usage.txt

LAME无疑是目前最优秀的MP3编码软件

-q 7:  same as -f.  Very fast, ok quality.  (psycho acoustics are
       used for pre-echo & M/S, but no noise shaping is done.  

-q 9:  disables almost all algorithms including psy-model.  poor quality.



=======================================================================
input file is raw pcm
=======================================================================
-r  

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.

Not supported if LAME is compiled to use LIBSNDFILE.



=======================================================================
slightly more accurate ReplayGain analysis and finding the peak sample
=======================================================================
--replaygain-accurate

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.


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/

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:
  * 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
  * decoding on the fly significantly slows down the encoding process
The apparent advantage is that:
  * 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.
    

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.

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).

See also: --replaygain-fast, --noreplaygain, --clipdetect


=======================================================================
fast ReplayGain analysis
=======================================================================
--replaygain-fast

Compute "Radio" ReplayGain of the input data stream after user-specified
volume scaling and/or resampling.

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/

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.

This switch is enabled by default.

See also: --replaygain-accurate, --noreplaygain



=======================================================================
output sampling frequency in kHz
=======================================================================
--resample  n

where n = 8, 11.025, 12, 16, 22.05, 24, 32, 44.1, 48

Output sampling frequency.  Resample the input if necessary.  

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 --resamle to set the output samplerate
equal to the inptu samplerate.  In that case, LAME will not
perform any extra computations.



=======================================================================
sampling frequency in kHz
=======================================================================
-s  n

where n = sampling rate in kHz.

Required for raw PCM input files.  Otherwise it will be determined
from the header information in the input file.

LAME will automatically resample the input file to one of the
supported MP3 samplerates if necessary.


=======================================================================
silent operation
=======================================================================
-S

don't print progress report

=======================================================================
scale
=======================================================================
--scale <arg>

Scales input by <arg>.  This just multiplies the PCM data
(after it has been converted to floating point) by <arg>.  

<arg> > 1:  increase volume
<arg> = 1:  no effect
<arg> < 1:  reduce volume

Use with care, since most MP3 decoders will truncate data
which decodes to values greater than 32768.  


=======================================================================
strict ISO complience
=======================================================================
--strictly-enforce-ISO   

With this option, LAME will enforce the 7680 bit limitation on
total frame size.  This results in many wasted bits for
high bitrate encodings.


=======================================================================
disable VBR tag
=======================================================================
-t              

Disable writing of the VBR Tag (only valid if -v flag is
specified) This tag in embedded in frame 0 of the MP3 file.  It lets
VBR aware players correctly seek and compute playing times of VBR
files.

When '--decode' is specified (decode mp3 to wav), this flag will 
disable writing the WAV header.  The output will be raw pcm,
native endian format.  Use -x to swap bytes.



=======================================================================
variable bit rate  (VBR)
=======================================================================
-v

Turn on VBR.  There are several ways you can use VBR.  I personally
like using VBR to get files slightly bigger than 128 kbps files, where
the extra bits are used for the occasional difficult-to-encode frame.
For this, try specifying a minimum bitrate to use with VBR:

lame -v      -b 112  input.wav output.mp3

If the file is too big, use -V n, where n = 0...9

lame -v -V n -b 112  input.wav output.mp3


If you want to use VBR to get the maximum compression possible,
and for this, you can try:  

lame -v  input.wav output.mp3
lame -v -V n input.wav output.mp3         (to vary quality/filesize)






=======================================================================
VBR quality setting
=======================================================================
-V n       

n = 0...9.  Specifies the value of VBR_q.
default = 4,  highest quality = 0, smallest files = 9

Using -V 5 or higher (lower quality) is NOT RECOMMENDED.  
ABR will produce better results.  


How is VBR_q used?

The value of VBR_q influences two basic parameters of LAME's psycho
acoustics:
 a) the absolute threshold of hearing
 b) the sample to noise ratio
The lower the VBR_q value the lower the injected quantization noise
will be.
 
*NOTE* No psy-model is perfect, so there can often be distortion which
is audible even though the psy-model claims it is not!  Thus using a
small minimum bitrate can result in some aggressive compression and
audible distortion even with -V 0.  Thus using -V 0 does not sound
better than a fixed 256 kbps encoding.  For example: suppose in the 1 kHz
frequency band the psy-model claims 20 dB of distortion will not be
detectable by the human ear, so LAME VBR-0 will compress that
frequency band as much as possible and introduce at most 20 dB of
distortion.  Using a fixed 256 kbps framesize, LAME could end up
introducing only 2 dB of distortion.  If the psy-model was correct,
they will both sound the same.  If the psy-model was wrong, the VBR-0
result can sound worse.


=======================================================================
voice encoding mode
=======================================================================
--voice

An experimental voice encoding mode.  Tuned for 44.1 kHz input files.
--voice is deprecated, use --preset voice instead


=======================================================================
swapbytes   
=======================================================================
-x

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.

=======================================================================
OS/2 process priority control   
=======================================================================
--priority <type>

(OS/2 only)

Sets the process priority for LAME while running under 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.

The valid parameters are:

     0 = Low priority (IDLE, delta = 0)
     1 = Medium priority (IDLE, delta = +31)
     2 = Regular priority (REGULAR, delta = -31)
     3 = High priority (REGULAR, delta = 0)
     4 = Maximum priority (REGULAR, delta = +31) 

Note that if you call '--priority' without a parameter, then 
priority 0 will be assumed.