Hello friends! In Week 7 we chose a sample rate of 48khz, because apparently that's what most audio files use nowadays. I don't know very much about audio, but I'm very curious what would have happened if we set our primary buffer's WaveFormat.nSamplesPerSec = 96000
Would it perhaps interpolate between the 2 samples that were actually written into whatever 48khz WAV file we end up loading? If so, is the reason we didn't set nSamplesPerSec to a higher value due to the performance cost of that interpolation?