    Type the name of the INFILE (file you wish to convert, a pre-existing .WAV file),
 or press the BROWSE button to the left of the edit box, to browse for the file. Next
 type the name of the OUTFILE (the converted form of the INFILE), or press the BROWSE
 button to the left of the edit box.

    Once you have done this, you can press GO to convert the INFILE to the OUTFILE.
 You can also edit the command line options to suit your needs. Here is the list as
 follows:

  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
    --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
    --ogginput      input file is a Ogg Vorbis file

  Operational options:
    -m <mode>       (s)tereo, (j)oint, (f)orce, (m)ono or (a)auto  
                    default is (s) or (j) depending on bitrate
                    force = force ms_stereo on all frames.
                    auto = jstereo, with varialbe mid/side threshold
    -a              downmix from stereo to mono file for mono encoding
    -d              allow channels to have different blocktypes
    --disptime <arg>print progress report every arg seconds
    --ogg           encode to Ogg Vorbis instead of MP3
    --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>
    --athonly       only use the ATH for masking
    --noath         disable the ATH for masking
    --athlower x    lower the ATH x dB
    --notemp        disable temporal masking effect
    --short         use short blocks
    --noshort       do not use short blocks
    --voice         experimental voice mode
    --preset type   type must be phone, voice, fm, tape, hifi, cd or studio
                    "--preset help" gives some more infos on these

  Verbosity:
    -S              don't print progress report, VBR histograms
    --silent        don't print anything on screen
    --quiet         don't print anything on screen
    --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

  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
    --vbr-mtrh      a merger of old and new (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 (Apex AD600-A DVD/mp3 player)
    -t              disable writing Xing VBR informational tag
    --nohist        disable VBR histogram display

  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
  --cwlimit <freq>    compute tonality up to freq (in kHz) default 8.8717

  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

    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.


    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

    This program is configured to use LAME, but you can use any codec you wish.