Steve Rooke wrote: > 2008/10/4 Bruce Griffiths <[EMAIL PROTECTED]>: > > >> 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. > > 73 > Steve > > Steve
Such interpolation is more effective when one channel has a reference signal input (eg PPS or a known frequency) and the other channel has the frequency to be measured. Bruce _______________________________________________ time-nuts mailing list -- [email protected] To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts and follow the instructions there.
