This converts 16/24 bit FLAC, M4A, MP4, SHN or WAV audio files to (VBR 256kbps) or (CBR 320kbps) MP3 using lame encoding.
If content is already MP3, it does not re-encode. However, it still proceeds with additional processing.
Aside from converting audio (if necessary), it proceeds with the following:
It looks to folder.jpg with the directory or the artwork image embedded within the first music file.
Then it optimizes the image, scaling it to 500px while maintaining aspect ratio.
This results in a small-sized jpeg image that is embedded within each MP3 file.
It preserves the text tags from each parent FLAC or ALAC file respectively, encoding them to ID3v2 version 4 specification.
It asks to confirm the artist tag.
It asks to confirm the album tag, which it assumes is the parent folder of the current path. This is logically how studio albums are organized. This also works for live music where the folders are named and sorted by YEAR.MON.DAY Venue, City, StateCode - CollectionName.
Install ffmpeg and lame, do:
$ sudo apt-get install ffmpeg lame libmp3lame0
From this path, symlink mct to user's bin directory:
ln -s `pwd`/mct ~/bin/
To install updates; from this path, run:
git pull
Open the path containing the files you wish to convert, then run:
mct