I have one and only one frequency signal, a sine signal at f = 10 kHz that is sampled at Fs = 320 kHz.
I have read about applying a simple analog anti aliasing (AAA) filter then oversample, do some digital anti aliasing (DAA) and decimate to get the correct signal frequency.
I understand how this is useful if signals of interest are close to Fs/2 as it might be hard to find an AAA filter that has sharp enough roll-off, low cost, simplicity etc. And therefore DAA filter in combination with a simple AAA filter is often preferred.
But in my case I don't have a signal close to Fs/2 at all. So I was thinking maybe I can get a slower/cheaper ADC and do no oversampling, since I might be able to properly attenuate frequencies that might alias even using a simple AAA? Is what I am thinking correct?
Additional related question: How would this work if I was to apply equivalent time sampling (ETS) technique? This means that the signal is 10 kHz and the sampling frequency appears to be 320 kHz as well, but 32 samples are taken over 32 periods of the 10 kHz signal to produce 1 period. So the sampling rate is actually 10 kHz + (10 kHz / 32) = 10.3125 kHz. What I am not able to wrap my head around: will Fs = 10.3125 kHz or 320 kHz? Will the same filter work for both cases? I assume Fs = 10.3125 kHz thus giving a problem as the transition area will be relatively very small: Fs - f = 3125 Hz.
Other thoughts: I am using a DDS to generate the sine signal, it should be quite pure and maybe AAA is not "needed" / very important?