I am designing an anti-aliasing filter for an MCP3202 ADC. I'm kind of confused by defining my corner frequency Fc, I'm working with a 60 Hz input signal with harmonics until around 3 kHz that comes from an air conditioner current signal, so I can say that the maximum frequency of my input signal is 3 kHz or 50 X 60 Hz.
The MCP3202 is a SAR ADC, with SPI output protocol, 0-5 Vdd, and its specification says "100kSps", that requires an input clock signal Fclk more or less of 1.8MHz, that is, to generate a sample it needs a clock signal of 18 X Fclk, supplied by a microcontroller.
As we know SAR ADC works with the "sample and hold" method to hold the samples and be converted.
I assumed that the internal R-C input of the ADC works with the ADC clock frequency Fclk, which generates my confusion because the output of the anti-aliasing filter is wired to the ADC (no bucket filter before the ADC,) so the filter "sees" the Fclk in the ADC's input.
Being more specific according to this information.
Does the cutoff frequency Fc have to be defined by the clock frequency Fclock if not, is Fc defined by the sampling frequency specified in the ADC datasheet (100 kSps?)
Best Answer
The MCP3202 has an on-chip sample and hold. Don't make assumptions about internal RCs that work at the Fclk frequency.
You can ignore Fclk, as this is an 'implementation detail' of the ADC. It's important if you're programming Fclk, it's irrelevant if you simply want to use the ADC at whatever rate Fclk sets it to.
The ADC has a maximum sample rate of 100 kS/s, but you can choose a lower rate if you don't want to handle that throughput.
You want to process signals up to 3 kHz. By Nyquist, that means you need an absolute minimum rate of 6 kS/s, at which you would need an unfeasibly good AA filter. To work with a practical AA filter, you need a somewhat higher sampling rate.
You need to know how you are going to be processing the results of your ADC measurements before you design the AA filter.
Are you going to do frequency selective analysis digitally? In this case you only need the AA filter to protect your samples from aliasing. For instance with a 10 kS/s sample rate, the AA filter would have to pass 0-3 kHz, and stop 7 kHz and above. This allows a very generous transition band, making it easier to design. For instance, any noise at 6 kHz that gets aliased to 4 kHz will be ignored by your subsequent analysis stopping at 3 kHz.
Alternatively, do you want every sample taken by your ADC to be valid as it is without further analysis? In this case with 10 kS/s sampling, your AA filter should pass 0-3 kHz, and stop at least 5 kHz (Nyquist) and above. However it will be beneficial to stop from as close as possible to 3 kHz as it can to remove as much noise above 3 kHz as possible. This then gives you a filter flatness/size/performance tradeoff.
You'll notice that in neither case do we use the Nyquist rate directly to specify the AA filter, but via the input bandwidth, and the sort of analysis we want to do subsequently.