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Not entirely related to your question on audio aliasing, but many moons ago I was an EE on a team using electro-optics to design a much better Analog-to Digital converter than you could buy commercially. The signals we sampled were in hundreds of MHz but the concepts are exactly the same as in audio frequencies.

What we effectively did was use a pulsed fiber laser, which produces _very_ short pulse widths of light. Connecting this to an electro-optical modulator, we can effectively capture a very fast snapshot of the electrical signal. Use some analog circuitry to hold that level constant for the slower analog-to-digital conversion, and we had an AD converter that while sampling relatively slowly, see's only a tiny portion of the signal for it's actually sample.

What was amazing about this was that the laser pulses were so short, and the 'timing jitter' of the pulses so low that we could cleanly see the expected aliased sampling frequency when saming a sine wave signal at large multiples of our sample frequency (iirc up to the 40's). Eg, inputting a sine wave around 10 GHZ and we'd see only a single sine wave in the sampled signal at it's row sampled frequency with minimal artifacts.

So if you are wondering how aliasing affects audio samping, draw a sine wave and see what happens if you sample it at a much lower frequency. You'll get a lower frequency sine wave. Without filtering the input signal, it's impossible to tell if this is a real low-frequency sine wave or something above Nyquist that has been down sampled.

N.B. If you're wondering why our AD converter was better than a commercial one, we also added optical demultiexing to get an extra order of magnitude improvement in sample rate. This is possible using our electro optical setup due to the laser's low jitter and the ability to hold the signal steady, a purely electronic version will have difficulty with the timing to see much actual improvement in SNR.




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