希望把檔案轉小

[雪風]

有什麼轉檔MP3 to MP3的因為之前用CD抓下來的是MP3 320KBPS的可是音樂越來越多我的IPOD MINI越來越不夠用所以我想要把MP3同一改成192KBPS的我發現ITUNES轉出來不是很好可是我也不知道還有什麼軟體可以轉的希望大大可以提供一點意見

[阿榮]

FOOBAR 2000如何?

FOOBAR 2000 + LAME.....

[雪風]

FOOBAR 2000如何?

FOOBAR 2000 + LAME.....

FOOBAR我有用來聽歌

LAME我也知道但我不會用要怎麼執行阿

[阿榮]

關鍵字都給你了.. 你自己不搜尋一下?

不只在這個站.. 其他友站(PCVCD或是BEARHONEY)也都去搜尋一下...

[雪風]

關鍵字都給你了.. 你自己不搜尋一下?

不只在這個站.. 其他友站(PCVCD或是BEARHONEY)也都去搜尋一下...

我之前爬過文了有找到一些可是並不符合我要的
我還是看不懂FOOBAR要怎麼跟LAME配合我也用字典查了很辛苦還是不知道怎麼把這2個軟體合在一起用

[oScARSr]

你直接執行lame.exe -?，
就會出現用法了：

代碼: 選擇全部

D:\My Documents\My Tools\Codec\Lame396>lame -?
LAME version 3.96 MMX  (http://www.mp3dev.org/)

usage: lame [options] <infile> [outfile]

    <infile> and/or <outfile> 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 II  file
    --mp3input      input file is a MPEG Layer III file
    --nogap <file1> <file2> <...>
                    gapless encoding for a set of contiguous files
    --nogapout <dir>
                    output dir for gapless encoding (must precede --nogap)
    --nogaptags     allow the use of VBR tags in gapless encoding

  Operational options:
    -m <mode>       (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  <arg>   choose bitrate to achive a compression ratio of <arg>
    --scale <arg>   scale input (multiply PCM data) by <arg>
    --scale-l <arg> scale channel 0 (left) input (multiply PCM data) by <arg>
    --scale-r <arg> scale channel 1 (right) input (multiply PCM data) by <arg>
    --replaygain-fast   compute RG fast but slightly inaccurately (default)
    --replaygain-accurate   compute RG more accurately and find the peak sample
    --noreplaygain  disable 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 <arg>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 & psycho acoustic algorithms:
    -q <arg>        <arg> = 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 <bitrate>    set the bitrate in kbps, default 128 kbps
    --cbr           enforce use of constant bitrate

  ABR options:
    --abr <bitrate> 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 <bitrate>    specify minimum allowed bitrate, default  32 kbps
    -B <bitrate>    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 [0-4]
    --athlower x    lowers ATH by x dB
    --athaa-type n  ATH auto adjust types 1-3, else no adjustment
    --athaa-loudapprox n   n=1 total energy or n=2 equal loudness curve
    --athaa-sensitivity x  activation 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 <freq>  compute tonality up to freq (in kHz) default 8.8717
    --notemp        disable temporal masking effect
    --nssafejoint   M/S switching criterion
    --nsmsfix <arg> M/S switching tuning [effective 0-3.5]
    --interch x     adjust inter-channel masking ratio
    --ns-bass x     adjust masking for sfbs  0 -  6 (long)  0 -  5 (short)
    --ns-alto x     adjust masking for sfbs  7 - 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,y  short 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 [n]          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 <emp>        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 <freq>        frequency(kHz), lowpass filter cutoff above freq
  --lowpass-width <freq>  frequency(kHz) - default 15% of lowpass freq
  --highpass <freq>       frequency(kHz), highpass filter cutoff below freq
  --highpass-width <freq> frequency(kHz) - default 15% of highpass freq
  --resample <sfreq>  sampling frequency of output file(kHz)- default=automatic


  ID3 tag options:
    --tt <title>    audio/song title (max 30 chars for version 1 tag)
    --ta <artist>   audio/song artist (max 30 chars for version 1 tag)
    --tl <album>    audio/song album (max 30 chars for version 1 tag)
    --ty <year>     audio/song year of issue (1 to 9999)
    --tc <comment>  user-defined text (max 30 chars for v1 tag, 28 for v1.1)
    --tn <track>    audio/song track number (1 to 255, creates v1.1 tag)
    --tg <genre>    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-errors  ignore 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 <type>     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 <instructions> disable assembly optimizations for mmx/3dnow/sse



MPEG-1   layer III sample frequencies (kHz):  32  48  44.1
bitrates (kbps): 32 40 48 56 64 80 96 112 128 160 192 224 256 320

MPEG-2   layer III sample frequencies (kHz):  16  24  22.05
bitrates (kbps):  8 16 24 32 40 48 56 64 80 96 112 128 144 160

MPEG-2.5 layer III sample frequencies (kHz):   8  12  11.025
bitrates (kbps):  8 16 24 32 40 48 56 64 80 96 112 128 144 160

[雪風]

Microsoft Windows XP [版本 5.1.2600]
(C) Copyright 1985-2001 Microsoft Corp.

F:\>lame4\lame
LAME 32bits version 3.97 (beta 1, Sep 12 2005) (http://www.mp3dev.org/)

usage: lame4\lame [options] <infile> [outfile]

<infile> and/or <outfile> can be "-", which means stdin/stdout.

Try:
"lame4\lame --help" for general usage information
or:
"lame4\lame --preset help" for information on suggested predefined setti
ngs
or:
"lame4\lame --longhelp"
or "lame4\lame -?" for a complete options list

為什麼我的跟你用的不一樣難道我拿裡用錯了嗎

[阿榮]

1. foobar2000 -> preferences -> components -> diskwriter -> 把所有參數設一設...

2. 在FOOBAR 2000的歌曲列表上把要轉換的歌選一選 -> 按滑鼠右鍵 -> CONVERT -> RUN CONVERSION

[雪風]

1. foobar2000 -> preferences -> components -> diskwriter -> 把所有參數設一設...

2. 在FOOBAR 2000的歌曲列表上把要轉換的歌選一選 -> 按滑鼠右鍵 -> CONVERT -> RUN CONVERSION

這樣轉出來不是WAV嗎??

[阿榮]

1. foobar2000 -> preferences -> components -> diskwriter -> 把所有參數設一設...

2. 在FOOBAR 2000的歌曲列表上把要轉換的歌選一選 -> 按滑鼠右鍵 -> CONVERT -> RUN CONVERSION

[阿榮]

不然再把你所有的CD拿出來用EAC重頭開始轉....