[time-nuts] Lucent RFTFm-II-XO

Charles Steinmetz csteinmetz at yandex.com
Thu Sep 22 16:04:19 EDT 2016

One additional point:  Many ordinary PC sound cards do not have proper 
input filtering, so anything as fast as a PPS pulse will cause severe 
aliasing, even after it has been voltage-clamped to a level that will 
not overload the ADC.  (Thus my mention of LP filtering under pulse 

The sound cards that do have adequate input filtering generally use 
rapid-cutoff filter alignments, which can ring like crazy -- very much 
what you do not want when you are trying to use a pulse for timing.

So:  Best to slow the pulse rise- and fall-times down so you are not 
banging on the input filter (if present), or causing aliasing.

Best regards,


> Bill wrote:
>> If you could,  would you see if you can take the pps out of your
>> lucent box and use it to sync the pc sound card clock via continuous
>> calibration for  using SL for fx measurements.  Maybe by lucent box
>> pps output is bad.
> The PPS directly out of most GPSDOs is too fast for sound cards.  But
> once you stretch it, there is another problem.  Assuming 5v logic, and
> knowing that the sound card is AC coupled, the leading edge of the
> (assumed positive-going) pulse will produce a 5v spike that decays over
> a time period that depends on the post-capacitor DC resistance inside
> the sound card.  It might decay fully (back to 0v) during the pulse, or
> less than that.
> The falling edge of the pulse will generate a negative-going spike with
> a peak negative voltage of [decayed voltage after leading edge minus
> 5v].  This could be anywhere from just slightly negative to the full
> logic supply (if the post-capacitor voltage decays fully before the
> falling edge occurs).  The same would be true for 3.3v logic,
> substituting "3.3" for "5" above.
> These peak voltages are way outside the input range of most sound cards,
> so the positive spike is almost certainly going to overload the ADC,
> with unpredictable (and, very likely, unfortunate) results.
> At the very least, you will need some sort of sound-card oscilloscope
> program so you can see what the sound card is actually capturing.  Then
> you can start shaping your trigger pulses (with voltage clamping and
> possibly LP filtering) to give predictable and stable results.
> Best regards,
> Charles

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