Hi

Ok, a little more data:

You can hook your flip flop up as a sampler or as a full blown mixer. Hooked up 
as a full blown mixer, you get the 20 MHz and 10 Hz signals. You also get more 
resolution on the 10 Hz. Either way, the 10 Hz is still a beat note. In the 
case of a sampler, the filter is there for edge jitter. 

With a sampler, your data is only modulo 100 ns. With a 100 ms beat note 
period, you only get 1x10^-6 at best. That’s very different than what you get 
with the same chip used as a mixer (or an XOR gate). The true mixer connection 
gives you data the instant the edge changes. The sampler goes to sleep and lets 
you know up to 100 ns later ...

Bob

On Oct 11, 2014, at 6:31 PM, Simon Marsh <subscripti...@burble.com> wrote:

> I (mostly) understand this when considering an analogue mixer, but I'm lost 
> on whether there are any similar effects going on with a digital signal ?
> 
> TBH, I'm not really sure 'mixing' is the right phrase in the digital case, 
> and my apologies if I got that wrong.
> 
> What's actually going on is sampling one (digital) signal at a rate close to 
> the signal frequency. This gives a vernier effect and the result is a purely 
> digital set of pulses at the beat frequency, aligned to when the signal and 
> sample clock are in phase. It does not have a high frequency component to 
> filter out.
> 
> Cheers
> 
> 
> Simon
> 
> On 11/10/2014 21:11, Bob Camp wrote:
>> Hi
>> 
>> Your glitches are (in part) coming from the 20 MHz (10 + 10) component on 
>> the mixed signal. Since they have no direct relation to the beat note, 
>> filtering them after limiting is not a simple task. It is far easier to keep 
>> filter the signal pre-limit than to do so post limit.
>> 
>> The other component of the glitches is related to the limiting process. The 
>> paper by Collins is a good one to read for information on gain, bandwidth 
>> and the limiting process. Again, there is very little you can do “post 
>> limit” to sort things out.  None of the zero crossings you are getting may 
>> be “correct”. It’s not simply a process of picking one out of the group.
>> 
>> ——————
>> 
>> Some math:
>> 
>> You have two 10 MHz signals and a (say) 10 Hz beat note. You are looking for 
>> 1x10^-13. You get 1x10^-6 from the downconversion. You need to get 1x10^-7 
>> out of the beat note.
>> 
>> Put another way, 1x10^-13 at 10 MHz is 1x10^-5 Hz.
>> 
>> If your beat note is 3 V p-p, it will cover 6V every 1/10 second. It’s about 
>> 1.2X faster than a triangle wave as it zero crosses (memory may be failing 
>> me here), so that makes it equal to a 7.2V triangle excursion.
>> 
>> 1x10^-6 of 7.2V is 7.2 microvolts.
>> 
>> That’s how accurate your limiter / filter combination needs to be, 
>> pre-limiting.
>> 
>> It can be in a fairly narrow bandwidth, so it’s not quite as daunting as a 
>> radio front end.
>> 
>> Since you have a very large signal, and very small noise, the normal 
>> “dithering will help me” effect of the noise can not be counted on.
>> 
>> The thing you *want* to come up with is essentially a random signal (ADEV), 
>> so massive filtering will not do the trick either.
>> 
>> Bob
>>  On Oct 11, 2014, at 3:33 PM, Robert Darby <bobda...@triad.rr.com> wrote:
>> 
> 
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