Thomas, I think that aliasing is a red herring for your application. I think that other people have pointed this out in postings to this list. If you are only interested in the settings of the pots at your sampling instants then you don't need any anti-aliasing filtering. What you want is what you will get. Aliasing would only be an issue if you wanted to be able to re-construct the signal (e.g. by digital processing and/or writing your samples to a D/A and filter) between the sampling instants. Cheers, Bill > -----Original Message----- > From: Thomas Brandon [SMTP:tom@PSY.UNSW.EDU.AU] > Sent: Monday, September 27, 1999 10:50 AM > To: PICLIST@MITVMA.MIT.EDU > Subject: A/D Challenge Summary > > Sorry, but I'm getting a little overloaded with information. Let me see if > I > can summarise the major things so far. > > There are 2 things that are possibly going to throw my readings of: > 1) Noise > 2) Aliasing > > The noise is something that will only really come out in practice but > should > be planned for from the start. Noise will come from the following sources: > 1) Clock > This will be at whatever clock frequencies are present. Robert said > "Don't forget that clock dividing is going on inside your chip. (E.g., > divide by 4), so lower frequencies will be present." but I'm planning to > use > a Scenix. I don't think these have a divided clock. They get 50MIPS at > 50MHz > so they can't be dividing the clock can they? Must be something like 4 > parallel paths running simultaneously. > > 2) Power frequency > 50\60Hz. > > 3) Any EMI my circuit chooses to recieve. > > I can try and avoid noise but the only way to know how much of a problem > it > will be is by trial and error. Can someone point me at a good source of > information. I have Art Of Electronics (lent out at the moment tho so > can't > look), is it's section on noise any good? Can anyone recommend a good > source > of information on topics related to analog data aquisition such as EMI, > grounding, filtering etc. > > > Aliasing is something I don't quite understand. I know vaguely how it > works > in theory but am having trouble linkling the theory to practice. I know it > is related to frequencies in the signal and the sampling rate. I can see > how > it works in a perfect audio system (i.e. the only frequencies are those of > the audio (no noise)) but have trouble linking that to a knob box. > > What frequencies will the knobs generate? What if you tweak the knob as > fast > as you can (say 5Hz (I don't think it'd be easy to change direction more > than 5 times a second so this seems pretty over the top)) between two > points > on the knob. Let's say it's a 7-bit knob (obviously the knob has infinite > resolution but I'm sampling at 7-bit) and the two positions are say 60 and > 70. So your tweaking it 5 times a second between 2 positions around the > middle of the knob. So if you sampled it you would get something like > 55->60->70->60->70->60->80. Now, if I filtered that with say a 4Hz rolloff > LPF. What is the signal now? Would it completely remove the 5Hz 60-70 > oscillation giving an output something like 55->0->0->0->0->0->80 or what? > Now if I took that signal and didn't LPF it and sampled at 8Hz say (less > than 2x5Hz) what would happen. According to my understanding of the > Nyquist > theorem I will get aliasing on components of the signal with a frequency > above the Nyquist freq. (1\2 the sampling rate = 4Hz). Thus the 5Hz signal > will be aliased down into the 1-4Hz range. What will this do to my signal? > I > don't care whether the 60->70 oscillation is at 4Hz or 80MHz I just care > about the amplitude of the signal at my sampling points. How will this be > effected? > > Tom.