Metal<br />
  7 Hip-Hop<br />
 35 House<br />
100 Humour<br />
131 Indie</td><td style="border-bottom-width:0px;">
 19 Industrial<br />
 33 Instrumental<br />
 46 Instrumental Pop<br />
 47 Instrumental Rock<br />
  8 Jazz<br />
 29 Jazz+Funk<br />
146 JPop<br />
 63 Jungle<br />
 86 Latin<br />
 71 Lo-Fi<br />
 45 Meditative<br />
142 Merengue<br />
  9 Metal<br />
 77 Musical<br />
 82 National Folk<br />
 64 Native US<br />
133 Negerpunk<br />
 10 New Age<br />
 66 New Wave<br />
 39 Noise<br />
 11 Oldies<br />
103 Opera<br />
 12 Other<br />
 75 Polka<br />
134 Polsk Punk<br />
 13 Pop<br />
 53 Pop-Folk<br />
 62 Pop/Funk<br />
109 Porn Groove<br />
117 Power Ballad<br />
 23 Pranks<br />
108 Primus<br />
 92 Progressive Rock<br />
 67 Psychedelic<br />
 93 Psychedelic Rock<br />
 43 Punk<br />
121 Punk Rock</td><td style="border-bottom-width:0px;">
 15 Rap<br />
 68 Rave<br />
 14 R&amp;B<br />
 16 Reggae<br />
 76 Retro<br />
 87 Revival<br />
118 Rhythmic Soul<br />
 17 Rock<br />
 78 Rock &amp; Roll<br />
143 Salsa<br />
114 Samba<br />
110 Satire<br />
 69 Showtunes<br />
 21 Ska<br />
111 Slow Jam<br />
 95 Slow Rock<br />
105 Sonata<br />
 42 Soul<br />
 37 Sound Clip<br />
 24 Soundtrack<br />
 56 Southern Rock<br />
 44 Space<br />
101 Speech<br />
 83 Swing<br />
 94 Symphonic Rock<br />
106 Symphony<br />
147 SynthPop<br />
113 Tango<br />
 18 Techno<br />
 51 Techno-Industrial<br />
130 Terror<br />
144 Thrash Metal<br />
 60 Top 40<br />
 70 Trailer<br />
 31 Trance<br />
 72 Tribal<br />
 27 Trip-Hop<br />
 28 Vocal</td></tr>
</table>

<p class="settingtitle">
<a name="help"><span class="hilight">--help</span></a> Print the command line help.
</p>
<p>
Print the short command line help and exit.
</p>

<p class="settingtitle">
<a name="highpass"><span class="hilight">--highpass number</span></a> Use a highpass filter when encoding
</p>
<p>
Enables a highpass filter of the specified frequency when encoding the source.
Range [0.001..50]kHz or [50..50000]Hz.<br/>
This is usually not required, and the gains are usually minimal. May be useful to
remove an interference signal on 50Hz or 60Hz, or a DC offset. (default: disabled)<br/>
Note: The current implementation has a minimum highpass frequency of (67.5/62)% of the sample rate
(I.e. 481Hz at 44Khz).
</p>

<p class="settingtitle">
<a name="highpass-width"><span class="hilight">--highpass-width width</span></a> Set the width of the decaying curve.
</p>
<p>
Specify the width in Hz of the decaying curve of the highpass. Range [16..50000]Hz<br/>
The minimum (and default) width is 75% of a band's width (which is 1/64th of the sample rate).<br/>
Note: See the remark in the --highpass command above.
</p>

<p class="settingtitle">
<a name="id3v1-only"><span class="hilight">--id3v1-only</span></a> Disable the use of id3v2.
</p>
<p>
Put it before any tag setting.<br/>
This setting tells LAME to use ID3 v1 tag only, and not create an ID3v2 even if
it thinks it should.
</p>

<p class="settingtitle">
<a name="id3v2-only"><span class="hilight">--id3v2-only</span></a> Disable the user ov id3v1.
</p>
<p>
Put it before any tag setting.<br/>
This setting tells LAME to use ID3 v2 tag only. An ID3 v1 tag would not be written.
</p>

<p class="settingtitle">
<a name="ignore-tag-errors"><span class="hilight">--ignore-tag-errors</span></a> Ignore tag information errors
</p>
<p>
Put it before any tag setting.<br/>
This tells lame to ignore the tag information that sees as erroneous and continue
encoding without those. Without this setting, errors are reported and encoding
does not start.
</p>

<p class="settingtitle">
<a name="license"><span class="hilight">--license</span></a> Print the license page
</p>
<p>
Prints version and license information about LAME encoder.
</p>

<p class="settingtitle">
<a name="little-endian"><span class="hilight">--little-endian</span></a> Set the byte order to little-endian.
</p>
<p>
This switch tells LAME that the RAW pcm input is encoded in little-endian. It is
the default setting.
</p>

<p class="settingtitle">
<a name="longhelp"><span class="hilight">--longhelp</span></a> Shows the detailed help 
</p>
<p>
Prints a help page with all of the command-line switches and a brief explanation
of them.
</p>

<p class="settingtitle">
<a name="lowpass"><span class="hilight">--lowpass number</span></a> Use a lowpass filter when encoding
</p>
<p>
Enables a lowpass filter of the specified frequency when encoding the source.
Range [0.001..50]kHz or [50..50000]Hz<br />
</p>
<p>
Using a lowpass filter helps reducing the amount of data to encode. This is
important in MP3 due to a limitation in very high frequencies (>16Khz). The
default value depends on the target bitrate/quality. It is not recommended to
change it as a general basis.
</p>

<p class="settingtitle">
<a name="lowpass-width"><span class="hilight">--lowpass-width width</span></a> Set the width of the decaying curve.
</p>
<p>
Specify the width in Hz of the decaying curve of the lowpass. Range [0.001..16]kHz or [16..50000]Hz <br/>
The lowpass is in the center of this curve. 
The minimum (and default) width is 75% of a band's width (which is 1/64th of the sample rate).
 
</p>

<p class="settingtitle">
<a name="m"><span class="hilight">-m x</span></a> Channel modes
</p>
<table cellpadding="4">
<tr><td width="35px">-m m</td><td>mono</td></tr>
<tr><td>-m l</td><td>get only the left channel of a stereo signal for a mono output</td></tr>
<tr><td>-m r</td><td>get only the right channel of a stereo signal for a mono output</td></tr>
<tr><td>-m s</td><td>forced L/R stereo</td></tr>
<tr><td>-m j /<br/>-m a</td><td>automatic switch between L/R and M/S stereo</td></tr>
<tr><td>-m f</td><td>forced mid/side stereo</td></tr>
<tr><td style="border-bottom:0px;">-m d</td><td style="border-bottom:0px;">dual (independent) channels.</td></tr>
</table>
<p>
MONO is the default mode for mono input files.  If "-m m" is specified
for a stereo input file, the two channels will be averaged into a mono
signal.  (Note: See comments about the -a switch for RAW PCM streams)
</p>
<p>
FORCED L/R STEREO encodes the left and the right signals independently, and
gives more or less bits to each, depending on the currently available. 
</p>
<p>
JOINT STEREO is the default mode of encoding. jstereo means the encoder can use
(on a frame by frame basis) either L/R stereo or mid/side stereo.
In mid/side stereo, the mid (L+R) and side (L-R)
channels are encoded, and more bits are allocated to the mid channel
than the side channel. When there isn't too much stereo separation, this effectively
increases the bandwidth, so having higher quality with the same amount of bits.<br/>
Using mid/side stereo inappropriately can result in audible
compression artifacts.  Too 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 the one
described in the ISO documentation.
</p>
<p>
FORCED MID/SIDE STEREO forces all frames to be encoded with mid/side stereo. It 
should only be used if you are sure every frame of the input file
has very little stereo seperation.
</p>
<p>
DUAL CHANNEL mode is similar to encode the left and right as two mono signals.
Its purpose was meant for Dual language streams where only one of them should be decoded.
Most decoders just decode them as a stereo stream
</p>

<p class="settingtitle">
<a name="mp1input"><span class="hilight">--mp1input</span></a>
 <a name="mp2input"><span class="hilight">--mp2input</span></a>
 <a name="mp3input"><span class="hilight">--mp3input</span></a> MPEG layer I, II or III input file
</p>
<p>
Assume the input file is an MP1/2/3 file.  LAME will decode the input file
before re-encoding it.  Since MP3 is a lossy format, this is 
not recommended in general.  But it is useful for creating low bitrate
mp3s from high bitrate mp3s.  If the filename ends in ".mp3" LAME will assume
it is an MP3.  For stdin or MP3 files which don't end in .mp3 you need
to use this switch.
</p>

<p class="settingtitle">
<a name="noasm"><span class="hilight">--noasm value</span></a> Disables the specified assembler 
instructions
</p>
<p>
When the encoder is compiled with assembler optimizations, this setting allows
to disable them at runtime (They are only enabled if the CPU supports them).
The purpose of this setting is to detect problems in those optimizations, and/or
check the speed difference.
</p>

<p class="settingtitle">
<a name="nogap"><span class="hilight">--nogap file1 file2 [...]</span></a> Encodes multiple continuous
files.
</p>
<p>
Encodes multiple files (ordered by position) which are meant to be played 
gaplessly.
</p>
<p>
By default, LAME will encode the files with accurate length, but the first and
last frame may contain a few erroneous samples for signals that don't fade-in/out
(as is the case of continuous playback).
</p>
<p>
This setting solves that by using the samples from the next/previous file to 
compute the encoding.
</p>

<p class="settingtitle">
<a name="nogapout"><span class="hilight">--nogapout dir</span></a> Specify a directory for the output
of the files encoded with --nogap
</p>
<p>
This setting should precede --nogap, and is used to specify the alternate 
directory where to store the encoded files. The default one is the input file
directory.
</p>

<p class="settingtitle">
<a name="nogaptags"><span class="hilight">--nogaptags</span></a> Enables the use of VBR tags with files
encoded with --nogap
</p>
<p>
Tells LAME to put VBR tags to encoded files if they are encoded in VBR or ABR
modes. Else, using the --nogap option doesn't generate it.
</p>

<p class="settingtitle">
<a name="nohist"><span class="hilight">--nohist</span></a> Disable bitrate Histogram
</p>
<p>
By default, LAME will display a bitrate histogram while producing
VBR mp3 files.  This will disable that feature.
</p>

<p class="settingtitle">
<a name="noreplaygain"><span class="hilight">--noreplaygain</span></a> Disable ReplayGain analysis
</p>
<p>
By default ReplayGain analysis is enabled. This switch disables it.
</p>
<p>
See also: --replaygain-accurate, --replaygain-fast
</p>

<p class="settingtitle">
<a name="o"><span class="hilight">-o</span></a> non-original   
</p>
<p>
Mark the encoded file as a copy
</p>

<p class="settingtitle">
<a name="p"><span class="hilight">-p</span></a> CRC error detection  
</p>
<p>
Turn on CRC error protection.
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.
</p>

<p class="settingtitle">
<a name="pad-id3v2"><span class="hilight">--pad-id3v2</span></a> Pad ID3v2 tag.  
</p>
<p>
Pads the ID3v2 tag with extra 128bytes to allow it to expand. 
</p>

<p class="settingtitle">
<a name="preset"><span class="hilight">--preset x</span></a> Enable one of the presets
</p>
<table cellpadding="4">
<tr><th>Setting</th><th>Meaning</th></tr>
<tr><td>--preset medium</td><td>-V 5 --vbr-old</td></tr>
<tr><td>--preset standard</td><td>-V 2 --vbr-old</td></tr>
<tr><td>--preset extreme</td><td>-V 0 --vbr-old</td></tr>
<tr><td>--preset insane</td><td>-b 320 --vbr-old</td></tr>
<tr><td>--preset fast xxx</td><td>In versions older than LAME 3.98, "fast" was needed to enable the vbr-new routine. It is no longer needed.</td></tr>
<tr><td colspan="2">&nbsp;</td></tr>
<tr><td>--preset number</td><td>--abr number</td></tr>
<tr><td>--preset cbr number</td><td>-b number</td></tr>
<tr><th colspan="2">Old compatibility settings. Meaningless</th></tr>
<tr><td>--preset phone</td><td>-b 16 -m m</td></tr>
<tr><td>--preset phon+ / lw / mw-eu</td><td>-b 24 -m m</td></tr>
<tr><td>--preset mw-us</td><td>-b 40 -m m</td></tr>
<tr><td>--preset voice</td><td>-b 56 -m m</td></tr>
<tr><td>--preset fm / radio</td><td>-b 112</td></tr>
<tr><td>--preset hifi</td><td>-b 160</td></tr>
<tr><td>--preset cd</td><td>-b 192</td></tr>
<tr><td style="border-bottom:0px;">--preset studio</td><td style="border-bottom:0px;">-b 256</td></tr>
</table>

<p class="settingtitle">
<a name="priority"><span class="hilight">--priority value</span></a> Set process priority
</p>
<p>
(Windows and OS/2 only)
</p>
<p>
Sets the process priority for LAME while running under Windows and/or IBM OS/2.
This can be very useful to avoid the system becoming slow and/or
unresponsive. By setting LAME to run in a lower priority, you leave
more time for the system to update basic processing (drawing windows,
polling keyboard/mouse, etc). The impact in LAME's performance is 
minimal if you use priority 0 to 2.
</p>
<p>
The valid parameters are:
</p>
<table cellpadding="4">
<thead>
<tr><th></th><th>Priority</th><th>Windows</th><th>OS/2</th></tr>
</thead>
<tbody>
<tr><td>0</td><td>Low priority </td><td>IDLE_PRIORITY_CLASS</td><td>IDLE, delta = 0</td></tr>
<tr><td>1</td><td>Medium priority </td><td>IDLE_PRIORITY_CLASS</td><td>IDLE, delta = +31</td></tr>
<tr><td>2</td><td>Regular priority </td><td>NORMAL_PRIORITY_CLASS</td><td>REGULAR, delta = -31</td></tr>
<tr><td>3</td><td>High priority </td><td>HIGH_PRIORITY_CLASS</td><td>REGULAR, delta = 0</td></tr>
<tr><td style="border-bottom:0px;">4</td><td style="border-bottom:0px;">Maximum priority </td><td style="border-bottom:0px;">HIGH_PRIORITY_CLASS</td><td style="border-bottom:0px;">REGULAR, delta = +31</td></tr>
</tbody>
</table>
<p>
Note that if you call '--priority' without a parameter, then 
priority 0 will be assumed.
</p>

<p class="settingtitle">
<a name="q"><span class="hilight">-q n</span></a> Algorithm quality selection
</p>  
<p>
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 coding (noise shaping).
</p>
<p>
For CBR, ABR and --vbr-old modes, the following table applies
</p>
<table cellpadding="4">
<tr><td>-q 0</td><td>Use the best algorithms (Best Huffman coding search, full outer loop, and the highest precision of several parameters).</td></tr>
<tr><td>-q 1 to -q 4</td><td>Similar to -q 0 without the full outer loop and with decreasing precision of parameters the further from -q 0. -q 3 is the default</td></tr>
<tr><td>-q 5 and -q 6</td><td>Same as -q 7, but enables noise shaping and increases subblock gain</td></tr>
<tr><td style="border-bottom:0px;">-q 7 to -q 9</td><td style="border-bottom:0px;">Same as -f. Very fast, OK quality. Psychoacoustics are used for pre-echo and mid/side stereo, but no noise-shaping is done.</td></tr>
</table>
<p>
For the default VBR mode since LAME 3.98, the following table applies
</p>
<table cellpadding="4">
<tr><td>-q 0 to -q 4</td><td>include all features of the other modes and additionally use the best search when applying Huffman coding.</td></tr>
<tr><td>-q 5 and -q 6</td><td> include all features of -q7, calculate and consider actual quantisation noise, and additionally enable subblock gain.</td></tr>
<tr><td style="border-bottom:0px;">-q 7 to -q 9</td><td style="border-bottom:0px;">This level uses a psymodel but does not calculate quantisation noise when encoding: it takes a quick guess.</td></tr>
</table>
<p class="settingtitle">
<a name="quiet"><span class="hilight">--quiet</span></a> Don't print any output in the screen  
</p>
<p>
Disables the screen output. Useful for job processing or other times where
screen output is undesirable.
</p>

<p class="settingtitle">
<a name="r"><span class="hilight">-r</span></a> Input file is raw pcm  
</p>
<p>
Assume the input file is raw pcm.  Sampling rate and mono/stereo
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.
</p>
<p>
Not supported if LAME is compiled to use LIBSNDFILE.
</p>

<p class="settingtitle">
<a name="replaygain-accurate"><span class="hilight">--replaygain-accurate</span></a> Slightly more accurate
 ReplayGain analysis and finding the peak sample
</p>
<p>
Compute "Radio" ReplayGain on the decoded data stream. Find the peak sample
by decoding on the fly the encoded data stream and store it in the file.
</p>
<p>
ReplayGain analysis does _not_ 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 http://www.replaygain.org/
</p>
<p>
By default, LAME performs ReplayGain analysis on the input data (after
the user-specified volume scaling). This behaviour 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:
</p>
<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>
<p>    
Only the "Radio" 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.
</p>
<p>
This option is not usable if the MP3 decoder was _explicitly_ 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).
</p>
<p>
See also: --replaygain-fast, --noreplaygain, --clipdetect
</p>

<p class="settingtitle">
<a name="replaygain-fast"><span class="hilight">--replaygain-fast</span></a> Fast ReplayGain analysis
</p>
<p>
Compute "Radio" ReplayGain of the input data stream after user-specified
volume scaling and/or resampling.
</p>
<p>
ReplayGain analysis does _not_ 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 http://www.replaygain.org/
</p>
<p>
Only the "Radio" 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.
</p>
<p>
This switch is enabled by default.
</p>
<p>
See also: --replaygain-accurate, --noreplaygain
</p>

<p class="settingtitle">
<a name="resample"><span class="hilight">--resample  n</span></a> Output sampling frequency in kHz
</p>
<p>
where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48
</p>
<p>
Output sampling frequency.  Resample the input if necessary.  
</p>
<p>
If not specified, LAME may sometimes resample automatically 
when faced with extreme compression conditions (like encoding
a 44.1 kHz input file at 32 kbps).  To disable this automatic
resampling, you have to use --resample to set the output sample rate
equal to the input sample rate.  In that case, LAME will not
perform any extra computations.
</p>

<p class="settingtitle">
<a name="s"><span class="hilight">-s  n</span></a> Sampling frequency in kHz
</p>
<p>
where n = sampling rate in kHz.
</p>
<p>
Required for raw PCM input files.  Otherwise it will be determined
from the header information in the input file.
</p>
<p>
LAME will automatically resample the input file to one of the
supported MP3 sample rates if necessary.
</p>

<p class="settingtitle">
<a name="scale"><span class="hilight">--scale arg</span></a>
<a name="scale-l"><span class="hilight">--scale-l arg</span></a>
<a name="scale-r"><span class="hilight">--scale-r arg</span></a> Scale the amplitude of the input file
</p>
<p>
Scales input by arg.  This just multiplies the PCM data
(after it has been converted to floating point) by arg.
</p>
<p>
arg &gt; 1:  increase volume<br/>
arg = 1:  no effect<br/>
arg &lt; 1:  reduce volume<br/>
</p>
<p>
Use with care, since most MP3 decoders will truncate data
which decodes to values greater than 32768.  
</p>
<p>
The <i>l</i> and <i>r</i> variants apply the scaling only to left (channel 0) or right
(channel 1) respectively.
</p>

<p class="settingtitle">
<a name="signed"><span class="hilight">--signed</span></a> Input RAW uses signed values   
</p>
<p>
Use with RAW pcm to indicate if the data is signed (values centered at zero) or
unsigned (all positive values). This is the default value.
</p>

<p class="settingtitle">
<a name="silent"><span class="hilight">--silent</span></a> Don't print any output in the screen  
</p>
<p>
Disables the screen output. Useful for job processing or other times where
screen output is undesirable.
</p>

<p class="settingtitle">
<a name="space-id3v1"><span class="hilight">--space-id3v1</span></a> Use spaces for padding.  
</p>
<p>
Use spaces instead of null values for padding data in an ID3v1 tag.
</p>

<p class="settingtitle">
<a name="strictly-enforce-ISO"><span class="hilight">--strictly-enforce-ISO</span></a> Strict ISO compliance   
</p>
<p>
With this option, LAME will enforce the 7680 bit limitation on
total frame size. This results in many wasted bits for
high bitrate encodings. Some decoders need it.
</p>

<p class="settingtitle">
<a name="t"><span class="hilight">-t</span></a> Disable VBR tag or disable WAV header            
</p>
<p>This setting has two different uses:</p>
<p>
When encoding to VBR, this setting disables writing the VBR Tag (also known as XING tag).
This tag is embedded by default in the frame 0 of the MP3 file. It lets VBR aware players 
to correctly seek and compute playing times in such files.<br/>
</p>
<p>
When decoding MP3 to WAV using --decode, this flag will 
disable writing the WAV header.  The output will be raw pcm,
native endian format.  Use -x to swap bytes.
</p>

<p class="settingtitle">
<a name="T"><span class="hilight">-T</span></a> Force the usage of the LAME tag            
</p>
<p>
By default, LAME already adds a LAME tag when encoding.
This setting overrides the -t setting, and implies --nogaptags.
</p>

<p class="settingtitle">
<a name="unsigned"><span class="hilight">--unsigned</span></a> Input RAW uses unsigned values   
</p>
<p>
Use with RAW pcm to indicate if the data is signed (values centered at zero) or
unsigned (all positive values).
</p>

<p class="settingtitle">
<a name="usage"><span class="hilight">--usage</span></a> Print the command line help.
</p>
<p>
Print the short command line help and exit.
</p>

<p class="settingtitle">
<a name="V"><span class="hilight">-V n</span></a> Enable VBR encoding
</p>       
<p>
Encodes using the VBR algorithm, at the indicated quality.<br/>
0=highest quality, bigger files. 9.999=lowest quality, smaller files. Decimal values can be specified, like: 4.51<br/>
On average, the resulting bitrates are as follows:
<table cellpadding="4">
<thead>
<tr><th>Setting</th><th>Average bitrate (kbps)</th></tr>
</thead>
<tbody>
<tr><td align="right">0</td><td align="right">245</td></tr>
<tr><td align="right">2</td><td align="right">190</td></tr>
<tr><td align="right">3</td><td align="right">175</td></tr>
<tr><td align="right">4</td><td align="right">165</td></tr>
<tr><td style="border-bottom:0px;" align="right">5</td><td style="border-bottom:0px;" align="right">130</td></tr>
</tbody>
</table>

</p>
<p>
Using -V 7 or higher (lower quality) is not recommended.  
ABR usually produces better results.  
</p>

<p class="settingtitle">
<a name="vbr-old"><span class="hilight">--vbr-old</span></a> Uses the old VBR method of encoding
</p>
<p>
Tells LAME to encode to VBR using the old (slower) method of encoding.
</p>

<p class="settingtitle">
<a name="verbose"><span class="hilight">--verbose</span></a>Program verbosity
</p>
<p>
Print a lot of information on screen. 
</p>

<p class="settingtitle">
<a name="version"><span class="hilight">--version</span></a> Prints the license page
</p>
<p>
Prints version and license information about LAME encoder.
</p>

<p class="settingtitle">
<a name="x"><span class="hilight">-x</span></a> Swap input bytes
</p>
<p>
swap bytes in the input file (and output file when using --decode).
For sorting out little endian/big endian type problems.  If your
encodings sound like static, try this first.
</p>

<p class="settingtitle">
<a name="Y"><span class="hilight">-Y</span></a> Ignore noise in sbf21, like CBR mode does
</p>
<p>
Allows -V2, -V1 and -V0 to not encode the highest frequencies accurately,
if doing so causes disproportional increases in bitrate. This is the same that
CBR and ABR modes do.<br/>
Due to the design of the MP3 format, to keep precision in the last scalefactor
band, an encoder needs to increase the precision in all the bands (not only in
this one).<br/>
The consequence is an increase of bitrate (+60kbps in some cases) compared to
not keeping that precision. Generally, this band should allow for distortions,
so using this switch shouldn't cause harm.
</p>

<p>&nbsp;
</p>

<div id="notes">
<ol>
<li><a name="note1"></a>If compiled with decoding capabilities</li>
<li><a name="note2"></a>HIP stands for Hip Isn't a Player and is based off of
 Michael Hipp's <a href="http://www.mpg123.de">mpglib 0.2a</a></li>

</ol>
</div>

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