Time-stamp: "2000-07-26 21:17:00 seiichi"
% lame [options] inputfile [outputfile]
=======================================================================
Examples:
=======================================================================
fixed bit rate jstereo 128kbs encoding:
% lame sample.wav sample.mp3
fixed bit rate jstereo 128kbs encoding, highest quality: (recommended)
% lame -h sample.wav sample.mp3
To disable joint stereo encoding (slightly faster, but less quality at bitrates<=128kbs)
ちょっとはやいがビットレートが128kbpsよりおちる
% lame -m s sample.wav sample.mp3
Fast encode, low quality (no psycho-acoustics)
% lame -f sample.wav sample.mp3
Variable Bitrate (VBR): (use -V n to adjust quality/filesize)
% lame -h -v sample.wav sample.mp3
Note: VBR is currently under heavy development. Right now it can
often result in too much compression. I would recommend using VBR
with a minimum bitrate of 112kbs. This will let LAME increase
the bitrate for difficult-to-encode frames, but prevent LAME from
being too aggressive for simple frames:
注: VBR は、いまのところかなりきつい進展です。現状でそれをつかうとしば
しば圧縮しすぎる結果になります。VBR をつかう場合には最低のビットレート
112kbs をすすめます。これは LAME が「エンコードするのにむずかしい」フ
レームにビットレートを増加させるためで、簡単なフレームへもとても積極的
なエンコードをおこなうためです(訳注:この文わけわか)。
% lame -h -v -b 112 sample.wav sample.mp3
=======================================================================
LOW BITRATES
=======================================================================
At lower bitrates, (like 24kbs per channel), it is recommended that
you use a 16kHz sampling rate combined with lowpass filtering. LAME,
as well as commercial encoders (FhG, Xing) will do this automatically.
However, if you feel there is too much (or not enough) lowpass
filtering, you may need to try different values of the lowpass cutoff
and passband width (--resample, --lowpass and --lowpass-width options).
低ビットレートは(チャネルあたり 24kbs のような)、16kHz サンプリングか
つローパスフィルタをとおした信号におすすめします。LAME は商用エンコー
ダ(FhG, Xing)とおなじようにこれを自動でおこないます。しかし、ローパス
フィルタのききすぎ(たりなさ)をかんじた場合には、べつなローパスカットオ
フとバンドパス幅をためすことができます。
=======================================================================
STREAMING EXAMPLES
=======================================================================
% cat inputfile | lame [options] - - > output
=======================================================================
For more options, just type:
% lame --help
Scripts are included to run lame on multiple files:
bach script: mlame Run "mlame -h" for instructions.
sh script: auenc Run auenc for instructions
=======================================================================
options guide:
=======================================================================
These options are explained in detail below.
Quality related:
-m m/s/j/f mode selection
-k disable all filtering
-d allow block types to differ between channels
--athonly ignore psy-model output, only use masking from the ATH
--voice experimental voice encoding mode
--noshort disable short blocks
Constant Bit Rate (CBR)
-b n set bitrate (8,16,24,...,320)
-h higher quality but slower
-f disable noise shaping. Encodes faster, but lower quality
--freeformat produce a free format bitstream. User must also specify
a bitrate with -b, between 8 and 320 kbs.
Variable Bit Rate (VBR)
-v VBR
--vbr-old use old variable bitrate (VBR) routine
-V n VBR quality setting (0=highest quality, 9=lowest)
-b n specify a minimum allowed bitrate (8,16,24,...,320)
-B n specify a maximum allowed bitrate (8,16,24,...,320)
-F strictly enforce minimum bitrate
-t disable Xing VBR informational tag
--nohist disable display of VBR bitrate histogram
--abr n specify average bitrate desired
Experimental (undocumented): may work better or worse:
-X n try different quality measures (when comparing quantizations)
-Y
-Z
Operational:
-r assume input file is raw PCM
--decode assume input file is an mp3 file, and decode to raw pcm.
-s n input sampling frequency in kHz (for raw PCM input files)
--resample n output sampling frequency
--mp3input input file is an MP3 file. decode using mpglib/mpg123
--ogginput input file is an Ogg Vorbis file. decode using libvorbis
-x swap bytes of input file
-a downmix stereo input file to mono .mp3
-e n/5/c de-emphasis
-p add CRC error protection
-c mark the encoded file as copyrighted
-o mark the encoded file as a copy
-S don't print progress report, VBR histogram
-g run MP3x, the graphical frame analyzer
--strictly-enforce-ISO comply as much as possible to ISO MPEG spec
--ogg Encode using Ogg Vorbis (.ogg) instead of mp3.
id3 tagging:
--tt "title" title of song (max 30 chars)
--ta "artist" artist who did the song (max 30 chars)
--tl "album" album where it came from (max 30 chars)
--ty "year" year in which the song/album was made (max 4 chars)
--tc "comment" additional info (max 30 chars)
--tg "genre" genre of song (name or number)
options not yet described:
--nores disable bit reservoir
--noath disable ATH
--cwlimit specify range of tonality calculation
--lowpass
--lowpass-width
--highpass
--highpass-width
=======================================================================
Detailed description of all options in alphabetical order
アルファベット順のぜんぶのオプションのくわしい説明
=======================================================================
=======================================================================
downmix
=======================================================================
-a
mix the stereo input file to mono and encode as mono.
ステレオ入力をモノラルにまぜて、そしてモノラルでエンコードする。
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 "lame -m s -a"
生 PCM ステレオ入力の場合にのみ必要なオプションです(LAME は入力ファイ
ルのチャンネル数は決定できません)。スレテオ PCM 入力ファイルをモノラル
でエンコードするときには、"lame -m s -a" をつかいます。
For WAV and AIFF input files, using "-m m" will always produce a
mono .mp3 file from both mono and stereo input.
WAV と AIFF 入力ファイルの場合は、"-m m" でモノラル入力でもステレオ入
力でもモノラルの .mp3 ファイルができます。
=======================================================================
average bitrate encoding (aka Safe VBR)
平均ビットレートエンコーディング(aka 安全VBR)
=======================================================================
--abr n
turns on encoding with a targeted average bitrate of n kbits, allowing
to use frames of different sizes. The allowed range of n is 4-310,
you can use any integer value within that range.
目標とする n kbits の平均ビットレートエンコードを有効にして、異ったお
おきさのフレーム使用を許可します。許可する範囲 n は 4から 310で、範囲
内のを整数値を指定できます。
It can be combined with the -b and -B switches like
lame --abr 123 -b 64 -B 192 a.wav a.mp3
which would limit the allowed frame sizes between 64 and 192 kbits.
-b と -B スイッチの次のようにあわせてつかいます。
lame --abr 123 -b 64 -B 192 a.wav a.mp3
これは許可する範囲を 64 から 192 kbits に制限しています。
Using -B is NOT RECOMMENDED. A 128kbs CBR bitstream, because of the
bit resevoir, can actually have frames which use as many bits as a
320kbs frame. VBR modes minimize the use of the bit reservoir, and
thus need to allow 320kbs frames to get the same flexability as CBR
streams.
-B は、おすすめしません。128kbps CBR ビットストームではビット蓄積の理
由で…。VBR モードはビット蓄積をつかうことにより最小化されて、そして
CBR ストリームとおなじ柔軟性を確保して 320kbs フレームがゆるされます。
(訳注:ここも意味不明)
=======================================================================
ATH only
=======================================================================
--athonly
This option causes LAME to ignore the output of the psy-model and
only use masking from the ATH. Might be useful at very high bitrates
or for testing the ATH.
このオプションは LAME を心理モデム出力無効にします。そして、ATH からの
マスキングを使う場合にのみ使用します(訳注:ATH ってなんじゃい)。とても
高いビットレートか ATH のテスト時に役立ちます。
=======================================================================
bitrate
=======================================================================
-b n
For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
The bitrate to be used. Default is 128kbs MPEG1, 80kbs MPEG2.
When used with variable bitrate encodings (VBR), -b specifies the
minimum bitrate to use. This is useful to prevent LAME VBR from
using some very aggressive compression which can cause some distortion
due to small flaws in the psycho-acoustic model.
可変ビットレートエンコーディング(VBR)を使う場合に -b には最低のビット
レートをつかうように指定します。これは LAME VBR が心理音響モデルのなか
のちいさな欠点によるいくつかのひずみを原因とする、いくつかの過剰な圧縮
をふせぐのに役立ちます(訳注:ここくどすぎる)。
=======================================================================
max bitrate
=======================================================================
-B n
For MPEG1 (sampling frequencies of 32, 44.1 and 48kHz)
n = 32,40,48,56,64,80,96,112,128,160,192,224,256,320
For MPEG2 (sampling frequencies of 16, 22.05 and 24kHz)
n = 8,16,24,32,40,48,56,64,80,96,112,128,144,160
Maximum allowed bitrate when using VBR.
=======================================================================
copyright
=======================================================================
-c
mark the encoded file as copyrighted
=======================================================================
block type control
=======================================================================
-d
Allows the left and right channels to use different block types.
Normally this is not allowed, only because the FhG encoder does
not seem to allow it either. If anyone finds a sample where -d
produces better results, let me know. (mt@sulaco.org)
左と右チャンネルで異なるブロックタイプをつかうのを許可します。ふつう、
これはゆるされませんが、FhG エンコーダだけは…(訳注:わけわか)。どなた
か -d でいい結果の例を見つけたら、わたしにおしえてください。
=======================================================================
mpglib decode capability
=======================================================================
--decode
This just uses LAME's mpg123/mpglib interface to decode an MP3 file to
a raw pcm file. The input file must be an MP3 file, and the raw pcm
data will be written to the output file in native endian format.
Under linux, on i386, to convert the output file to a wav, use:
これは LAME の mpg123/mpglib インタフェースで MP3 ファイルを生 pcm ファ
イルにデコードすることです。入力ファイルはかならず MP3 ファイルで、そ
して生 pcm データは i386 上の linux では native endian 形式で出力され
ます。出力を wav ファイルに変換するには:
% lame --decode input.mp3 output.pcm
% sox -c 2 -t raw -r 44100 -s -w output.pcm output.wav
(assuming output.pcm came from a stereo, 44.1khz mp3 file)
=======================================================================
de-emphasis
=======================================================================
-e n/5/c
n = (none, default)
5 = 0/15 microseconds
c = citt j.17
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.
A better solution would be to apply the de-emphasis with a standalone
utility before encoding, and then encode without -e.
=======================================================================
fast mode
=======================================================================
-f
disable noise shaping. Encodes faster, but lower quality.
Psycho acoustics are computed for bit allocation and
pre-echo detection.
=======================================================================
strictly enforce VBR minimum bitrate
=======================================================================
-F
strictly enforce VBR minimum bitrate. With out this optioni, the minimum
bitrate will be ignored for passages of analog silence.
=======================================================================
free format bitstreams
=======================================================================
--freeformat
LAME will produce a fixed bitrate, free format bitstream.
User must specify the desired bitrate in kbs, which can
be any integer between 8 and 320.
Not supported by most decoders. Decoders only required to support
up to 320kbs.
Decoders which can handle free format:
supports up to
"lame --decode" 560kbs
Freeamp: 440kbs
l3dec: 310kbs
=======================================================================
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
=======================================================================
--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.