[time-nuts] Sub Pico Second Phase logger

Bruce Griffiths bruce.griffiths at xtra.co.nz
Wed Dec 17 01:43:29 UTC 2008


Joe

Joseph M Gwinn wrote:
> Bruce,
>
>
> time-nuts-bounces at febo.com wrote on 12/15/2008 06:42:59 PM:
>
>   
>> Joe
>> Joseph M Gwinn wrote:
>>     
>>> Bruce,
>>>
>>>
>>> time-nuts-bounces at febo.com [Bruce] wrote on 12/15/2008 04:56:26 PM:
>>>
>>>       
>  
>   
>>>>>>> [snip]
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>>>> The only configuration for which it makes any sense is an 
>>>>>>>>                 
> inverting
>   
>>>>>>>> input amplifier with a finite input voltage offset.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> Why would non-inverting not work?  Both inputs source or sink bias 
>>>>>>>               
>
>   
>>>>>>> currents, and non-inverting presents a very high impedance.
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> Non inverting amplifiers usually have lower noise and generally 
>>>>>>             
> work 
>   
>>>>>> very well.
>>>>>>
>>>>>> I was only trying to come up with a preamp circuit for which the
>>>>>> comments in the Minicircuits application note on the effect of 
>>>>>> amplifier input offset voltage made any sense.
>>>>>>
>>>>>>
>>>>>>             
>>>>> Ah.  It may be hopeless.
>>>>>
>>>>> My reading was that they were worried about bias currents from the 
>>>>>           
> amp 
>   
>>>>> flowing into the mixer and causing offsets, not amplifier offset 
>>>>>           
> voltages 
>   
>>>>> per se.  The amplifier offset voltage does not cause a mixer offset, 
>>>>>           
>
>   
>>>>> and may be reduced by use of a chopper amp or very good balance.
>>>>>
>>>>>           
>>> By the way, I've noticed that Tek TDS3012B oscilloscope inputs can 
>>>       
> cause 
>   
>>> offsets as well, again I assume from the bias currents.  The circuit 
>>>       
> has 
>   
>>> the scope input in parallel with the Agilent 34410A 6.5-digit 
>>>       
> voltmeter. 
>   
>>> With scope input set to DC, big effect.  Set to AC, small effect.  Set 
>>>       
> to 
>   
>>> Gnd, no effect.  (Input is not grounded, so voltmeter is still happy.) 
>>>       
>
>   
>>> Didn't try changing the input volts/cm scale.  Anyway, I think that 
>>>       
> this 
>   
>>> effect is what the mystery app note was trying to say.  A bias current 
>>>       
>
>   
>>> from the scope would cause a voltage offset that depended on the DC 
>>> resistance through which the bias current flowed, the DC load of the 
>>>       
> mixer 
>   
>>> in this case.
>>>
>>>
>>>       
>> However the proposed remedy has little or no effect on the errors caused
>> by such bias currents (eg transistor base currents).
>> The series resistor could be reduced to zero without effect on the mixer
>> offset due to the bias current. However the preamp offset due to the
>> source resistance would be reduced.
>>     
>
> Hmm.  It may be simpler than that.  With the TDS3012B and 34410A connected 
> in parallel across the IF output of a mixer, the bias currents from the 
> TDS3012B developed a voltage across the mixer load resistor, and this 
> voltage was sensed by the 34410A.  All the phase detector had to do was 
> not short the bias current to ground.
>
>
>   
AC coupling?
>   
>>>>>> If we design our own PCB then the AD7760 series ADCs are another
>>>>>> possible option.
>>>>>> These have a built in differential input differential output 
>>>>>>             
> amplifier.
>   
>>>>> Yes.  But aren't we trying to use commonly available soundcards?
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> Ideally yes, but they all seem to have built in performance 
>>>>         
> limitations.
>   
>>>> AFAIK the AP192 with its 4Vrms full scale balanced inputs with no
>>>> variable gain preamps or +48V phantom supplies seems to be one of the
>>>> best for this application.
>>>> Its major drawback is that its a PCI card located within a noisy PC.
>>>>
>>>>         
>>> I think that there are many top-end firewire soundcards.  Whatever the 
>>>       
>
>   
>>> music folk like the sound of would be a good place to start - 
>>>       
> musicians' 
>   
>>> well-trained hearing can be quite good.  At least above 20 Hz.
>>>
>>> Actually, the people that make the AP192 do have firewire and usb 
>>> offerings:
>>>
>>> <http://www.m-audio.com/index.php?do=products.family&ID=recording>
>>>
>>>
>>>       
>> I've looked at all of the M-Audio offerings.
>> The more expensive ones have built in preamps plus 48V phantom supplies,
>> which can be switched off, however the presence of the switched +48V
>> supply is perhaps an invitation to disaster.
>>     
>
> Given that capacitance to ground is more benefit than problem in this 
> application, I would be tempted to use a pair of back-to-back rectifier 
> diodes as a clamp to protect the mixer IF output et al.  The 48 volt 
> phantom supply will be short-circuit protected, so current will 
> automatically limit.
>
>   
A pair of coupling capacitors at the preamp output combined with clamp
diodes to the amplifier power supply rails would work well even if the
+48V cant be switched off.
The +48V appears between the balanced pair conductors and ground.
Unfortunately the power available  from the phantom supply may not be
sufficient to power the mixer preamp.
>   
>> I've also looked at the specs for several other high end sound cards.
>> Quite a few only have single ended inputs.
>> Maybe, I should document the various cards and highlight their
>> shortcomings etc for this application.
>>     
>
> That would be very useful.
>   
I'll start on this shortly.
>  
>   
>>>> The 4V rms input allows the mixer preamp to use devices like the THAT
>>>> 1646 to drive the balanced sound card inputs without degrading the 
>>>>         
> noise
>   
>>>> floor too much.
>>>>
>>>>         
>>> Or build an isolation amp with some gain, and kill two birds with one 
>>> stone?
>>>
>>>
>>>
>>>       
>> A low noise isolation amplifier with a frequency response down to 1Hz or
>> so without using a transformer may be difficult to do.
>>     
>
> People do make low noise common-base RF amplifiers, but 1 Hz would yield 
> some pretty large bypass capacitors, even if the flicker noise can be 
> controlled well enough.  I would consider using ultracapacitors, which 
> didn't exist until very recently, and of course have very large 
> capacitance values.
>
>   
A CB stage probably isnt optimum for the mixer preamp so that lower
value caps can be used provided that they effectively short the
amplifier input resistor Johnson noise at the frequencies of interest.
>   
>>>> With a 1V rms full scale the noise floor degradation would be very
>>>> obvious when using a THAT 1646 (equivalent devices are even noisier).
>>>> It may be better to use a mixer preamp with a transformer coupled 
>>>>         
> output
>   
>>>> stage using hybrid feedback to achieve a low frequency cutoff below 
>>>>         
> 1Hz
>   
>>>> together with low noise.
>>>>
>>>>         
>>> With a transformer, even if toroidal, keeping hum out may prove quite 
>>> difficult.
>>>
>>>
>>>       
>> High end (eg Lundahl LL1517) line output audio transformers come with mu
>> metal screens and metal foil interwinding shields.
>>     
>
> They don't pass 1 Hz very well. I bet the rolloff is ~20 Hz. 
>
>   

When driven conventionally the transformer cutoff is around 20Hz,
however if one uses the appropriate driver with a controlled negative
output R to cancel the transformer primary internal winding resistance,
the low frequency response can be extended significantly. This also
reduces the low frequency distortion.
However individual adjustment of driver to suit transformer is required
and tracking the winding resistance over temperature may be an issue.

> Certainly one can build a VLF transformer, but it will be a project for 
> sure, and the transformer may be quite large.
>
>   
The transformers only weigh about 65g.

It may be simpler just to select a mixer for which the IF ground can be
isolated from the RF and IF grounds.
However a preamp with a transformer output may be useful if one uses a
mixer where all the grounds are connected together by the package.
>
>   
>>>>>>> The [5120A] paper is also worthwhile, and available on the web 
>>>>>>>               
> somewhere 
>   
>>>>>>> (don't recall where, but google found the pdf).  I had to read the 
>>>>>>>               
> patent 
>   
>>>>>>> multiple times to figure out what's going on.  The correlation 
>>>>>>> approach is old as the hills, and only the digital phase detector 
>>>>>>>               
> was patentable.
>   
>>>>>> It may be feasible to achieve the same effect by purely digital 
>>>>>>             
> means 
>   
>>>>>> at least for low sample rates where FIR filters with tens of 
>>>>>>             
> thousands 
>   
>>>>>> of taps are feasible.
>>>>>>
>>>>>>
>>>>>>             
>>>>> It *is* feasible, and Sam Stein is doing it.  I've perhaps lost the 
>>>>>           
> thread 
>   
>>>>> here.
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> No, I meant replace his 90 degree hybrids with a digital equivalent.
>>>>
>>>>         
>>> I believe that his 90-degree hybrids are already digital.
>>>       
>> I'm not convinced of that, if only because real time 10,000+ tap FIR
>> filters at 30+MSPS are probably still impractical.
>>     
>
> I'm not convinced that one needs a 10,000-tap FIR to achieve this, and Sam 
> Stein is one smart fellow.  I recall some NASA patents from twenty years 
> ago on how to get I+Q data from a single ADC, and while there was FIR 
> processing of some kind, there were only maybe 8 or 16 taps.  And Tayloe 
> (US patent 6,230,000) gets much the same effect with one resistor, four 
> capacitors, an analog mux, and two differential amplifiers.
>
>   
I have read similar papers from that era on radar signal processing.
They either used a Hilbert transform or a pair of digital filters whose
outputs were in phase quadrature.
The quadrature accuracy for a given bandwidth depends on the the number
of taps.
The beat frequencies (in a dual mixer system) won't match exactly and
some correction for the resultant phase shift errors will need to be made.
This may be less of a problem when the 2 beat frequency signals are
identical in frequency and just differ in phase.
> How accurately must the quadrature delay be achieved?  If I recall, the 
> patent or paper implies that it need not be exact.
>
> I also recall thinking that he could implement the quadrature delay 
> digitally.  I don't recall the details, but it depended on cutting time up 
> into one-second batches and processing each batch independently of the 
> others.  I suppose one can use a FFT-Multiply-IFFT to do it directly.
>
>
> Joe
>
>
>   
Bruce



More information about the time-nuts mailing list