Oz, On 27 June 2010 01:09, Oz-in-DFW <li...@ozindfw.net> wrote: > > > On 6/26/2010 7:12 AM, Steve Rooke wrote: >> <Deletia> >> I next thought about turning the DC into AC by chopping it, IE. >> inverting 50% of the voltage via an oscillator. This way I could pass >> the square wave directly into an unmodified sound card, take >> measurements and then do an RMS calculation on them (really just need >> to flip the sign on, say, the negative readings). >> > I've done similar stuff in work projects, but never written code. I've > thought about this some as well. I'd consider a few things; > > 1. Use the sound card output as the chopper control signal instead of > the discrete unit. You'll have more control and phase sync will > be easier. > * I'd be temped to take the sound card output and run it > through a comparator to square it up, but I'm almost certain > this isn't needed.
Sorry, not sure what you mean here. Are you saying that I should derive the chopper frequency directly from a connection to the sound card? I was hoping not to modify the sound card in any way so as to keep it simple. > 2. Buffer the input so that your waveform is not so dependent on > source impedance. Good idea, thanks. > 3. Make the input buffer differential so that you can get some small > amount of ground isolation and CMRR If you look closely at it you can see that it is a differential input. > 4. look at the 4053 mux, it might make your interconnect life easier. Thanks, will do that. > 5. The probelm with chopping is that signal levels around zero don't > have much amplitude and are a challenge to extract from noise. I was under the impression that this was the idea that is used to amplify very low level signals like the output from the likes of strain-gauges. It would surely seem to me to be a problem to amplify small signals around zero due to offsets in the amp unless you do this sort of thing. > 6. If you mix (in the RF receiver sense, not sum in the audio studio > sense) rather than chop the DC offset becomes a phase shift, > generally pretty easy to calibrate for and decode from the output > samples of a sound card. See > http://en.wikipedia.org/wiki/Frequency_mixer Interesting idea, hadn't thought of doing it that way but it's a good idea, thanks. >> I wonder if anyone has done something like this before and could share >> their experiences. I've attached a diagram image (hope it is accepted >> by the list) which is my first go with Eagle so I'm not exactly very >> familiar with it, sorry. The R's and C's in the astable would be set >> to a clock frequency that enables this to work without bias given the >> sampling frequency. I'm not sure if this clock should be slower than >> the sampling frequency or higher, just haven't got my head around that >> yet. > The clock needs to be much higher than the highest frequency of the > input waveform to keep Nyquist happy and things simple. You can do this > inband, but you don't want to. > > If you chop very close to half the soundcard sample rate I suspect > you'll get no output because you'll be in the roofing filter cutoff and > your waveform will integrate to zero. I suspect you want to be 5 - 10X > below that to make waveform recovery easier, and even lower is better. > > So, if you use a 44.1 ksps default rate, Nyquist is 22.05. I'd run the > chopper at less than 1 kHz. The good news is that your input waveform > period is hours (maybe ~100 microhertz) and chopping at 1 Khz will make > 100 Hz response easy and 500 Hz possible with great care and some effort. Right, that makes sense, thanks. >> The R's around the op-amp would need to be set in a ratio that >> transforms the EFC voltage into the range that the sound card can >> handle (that is yet to be calculated by measuring the limits). > Most sound cards I've seen are ~ 1V pk to peak, though some are MUCH > higher. Gives me a ball-park to start work with, thanks. >> If you >> have any suggestions or ways of doing this in a better way, I'd be >> very grateful for the advice. >> > It's worth exactly what you've paid for it... And worth every penny :) Thanks, Steve >> Thanks, >> Steve >> >> > Oz > > -- > mailto:o...@ozindfw.net > Oz > POB 93167 > Southlake, TX 76092 (Near DFW Airport) > > > > > _______________________________________________ > time-nuts mailing list -- time-nuts@febo.com > To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts > and follow the instructions there. > -- Steve Rooke - ZL3TUV & G8KVD The only reason for time is so that everything doesn't happen at once. - Einstein _______________________________________________ time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.