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A.C
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LAME MP3 Encoder

LAME 3.98 beta 7 April 6 2008

 Robert Hegemann:
  • libmp3lame API: allow frontends to separately retrieve LAME/Xing and ID3 data, because the old library automatism makes it impossible to make fully buffered encodes.
  • libmp3lame API: added some experimental unicode ID3 tagging code.
  • frontends: write itself final ID3 tags and LAME/Xing header frame
  • lame_enc.dll: writes itself final LAME/Xing header frame
  • Latest changes to the new VBR psymodel:
    •uses a different spreading function
    bug-fix for out-of-bounds array access (program stack corruption possible)

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舊 2008-04-08, 11:24 PM #1
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說明列表

代碼:
LAME 32bits version 3.98 (beta 7, Apr  6 2008) (http://www.mp3dev.org/)

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

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

RECOMMENDED:
    lame -V2 input.wav output.mp3

OPTIONS:
  Input options:
    --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

  Input options for RAW PCM:
    -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)
    --signed        input is signed (default)
    --unsigned      input is unsigned
    --little-endian input is little-endian (default)
    --big-endian    input is big-endian


  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
    --flush         flush output stream as soon as possible
    --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 n            quality setting for VBR.  default n=4
                    0=high quality,bigger files. 9=smaller files
    -v              the same as -V 4
    --vbr-old       use old variable bitrate (VBR) routine
    --vbr-new       use new variable bitrate (VBR) routine (default)
    -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:
    --temporal-masking x   x=0 disables, x=1 enables 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:
    -Y              lets LAME ignore noise in sfb21, like in CBR


  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:
  --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[/total]>   audio/song track number and (optionally) the total
                           number of tracks on the original recording. (track
                           and total each 1 to 255. just the track number
                           creates v1.1 tag, providing a total forces v2.0).
    --tg <genre>    audio/song genre (name or number in list)
    --ti <file>     audio/song albumArt (jpeg/png/gif file, 128KB max, v2.3)
    --tv <id=value> user-defined frame specified by id and value (v2.3 tag)
    --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, default)
                         3,4 = High priority (HIGH_PRIORITY_CLASS))
    Note: Calling '--priority' without a parameter will select priority 0.

Misc:
    --license       print License information



  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
 

此文章於 2008-04-13 09:07 PM 被 A.C 編輯.
舊 2008-04-13, 08:59 PM #2
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A.C離線中  
A.C
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preset help

代碼:
LAME 32bits version 3.98 (beta 7, Apr  6 2008) (http://www.mp3dev.org/)


The --preset switches are designed to provide the highest possible quality.

They have for the most part been subject to and tuned via rigorous double blind
listening tests to verify and achieve this objective.

These are continually updated to coincide with the latest developments that
occur and as a result should provide you with nearly the best quality
currently possible from LAME.

To activate these presets:

   For VBR modes (generally highest quality):

     "--preset medium" This preset should provide near transparency
                             to most people on most music.

     "--preset standard" This preset should generally be transparent
                             to most people on most music and is already
                             quite high in quality.

     "--preset extreme" If you have extremely good hearing and similar
                             equipment, this preset will generally provide
                             slightly higher quality than the "standard"
                             mode.

   For CBR 320kbps (highest quality possible from the --preset switches):

     "--preset insane"  This preset will usually be overkill for most
                             people and most situations, but if you must
                             have the absolute highest quality with no
                             regard to filesize, this is the way to go.

   For ABR modes (high quality per given bitrate but not as high as VBR):

     "--preset <kbps>"  Using this preset will usually give you good
                             quality at a specified bitrate. Depending on the
                             bitrate entered, this preset will determine the
                             optimal settings for that particular situation.
                             While this approach works, it is not nearly as
                             flexible as VBR, and usually will not attain the
                             same level of quality as VBR at higher bitrates.

The following options are also available for the corresponding profiles:

   <fast>        standard
   <fast>        extreme
                 insane
   <cbr> (ABR Mode) - The ABR Mode is implied. To use it,
                      simply specify a bitrate. For example:
                      "--preset 185" activates this
                      preset and uses 185 as an average kbps.

   "fast" - Enables the new fast VBR for a particular profile. The
            disadvantage to the speed switch is that often times the
            bitrate will be slightly higher than with the normal mode
            and quality may be slightly lower also.

   "cbr"  - If you use the ABR mode (read above) with a significant
            bitrate such as 80, 96, 112, 128, 160, 192, 224, 256, 320,
            you can use the "cbr" option to force CBR mode encoding
            instead of the standard abr mode. ABR does provide higher
            quality but CBR may be useful in situations such as when
            streaming an mp3 over the internet may be important.

    For example:

    "--preset fast standard <input file> <output file>"
 or "--preset cbr 192 <input file> <output file>"
 or "--preset 172 <input file> <output file>"
 or "--preset extreme <input file> <output file>"


A few aliases are available for ABR mode:
phone => 16kbps/mono        phon+/lw/mw-eu/sw => 24kbps/mono
mw-us => 40kbps/mono        voice => 56kbps/mono
fm/radio/tape => 112kbps    hifi => 160kbps
cd => 192kbps               studio => 256kbps
舊 2008-04-13, 09:06 PM #3
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LAME 3.98 beta 8

很快的,LAME 3.98 beta 8 出來了。

LAME 3.98 beta 8 April 13 2008
 Robert Hegemann:
  • LAME now accepts a floating point value in the range [0,...,10[ as VBR quality setting, like -V5.678
  • Found and fixed some suspicious code in additive masking calculation for VBR-NEW
  • bug-fix:experimental code was defaulted by accident for VBR-NEW
  • fix for some endianess problem on big-endian machines

比較重要的是第一點更新,VBR 增加小數點設定,可更自由的選擇位元率。
舊 2008-04-16, 11:31 PM #4
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用 VBR 模式轉檔的使用者,可以試試加上 Z2。VBR 的小數點範圍為 0~9.999999。
舊 2008-04-17, 12:24 PM #5
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LAME 3.98 Final

經過近兩年的測試與開發,LAME 3.98 終於釋出正式版。正式版的 VBR 模式預設為 VBR-New。

下載/原始碼/網頁
舊 2008-07-04, 11:01 PM #6
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蝦米碗糕
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您的住址: 潛水中
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我頭一次看到

能這樣拉網頁

東西真的學不完

感謝分享編碼器




.

此文章於 2008-07-05 12:48 PM 被 蝦米碗糕 編輯.
舊 2008-07-05, 12:44 PM #7
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pioneersu
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Smile

http://www.free-codecs.com/download...Show_Filter.htm
幫貼 3.98正式版 DirectShow 用的 encoder...
舊 2008-07-05, 11:03 PM #8
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MyAngelism
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我也用beta好久了…

一直都是用EAC+Lame壓mp3,非常棒的encoder。
舊 2008-07-07, 11:31 AM #9
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