Ok, found solution for MP3s as well. Since I have music only in flac and
mp3 my solution is complete.

  mp3 flc * bc:5f:f4:bf:3e:70
        # FT:{START=-ss %t}
        [ffmpeg] -loglevel quiet $START$ -i $FILE$ -f wav - | [sox] -D -q -t 
wav - -t flac -C 0 -b 24 - gain -6.9 equalizer 30 0.43q +6.2 equalizer 699 
2.00q +1.0 equalizer 2924 4.11q -1.7 equalizer 6031 5.05q -6.2 equalizer 9825 
4.08q +2.2 equalizer 83 4.24q +1.3 equalizer 209 1.54q -1.2 equalizer 2842 0.1q 
+0.2 equalizer 12443 1.29q +2.7 equalizer 19698 0.46q -11.8

I had to use ffmpeg because, as far as I can see, LAME does not accept
time offsets when decoding
(https://svn.code.sf.net/p/lame/svn/trunk/lame/USAGE). Don't know how
the internal functionality of mp3 transcoding works (when lowering
bitrates) because that is capable of seeking but the convert.conf line
does not show any instruction for that. I could probably be using ffmpeg
for flac files too but I guess there's no point or added value.

If there are mistakes in my solution above, please do point them out.

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