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mp3エンコーダのインストール記録

[root@fitPC2i ~]# yum list | grep lame
yum -y install lame
で入れられないケース。
●mp3エンコーダ『lame』の取得
[[taro@fitPC2i lame]$ wget http://sourceforge.net/projects/lame/files/lame/3.99/lame-3.99.5.tar.gz/download
	:
●『lame』の解凍 ▼
[taro@fitPC2i lame]$ ll
合計 1412
-rw-rw---- 1 taro apache 1445348  2月 29 04:00 2012 lame-3.99.5.tar.gz
[taro@fitPC2i lame]$ tar zxvf lame-3.99.5.tar.gz
	:
[taro@fitPC2i lame]$ ll
合計 1416
drwxr-x--- 15 taro apache    4096  2月 29 03:56 2012 lame-3.99.5
-rw-rw----  1 taro apache 1445348  2月 29 04:00 2012 lame-3.99.5.tar.gz
●『lame』のコンパイル
[taro@fitPC2i lame]$ cd lame-3.99.5
[taro@fitPC2i lame-3.99.5]$ ./configure
	:

[taro@fitPC2i lame-3.99.5]$ su
パスワード:
[root@fitPC2i lame-3.99.5]# make install
Making install in mpglib
	:
[root@fitPC2i lame-3.99.5]# exit
exit
●インストール完了
[root@fitPC2i lame-3.99.5]# cd ~/
[taro@fitPC2i ~]$ which lame
/usr/local/bin/lame
●コマンドの使い方
[taro@fitPC2i ~]$ lame --longhelp
LAME 32bits version 3.99.5 (http://lame.sf.net)

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:
    --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)ual-mono, (m)ono (l)eft (r)ight
                    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 achieve 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
    --swap-channel  swap L/R channels
    -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
    --quiet         don't print anything on screen
    --silent        don't print anything on screen, but fatal errors
    --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)
    -Y              lets LAME ignore noise in sfb21, like in CBR
    -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


  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
    --buffer-constraint <constraint> available values for constraint:
                                     default, strict, maximum

  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, v2.3 tag)
    --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
    --id3v2-utf16   add following options in unicode text encoding
    --id3v2-latin1  add following options in latin-1 text encoding
    --space-id3v1   pad version 1 tag with spaces instead of nulls
    --pad-id3v2     same as '--pad-id3v2-size 128'
    --pad-id3v2-size <value> adds version 2 tag, pad with extra <value> 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.

Misc:
    --license       print License information


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
●例:モノラル、ID3タグ アーチスト名 "TOKYO FM"、固定ビットレート48kbps
[taro@fitPC2i ~]$ lame -b 48 -m m --ta "TOKYO FM" AD13051423.wav
LAME 3.99.5 32bits (http://lame.sf.net)
Resampling:  input 44.1 kHz  output 32 kHz
Using polyphase lowpass filter, transition band: 10968 Hz - 11355 Hz
Encoding AD13051423.wav to AD13051423.mp3
Encoding as 32 kHz single-ch MPEG-1 Layer III (10.7x)  48 kbps qval=3
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA
 99914/99914 (100%)|    7:31/    7:31|    7:34/    7:34|   7.9694x|    0:00
-------------------------------------------------------------------------------
   kbps       mono %     long switch short %
   48.0      100.0        94.1   3.2   2.7
Writing LAME Tag...done
ReplayGain: -15.3dB
モノラルなので固定ビットレートは通常の半分。

『午後のこーだ』に比べ、エンコードはかなり遅い。モノラル1時間で7分31秒。
(Win7稼動時のパフォーマンス評価、プロセッサ2.3、メモリ4.3、ハードディスク4.2)

●環境
CentOS release 6.2 (Final)

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