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Originally Posted by TweakHead
Your last sentence kind of confirms what I was trying to say with the point numbers, I can picture a scenario where a perfectly contoured waveform in analogue wouldn't translate well in digital due to it not being aligned with the points and where that signal would happen to translate into that grid - this is interesting and honestly haven't thought much about it just yet.
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This is very simple thankfully. Any analog waveform can be captured and almost perfectly reproduced with digital sampling as long as the sample rate is 2x the lowest frequency. But like was pointed out, there is a small catch. If you were to loop the digital signal it's not the same thing as looping the analog signal. So if the original analog signal is 2.5 sample points long, looping the digital signal will produce something different than looping the analog signal. The FFT would also be different because in digtal you could not take 2.5 sample point length FFT. Once you oversample though, there is no problem. This is also why wavetable oscillators of most frequencies need to be longer than one analog cycle and possibly one reason why extreme pitch resolution requires a lot of processing power - the digital oscillators can't be periodic even if the resulting analog waveform is.
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No developer here either, but it feels like some algorithm could be implemented to make a good guess based on some results. Meaning, that the estimative could be almost spot on based on the behaviour of the wave at certain points. Always thought that was what interpolation means and does to the waveform. But I'm guessing efforts in that are made to work within a minor error margin, but we're never talking about absolute precision here, just not so good guesses and better guesses - and we still have to factor further latency introduced by the processing of this somehow, I guess.
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I am not a developer (far from it, lol) either. There certainly seem to be a lot of unexplored waters in audio. In the end, I guess it's not that big of a market.
Anyway, oversampling or in other words interpolation with sinc function (also goes by the name of low pass filter) gives you more points along the analog signal - well extremely close anyway (analog filters are usually worse than digital ones, and won't be linear phase). But once again, oversampling affects the frequency domain so that higher frequencies can be produced, the frequency resolution remains the same. Only way to increase frequency resolution is to have a longer signal.
If you want to know how the analog signal is constructed from digital check:
http://lavryengineering.com/pdfs/lav...ing-theory.pdf