0xAA55 发表于 2014-2-4 00:43:59

MP3转码工具:Lame下载(命令行程序)

Lame主页:
http://www.mp3dev.org/
http://lame.sourceforge.net/

下载链接在1楼底部。

这个是Lame转码工具,是个命令行程序,使用需要有命令行的使用基础。调用:
lame 命令参数 输入文件 -o 输出文件
我们经常需要用到的编码WAV为MP3命令参数是:
设置音质:
-qN   N是音质参数,0到9。0是音质最好,9是编码最快。
这个命令指定了编码的侧重点,你要编码快一些还是音质好一些。
举例:
-q0(这个是音质好的)。
-q9(这个是速度快的)。
这里对音质的影响和位率不一样,它不影响文件大小。
-bN   N是位率。最低8,最高320
这里对音质的影响非常大。位率8的音质不敢恭维,但是文件非常小。位率320的音质最好,但是产生的MP3文件一般都比较大。
一首6分钟的歌,如果位率是320,文件大小应该是16 MB左右。百度音乐下载到的“高品质”一般指的就是位率。

你要是实在拿不定注意,你就直接输-q0 -b320就行了。这个编码的是最好音质。
举例:
编码1.WAV到1.MP3,最好的音质:(最慢编码,最大文件)
lame -q0 -b320 1.WAV -o 1.MP3
编码2.WAV到2.MP3,最快编码,最小文件:(最烂音质)
lame -q9 -b8 2.WAV -o 2.MP3

这里我们用到的解码MP3为WAV的参数是:
--decode
解码很简单,这样的命令:
lame --decode 输入文件 -o 输出文件
举例:
将1.MP3解码为1.WAV:
lame --decode 1.MP3 -o 1.WAV
就可以了。


FAQ:
1、为什么提示文件没找到?
答:长文件名如果路径包含空格,那么你就必须用半角双引号将路径括起来,如下:
把D:\Halo 2 CD\1.WAV编码为D:\Halo 2 CD\1.MP3:
lame -q0 -b320 D:\Halo 2 CD\1.WAV -o D:\Halo 2 CD\1.MP3
这是不行的!
必须写成这样:
lame -q0 -b320 "D:\Halo 2 CD\1.WAV" -o "D:\Halo 2 CD\1.MP3"
这就OK啦。


exe下载地址(32位):(版本:3.99.5)
http://www.rarewares.org/files/mp3/lame3.99.5.zip
exe下载地址(64位):(版本:3.99.5)
http://www.rarewares.org/files/mp3/lame3.99.5-64.zip
论坛备份:(版本貌似是3.99)
http://pan.baidu.com/s/1bnqwRrL
源码下载地址:(版本:3.99.5)
http://nchc.dl.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz

0xAA55 发表于 2014-2-4 00:46:59

Lame命令行

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<div style="float: left; white-space: pre; line-height: 1; background: #FFFFFF; "><span class="sc0">Microsoft Windows [版本 6.1.7601]
版权所有 (c) 2009 Microsoft Corporation。保留所有权利。

C:\Users\0xAA55&gt;lame
LAME version 3.97 MMX (alpha 2, May 28 2004 06:00:28) (</span><span class="sc64">http://www.mp3dev.org/)</span><span class="sc0">

warning: alpha versions should be used for testing only

usage: lame &lt;infile&gt;

    &lt;infile&gt; and/or &lt;outfile&gt; can be "-", which means stdin/stdout.

Try:
   "lame --help"         for general usage information
or:
   "lame --preset help"    for information on suggested predefined settings
or:
   "lame --longhelp"
or "lame -?"            for a complete options list


C:\Users\0xAA55&gt;lame --longhelp
LAME version 3.97 MMX (alpha 2, May 28 2004 06:00:28) (</span><span class="sc64">http://www.mp3dev.org/)</span><span class="sc0">

warning: alpha versions should be used for testing only

usage: lame &lt;infile&gt;

    &lt;infile&gt; and/or &lt;outfile&gt; can be "-", which means stdin/stdout.

RECOMMENDED:
    lame -h input.wav output.mp3

OPTIONS:
Input options:
    -r            input is raw pcm
    -x            force byte-swapping of input
    -s sfreq      sampling frequency of input file (kHz) - default 44.1 kHz
    --bitwidth w    input bit width is w (default 16)
    --mp1input      input file is a MPEG Layer I   file
    --mp2input      input file is a MPEG Layer IIfile
    --mp3input      input file is a MPEG Layer III file
    --nogap &lt;file1&gt; &lt;file2&gt; &lt;...&gt;
                  gapless encoding for a set of contiguous files
    --nogapout &lt;dir&gt;
                  output dir for gapless encoding (must precede --nogap)
    --nogaptags   allow the use of VBR tags in gapless encoding

Operational options:
    -m &lt;mode&gt;       (s)tereo, (j)oint, (f)orce, (m)ono
                  default is (s) or (j) depending on bitrate
                  force = force ms_stereo on all frames.
    -a            downmix from stereo to mono file for mono encoding
    --freeformat    produce a free format bitstream
    --decode      input=mp3 file, output=wav
    -t            disable writing wav header when using --decode
    --comp&lt;arg&gt;   choose bitrate to achive a compression ratio of &lt;arg&gt;
    --scale &lt;arg&gt;   scale input (multiply PCM data) by &lt;arg&gt;
    --scale-l &lt;arg&gt; scale channel 0 (left) input (multiply PCM data) by &lt;arg&gt;
    --scale-r &lt;arg&gt; scale channel 1 (right) input (multiply PCM data) by &lt;arg&gt;
    --replaygain-fast   compute RG fast but slightly inaccurately (default)
    --replaygain-accurate   compute RG more accurately and find the peak sample
    --noreplaygaindisable ReplayGain analysis
    --clipdetect    enable --replaygain-accurate and print a message whether
                  clipping occurs and how far the waveform is from full scale
    --preset type   type must be "medium", "standard", "extreme", "insane",
                  or a value for an average desired bitrate and depending
                  on the value specified, appropriate quality settings will
                  be used.
                  "--preset help" gives more info on these


Verbosity:
    --disptime &lt;arg&gt;print progress report every arg seconds
    -S            don't print progress report, VBR histograms
    --nohist      disable VBR histogram display
    --silent      don't print anything on screen
    --quiet         don't print anything on screen
    --brief         print more useful information
    --verbose       print a lot of useful information

Noise shaping &amp; psycho acoustic algorithms:
    -q &lt;arg&gt;      &lt;arg&gt; = 0...9.Default-q 5
                  -q 0:Highest quality, very slow
                  -q 9:Poor quality, but fast
    -h            Same as -q 2.   Recommended.
    -f            Same as -q 7.   Fast, ok quality


CBR (constant bitrate, the default) options:
    -b &lt;bitrate&gt;    set the bitrate in kbps, default 128 kbps
    --cbr         enforce use of constant bitrate

ABR options:
    --abr &lt;bitrate&gt; specify average bitrate desired (instead of quality)

VBR options:
    -v            use variable bitrate (VBR) (--vbr-old)
    --vbr-old       use old variable bitrate (VBR) routine
    --vbr-new       use new variable bitrate (VBR) routine
    -V n            quality setting for VBR.default n=4
                  0=high quality,bigger files. 9=smaller files
    -b &lt;bitrate&gt;    specify minimum allowed bitrate, default32 kbps
    -B &lt;bitrate&gt;    specify maximum allowed bitrate, default 320 kbps
    -F            strictly enforce the -b option, for use with players that
                  do not support low bitrate mp3
    -t            disable writing LAME Tag
    -T            enable and force writing LAME Tag


ATH related:
    --noath         turns ATH down to a flat noise floor
    --athshort      ignore GPSYCHO for short blocks, use ATH only
    --athonly       ignore GPSYCHO completely, use ATH only
    --athtype n   selects between different ATH types
    --athlower x    lowers ATH by x dB
    --athaa-type nATH auto adjust types 1-3, else no adjustment
    --athaa-loudapprox n   n=1 total energy or n=2 equal loudness curve
    --athaa-sensitivity xactivation offset in -/+ dB for ATH auto-adjustment

PSY related:
    --short         use short blocks when appropriate
    --noshort       do not use short blocks
    --allshort      use only short blocks
    --cwlimit &lt;freq&gt;compute tonality up to freq (in kHz) default 8.8717
    --notemp      disable temporal masking effect
    --nssafejoint   M/S switching criterion
    --nsmsfix &lt;arg&gt; M/S switching tuning
    --interch x   adjust inter-channel masking ratio
    --ns-bass x   adjust masking for sfbs0 -6 (long)0 -5 (short)
    --ns-alto x   adjust masking for sfbs7 - 13 (long)6 - 10 (short)
    --ns-treble x   adjust masking for sfbs 14 - 21 (long) 11 - 12 (short)
    --ns-sfb21 x    change ns-treble by x dB for sfb21
    --shortthreshold x,yshort block switching threshold, x for L/R/M channel,
y for S channel
Noise Shaping related:
    --substep n   use pseudo substep noise shaping method types 0-2


experimental switches:
    -X n[,m]      selects between different noise measurements
                  n for long block, m for short. if m is omitted, m = n
    -Y            lets LAME ignore noise in sfb21, like in CBR
    -Z           toggles the scalefac-scale and subblock gain feature on
                  if n is set and minus, only scalefac-scale is enabled


MP3 header/stream options:
    -e &lt;emp&gt;      de-emphasis n/5/c(obsolete)
    -c            mark as copyright
    -o            mark as non-original
    -p            error protection.adds 16 bit checksum to every frame
                  (the checksum is computed correctly)
    --nores         disable the bit reservoir
    --strictly-enforce-ISO   comply as much as possible to ISO MPEG spec

Filter options:
    -k            keep ALL frequencies (disables all filters),
                  Can cause ringing and twinkling
--lowpass &lt;freq&gt;      frequency(kHz), lowpass filter cutoff above freq
--lowpass-width &lt;freq&gt;frequency(kHz) - default 15% of lowpass freq
--highpass &lt;freq&gt;       frequency(kHz), highpass filter cutoff below freq
--highpass-width &lt;freq&gt; frequency(kHz) - default 15% of highpass freq
--resample &lt;sfreq&gt;sampling frequency of output file(kHz)- default=automatic


ID3 tag options:
    --tt &lt;title&gt;    audio/song title (max 30 chars for version 1 tag)
    --ta &lt;artist&gt;   audio/song artist (max 30 chars for version 1 tag)
    --tl &lt;album&gt;    audio/song album (max 30 chars for version 1 tag)
    --ty &lt;year&gt;   audio/song year of issue (1 to 9999)
    --tc &lt;comment&gt;user-defined text (max 30 chars for v1 tag, 28 for v1.1)
    --tn &lt;track&gt;    audio/song track number (1 to 255, creates v1.1 tag)
    --tg &lt;genre&gt;    audio/song genre (name or number in list)
    --add-id3v2   force addition of version 2 tag
    --id3v1-only    add only a version 1 tag
    --id3v2-only    add only a version 2 tag
    --space-id3v1   pad version 1 tag with spaces instead of nulls
    --pad-id3v2   pad version 2 tag with extra 128 bytes
    --genre-list    print alphabetically sorted ID3 genre list and exit
    --ignore-tag-errorsignore errors in values passed for tags

    Note: A version 2 tag will NOT be added unless one of the input fields
    won't fit in a version 1 tag (e.g. the title string is longer than 30
    characters), or the '--add-id3v2' or '--id3v2-only' options are used,
    or output is redirected to stdout.


MS-Windows-specific options:
    --priority &lt;type&gt;   sets the process priority:
                               0,1 = Low priority (IDLE_PRIORITY_CLASS)
                               2 = normal priority (NORMAL_PRIORITY_CLASS, defau
lt)
                               3,4 = High priority (HIGH_PRIORITY_CLASS))
    Note: Calling '--priority' without a parameter will select priority 0.


Platform specific:
    --noasm &lt;instructions&gt; disable assembly optimizations for mmx/3dnow/sse



MPEG-1   layer III sample frequencies (kHz):324844.1
bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320

MPEG-2   layer III sample frequencies (kHz):162422.05
bitrates (kbps):8 16 24 32 40 48 56 64 80 96 112 128 144 160

MPEG-2.5 layer III sample frequencies (kHz):   81211.025
bitrates (kbps):8 16 24 32 40 48 56 64 80 96 112 128 144 160


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