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diff --git a/lib/liblame/doc/html/switchs.html b/lib/liblame/doc/html/switchs.html new file mode 100644 index 0000000000..d22521c99d --- /dev/null +++ b/lib/liblame/doc/html/switchs.html @@ -0,0 +1,1133 @@ +<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 3.2//EN"> +<HTML> +<HEAD> +<TITLE>Full command line switch reference</TITLE> +<META NAME="description" CONTENT="Command line switch reference"> +<META NAME="keywords" CONTENT="lame"> +<META NAME="resource-type" CONTENT="document"> +<META NAME="distribution" CONTENT="global"> +<META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso_8859_1"> +<LINK REL="STYLESHEET" HREF="lame.css"> +</HEAD> +<BODY TEXT=#000000 + BGCOLOR=#F9FBFB LINK=#006666 VLINK=#4C4C4C + ALINK=#995500> +<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="#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="#-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="#-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="#-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="#-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="#-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, integer or floating point number [0,...,10[</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>-b n</kbd><a name="b"> bitrate</a></strong> + </dt> +</dl> +<dl> + <dd>For MPEG-1 (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 MPEG-2 (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> + For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)<br> + n = 8,16,24,32,40,48,56,64<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 MPEG-1 (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 MPEG-2 (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> + For MPEG-2.5 (sampling frequencies of 8, 11.025 and 12 kHz)<br> + n = 8,16,24,32,40,48,56,64<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>--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>--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>--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>--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>--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>--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,...,10[</kbd><a name="V"> VBR quality + setting, integer or floating point number</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> |