There are lots of tools for dealing with audio files of various formats (MP3, FLAC, AAC, Ogg, WMA, etc), but most of them fall down when it comes to letting you deal with an entire library of music files. (I suppose some of them must be good, but there are so many, you get tired of trying them.)
I've mostly fallen back on scripting-based approaches. When I rip using EAC (which rips to WAV, which has no tagging features), I instruct EAC to create a file that has all the relevant tags (title, artist, genre, etc) embedded in the file name, and I've got a script that feeds these into the flac encoder, creates files using a directory hierachy of the form /artist/album/track.flac, and sets the appropriate flac tags.
I downloaded another script (http://robinbowes.com/projects/flac2mp3) that takes this directory and creates a parallel directory of MP3 files, transcoded from the flac. It uses LAME to do the encoding, using the "--alt-preset standard" settings, and I point iTunes at that directory. So far, so good.
Well, iTunes seems confused by VBR (variable bit rate) MP3s that it didn't encode itself. The symptom is that iTunes thinks the tracks are way longer than they are.
After much searching, the best alternative I came up with was to use a program called 'vbrfix' (http://www.willwap.co.uk/Programs/vbrfix.php), which rewrites the mp3 headers in a way that iTunes is happy with. (It claims to fix other problems too. The documentation sucks, and I had to download the source and build it (with no make file, just guessed, but it wasn't hard), and it does appear to render the MP3s compatible with iTunes.