Apologies if this sounds like a stupid question, I’m new to VST development. I’m trying to build a plugin and I’m currently trying to test it with a sine wav. When I open the .wav file in Audacity it tells me it’s 44100Hz and it’s a 32 bit float. When I load this same file into matlab the first three samples are something like 0.00, 0.0443, 0.0884… However, when I put the same file into Ableton and try to step through the code I find the first three samples of the same file are 0.00000000, 0.00012068315, 0.00048156900… I see this when I peak into the memory in VS and look at it in 32 bit floating point view.
My problem is that I need the audio to have the same sample values as they are in Matlab for my algo to work. Obviously there’s a conversion happening that I have no control of. Can anyone shed any light on this problem and how I should go about fixing it.