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歌曲转换格式问题

歌曲转换格式问题

虽然感觉在这里问不太合适。。。但好像只可以在这问了。。。

偶把CD转成320码率AAC和320码率MP3   哪种编码方式音质比较好啊???

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相对来说,AAC 理论上比 MP3 默认编码方式要好。但是如果懂细节调整的话,那就不一样了。

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细节调整???不是只有码率 大小可以选嘛~~~还有什么可以选择吗???

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LAME 32bits version 3.97 (http://www.mp3dev.org/)

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

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

RECOMMENDED:
    lame -V2 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)
    --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>
    --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:
    -a              downmix from stereo to mono file for mono encoding
    -m <mode>       (j)oint, (s)imple, (f)orce, (d)dual-mono, (m)ono
                    default is (j) or (s) depending on bitrate
                    joint  = joins the best possible of MS and LR stereo
                    simple = force LR stereo on all frames
                    force  = force MS stereo on all frames.
    --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
    --comp  <arg>   choose bitrate to achive a compression ratio of <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
    --freeformat    produce a free format bitstream
    --decode        input=mp3 file, output=wav
    -t              disable writing wav header when using --decode


  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


  PSY related:
    --short         use short blocks when appropriate
    --noshort       do not use short blocks
    --allshort      use only short blocks
    --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


  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]          currently no effects


  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

- -看得懂么?
中学生,副教授。博不精,专不透。名虽扬,实不够。高不成,低不就。
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不为如“腾讯”之流的渣公司的软件提供技术支持服务

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偶不怎么懂洋文。。。

麻烦直接说下哪个好点~~~通过怎么调整才可以转出最好。。。

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嘛- -首先,位比率不是绝对的音质表现,比如 MP3 还有一个“质量”的选项。通常情况下,同等位比率的情况下,MP3 音质就不是太突出了0 0
中学生,副教授。博不精,专不透。名虽扬,实不够。高不成,低不就。
——————————————
不为如“腾讯”之流的渣公司的软件提供技术支持服务

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那看来我还是全转AAC好了

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