[time-nuts] Sub Pico Second Phase logger
Bruce Griffiths
bruce.griffiths at xtra.co.nz
Sun Dec 21 22:19:06 UTC 2008
Joe
Joe Gwinn wrote:
> Bruce,
>
> At 9:50 PM +0000 12/19/08, time-nuts-request at febo.com wrote:
>
>> Message: 7
>> Date: Sat, 20 Dec 2008 10:38:26 +1300
>> From: Bruce Griffiths <bruce.griffiths at xtra.co.nz>
>> Subject: Re: [time-nuts] Sub Pico Second Phase logger
>> To: Discussion of precise time and frequency measurement
>> <time-nuts at febo.com>
>>
>> Joe
>>
>> Joe Gwinn wrote:
>>
>>> Bruce,
>>>
>>> At 3:54 AM +0000 12/19/08, time-nuts-request at febo.com wrote:
>>>
>>>
>>>> Message: 5
>>>> Date: Fri, 19 Dec 2008 16:51:55 +1300
>>>> From: Bruce Griffiths <bruce.griffiths at xtra.co.nz>
>>>> Subject: Re: [time-nuts] Sub Pico Second Phase logger
>>>> To: Discussion of precise time and frequency measurement
>>>> <time-nuts at febo.com>
>>>>
>>>> Joe
>>>>
>>>> Joe Gwinn wrote:
>>>>
>>>>
>>>>> At 11:48 PM +0000 12/18/08, time-nuts-request at febo.com wrote:
>>>>>
>> >>>
>> [snip]
>>
>>
>>>
>>>
>>>> Some claim to be able to sync to an SPDIF input but the resultant jitter
>>>> may be large.
>>>>
>>>>
>>> Why large jitter? Bad implementation?
>>>
>>>
>>>
>>>
>> I'm just suspicious, although I did see some data somewhere that seemed
>> to confirm my suspicions.
>> The S/PDIF signal has to be a valid SPDIF signal not just a square or
>> sine wave clock.
>> Output sample rates (for the AP192) are then identical to that of the
>> the S/PDIF source which is limited to
>> 192, 176.4, 96,88.2 48, 44.1 32 KSPS.
>>
>
> I did a little looking. I bet that the sync quite well, but this
> signal is pretty complex. One assumes that there is a box that takes
> in a 10 MHz ref and does the rest, because the broadcast industry
> does use atomic clocks.
>
>
>
Ulrich has built a circuit that takes a sampling frequency input derived
from a 10MHz GPSDO output and produces an S/PDIF output for this
application.
Its certainly worth trying since all the specs for the sound card aren't
readily available.
>> >
>> >> >> >> A few divide and mix stages will be required to achieve
>> a spur free resolution of 10Hz.
>> >>>>>>
>>
>>>>>>> That is a traditional approach. But are there alternate
>>>>>>> approaches that
>>>>>>>
>> >>>>> have now become practical?
>> >>>>>
>> >>>> Diophantine frequency synthesis?
>> >>>>
>>
>>>>> From the sound of the name I think so, at least in the last DDS
>>>>> stage, as done by that patent.
>>>>>
>>>>>
>> >>> But I was fishing.
>> >>>
>>
>>>> Conventional Diophantine synthesis uses number theory together with 2 or
>>>> 3 conventional synthesiser loops to achieve very fine resolution whilst
>>>>
>> >> maintaining a high PLL phase detector input frequency.
>> >
>>
>>> In a sense, the concatenated DDS approach is a divide-and-mix chain.
>>>
>> > Perhaps there is a parallel here.
>> >
>> The DDS based equivalent (of the dual PLL Diophantine synthesiser) would
>> use a pair of DDS chips each replacing a conventional PLL in the
>> Diophantine frequency synthesiser, the output frequency of each having
>> zero phase truncation spurs.
>>
>> Both DDS clock sources should be spur free and have a frequency ratio
>> that is a selected fixed rational fraction.
>>
>
> A M/N PLL chip can arrange this. I recall that Silicon Labs makes
> such a chip, which requires a parameter load on power-up, so a
> computer or FPGA is needed.
>
>
>
M and N only have to be relatively prime (ie the GCD of M and N is 1).
The ratio of M/N should also be close to 1.
If the spacing of the phase truncation spur free output frequencies is
about 10kHz (for either DDS) and M, N ~ 1000 the resultant mixer output
frequencies would have a spacing of about 10Hz which may be adequate for
this application.
>> A conventional mixer would then be used to either add or subtract the
>> two DDS output frequencies.
>> If the ratio of the 2 DDS clock source frequencies is appropriately
>> chosen the spacing between the resultant mixer output frequencies can be
>> much finer than the spacing between the truncation spur free outputs of
>> either DDS chip.
>> The DDS and mixer outputs should be filtered to remove harmonics and
>> other unwanted frequencies.
>>
>
> If the DDS chips are well chosen, we will get sin and cos outputs,
> and so can implement a dual-mixer phasing scheme to yield only the
> sum frequency or only the difference frequency, greatly reducing the
> amount of filtering needed. The better balanced the channels are the
> better the cancellation of the unwanted term. This is basically the
> phasing method of single-sideband signal generation.
>
>
>
>> One drawback is that selecting the output frequencies of the 2 DDS chips
>> required to produce the desired output frequency is somewhat complex.
>> Since one almost certainly needs a computer of some sort to set the DDS
>> frequencies this shouldn't be a significant issue.
>>
>
> Yes. Basically, every digital chip in this scheme requires a
> computer of some kind. What I've seen used is a small CPLD to do the
> initial parameter loads, and then to run the show.
>
> Joe
>
>
Bruce
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