thinking about audiobuffers and how to improve my Orca DSP technique.
so far I've tried to naively send data to the soundcard at the target sample rate, but even a slight variance in speed would lead to desynchonisation.
I think the way to fix this is to to utilise a buffer: generate samples as fast as you can and then idle for the rest of the time. that way, hopefully, you're never too slow because you frontload the work, and never too fast because you wait before generaing the next buffer

now, I know that audiobuffers aren't a new concept or anything (in fact the are pretty standard), but I like trying the KISS thing first to see if it isn't good enough for the job

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