Hi

With *all* of these “drop to a lower frequency” approaches, the theoretical 
resolution is very good compared to the useful resolution. A straight mix to 1 
Hz into a 5370 is a great example. The filter / limiter is the thing that sets 
the useful resolution rather than the theoretical 1x10^-17 the setup provides. 

Bob

On Oct 12, 2014, at 12:14 AM, Robert Darby <[email protected]> wrote:

> Bob Camp,
> 
> Bob, Simon is talking about the sampler versus a true mixer.  This is the 
> idea I asked you about some months ago when I asked about how the digital 
> filter functions.  You were kind to explain the filter method in terms of  
> buckets. You are of course correct that the resolution is low, 100 ns for a 
> 10 MHz DUT with a 10 Hz frequency offset but the hetrodyne factor takes the 
> theoretical resolution to 100 fs.  That's not shabby for a very low cost 
> DDMTD.  And of course, the actual noise floor will not be close to this but 
> potentially it's better than a 5370 and a lot easier to maintain. :o)
> 
> Simon,
> 
> I have a 4 channel 1 ns tagger "working" but I can't successfully link the 
> FTDI library to a c program so doing this in hardware looks far more 
> attractive to me.  Here's how I see it at this point:
> 
> -- Objective:
> --        A four channel DDMTD with 44 bit time tags delivered over the USB 
> port
> --        At least 100 Hz beat frquency on each channel
> --        The hardware is capable of much higher rates but increasing the 
> beat frequency offset
> --            degrades resolution and realistically the device will probably 
> be used at 5 or 10 Hz
> --
> -- Additional Hardware Required:
> --        A "wing" with three or five LTC6957-1 low phase noise buffers to 
> convert sine inputs into
> --            high speed low-jitter square waves using LVPECL differential 
> outputs
> --        Either an oscillator offset by the beat frequency or a DDS 
> frequency generator
> --        A USB equipped computer
> --
> --Architecture
> --        Differential inputs are fed to the master clock, thence to the D 
> flip-flops clocks
> --        Differential inputs for each channel are fed to the data inputs for 
> each flip-flop
> --        The master clock drives a 44 bit counter which is common to all 
> four channels
> --        Each channel has two independent counters, provisionally 14 bit, 
> designated high and low
> --        The low counter first establishes a low state without transitions 
> i.e. it times out
> --        After the low counter times out, the flip-flop is armed
> --        The first high output at q resets and starts both high and low 
> counters - whichever counts depends on whether q is high or low
> --        Every time the high and low counters match we store the 44 bit 
> count; each new match replaces the previous one
> --        At some point (2^14 highs) the high counter will roll over - 
> hopefully low will have stopped counting much earlier
> --        The highest stored match should meet the equal count criteria as 
> described in the P. Moreira and I. Darwazeh paper
> --        Since there are four channels it will be necessary to multiplex the 
> time tags into the fifo
> --        The multiplexer will add 1 bit per channel for one-hot channel id 
> coding
> --        The 48 bits will clock into a 48 bit to 8 bit fifo thence to an 8 
> bit USB port
> 
> I believe you can have multiple points where the two counts match but I don't 
> have any data to confirm that. I played with this in excel and when you feed 
> it ones and zeros in a distribution that "looks" like the typical  output out 
> of a digital sampler it is possible to get multiple matches.  My intention is 
> to go with the last crossing and the scheme mentioned above does this rather 
> trivially. Unless, of course, I'm missing something and I usually do.
> 
> I've got a Pipistrello board and it has the option of an asynchronous fifo 
> USB interface; since I've already paid my dues on that I'll just use that 
> code again.  The data rate is so low that snail mail would work.  The 
> computer gets a series of time tags and your program has to pair up the 
> channels to get the deltas.  Getting time tags lets you compare three or four 
> devices simultaneously and facilitates three-cornered hat calculations.  I 
> suspect that's a lot easier to say than do but we'll cross that bridge if we 
> ever get there. Also time tags permit continuous sampling; there's no counter 
> dead-time which I think can be an issue when it causes variable data sampling 
> rates.
> 
> Bob Camp mention Collins low jitter hard limiters but I suspect that's much 
> more of an issue on the very shallow slopes you see on 5 or 10 Hz mixer 
> outputs.  The LTC6957 is probably overkill on 10 MHz inputs but I believe 
> they're a tad better than a 74AC gate, but then again maybe not all that much 
> better.  Lot more expensive.  Bob C discussed sine to square conversion in a 
> recent post (IIRC) perhaps in connection with 5V to 3.3V conversion, and for 
> a low cost solution the 74AC gate looks pretty good and they're easy to dead 
> bug.
> 
> I'm out of spit. Later
> 
> bob
> 
> 
> 
> 
> On 10/11/2014 9:17 PM, Bob Camp wrote:
>> 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 <[email protected]> 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 <[email protected]> wrote:
>>>> 
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