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15<H1>Full command line switch reference</H1>
16<P> <font size="-1">note: Options which could exist without being documented
17 here are considered as experimental ones. Such experimental options should usually
18 not be used.</font>
19<P>
20<TABLE CELLPADDING=3 BORDER="1">
21 <TR VALIGN="TOP">
22 <TD ALIGN="LEFT" nowrap><b>switch</b></TD>
23 <TD ALIGN="LEFT" nowrap><b>parameter</b></TD>
24 </TR>
25 <tr valign="TOP">
26 <td align="LEFT" nowrap><kbd><a href="#a">-a</a></kbd></td>
27 <td align="LEFT" nowrap>downmix stereo file to mono</td>
28 </tr>
29 <tr valign="TOP">
30 <td align="LEFT" nowrap><kbd><a href="#-abr">--abr</a></kbd></td>
31 <td align="LEFT" nowrap>average bitrate encoding</td>
32 </tr>
33 <tr valign="TOP">
34 <td align="LEFT" nowrap><kbd><a href="#-allshort">--allshort</a></kbd></td>
35 <td align="LEFT" nowrap>use short blocks only</td>
36 </tr>
37 <tr valign="TOP">
38 <td align="LEFT" nowrap><kbd><a href="#-athlower">--athlower</a></kbd></td>
39 <td align="LEFT" nowrap>lower the ATH</td>
40 </tr>
41 <tr valign="TOP">
42 <td align="LEFT" nowrap><kbd><a href="#-athonly">--athonly</a></kbd></td>
43 <td align="LEFT" nowrap>ATH only</td>
44 </tr>
45 <tr valign="TOP">
46 <td align="LEFT" nowrap><kbd><a href="#-athshort">--athshort</a></kbd></td>
47 <td align="LEFT" nowrap>ATH only for short blocks</td>
48 </tr>
49 <tr valign="TOP">
50 <td align="LEFT" nowrap><kbd><a href="#-athtype">--athtype</a></kbd></td>
51 <td align="LEFT" nowrap>select ATH type</td>
52 </tr>
53 <tr valign="TOP">
54 <td align="LEFT" nowrap><kbd><a href="#b">-b</a></kbd></td>
55 <td align="LEFT" nowrap>bitrate (8...320)</td>
56 </tr>
57 <tr valign="TOP">
58 <td align="LEFT" nowrap><kbd><a href="#Bmax">-B</a></kbd></td>
59 <td align="LEFT" nowrap>max VBR/ABR bitrate (8...320)</td>
60 </tr>
61 <tr valign="TOP">
62 <td align="LEFT" nowrap><kbd><a href="#-bitwidth">--bitwidth</a></kbd></td>
63 <td align="LEFT" nowrap>input bit width</td>
64 </tr>
65 <tr valign="TOP">
66 <td align="LEFT" nowrap><kbd><a href="#c">-c</a></kbd></td>
67 <td align="LEFT" nowrap>copyright</td>
68 </tr>
69 <tr valign="TOP">
70 <td align="LEFT" nowrap><kbd><a href="#-cbr">--cbr</a></kbd></td>
71 <td align="LEFT" nowrap>enforce use of constant bitrate</td>
72 </tr>
73 <tr valign="TOP">
74 <td align="LEFT" nowrap><kbd><a href="#-clipdetect">--clipdetect</a></kbd></td>
75 <td align="LEFT" nowrap>clipping detection</td>
76 </tr>
77 <tr valign="TOP">
78 <td align="LEFT" nowrap><kbd><a href="#-comp">--comp</a></kbd></td>
79 <td align="LEFT" nowrap>choose compression ratio</td>
80 </tr>
81 <tr valign="TOP">
82 <td align="LEFT" nowrap><kbd><a href="#-cwlimit">--cwlimit</a></kbd></td>
83 <td align="LEFT" nowrap>tonality limit</td>
84 </tr>
85 <tr valign="TOP">
86 <td align="LEFT" nowrap><kbd><a href="#d">-d</a></kbd></td>
87 <td align="LEFT" nowrap>block type control</td>
88 </tr>
89 <tr valign="TOP">
90 <td align="LEFT" nowrap><kbd><a href="#-decode">--decode</a></kbd></td>
91 <td align="LEFT" nowrap>decoding only</td>
92 </tr>
93 <tr valign="TOP">
94 <td align="LEFT" nowrap><kbd><a href="#-disptime">--disptime</a></kbd></td>
95 <td align="LEFT" nowrap>time between display updates</td>
96 </tr>
97 <tr valign="TOP">
98 <td align="LEFT" nowrap><kbd><a href="#e">-e</a></kbd></td>
99 <td align="LEFT" nowrap>de-emphasis (<b>n</b>, 5, c)</td>
100 </tr>
101 <tr valign="TOP">
102 <td align="LEFT" nowrap><kbd><a href="#f">-f</a></kbd></td>
103 <td align="LEFT" nowrap> fast mode</td>
104 </tr>
105 <tr valign="TOP">
106 <td align="LEFT" nowrap><kbd><a href="#FF">-F</a></kbd></td>
107 <td align="LEFT" nowrap> strictly enforce the -b option</td>
108 </tr>
109 <tr valign="TOP">
110 <td align="LEFT" nowrap><kbd><a href="#-freeformat">--freeformat</a></kbd></td>
111 <td align="LEFT" nowrap> free format bitstream</td>
112 </tr>
113 <tr valign="TOP">
114 <td align="LEFT" nowrap><kbd><a href="#h">-h</a></kbd></td>
115 <td align="LEFT" nowrap>high quality</td>
116 </tr>
117 <tr valign="TOP">
118 <td align="LEFT" nowrap><kbd><a href="#-help">--help</a></kbd></td>
119 <td align="LEFT" nowrap> help</td>
120 </tr>
121 <tr valign="TOP">
122 <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass</a></kbd></td>
123 <td align="LEFT" nowrap> highpass filtering frequency in kHz</td>
124 </tr>
125 <tr valign="TOP">
126 <td align="LEFT" nowrap><kbd><a href="#-highpass">--highpass-width</a></kbd></td>
127 <td align="LEFT" nowrap> width of highpass filtering in kHz</td>
128 </tr>
129 <tr valign="TOP">
130 <td align="LEFT" nowrap><kbd><a href="#k">-k</a></kbd></td>
131 <td align="LEFT" nowrap> full bandwidth</td>
132 </tr>
133 <tr valign="TOP">
134 <td align="LEFT" nowrap><kbd><a href="#-lowpass">--lowpass</a></kbd></td>
135 <td align="LEFT" nowrap> lowpass filtering frequency in kHz</td>
136 </tr>
137 <tr valign="TOP">
138 <td align="LEFT" nowrap><kbd><a href="#-lowpass-width">--lowpass-width</a></kbd></td>
139 <td align="LEFT" nowrap> width of lowpass filtering in kHz</td>
140 </tr>
141 <tr valign="TOP">
142 <td align="LEFT" nowrap><kbd><a href="#m">-m</a></kbd></td>
143 <td align="LEFT" nowrap>stereo mode (s, <b>j</b>, f, m)</td>
144 </tr>
145 <tr valign="TOP">
146 <td align="LEFT" nowrap><kbd><a href="#-mp1input">--mp1input</a></kbd></td>
147 <td align="LEFT" nowrap>MPEG Layer I input file</td>
148 </tr>
149 <tr valign="TOP">
150 <td align="LEFT" nowrap><kbd><a href="#-mp2input">--mp2input</a></kbd></td>
151 <td align="LEFT" nowrap>MPEG Layer II input file</td>
152 </tr>
153 <tr valign="TOP">
154 <td align="LEFT" nowrap><kbd><a href="#-mp3input">--mp3input</a></kbd></td>
155 <td align="LEFT" nowrap>MPEG Layer III input file</td>
156 </tr>
157 <tr valign="TOP">
158 <td align="LEFT" nowrap><kbd><a href="#-noath">--noath</a></kbd></td>
159 <td align="LEFT" nowrap>disable ATH</td>
160 </tr>
161 <tr valign="TOP">
162 <td align="LEFT" nowrap><kbd><a href="#-noasm">--noasm</a></kbd></td>
163 <td align="LEFT" nowrap>disable assembly optimizations (mmx/3dnow/sse)</td>
164 </tr>
165 <tr valign="TOP">
166 <td align="LEFT" nowrap><kbd><a href="#-nohist">--nohist</a></kbd></td>
167 <td align="LEFT" nowrap>disable histogram display</td>
168 </tr>
169 <tr valign="TOP">
170 <td align="LEFT" nowrap><kbd><a href="#-noreplaygain">--noreplaygain</a></kbd></td>
171 <td align="LEFT" nowrap>disable ReplayGain analysis</td>
172 </tr>
173 <tr valign="TOP">
174 <td align="LEFT" nowrap><kbd><a href="#-nores">--nores</a></kbd></td>
175 <td align="LEFT" nowrap>disable bit reservoir</td>
176 </tr>
177 <tr valign="TOP">
178 <td align="LEFT" nowrap><kbd><a href="#-noshort">--noshort</a></kbd></td>
179 <td align="LEFT" nowrap>disable short blocks frames</td>
180 </tr>
181 <tr valign="TOP">
182 <td align="LEFT" nowrap><kbd><a href="#-notemp">--notemp</a></kbd></td>
183 <td align="LEFT" nowrap>disable temporal masking</td>
184 </tr>
185 <TR VALIGN="TOP">
186 <TD ALIGN="LEFT" nowrap><kbd><a href="#o">-o</a></kbd></TD>
187 <TD ALIGN="LEFT" nowrap>non-original</TD>
188 </TR>
189 <tr valign="TOP">
190 <td align="LEFT" nowrap><kbd><a href="#p">-p</a></kbd></td>
191 <td align="LEFT" nowrap>error protection</td>
192 </tr>
193 <tr valign="TOP">
194 <td align="LEFT" nowrap><kbd><a href="#-preset">--preset</a></kbd></td>
195 <td align="LEFT" nowrap>use built-in preset</td>
196 </tr>
197 <tr valign="TOP">
198 <td align="LEFT" nowrap><kbd><a href="#-priority">--priority</a></kbd></td>
199 <td align="LEFT" nowrap>OS/2 process priority control</td>
200 </tr>
201 <tr valign="TOP">
202 <td align="LEFT" nowrap><kbd><a href="#q">-q</a></kbd></td>
203 <td align="LEFT" nowrap>algorithm quality selection</td>
204 </tr>
205 <tr valign="TOP">
206 <td align="LEFT" nowrap><kbd><a href="#-silent">--quiet</a></kbd></td>
207 <td align="LEFT" nowrap>silent operation</td>
208 </tr>
209 <tr valign="TOP">
210 <td align="LEFT" nowrap><kbd><a href="#r">-r</a></kbd></td>
211 <td align="LEFT" nowrap>input file is raw PCM</td>
212 </tr>
213 <tr valign="TOP">
214 <td align="LEFT" nowrap><kbd><a href="#-replaygain-accurate">--replaygain-accurate</a></kbd></td>
215 <td align="LEFT" nowrap>compute ReplayGain more accurately and find the peak sample</td>
216 </tr>
217 <tr valign="TOP">
218 <td align="LEFT" nowrap><kbd><a href="#-replaygain-fast">--replaygain-fast</a></kbd></td>
219 <td align="LEFT" nowrap>compute ReplayGain fast but slightly inaccurately (default)</td>
220 </tr>
221 <tr valign="TOP">
222 <td align="LEFT" nowrap><kbd><a href="#-resample">--resample</a></kbd></td>
223 <td align="LEFT" nowrap>output sampling frequency in kHz (encoding only)</td>
224 </tr>
225 <TR VALIGN="TOP">
226 <TD ALIGN="LEFT" nowrap><kbd><a href="#s">-s</a></kbd></TD>
227 <TD ALIGN="LEFT" nowrap>sampling frequency in kHz</TD>
228 </TR>
229 <tr valign="TOP">
230 <td align="LEFT" nowrap><kbd><a href="#-silent">-S</a></kbd></td>
231 <td align="LEFT" nowrap>silent operation</td>
232 </tr>
233 <tr valign="TOP">
234 <td align="LEFT" nowrap><kbd><a href="#-scale">--scale</a></kbd></td>
235 <td align="LEFT" nowrap>scale input</td>
236 </tr>
237 <tr valign="TOP">
238 <td align="LEFT" nowrap><kbd><a href="#-scale-l">--scale-l</a></kbd></td>
239 <td align="LEFT" nowrap>scale input channel 0 (left)</td>
240 </tr>
241 <tr valign="TOP">
242 <td align="LEFT" nowrap><kbd><a href="#-scale-r">--scale-r</a></kbd></td>
243 <td align="LEFT" nowrap>scale input channel 1 (right)</td>
244 </tr>
245 <tr valign="TOP">
246 <td align="LEFT" nowrap><kbd><a href="#-short">--short</a></kbd></td>
247 <td align="LEFT" nowrap>use short blocks</td>
248 </tr>
249 <tr valign="TOP">
250 <td align="LEFT" nowrap><kbd><a href="#-silent">--silent</a></kbd></td>
251 <td align="LEFT" nowrap>silent operation</td>
252 </tr>
253 <tr valign="TOP">
254 <td align="LEFT" nowrap><kbd><a href="#-strictly-enforce-ISO">--strictly-enforce-ISO</a></kbd></td>
255 <td align="LEFT" nowrap>strict ISO compliance</td>
256 </tr>
257 <tr valign="TOP">
258 <td align="LEFT" nowrap><kbd><a href="#t">-t</a></kbd></td>
259 <td align="LEFT" nowrap>disable INFO/WAV header</td>
260 </tr>
261 <tr valign="TOP">
262 <td align="LEFT" nowrap><kbd><a href="#V">-V</a></kbd></td>
263 <td align="LEFT" nowrap>VBR quality setting (0...9)</td>
264 </tr>
265 <tr valign="TOP">
266 <td align="LEFT" nowrap><kbd><a href="#-vbr-new">--vbr-new</a></kbd></td>
267 <td align="LEFT" nowrap>new VBR mode</td>
268 </tr>
269 <tr valign="TOP">
270 <td align="LEFT" nowrap><kbd><a href="#-vbr-old">--vbr-old</a></kbd></td>
271 <td align="LEFT" nowrap>older VBR mode</td>
272 </tr>
273 <tr valign="TOP">
274 <td align="LEFT" nowrap><kbd><a href="#-verbose">--verbose</a></kbd></td>
275 <td align="LEFT" nowrap>verbosity</td>
276 </tr>
277 <tr valign="TOP">
278 <td align="LEFT" nowrap><kbd><a href="#x">-x</a></kbd></td>
279 <td align="LEFT" nowrap>swapbytes</td>
280 </tr>
281 <tr valign="TOP">
282 <td align="LEFT" nowrap><kbd><a href="#Xquant">-X</a></kbd></td>
283 <td align="LEFT" nowrap>change quality measure</td>
284 </tr>
285</TABLE>
286<BR>
287<dl>
288 <dt><strong>* <kbd>-a</kbd><a name="a">&nbsp;&nbsp;&nbsp;&nbsp;downmix&#160;</a></strong>
289 <dd>Mix the stereo input file to mono and encode as mono.<br>
290 The downmix is calculated as the sum of the left and right channel, attenuated
291 by 6 dB. <br>
292 <br>
293 This option is only needed in the case of raw PCM stereo input (because LAME
294 cannot determine the number of channels in the input file).<br>
295 To encode a stereo PCM input file as mono, use "lame -m s -a".<br>
296 <br>
297 For WAV and AIFF input files, using "-m m" will always produce a mono .mp3
298 file from both mono and stereo input.
299 <dt><br>
300 </dt>
301 <hr width="50%" noshade align="center">
302 <br>
303</dl>
304<dl>
305 <dt><strong>* <kbd>--abr n</kbd><a name="-abr">&nbsp;&nbsp;&nbsp;&nbsp;average
306 bitrate encoding</a></strong> </dt>
307</dl>
308<dl>
309 <dd>Turns on encoding with a targeted average bitrate of n kbits, allowing to
310 use frames of different sizes. The allowed range of n is 8-310, you can use
311 any integer value within that range.<br>
312 <br>
313 It can be combined with the -b and -B switches like:<br>
314 lame --abr 123 -b 64 -B 192 a.wav a.mp3<br>
315 which would limit the allowed frame sizes between 64 and 192 kbits. <br>
316 <br>
317 </dt>
318 <hr width="50%" noshade align="center">
319 <br>
320</dl>
321<dl>
322 <dt><strong>* <kbd>--allshort</kbd><a name="-allshort">&nbsp;&nbsp;&nbsp;&nbsp;use
323 short blocks only</a></strong> </dt>
324</dl>
325<dl>
326 <dd>Use only short blocks, no long ones.
327</dl>
328<dl>
329 <dd>&nbsp;
330 <dt><br>
331 </dt>
332 <hr width="50%" noshade align="center">
333 <br>
334</dl>
335<dl>
336 <dt><strong>* <kbd>--athlower n</kbd><a name="-athlower">&nbsp;&nbsp;&nbsp;&nbsp;lower
337 the ATH</a></strong> </dt>
338</dl>
339<dl>
340 <dd>Lower the ATH (absolute threshold of hearing) by n dB.<br>
341 Normally, humans are unable to hear any sound below this threshold, but for
342 music recorded at very low level this option might be useful.
343</dl>
344<dl>
345 <dd>&nbsp;
346 <dt><br>
347 </dt>
348 <hr width="50%" noshade align="center">
349 <br>
350</dl>
351<dl>
352 <dt><strong>* <kbd>--athonly</kbd><a name="-athonly">&nbsp;&nbsp;&nbsp;&nbsp;ATH
353 only</a></strong> </dt>
354</dl>
355<dl>
356 <dd>This option causes LAME to ignore the output of the psy-model and only use
357 masking from the ATH (absolute threshold of hearing). Might be useful at very
358 high bitrates or for testing the ATH.
359</dl>
360<dl>
361 <dd>&nbsp;
362 <dt><br>
363 </dt>
364 <hr width="50%" noshade align="center">
365 <br>
366</dl>
367<dl>
368 <dt><strong>* <kbd>--athshort</kbd><a name="-athshort">&nbsp;&nbsp;&nbsp;&nbsp;ATH
369 only for short blocks</a></strong> </dt>
370</dl>
371<dl>
372 <dd>Ignore psychoacoustic model for short blocks, use ATH only.
373</dl>
374<dl>
375 <dd>&nbsp;
376 <dt><br>
377 </dt>
378 <hr width="50%" noshade align="center">
379 <br>
380</dl>
381<dl>
382 <dt><strong>* <kbd>--athtype 0/1/2</kbd><a name="-athtype">&nbsp;&nbsp;&nbsp;&nbsp;select
383 ATH type</a></strong> </dt>
384</dl>
385<dl>
386 <dd>The Absolute Threshold of Hearing is the minimum threshold under which humans
387 are unable to hear any sound. In the past, LAME was using ATH shape 0 which
388 is the Painter & Spanias formula. Tests have shown that this formula is innacurate
389 for the 13-22 kHz area, leading to audible artifacts in some cases. Shape 1
390 was thus implemented, which is over sensitive, leading to very high bitrates.
391 Shape 2 formula was accurately modelized from real data in order to real optimal
392 quality while not wasting bitrate. In CBR and ABR modes, LAME uses ATH shape
393 2 by default. <br>
394 <br>
395 In VBR mode, LAME is adapting its shape according to the
396 -V value, going gradually from the 0 shape at -V9 up to shape 2 at -V0.
397</dl>
398<dl>
399 <dd>&nbsp;
400 <dt><br>
401 </dt>
402 <hr width="50%" noshade align="center">
403 <br>
404</dl>
405<dl>
406 <dt><strong>* <kbd>-b n</kbd><a name="b">&nbsp;&nbsp;&nbsp;&nbsp;bitrate</a></strong>
407 </dt>
408</dl>
409<dl>
410 <dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
411 n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
412 <br>
413 For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
414 n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
415 <br>
416 Default is 128 kbps for MPEG1 and 64 kbps for MPEG2. <br>
417 <br>
418 When used with variable bitrate encoding (VBR), -b specifies the minimum bitrate
419 to be used. However, in order to avoid wasted space, the smallest frame size
420 available will be used during silences.
421 <dt><br>
422 </dt>
423 <hr width="50%" noshade align="center">
424 <br>
425</dl>
426<dl>
427 <dt><strong>* <kbd>-B n</kbd><a name="Bmax">&nbsp;&nbsp;&nbsp;&nbsp;maximum
428 VBR/ABR bitrate&nbsp;</a></strong> </dt>
429</dl>
430<dl>
431 <dd>For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)<br>
432 n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320<br>
433 <br>
434 For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)<br>
435 n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160<br>
436 <br>
437 Specifies the maximum allowed bitrate when using VBR/ABR <br>
438 <br>
439 The use of -B is NOT RECOMMENDED. A 128kbps CBR bitstream, because of the bit reservoir,
440 can actually have frames which use as many bits as a 320kbps frame. VBR modes
441 minimize the use of the bit reservoir, and thus need to allow 320kbps frames
442 to get the same flexibility as CBR streams.<br>
443 <br>
444 <i>note: If you own an mp3 hardware player build upon a MAS 3503 chip, you
445 must set maximum bitrate to no more than 224 kpbs.</i> <br>
446</dl>
447<dl>
448 <dt><strong>* <kbd>--bitwidth 8/16/24/32</kbd><a name="-bitwidth">&nbsp;&nbsp;&nbsp;&nbsp;input
449 bit width&nbsp;</a></strong> </dt>
450</dl>
451<dl>
452 <dd> Required only for raw PCM input files. Otherwise it will be determined
453 from the header of the input file. <br>
454</dl>
455<dl>
456 <hr width="50%" noshade align="center">
457 <br>
458 <dl> </dl>
459 <dt><strong>* <kbd>--clipdetect</kbd><a name="-clipdetect">&nbsp;&nbsp;&nbsp;&nbsp;clipping detection</a></strong>
460 </dt>
461</dl>
462<dl>
463 <dd>
464 Enable --replaygain-accurate and print a message whether clipping
465 occurs and how far in dB the waveform is from full scale.<br>
466 <br>
467 This option is not usable if the MP3 decoder was <b>explicitly</b>
468 disabled in the build of LAME.<br>
469 <br>
470 See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>
471 <dt><br>
472 <br>
473 <hr width="50%" noshade align="center">
474 <br>
475 <dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
476 &nbsp;&nbsp;&nbsp;&nbsp;enforce use of constant bitrate</a></strong>
477 </dt>
478</dl>
479<dl>
480 <dd>This switch enforces the use of constant bitrate encoding.
481 <dt><br>
482 <br>
483 <hr width="50%" noshade align="center">
484 <br>
485 <dt><strong>* <kbd>--cbr</kbd><a name="-cbr">
486 &nbsp;&nbsp;&nbsp;&nbsp;enforce use of constant bitrate</a></strong>
487 </dt>
488</dl>
489<dl>
490 <dd>This switch enforces the use of constant bitrate encoding.
491 <dt><br>
492 <br>
493 <hr width="50%" noshade align="center">
494 <br>
495 <dt><strong>* <kbd>--comp</kbd><a name="-comp">&nbsp;&nbsp;&nbsp;&nbsp;choose
496 compression ratio</a></strong> </dt>
497</dl>
498<dl>
499 <dd>Instead of choosing bitrate, using this option, user can choose compression
500 ratio to achieve.
501 <dt><br>
502 <br>
503 <hr width="50%" noshade align="center">
504 <br>
505 <dt><strong>* <kbd>--cwlimit n</kbd><a name="-cwlimit">&nbsp;&nbsp;&nbsp;tonality
506 limit</a></strong> </dt>
507</dl>
508<dl>
509 <dd>Compute tonality up to freq (in kHz). Default setting is 8.8717.
510 <dt><br>
511 <br>
512 <hr width="50%" noshade align="center">
513 <br>
514 <dt><strong>* <kbd>-d</kbd><a name="d">&nbsp;&nbsp;&nbsp;&nbsp;block type control</a></strong>
515 </dt>
516</dl>
517<dl>
518 <dd>Allows the left and right channels to use different block size types.
519 <dt><br>
520 <br>
521 <hr width="50%" noshade align="center">
522 <br>
523 <dt><strong>* <kbd>--decode</kbd><a name="-decode">&nbsp;&nbsp;&nbsp;&nbsp;decoding
524 only</a></strong> </dt>
525</dl>
526<dl>
527 <dd>Uses LAME for decoding to a WAV file. The input file can be any input type
528 supported by encoding, including layer I,II,III (MP3) and OGG files. In case
529 of MPEG files, LAME uses a bugfixed version of mpglib for decoding.<br>
530 <br>
531 If -t is used (disable WAV header), Lame will output raw PCM in native endian
532 format. You can use -x to swap bytes order. <br>
533 <br>
534 This option is not usable if the MP3 decoder was <b>explicitly</b>
535 disabled in the build of LAME.
536 <dt><br>
537 <br>
538 </dt>
539 <hr width="50%" noshade align="center">
540 <br>
541 <dl> </dl>
542 <dt><strong>* <kbd>--disptime n</kbd><a name="-disptime">&nbsp;&nbsp;&nbsp;&nbsp;time
543 between display updates</a></strong> </dt>
544</dl>
545<dl>
546 <dd>Set the delay in seconds between two display updates.
547 <dt><br>
548 <br>
549 </dt>
550 <hr width="50%" noshade align="center">
551 <br>
552 <dl> </dl>
553 <dt><strong>* <kbd>-e n/5/c</kbd><a name="e">&nbsp;&nbsp;&nbsp;&nbsp;de-emphasis</a></strong>
554 </dt>
555</dl>
556<dl>
557 <dd> <br>
558 n = (none, default)<br>
559 5 = 0/15 microseconds<br>
560 c = citt j.17<br>
561 <br>
562 All this does is set a flag in the bitstream. If you have a PCM input file
563 where one of the above types of (obsolete) emphasis has been applied, you
564 can set this flag in LAME. Then the mp3 decoder should de-emphasize the output
565 during playback, although most decoders ignore this flag.<br>
566 <br>
567 A better solution would be to apply the de-emphasis with a standalone utility
568 before encoding, and then encode without -e.
569 <dt><br>
570 <br>
571 </dt>
572 <hr width="50%" noshade align="center">
573 <br>
574 <dl> </dl>
575 <dt><strong>* <kbd>-f</kbd><a name="f">&nbsp;&nbsp;&nbsp;&nbsp;fast mode</a></strong>
576 </dt>
577</dl>
578<dl>
579 <dd> This switch forces the encoder to use a faster encoding mode, but with
580 a lower quality. The behaviour is the same as the -q7 switch.<br>
581 <br>
582 Noise shaping will be disabled, but psycho acoustics will still be computed
583 for bit allocation and pre-echo detection.
584 <dt><br>
585 <br>
586 </dt>
587 <hr width="50%" noshade align="center">
588 <br>
589 <dl> </dl>
590 <dt><strong>* <kbd>-F</kbd><a name="FF">&nbsp;&nbsp;&nbsp;strictly enforce the
591 -b option</a></strong> </dt>
592</dl>
593<dl>
594 <dd> This is mainly for use with hardware players that do not support low bitrate
595 mp3.<br>
596 <br>
597 Without this option, the minimum bitrate will be ignored for passages of analog
598 silence, ie when the music level is below the absolute threshold of human
599 hearing (ATH).
600 <dt><br>
601 <br>
602 </dt>
603 <hr width="50%" noshade align="center">
604 <br>
605 <dl> </dl>
606 <dt><strong>* <kbd>--freeformat</kbd><a name="-freeformat">&nbsp;&nbsp;&nbsp;&nbsp;free
607 format bitstream</a></strong> </dt>
608</dl>
609<dl>
610 <dd> Produces a free format bitstream. With this option, you can use -b with
611 any bitrate higher than 8 kbps.<br>
612 <br>
613 However, even if an mp3 decoder is required to support free bitrates at least
614 up to 320 kbps, many players are unable to deal with it.<br>
615 <br>
616 Tests have shown that the following decoders support free format:<br>
617 <br>
618 FreeAmp up to 440 kbps<br>
619 in_mpg123 up to 560 kbps<br>
620 l3dec up to 310 kbps<br>
621 LAME up to 560 kbps<br>
622 MAD up to 640 kbps<br>
623 <dt><br>
624 <br>
625 </dt>
626 <hr width="50%" noshade align="center">
627 <br>
628 <dl> </dl>
629 <dt><strong>* <kbd>-h</kbd><a name="h">&nbsp;&nbsp;&nbsp;&nbsp;high quality</a></strong>
630 </dt>
631</dl>
632<dl>
633 <dd> Use some quality improvements. Encoding will be slower, but the result
634 will be of higher quality. The behaviour is the same as the -q2 switch.<br>
635 This switch is always enabled when using VBR.
636 <dt><br>
637 <br>
638 </dt>
639 <hr width="50%" noshade align="center">
640 <br>
641 <dl> </dl>
642 <dt><strong>* <kbd>--help</kbd><a name="-help">&nbsp;&nbsp;&nbsp;&nbsp;help</a></strong>
643 </dt>
644</dl>
645<dl>
646 <dd> Display a list of all available options.
647 <dt><br>
648 <br>
649 </dt>
650 <hr width="50%" noshade align="center">
651 <br>
652 <dl> </dl>
653 <dt><strong>* <kbd>--highpass</kbd><a name="-highpass">&nbsp;&nbsp;&nbsp;&nbsp;highpass
654 filtering frequency in kHz</a></strong> </dt>
655</dl>
656<dl>
657 <dd> Set an highpass filtering frequency. Frequencies below the specified one
658 will be cutoff.
659 <dt><br>
660 <br>
661 </dt>
662 <hr width="50%" noshade align="center">
663 <br>
664 <dl> </dl>
665 <dt><strong>* <kbd>--highpass-width</kbd><a name="-highpass-width">&nbsp;&nbsp;&nbsp;&nbsp;width
666 of highpass filtering in kHz</a></strong> </dt>
667</dl>
668<dl>
669 <dd> Set the width of the highpass filter. The default value is 15% of the highpass
670 frequency.
671 <dt><br>
672 <br>
673 </dt>
674 <hr width="50%" noshade align="center">
675 <br>
676 <dl> </dl>
677 <dt><strong>* <kbd>-k</kbd><a name="k">&nbsp;&nbsp;&nbsp;&nbsp;full bandwidth</a></strong>
678 </dt>
679</dl>
680<dl>
681 <dd> Tells the encoder to use full bandwidth and to disable all filters. By
682 default, the encoder uses some lowpass filtering at lower bitrates, in order
683 to keep a good quality by giving more bits to more important frequencies.<br>
684 Increasing the bandwidth from the default setting might produce ringing artefacts
685 at low bitrates. Use with care!
686 <dt><br>
687 <br>
688 </dt>
689 <hr width="50%" noshade align="center">
690 <br>
691 <dl> </dl>
692 <dt><strong>* <kbd>--lowpass</kbd><a name="-lowpass">&nbsp;&nbsp;&nbsp;&nbsp;lowpass
693 filtering frequency in kHz</a></strong></dt>
694</dl>
695<dl>
696 <dd> Set a lowpass filtering frequency. Frequencies above the specified one
697 will be cutoff.
698 <dt><br>
699 <br>
700 </dt>
701 <hr width="50%" noshade align="center">
702 <br>
703 <dl> </dl>
704 <dt><strong>* <kbd>--lowpass-width</kbd><a name="-lowpass-width">&nbsp;&nbsp;&nbsp;&nbsp;width
705 of lowpass filtering in kHz</a></strong></dt>
706</dl>
707<dl>
708 <dd> Set the width of the lowpass filter. The default value is 15% of the lowpass
709 frequency.
710 <dt><br>
711 <br>
712 </dt>
713 <hr width="50%" noshade align="center">
714 <br>
715 <dl> </dl>
716 <dt><strong>* <kbd>-m s/<b>j/</b>f/d/m</kbd><a name="m">&nbsp;&nbsp;&nbsp;&nbsp;stereo
717 mode</a></strong> </dt>
718</dl>
719<dl>
720 <dd> Joint-stereo is the default mode for input files featuring two channels..
721 <b><i><br>
722 <br>
723 stereo</i></b> <br>
724 In this mode, the encoder makes no use of potentially existing correlations
725 between the two input channels. It can, however, negotiate the bit demand
726 between both channel, i.e. give one channel more bits if the other contains
727 silence or needs less bits because of a lower complexity.<br>
728 <br>
729 <i><b>joint stereo</b></i><br>
730 In this mode, the encoder will make use of correlation between both channels.
731 The signal will be matrixed into a sum ("mid"), computed by L+R, and difference
732 ("side") signal, computed by L-R, and more bits are allocated to the mid channel.<br>
733 This will effectively increase the bandwidth if the signal does not have too
734 much stereo separation, thus giving a significant gain in encoding quality.
735 In joint stereo, the encoder can select between Left/Right and Mid/Side representation
736 on a frame basis.<br>
737 <br>
738 Using mid/side stereo inappropriately can result in audible compression artifacts.
739 To much switching between mid/side and regular stereo can also sound bad.
740 To determine when to switch to mid/side stereo, LAME uses a much more sophisticated
741 algorithm than that described in the ISO documentation, and thus is safe to
742 use in joint stereo mode.<br>
743 <br>
744 <b><i>forced joint stereo </i></b><br>
745 This mode will force MS joint stereo on all frames. It's slightly faster than
746 joint stereo, but it should be used only if you are sure that every frame
747 of the input file has very little stereo separation.<br>
748 <br>
749 <b><i>dual channels </i></b><br>
750 In this mode, the 2 channels will be totally independently encoded. Each
751 channel will have exactly half of the bitrate. This mode is designed for applications
752 like dual languages encoding (ex: English in one channel and French in the
753 other). Using this encoding mode for regular stereo files will result in a
754 lower quality encoding.<br>
755 <br>
756 <b><i>mono</i></b><br>
757 The input will be encoded as a mono signal. If it was a stereo signal, it
758 will be downsampled to mono. The downmix is calculated as the sum of the left
759 and right channel, attenuated by 6 dB.
760 <dt><br>
761 <br>
762 </dt>
763 <hr width="50%" noshade align="center">
764 <br>
765 <dl> </dl>
766 <dt><strong>* <kbd>--mp1input</kbd><a name="-mp1input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG
767 Layer I input file</a></strong> </dt>
768</dl>
769<dl>
770 <dd> Assume the input file is a MPEG Layer I file.<br>
771 If the filename ends in ".mp1" or &quot;.mpg&quot; LAME will assume it is
772 a MPEG Layer I file. For stdin or Layer I files which do not end in .mp1 or .mpg
773 you need to use this switch.
774 <dt><br>
775 </dt>
776</dl>
777<dl>
778 <hr width="50%" noshade align="center">
779 <br>
780 <dl> </dl>
781 <dt><strong>* <kbd>--mp2input</kbd><a name="-mp2input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG
782 Layer II input file</a></strong> </dt>
783</dl>
784<dl>
785 <dd> Assume the input file is a MPEG Layer II (ie MP2) file.<br>
786 If the filename ends in ".mp2" LAME will assume it is a MPEG Layer II file. For
787 stdin or Layer II files which do not end in .mp2 you need to use this switch.
788 <dt><br>
789 </dt>
790</dl>
791<dl>
792 <hr width="50%" noshade align="center">
793 <br>
794 <dl> </dl>
795 <dt><strong>* <kbd>--mp3input</kbd><a name="-mp3input">&nbsp;&nbsp;&nbsp;&nbsp;MPEG
796 Layer III input file</a></strong> </dt>
797</dl>
798<dl>
799 <dd> Assume the input file is a MP3 file. Useful for downsampling from one
800 mp3 to another. As an example, it can be useful for streaming through an
801 IceCast server.<br>
802 If the filename ends in ".mp3" LAME will assume it is an MP3 file. For stdin or
803 MP3 files which do not end in .mp3 you need to use this switch.
804 <dt><br>
805 </dt>
806</dl>
807<dl>
808 <hr width="50%" noshade align="center">
809 <br>
810 <dl> </dl>
811 <dt><strong>* <kbd>--noath</kbd><a name="-noath">&nbsp;&nbsp;&nbsp;&nbsp;disable
812 ATH</a></strong> </dt>
813</dl>
814<dl>
815 <dd> Disable any use of the ATH (absolute threshold of hearing) for masking.
816 Normally, humans are unable to hear any sound below this threshold.
817 <dt><br>
818 </dt>
819</dl>
820<dl>
821 <hr width="50%" noshade align="center">
822 <br>
823 <dl> </dl>
824 <dt><strong>* <kbd>--noasm mmx/3dnow/sse</kbd><a name="-noasm">
825 &nbsp;&nbsp;&nbsp;&nbsp;disable assembly optimizations</a></strong> </dt>
826</dl>
827<dl>
828 <dd>Disable specific assembly optimizations. Quality will not increase, only
829 speed will be reduced. If you have problems running Lame on a Cyrix/Via
830 processor, disabling mmx optimizations might solve your problem.
831 <dt><br>
832 </dt>
833</dl>
834<dl>
835 <hr width="50%" noshade align="center">
836 <br>
837 <dl> </dl>
838 <dt><strong>* <kbd>--nohist</kbd><a name="-nohist">&nbsp;&nbsp;&nbsp;&nbsp;disable
839 histogram display</a></strong> </dt>
840</dl>
841<dl>
842 <dd> By default, LAME will display a bitrate histogram while producing VBR mp3
843 files. This will disable that feature.<br>
844 Histogram display might not be available on your release.
845 <dt><br>
846 </dt>
847</dl>
848<dl>
849 <hr width="50%" noshade align="center">
850 <br>
851 <dl> </dl>
852 <dt><strong>* <kbd>--noreplaygain</kbd><a name="-noreplaygain">&nbsp;&nbsp;&nbsp;&nbsp;disable
853 ReplayGain analysis</a></strong></dt>
854</dl>
855<dl>
856 <dd> By default ReplayGain analysis is enabled. This switch disables it.<br>
857 <br>
858 See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>,
859 <a href="#-replaygain-fast">--replaygain-fast</a>
860 <dt><br>
861 </dt>
862</dl>
863<dl>
864 <hr width="50%" noshade align="center">
865 <br>
866 <dl> </dl>
867 <dt><strong>* <kbd>--nores</kbd><a name="-nores">&nbsp;&nbsp;&nbsp;&nbsp;disable
868 bit reservoir</a></strong></dt>
869</dl>
870<dl>
871 <dd> Disable the bit reservoir. Each frame will then become independent from
872 previous ones, but the quality will be lower.
873 <dt><br>
874 </dt>
875</dl>
876<dl>
877 <hr width="50%" noshade align="center">
878 <br>
879 <dl> </dl>
880 <dt><strong>* <kbd>--noshort</kbd><a name="-noshort">&nbsp;&nbsp;&nbsp;&nbsp;disable
881 short blocks frames</a></strong></dt>
882</dl>
883<dl>
884 <dd> Encode all frames using long blocks only. This could increase quality when
885 encoding at very low bitrates, but can produce serious pre-echo artefacts.
886 <dt><br>
887 </dt>
888</dl>
889<dl>
890 <hr width="50%" noshade align="center">
891 <br>
892 <dl> </dl>
893 <dt><strong>* <kbd>--notemp</kbd><a name="-notemp">&nbsp;&nbsp;&nbsp;&nbsp;disable
894 temporal masking</a></strong></dt>
895</dl>
896<dl>
897 <dd>Don't make use of the temporal masking effect.
898 <dt><br>
899 </dt>
900</dl>
901<dl>
902 <hr width="50%" noshade align="center">
903 <br>
904 <dl> </dl>
905 <dt><strong>* <kbd>-o</kbd><a name="o">&nbsp;&nbsp;&nbsp;&nbsp;non-original</a></strong>
906 </dt>
907</dl>
908<dl>
909 <dd> Mark the encoded file as being a copy.
910 <dt><br>
911 <br>
912 </dt>
913 <hr width="50%" noshade align="center">
914 <br>
915 <dl> </dl>
916 <dt><strong>* <kbd>-p</kbd><a name="p">&nbsp;&nbsp;&nbsp;&nbsp;error protection</a></strong></dt>
917</dl>
918<dl>
919 <dd> Turn on CRC error protection.<br>
920 It will add a cyclic redundancy check (CRC) code in each frame, allowing to
921 detect transmission errors that could occur on the MP3 stream. However, it
922 takes 16 bits per frame that would otherwise be used for encoding, and then
923 will slightly reduce the sound quality.
924 <dt><br>
925 <br>
926 </dt>
927 <hr width="50%" noshade align="center">
928 <br>
929 <dl> </dl>
930 <dt><strong>* <kbd>--preset presetName</kbd> <a name="-preset">&nbsp;&nbsp;&nbsp;&nbsp;use
931 built-in preset</a></strong></dt>
932</dl>
933<dd> Use one of the built-in presets (standard, fast standard, extreme, fast extreme, insane, or the abr/cbr modes).
934<br>
935<dd> "--preset help" gives more information about the usage possibilities for these presets.
936<dt><br>
937 <br>
938<hr width="50%" noshade align="center">
939<br>
940<dl> </dl>
941<dt><strong>* <kbd>--priority 0...4</kbd><a name="-priority">&nbsp;&nbsp;&nbsp;&nbsp;OS/2
942 process priority control</a></strong> </dt>
943<dl>
944 <dd> With this option, LAME will run with a different process priority under
945 IBM OS/2.<br>
946 This will greatly improve system responsiveness, since OS/2 will have more
947 free time to properly update the screen and poll the keyboard/mouse. It should
948 make quite a difference overall, especially on slower machines. LAME's performance
949 impact should be minimal.<br>
950 <br>
951 <dd><b>0 (Low priority)</b><br>
952 Priority 0 assumes "IDLE" class, with delta 0.<br>
953 LAME will have the lowest priority possible, and the encoding may be suspended
954 very frequently by user interaction.<br>
955 <br>
956 <dd><b>1 (Medium priority)</b><br>
957 Priority 1 assumes "IDLE" class, with delta +31.<br>
958 LAME won't interfere at all with what you're doing.<br>
959 Recommended if you have a slower machine. <br>
960 <br>
961 <dd><b>2 (Regular priority)</b><br>
962 Priority 2 assumes "REGULAR" class, with delta -31.<br>
963 LAME won't interfere with your activity. It'll run just like a regular process,
964 but will spare just a bit of idle time for the system. Recommended for most
965 users. <br>
966 <br>
967 <dd><b>3 (High priority)</b><br>
968 Priority 3 assumes "REGULAR" class, with delta 0.<br>
969 LAME will run with a priority a bit higher than a normal process. <br>
970 Good if you're just running LAME by itself or with moderate user interaction.<br>
971 <br>
972 <dd><b>4 (Maximum priority)</b><br>
973 Priority 4 assumes "REGULAR" class, with delta +31.<br>
974 LAME will run with a very high priority, and may interfere with the machine
975 response.<br>
976 Recommended if you only intend to run LAME by itself, or if you have a fast
977 processor. <br>
978 <br>
979 <br>
980 Priority 1 or 2 is recommended for most users.
981 <dt><br>
982 <br>
983 </dt>
984 <hr width="50%" noshade align="center">
985 <br>
986 <dl> </dl>
987 <dt><strong>* <kbd>-q 0..9</kbd><a name="q">&nbsp;&nbsp;&nbsp;&nbsp;algorithm
988 quality selection</a></strong></dt>
989</dl>
990<dl>
991 <dd> Bitrate is of course the main influence on quality. The higher the bitrate,
992 the higher the quality. But for a given bitrate, we have a choice of algorithms
993 to determine the best scalefactors and Huffman encoding (noise shaping).<br>
994 <br>
995 -q 0: use slowest &amp; best possible version of all algorithms. -q 0 and -q 1
996 are slow and may not produce significantly higher quality.<br>
997 <br>
998 -q 2: recommended. Same as -h.<br>
999 <br>
1000 -q 5: default value. Good speed, reasonable quality.<br>
1001 <br>
1002 -q 7: same as -f. Very fast, ok quality. (psycho acoustics are used for pre-echo
1003 &amp; M/S, but no noise shaping is done.<br>
1004 <br>
1005 -q 9: disables almost all algorithms including psy-model. poor quality.
1006 <dt><br>
1007 <br>
1008 </dt>
1009 <hr width="50%" noshade align="center">
1010 <br>
1011 <dl> </dl>
1012 <dt><strong>* <kbd>-r</kbd><a name="r">&nbsp;&nbsp;&nbsp;&nbsp;input file is
1013 raw PCM</a></strong></dt>
1014</dl>
1015<dl>
1016 <dd> Assume the input file is raw PCM. Sampling rate and mono/stereo/jstereo
1017 must be specified on the command line. Without -r, LAME will perform several
1018 fseek()'s on the input file looking for WAV and AIFF headers.<br>
1019 Might not be available on your release.
1020 <dt><br>
1021 <br>
1022 </dt>
1023 <hr width="50%" noshade align="center">
1024 <br>
1025 <dl> </dl>
1026 <dt><strong>* <kbd>--replaygain-accurate</kbd><a name="-replaygain-accurate">&nbsp;&nbsp;&nbsp;&nbsp;compute
1027 ReplayGain more accurately and find the peak sample</a></strong></dt>
1028</dl>
1029<dl>
1030 <dd>
1031 Enable decoding on the fly. Compute "Radio" ReplayGain on the decoded
1032 data stream. Find the peak sample of the decoded data stream and store
1033 it in the file.<br>
1034 <br>
1035 ReplayGain analysis does <i>not</i> affect the content of a
1036 compressed data stream itself, it is a value stored in the header
1037 of a sound file. Information on the purpose of ReplayGain and the
1038 algorithms used is available from
1039 <a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
1040 <br>
1041 By default, LAME performs ReplayGain analysis on the input data
1042 (after the user-specified volume scaling). This
1043 behavior might give slightly inaccurate results because the data on
1044 the output of a lossy compression/decompression sequence differs from
1045 the initial input data. When --replaygain-accurate is specified the
1046 mp3 stream gets decoded on the fly and the analysis is performed on the
1047 decoded data stream. Although theoretically this method gives more
1048 accurate results, it has several disadvantages:
1049 <ul>
1050 <li> tests have shown that the difference between the ReplayGain values
1051 computed on the input data and decoded data is usually no greater
1052 than 0.5dB, although the minimum volume difference the human ear
1053 can perceive is about 1.0dB
1054 </li>
1055 <li> decoding on the fly significantly slows down the encoding process
1056 </li>
1057 </ul>
1058 The apparent advantage is that:
1059 <ul>
1060 <li> with --replaygain-accurate the peak sample is determined and
1061 stored in the file. The knowledge of the peak sample can be useful
1062 to decoders (players) to prevent a negative effect called 'clipping'
1063 that introduces distortion into sound.
1064 </li>
1065 </ul>
1066 <br>
1067 Only the "RadioGain" ReplayGain value is computed. It is stored in the
1068 LAME tag. The analysis is performed with the reference volume equal
1069 to 89dB. Note: the reference volume has been changed from 83dB on
1070 transition from version 3.95 to 3.95.1.<br>
1071 <br>
1072 This option is not usable if the MP3 decoder was <b>explicitly</b>
1073 disabled in the build of LAME. (Note: if LAME is compiled without the
1074 MP3 decoder, ReplayGain analysis is performed on the input data after
1075 user-specified volume scaling).<br>
1076 <br>
1077 See also: <a href="#-replaygain-fast">--replaygain-fast</a>,
1078 <a href="#-noreplaygain">--noreplaygain</a>, <a href="#-clipdetect">--clipdetect</a>
1079 <dt><br>
1080 </dt>
1081</dl>
1082<dl>
1083 <hr width="50%" noshade align="center">
1084 <br>
1085 <dl> </dl>
1086 <dt><strong>* <kbd>--replaygain-fast</kbd><a name="-replaygain-fast">&nbsp;&nbsp;&nbsp;&nbsp;compute
1087 ReplayGain fast but slightly inaccurately (default)</a></strong></dt>
1088</dl>
1089<dl>
1090 <dd>
1091 Compute "Radio" ReplayGain on the input data stream after user-specified
1092 volume scaling and/or resampling.<br>
1093 <br>
1094 ReplayGain analysis does <i>not</i> affect the content of a
1095 compressed data stream itself, it is a value stored in the header
1096 of a sound file. Information on the purpose of ReplayGain and the
1097 algorithms used is available from
1098 <a href="http://www.replaygain.org/">http://www.replaygain.org/</a><br>
1099 <br>
1100 Only the "RadioGain" ReplayGain value is computed. It is stored in the
1101 LAME tag. The analysis is performed with the reference volume equal
1102 to 89dB. Note: the reference volume has been changed from 83dB on
1103 transition from version 3.95 to 3.95.1.<br>
1104 <br>
1105 This switch is enabled by default.<br>
1106 <br>
1107 See also: <a href="#-replaygain-accurate">--replaygain-accurate</a>,
1108 <a href="#-noreplaygain">--noreplaygain</a>
1109 <dt><br>
1110 </dt>
1111</dl>
1112<dl>
1113 <hr width="50%" noshade align="center">
1114 <br>
1115 <dl> </dl>
1116 <dt><strong>* <kbd>--resample 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="-resample">&nbsp;&nbsp;&nbsp;&nbsp;output
1117 sampling frequency in kHz</a></strong></dt>
1118</dl>
1119<dl>
1120 <dd> Select output sampling frequency (for encoding only). <br>
1121 If not specified, LAME will automatically resample the input when using high
1122 compression ratios.
1123 <dt><br>
1124 </dt>
1125</dl>
1126<dl>
1127 <hr width="50%" noshade align="center">
1128 <br>
1129 <dl> </dl>
1130 <dt><strong>* <kbd>-s 8/11.025/12/16/22.05/24/32/44.1/48</kbd><a name="s">&nbsp;&nbsp;&nbsp;&nbsp;sampling
1131 frequency</a></strong> </dt>
1132</dl>
1133<dl>
1134 <dd> Required only for raw PCM input files. Otherwise it will be determined
1135 from the header of the input file.<br>
1136 <br>
1137 LAME will automatically resample the input file to one of the supported MP3
1138 samplerates if necessary.
1139 <dt><br>
1140 <br>
1141 </dt>
1142 <hr width="50%" noshade align="center">
1143 <br>
1144 <dl> </dl>
1145 <dt><strong>* <kbd>-S / --silent / --quiet</kbd><a name="-silent">&nbsp;&nbsp;&nbsp;&nbsp;silent
1146 operation</a></strong> </dt>
1147</dl>
1148<dl>
1149 <dd> Don't print progress report.
1150 <dt><br>
1151 <br>
1152 </dt>
1153 <hr width="50%" noshade align="center">
1154 <br>
1155 <dl> </dl>
1156 <dt><strong>* <kbd>--scale n</kbd><a name="-scale">&nbsp;&nbsp;&nbsp;&nbsp;scales
1157 input by n</a></strong> </dt>
1158 <dt><strong>* <kbd>--scale-l n</kbd><a name="-scale-l">&nbsp;&nbsp;&nbsp;&nbsp;scales
1159 input channel 0 (left) by n</a></strong> </dt>
1160 <dt><strong>* <kbd>--scale-r n</kbd><a name="-scale-r">&nbsp;&nbsp;&nbsp;&nbsp;scales
1161 input channel 1 (right) by n</a></strong> </dt>
1162</dl>
1163<dl>
1164 <dd>Scales input by n. This just multiplies the PCM data (after it has been
1165 converted to floating point) by n. <br>
1166 <br>
1167 n > 1: increase volume<br>
1168 n = 1: no effect<br>
1169 n < 1: reduce volume<br>
1170 <br>
1171 Use with care, since most MP3 decoders will truncate data which decodes to
1172 values greater than 32768.
1173 <dt><br>
1174 <br>
1175 </dt>
1176 <hr width="50%" noshade align="center">
1177 <br>
1178 <dl> </dl>
1179 <dt><strong>* <kbd>--short</kbd><a name="-short">&nbsp;&nbsp;&nbsp;&nbsp;use
1180 short blocks</a></strong> </dt>
1181</dl>
1182<dl>
1183 <dd>Let LAME use short blocks when appropriate. It is the default setting.
1184</dl>
1185<dl>
1186 <dd>&nbsp;
1187 <dt><br>
1188 <br>
1189 </dt>
1190 <hr width="50%" noshade align="center">
1191 <br>
1192 <dl> </dl>
1193 <dt><strong>* <kbd>--strictly-enforce-ISO</kbd><a name="-strictly-enforce-ISO">&nbsp;&nbsp;&nbsp;&nbsp;strict
1194 ISO compliance</a></strong> </dt>
1195</dl>
1196<dl>
1197 <dd> With this option, LAME will enforce the 7680 bit limitation on total frame
1198 size.<br>
1199 This results in many wasted bits for high bitrate encodings but will ensure
1200 strict ISO compatibility. This compatibility might be important for hardware
1201 players.
1202</dl>
1203<dl>
1204 <dd>&nbsp;
1205 <dt><br>
1206 <br>
1207 </dt>
1208 <hr width="50%" noshade align="center">
1209 <br>
1210 <dl> </dl>
1211 <dt><strong>* <kbd>-t</kbd><a name="t">&nbsp;&nbsp;&nbsp;&nbsp;disable INFO/WAV
1212 header </a></strong></dt>
1213</dl>
1214<dl>
1215 <dd> Disable writing of the INFO Tag on encoding.<br>
1216 This tag in embedded in frame 0 of the MP3 file. It includes some information
1217 about the encoding options of the file, and in VBR it lets VBR aware players
1218 correctly seek and compute playing times of VBR files.<br>
1219 <br>
1220 When '--decode' is specified (decode to WAV), this flag will disable writing
1221 of the WAV header. The output will be raw PCM, native endian format. Use -x
1222 to swap bytes.
1223 <dt><br>
1224 <br>
1225 </dt>
1226 <hr width="50%" noshade align="center">
1227 <br>
1228 <dl> </dl>
1229 <dt><strong>* <kbd>-V 0...9</kbd><a name="V">&nbsp;&nbsp;&nbsp;&nbsp;VBR quality
1230 setting</a></strong></dt>
1231</dl>
1232<dl>
1233 <dd> Enable VBR (Variable BitRate) and specifies the value of VBR quality.<br>
1234 default=4<br>
1235 0=highest quality.
1236 <dt><br>
1237 <br>
1238 </dt>
1239 <hr width="50%" noshade align="center">
1240 <br>
1241 <dl> </dl>
1242 <dt><strong>* <kbd>--vbr-new</kbd><a name="-vbr-new">&nbsp;&nbsp;&nbsp;&nbsp;new
1243 VBR mode</a></strong></dt>
1244</dl>
1245<dl>
1246 <dd> Invokes the newest VBR algorithm. During the development of version 3.90,
1247 considerable tuning was done on this algorithm, and it is now considered to
1248 be on par with the original --vbr-old. <br>
1249 It has the added advantage of being very fast (over twice as fast as --vbr-old).
1250 <dt><br>
1251 <br>
1252 </dt>
1253 <hr width="50%" noshade align="center">
1254 <br>
1255 <dl> </dl>
1256 <dt><strong>* <kbd>--vbr-old</kbd><a name="-vbr-old">&nbsp;&nbsp;&nbsp;&nbsp;older
1257 VBR mode</a></strong></dt>
1258</dl>
1259<dl>
1260 <dd> Invokes the oldest, most tested VBR algorithm. It produces very good quality
1261 files, though is not very fast. This has, up through v3.89, been considered
1262 the "workhorse" VBR algorithm.
1263 <dt><br>
1264 <br>
1265 </dt>
1266 <hr width="50%" noshade align="center">
1267 <br>
1268 <dl> </dl>
1269 <dt><strong>* <kbd>--verbose</kbd><a name="-verbose">&nbsp;&nbsp;&nbsp;&nbsp;verbosity</a></strong></dt>
1270</dl>
1271<dl>
1272 <dd> Print a lot of information on screen.
1273 <dt><br>
1274 <br>
1275 </dt>
1276 <hr width="50%" noshade align="center">
1277 <br>
1278 <dl> </dl>
1279 <dt><strong>* <kbd>-x</kbd><a name="x">&nbsp;&nbsp;&nbsp;&nbsp;swapbytes</a></strong>
1280 </dt>
1281</dl>
1282<dl>
1283 <dd> Swap bytes in the input file or output file when using --decode.<br>
1284 For sorting out little endian/big endian type problems. If your encodings
1285 sounds like static, try this first.
1286 <dt><br>
1287 <br>
1288 </dt>
1289 <hr width="50%" noshade align="center">
1290 <br>
1291 <dl> </dl>
1292 <dt><strong>* <kbd>-X 0...7</kbd><a name="Xquant">&nbsp;&nbsp;&nbsp;&nbsp;change
1293 quality measure</a></strong> </dt>
1294</dl>
1295<dl>
1296 <dd> When LAME searches for a "good" quantization, it has to compare the actual
1297 one with the best one found so far. The comparison says which one is better,
1298 the best so far or the actual. The -X parameter selects between different
1299 approaches to make this decision, -X0 being the default mode:<br>
1300 <br>
1301 <b>-X0 </b><br>
1302 The criterions are (in order of importance):<br>
1303 * less distorted scalefactor bands<br>
1304 * the sum of noise over the thresholds is lower<br>
1305 * the total noise is lower<br>
1306 <br>
1307 <b>-X1</b><br>
1308 The actual is better if the maximum noise over all scalefactor bands is less
1309 than the best so far .<br>
1310 <br>
1311 <b>-X2</b><br>
1312 The actual is better if the total sum of noise is lower than the best so far.<br>
1313 <br>
1314 <b>-X3</b><br>
1315 The actual is better if the total sum of noise is lower than the best so far
1316 and the maximum noise over all scalefactor bands is less than the best so
1317 far plus 2db.<br>
1318 <br>
1319 <b>-X4</b> <br>
1320 Not yet documented.<br>
1321 <br>
1322 <b>-X5</b><br>
1323 The criterions are (in order of importance):<br>
1324 * the sum of noise over the thresholds is lower <br>
1325 * the total sum of noise is lower<br>
1326 <br>
1327 <b>-X6</b> <br>
1328 The criterions are (in order of importance):<br>
1329 * the sum of noise over the thresholds is lower<br>
1330 * the maximum noise over all scalefactor bands is lower<br>
1331 * the total sum of noise is lower<br>
1332 <br>
1333 <b>-X7</b> <br>
1334 The criterions are:<br>
1335 * less distorted scalefactor bands<br>
1336 or<br>
1337 * the sum of noise over the thresholds is lower
1338</dl>
1339</BODY>
1340</HTML>
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