I'm currently learning about Analogue to Digital Converters. From what I understand, aliasing occurs if the input signal being sampled has power above the "Nyquist frequency" of the ADC(Sampling frequency of the ADC/2). So, then logically, I assume then that all the anti aliasing filters have to be analogue. Or the same problem of aliasing due to sampling would exist, no? Either that of the digital anti aliasing filter has to sample at very high frequencies? Not to mention, have an DAC in the filter module to put the signal back into the analogue domain. Which would ask the question, why wouldn't we then simply just use a high frequency digital sampler and then have digital filtering done afterwards.
I'm asking because most of the tutorials and lectures I come across seem to miss how the signal is actually filtered before the sampling. Rather they just simply say the type and order, e.g. Butterworth 4th order. But no mention of how this would be implemented.
Where usually do people put these filters?. On the IC for the ADC itself? Or as an external filter before the sampling stage. Are physically big filters (e.g. through hole capacitors, inductors and resistors) the norm? Or rather integrated filters? For example, I'm guessing the ATMEGA328 (used on Arduino UNO) has an integrated anti aliasing filter since we usually don't filter the signal.
Particularly, I am referring to audio band ADCs, especially regarding the types of the filters. But the rest of the question, I am asking in a general sense.