Hi Kevin. I'm the least-expert guy here I'm sure, but as a fellow newbie I might have some newbie-level ideas for you.
Just to mention something simple: linear interpolating between samples is a huge improvement in reducing aliasing over not interpolating. For instance if your playback math says to get sample 171.25 out of the buffer, use .75 of sample 171 and .25 of sample 172. I don't know the math but my listening tests of tricky test waveforms (eg, a waveform with a fundamental at 100Hz and a 60th harmonic at 6kHz pumped up to 10, 20,30x the power, showed aliasing reduced by about the same amount that quadrupling the buffer sample rate did. I looked into other interpolations besides linear, and they didn't seem effective enough to bother programming. Just to give a feeling for it, if linear eliminated 90% of aliasing, then then more complex interpolation might eliminate 95%. So they might reduce it by half compared to linear, but only a tiny bit better compared to none at all. (The percentages are just meant to be illustrative and actually everything totally depends on your input data.) Another thing is to make sure your input data has been filtered out such that there's no frequencies over the Nyquist frequency. But if you're dealing with a PC or smart phone I'd imagine the computer hardware handles that for you. Once the data is in memory you cannot filter it out, as it will already have morphed into an alias in your input buffer and be "baked in" no matter how you try to play the data back. Finally, listen with good headphones; you'll hear things you probably won't in a room even if you have a good amp and speakers. Frank http://moselle-synth.com/
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