[time-nuts] And, for my next trick, 50Hz
sar10538 at gmail.com
Sat Oct 4 10:10:49 UTC 2008
2008/10/4 Bruce Griffiths <bruce.griffiths at xtra.co.nz>:
> With interpolation you dont need to slow the signal transitions too much.
> A transition suficient to allow 3 or 4 samples to be taken during the
> transition is adequate.
> If the signal slew rate is too slow sound card input noise and the
> finite ADC resolution will increase the measuremnet noise.
OK, I understand what you mean. Guess it will have to slewed somewhat
to try and get the 3 to 4 samples on a rising/falling edge though. If
we are looking at a 1KHz signal and sampling at 44KHz, that means we
have to slew the transition to something like 100us to guarantee
getting enough samples during the edge. So that is a 1/10 of the input
signal or a roll-off of 10khz. That happens to fall nicely inside the
bandwidth of the sound card.
So how do we time-stamp these samples considering they will be
buffered by the card and not read independently. Perhaps we don't have
to as we can read each buffer when it it is full and time-stamp at
that point. We know that each sample is taken at 44KHz and can easily
count the sample number to calculate the time.
I'll have a look at this and see what I get. I really need to check it
against a known standard so will solve that problem first but this
would be an interesting way of using a PC to check frequency.
Steve Rooke - ZL3TUV & G8KVD
Omnium finis imminet
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