lame.1

lame 3.97源码.最好的mp3解码和压缩软件

.TH lame 1 "August 03, 2005" "" "LAME audio compressor"
.SH NAME
lame \- create mp3 audio files
.SH SYNOPSIS
lame [options] <infile> <outfile>
.SH DESCRIPTION
.PP
LAME is a program which can be used to create compressed audio files.
(Lame ain't an MP3 encoder).
These audio files can be played back by popular MP3 players such as
mpg123 or madplay.
To read from stdin, use "\-" for <infile>.
To write to stdout, use a "\-" for <outfile>.
.SH OPTIONS
Input options:
.TP
.B \-r
Assume the input file is raw pcm.
Sampling rate and mono/stereo/jstereo must be specified on the command line.
Without
.B \-r,
LAME will perform several
.I fseek()'s
on the input file looking for WAV and AIFF headers.
.br
Might not be available on your release. 
.TP
.B \-x
Swap bytes in the input file or output file when using
.B \-\-decode.
.br
For sorting out little endian/big endian type problems.
If your encodings sounds like static,
try this first.
.br
Without using
.B \-x,
LAME will treat input file as native endian.
.TP
.BI \-s " sfreq"
.I sfreq
= 8/11.025/12/16/22.05/24/32/44.1/48

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

LAME will automatically resample the input file to one of the supported
MP3 samplerates if necessary.
.TP
.BI \-\-bitwidth " n"
Input bit width per sample.
.br
.I n
= 8, 16, 24, 32 (default 16)

Required only for raw PCM input files.
Otherwise it will be determined from the header of the input file.
.TP
.BI \-\-signed
Instructs LAME that the samples from the input are signed (the default).

Required only for raw PCM input files
and only available if LAME was compiled with libsndfile.
.TP
.BI \-\-unsigned
Instructs LAME that the samples from the input are unsigned.

Required only for raw PCM input files
and only available if LAME was compiled with libsndfile.
.TP
.BI \-\-little-endian
Instructs LAME that the samples from the input are in little-endian form.

Required only for raw PCM input files
and only available if LAME was compiled with libsndfile.
.TP
.BI \-\-big-endian
Instructs LAME that the samples from the input are in big-endian form.

Required only for raw PCM input files
and only available if LAME was compiled with libsndfile.
.TP
.B \-\-mp2input
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. 
.TP
.B \-\-mp3input
Assume the input file is a MP3 file.
.br
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.
For stdin or MP3 files which do not end in .mp3 you need to use this switch. 
.TP
.BI \-\-nogap " file1 file2 ..."
gapless encoding for a set of contiguous files
.TP
.BI \-\-nogapout " dir"
output dir for gapless encoding (must precede \-\-nogap)

.PP
Operational options:
.TP
.BI \-m  " mode"
.I mode
= s, j, f, d, m

Joint-stereo is the default mode for stereo files with VBR when
.B \-V
is more than 4 or fixed bitrates of 160kbs or less.
At higher fixed bitrates or higher VBR settings,
the default is stereo. 

.B (s)tereo 
.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.

.B (j)oint stereo
.br
In this mode,
the encoder will make use of a 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.
This will effectively increase the bandwidth if the signal does not
have too much stereo separation,
thus giving a significant gain in encoding quality.

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.

.B (f)orced joint stereo 
.br
This mode will force MS joint stereo on all frames.
It is 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.

.B (d)ual channels
.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 (for example: English in one channel and French in the other).
Using this encoding mode for regular stereo files will result in a
lower quality encoding.

.B (m)ono
.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.
.TP
.B \-a
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. 

This option is only needed in the case of raw PCM stereo input
(because LAME cannot determine the number of channels in the input file).
To encode a stereo PCM input file as mono,
use
.B lame \-m
.I s
.B \-a.

For WAV and AIFF input files,
using
.B \-m
will always produce a mono .mp3 file from both mono and stereo input. 
.TP
.B \-d
Allows the left and right channels to use different block size types. 
.TP
.B \-\-freeformat
Produces a free format bitstream.
With this option,
you can use
.B \-b
with any bitrate higher than 8 kbps.

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.

Tests have shown that the following decoders support free format:
.br
.B FreeAmp
up to 440 kbps
.br
.B in_mpg123
up to 560 kbps
.br
.B l3dec
up to 310 kbps
.br
.B LAME
up to 560 kbps
.br
.B MAD
up to 640 kbps
.TP
.B \-\-decode
Uses LAME for decoding to a wav file.
The input file can be any input type supported by encoding,
including layer II files.
LAME uses a bugfixed version of mpglib for decoding.

If
.B \-t
is used (disable wav header),
LAME will output raw pcm in native endian format.
You can use
.B \-x
to swap bytes order.

This option is not usable if the MP3 decoder was
.B explicitly
disabled in the build of LAME.
.TP
.BI \-t
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.

When
.B \-\-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
.B \-x
to swap bytes.
.TP
.BI \-\-comp " arg"
Instead of choosing bitrate,
using this option,
user can choose compression ratio to achieve.
.TP
.BI \-\-scale " n"
.PD 0
.TP
.BI \-\-scale\-l " n"
.TP
.BI \-\-scale\-r " n"
Scales input (every channel, only left channel or only right channel) by
.I n.
This just multiplies the PCM data (after it has been converted to floating
point) by
.I n. 

.I n
> 1: increase volume
.br
.I n
= 1: no effect
.br
.I n
< 1: reduce volume

Use with care,
since most MP3 decoders will truncate data which decodes to values
greater than 32768.
.PD
.TP
.B \-\-replaygain\-fast
Compute ReplayGain fast but slightly inaccurately.

This computes "Radio" ReplayGain on the input data stream after
user\(hyspecified volume\(hyscaling and/or resampling.

The ReplayGain analysis does
.I 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
.B http://www.replaygain.org/.

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.

This switch is enabled by default.

See also:
.B \-\-replaygain\-accurate, \-\-noreplaygain
.TP
.B \-\-replaygain\-accurate
Compute ReplayGain more accurately and find the peak sample.

This enables decoding on the fly, computes "Radio" ReplayGain on the
decoded data stream,
finds the peak sample of the decoded data stream and stores it in the file.
 
The ReplayGain analysis does
.I 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
.B http://www.replaygain.org/.

 
By default, LAME performs ReplayGain analysis on the input data
(after the user\(hyspecified 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
.B \-\-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:
.RS 8
.IP "*" 4
tests have shown that the difference between the ReplayGain values computed
on the input data and decoded data is usually not greater than 0.5dB,
although the minimum volume difference the human ear can perceive is
about 1.0dB
.IP "*" 4
decoding on the fly significantly slows down the encoding process
.RE
.RS 7

The apparent advantage is that:
.RE
.RS 8
.IP "*" 4
with
.B \-\-replaygain-accurate
the real 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 the sound.
.RE
.RS 7
 
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.
 
This option is not usable if the MP3 decoder was
.B 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:
.B \-\-replaygain-fast, \-\-noreplaygain \-\-clipdetect
.RE
.TP
.B \-\-noreplaygain
Disable ReplayGain analysis.

By default ReplayGain analysis is enabled. This switch disables it.

See also:
.B \-\-replaygain-fast, \-\-replaygain-accurate
.TP
.B \-\-clipdetect
Clipping detection.

Enable
.B \-\-replaygain-accurate
and print a message whether clipping occurs and how far in dB the waveform
is from full scale.
  
This option is not usable if the MP3 decoder was
.B explicitly
disabled in the build of LAME.

See also:
.B \-\-replaygain-accurate
.TP
.B \-\-preset " [fast] type | [cbr] kbps"
Use one of the built-in presets.

Have a look at the PRESETS section below.

.B Warning:
with the current version fast presets might result in too high bitrate
compared to regular presets.

.B \-\-preset help
gives more infos about the the used options in these presets.
.TP
.B \-\-alt-preset " [fast] type | [cbr] kbps"
Use one of the built-in  presets.

This option is deprecated and offers the same as the
.B \-\-preset
option above. Do not use it anymore, it will go away in a later version.
.TP
.B \-\-r3mix
Uses r3mix VBR preset.
.br
See http://www.r3mix.net/ for more details. 
.TP
.B \-\-noasm " type"
Disable specific assembly optimizations (
.B mmx
/
.B 3dnow
/
.B sse
).
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.

.PP
Verbosity:
.TP
.BI \-\-disptime " n"
Set the delay in seconds between two display updates. 
.TP
.B \-\-nohist
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. 
.TP
.B -S
.PD 0
.TP
.B \-\-silent
.TP
.B \-\-quiet
Do not print anything on the screen.
.PD
.TP
.B \-\-verbose
Print a lot of information on the screen.
.TP
.B \-\-help
Display a list of available options.

.PP
Noise shaping & psycho acoustic algorithms:
.TP
.BI -q " qual"
0 <=
.I qual
<= 9

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).

.B -q 0:
.br
use slowest & best possible version of all algorithms.
.B -q 0
and
.B -q 1
are slow and may not produce significantly higher quality.

.B -q 2:
.br
recommended.
Same as
.B -h.

.B -q 5:
.br
default value.
Good speed,
reasonable quality.

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

.B -q 9:
.br
disables almost all algorithms including psy-model.
Poor quality.
.TP
.B -h
Use some quality improvements.
Encoding will be slower,
but the result will be of higher quality.
The behavior is the same as the
.B -q 2
switch.
.br
This switch is always enabled when using VBR. 
.TP
.B -f
This switch forces the encoder to use a faster encoding mode,
but with a lower quality.
The behavior is the same as the
.B -q 7
switch.

Noise shaping will be disabled,
but psycho acoustics will still be computed for bit allocation
and pre-echo detection. 

.PP
CBR (constant bitrate, the default) options:
.TP
.BI -b  " n"
For MPEG1 (sampling frequencies of 32, 44.1 and 48 kHz)
.br
.I n
= 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320

For MPEG2 (sampling frequencies of 16, 22.05 and 24 kHz)
.br
.I n
= 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160

Default is 128 for MPEG1 and 64 for MPEG2. 
.TP
.BI \-\-cbr
enforce use of constant bitrate

.PP
ABR (average bitrate) options:
.TP
.BI \-\-abr " n"
Turns on encoding with a targeted average bitrate of n kbits,
allowing to use frames of different sizes.
The allowed range of
.I n
is 8 - 310,
you can use any integer value within that range.

It can be combined with the
.B -b
and
.B -B
switches like:
.B lame \-\-abr
.I 123
.B -b
.I 64
.B -B
.I 192 a.wav a.mp3
which would limit the allowed frame sizes between 64 and 192 kbits.

The use of
.B -B
is NOT RECOMMENDED.
A 128 kbps CBR bitstream,
because of the bit reservoir,
can actually have frames which use as many bits as a 320 kbps frame.
VBR modes minimize the use of the bit reservoir,
and thus need to allow 320 kbps frames to get the same flexibility
as CBR streams. 

.PP
VBR (variable bitrate) options:
.TP
.B -v
use variable bitrate
.B (\-\-vbr-old)
.TP
.B \-\-vbr-old
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.
.TP
.B \-\-vbr-new
Invokes the newest VBR algorithm.