usage

MP3编码程序和资料

=======================================================================
graphical frame analyzer
=======================================================================
-g

run MP3x, the graphical frame analyzer analysis on the inputfile.  The
inputfile can be either an .mp3 file or uncompressed audio file.  MP3x
support must be compiled into LAME, and requires GTK 1.2.
Documentation is under the About pull down menu.



=======================================================================
high quality
=======================================================================
-h

use (maybe) some quality improvements

LAME 3.21 and up:  -h enables specialized mid/side masking thresholds to
be used in jstereo mode.  Will sound better in jstereo mode
but is 20% slower.  No effect for mono files.

LAME 3.58beta and up: -h also enables a more accurate but slightly
slower quantization formula.  



=======================================================================
sfb=21 cutoff
=======================================================================
-k   

keep all frequencies.  (Disable all filters)

Without -k, LAME will automatically apply various types of lowpass
filters.  This is because the high frequency coefficients can take up
a lot of bits that would be better used for lower, more important
frequencies.




=======================================================================
Modes:
=======================================================================

-m m           mono.  
-m s           stereo
-m j           jstereo
-m f           forced mid/side stereo

mono is the default mode for mono input files.  If "-m m" is specified
for a stereo input file, the two channels will be averaged into a mono
signal.  

jstereo is the default mode for stereo files with fixed bitrates of
128kbs or less.  At higher fixed bitrates, the default is stereo.
For VBR encoding, jstereo is the default for VBR_q >4, and stereo
is the default for VBR_q <=4.  You can override all of these defaults
by specifing the mode on the command line.  

jstereo means the encoder can use (on a frame by frame bases) either
regular stereo (just encode left and right channels independently)
or mid/side stereo.  In mid/side stereo, the mid (L+R) and side (L-R)
channels are encoded, and more bits are allocated to the mid channel
than the side channel.  This will effectively increase the bandwidth
if the signal does not have too much stereo separation.  

Mid/side stereo is basically a trick to increase bandwidth.  At 128kbs,
it is clearly worth while.  At higher bitrates it is less usefull.  

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.

-m f forces all frames to be encoded mid/side stereo.  It 
should only be used if you are sure every frame of the input file
has very little stereo seperation.  



=======================================================================
MP3 input file
=======================================================================
--mp3input

Assume the input file is a MP3 file.  Usefull for downsampling from
one mp3 to another.  If the filename ends in ".mp3" LAME will assume
it is an MP3.  For stdin or MP3 files which dont end in .mp3 you need
to use this switch.


=======================================================================
disable historgram display
=======================================================================
--nohist

By default, LAME will display a bitrate histogram while producing
VBR mp3 files.  This will disable that feature.


=======================================================================
disable short blocks
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--noshort

Encode all frames using long blocks.



=======================================================================
non-original
=======================================================================
-o   

mark the encoded file as a copy



=======================================================================
Ogg Vorbis encoding
=======================================================================
--ogg

Encode using the Ogg Vobis codec (using libvorbis) instead of
LAME's internal mp3 codec.

Assume the input file is an Ogg Vorbis file.  Mostly usefull
with --decode for playing back .ogg files.  
If the filename ends in ".ogg" LAME will assume
it is an Ogg.  For stdin or files which dont end in .ogg you need
to use this switch.


=======================================================================
Ogg Vorbis input file
=======================================================================
--ogginput

Assume the input file is an Ogg Vorbis file.  Mostly usefull
with --decode for playing back .ogg files, or converting
your .ogg collection to MP3 :-)

If the filename ends in ".ogg" LAME will assume it is Ogg.  For
stdin or files which dont end in .ogg you need to use this switch.




=======================================================================
CRC error protection
=======================================================================
-p  

turn on CRC error protection.  
Yes this really does work correctly in LAME.  However, it takes 
16 bits per frame that would otherwise be used for encoding.



=======================================================================
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.



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

where n = 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 44.1khz input file at 16kbs).  



=======================================================================
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


=======================================================================
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 Xing VBR tag
=======================================================================
-t              

Disable writing of the Xing 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.



=======================================================================
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 128kbs 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 wan 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 very quality/filesize)




=======================================================================
old variable bit rate  (VBR)
=======================================================================
--vbr-old

same as -v but turns on the old VBR routine



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

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



How is VBR_q used?

OVER = number of scalefactor bands with distortion that exceeds the
allowed distortion given by the masking thresholds.  OVER is computed
by outer_loop, and the masking thresholds are computed by the
psycho-acoustic model. 

VBR_q = the minimum value of OVER which is to be allowed.
LAME will choose the smallest bitrate for which OVER <= VBR_q.  
(a minimum allowed bitrate can be set with -b.  default=64kbs)

If the frame contains short blocks, then the minimum bitrate is made
much larger since the OVER does not adequately measure distortion
caused by pre-echo.  LAME uses bitrates of at least 160kbs for short
blocks to make sure they sound good.

*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 256kbs encoding.  For example: suppose in the 1kHz
frequency band the psy-model claims 20db 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 20db of
distortion.  Using a fixed 256kbit framesize, LAME could end up
introducing only 2db 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 44.1kHz input files.


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

swap bytes in the input file.  for sorting out little endian/big endian
type problems.  If your encodings sound like static, try this first.