WinFF is a frontend for avconv (well, strictly speaking it's a frontend for ffmpeg, but in Ubuntu 'ffmpeg' just links to avconv).
Go ahead and use WinFF, ignore the dumb message.
These are words of truth. But if you want to use avconv (as always, command-line options are faster & more adaptable... once you get past the relatively steep learning curve)...
Basic usage is
Code:
avconv -i <input options> input <output options> output
In your case
Code:
avconv -i input.ogg -c:a libmp3lame -q:a 4 output.mp3
will give you an MP3 file with a variable bit rate in the 140-185 kbits/second range; for a higher quality use a lower -q:a number. This page has a brief guide to encoding MP3s (just replace 'ffmpeg' with 'avconv', they're the same for most practical purposes). Of course, encoding from one lossy format (Ogg Vorbis) to another (MP3) is not ideal, you should use a lossless (WAV, FLAC, and a few others) source when possible.
To do every file in a directory, you should use a for loop.
Code:
for f in *.ogg; do avconv -i $f -c:a libmp3lame -q:a 4 ${f/%ogg/mp3}; done
This will convert every file ending in '.ogg' to an MP3 file, while leaving the originals in place. Change the -q:a option if you want... I'd check out the quality on a single file, then see what you want to do.
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