[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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