I've got a very simple script that opens up a mic in 8khz mono and plays it back live using the default wave out device. The live playback starts off with a 100ms delay which gradually increases until the bufferedwaveprovider hits its limit at 5 seconds and starts chopping off the sound. I've done some digging and it seems that the mic is returning around 8.7khz instead of 8khz - it's about 9% out. The same goes if it's stereo or 11khz or 44khz it is always returning around 9% more samples than i've asked it to. Any ideas on whats going on here? Other mics are working fine. Is there a way to tell naudio to drop extra bytes? I'm thinking of writing some code to work out how many bytes should have arrived based on the waveformat and the time since recording but i'm hoping there is something better i can do about it... Thanks!
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