AudioFormatReader and WAV files question

Yo. Thicko here again.

I need to get a WAV off disk and into a float array. AudioFormatReader gives me int arrays (int**) but I need floats.

I’m mighty confused. If the WAVs fixed point (32 bit signed ints) how do I convert the samples to floats (in -1 to 1 range)

And whats the story with WAVs anyway? I didn’t realise they came in both fixed or floating format. Anyone know a link that’ll explain all this wav crap?

And the “zero-terminated” thing for the arrays? I take it I just make them 1 bigger than the number of channels I need and plonck a 0 in the last element?

Any help appreciated.

klf

there is some good data on wav here:

http://www.borg.com/~jglatt/tech/wave.htm

he doesn’t deal with floating point wavs though.

to convert int data to float data can’t you just divide the int value by the maximum int for the bit depth? IE a 16bit wav in signed form would have positive numbers divided by 32767 and negative divided by 32768