Use the 'Virtual' variants of afGetSampleFormat, afGetChannels,
afGetVirtualFrameSize in the audiofile library, since it already does
the necessary abstraction for us.

Of course, I've regression tested these changes against my
standard 44100Hz/2ch/16bit wave files and they continue to play
fine.

Files tested:
english.au (Linus Torvalds pronouncing 'Linux' in English)
B01.Red_Bright_Heart.au (Chinese opera, sounds correct to me even though
I don't actually understand the words)

git-svn-id: https://svn.musicpd.org/mpd/trunk@4681 09075e82-0dd4-0310-85a5-a0d7c8717e4f