[time-nuts] Digital Mixing with a BeagleBone Black and D Flip Flop

Robert Darby bobdarby at triad.rr.com
Sun Oct 12 00:14:27 EDT 2014


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 <subscriptions at 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 <bobdarby at triad.rr.com> wrote:
>>>
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