I've been developing an audio synthesizer and it's pretty much all up and running, but I've run into a bit of an issue while trying to load various audio.
Everything works as intended to the best of my debugging abilities. I think that the problem is not in the implementation, but that wav files can have samples represented in different ways. For example, a 32-bit wav file may use 32-bit integers or 32-bit floats (especially from the range of -1.0 to 1.0). There are similar cases for other bit depths (16 & 24).
My question is, how can these differences be detected? I don't know of any tag that describes the format of how audio samples should be saved. If it is safe to assume certain forms over others, such as floats from -1 to +1 being more common than floats in their full range, then may I get help telling me where to find these preferred formats?