[time-nuts] Sub Pico Second Phase logger

[Bruce Griffiths]

Joe

Joseph M Gwinn wrote:
> Bruce,
>
> time-nuts-bounces at febo.com wrote on 12/17/2008 03:43:16 PM:
>
>   
>> Joe
>>
>> Joseph M Gwinn wrote:
>>     
>>> Bruce,
>>>
>>>
>>> time-nuts-bounces at febo.com wrote on 12/16/2008 10:21:55 PM:
>>>
>>>       
> [snip]
>   
>>>> [BG] Obtaining suitable mixers for 5MHz and 10MHz input frequencies 
>>>>         
> or even 
>   
>>>> 100MHz is easy.
>>>>         
>>>> However for the higher microwave frequencies most mixers come 
>>>>         
> complete
>   
>>>> with connectors attached and share a common ground.
>>>>
>>>>         
>>> [JG] True.  However, I don't think we will be going from 1 GHz to 1 Hz 
>>>       
> in a 
>   
>>> single step, and the last mixer can have separate grounds.
>>>
>>>
>>>
>>>       
>> An upper limit of at least 100MHz should be feasible for the final 
>>     
> mixer.
>   
>> A dual conversion scheme will be essential if one uses a triple balanced
>> or similar first mixer that has an IF response that doesn't extend down
>> to the low frequencies that a sound card can use.
>>     
>
> Yes.  We will see if it's needed.
>
>
>   
>>>> [BG] If we can devise a suitable test setup then one could just log 
>>>>         
> the
>   
>>>> samples to a file for whatever sound card one has and make the data
>>>> available to others for analysis.
>>>>
>>>>         
>>> Yes.
>>>
>>>
>>>
>>>       
>>>> This allows a wide variety of sound cards to be evaluated without one
>>>> person having to test them all.
>>>>
>>>>         
>>> And evaluation of the same test data by multiple people usingdifferent 
>>>       
>
>   
>>> tools also allows us to distinguish test artifacts from processing 
>>> artifacts.
>>>
>>>       
>> [BG] Proposed test setup:
>> (preliminary to be refined)
>>
>> Drive 2 sound card inputs in parallel with the same source.
>>
>> Source amplitude:
>> Max sound card input -3dB
>>     
>
> What kind of dB?
>
>   
Peak input signal voltage = 70% of sound card maximum peak input voltage.
Just to leave some margin for gain tolerances.

>  
>   
>> Sources:
>>
>> 1) Wien bridge or equivalent (eg state variable oscillator with soft
>> clamping) relatively low distortion oscillator.
>>
>> 2) Buffered low pass filtered output of binary divider driven by a
>> crystal oscillator
>>     
>
> RC oscillator sounds far simpler and more flexible.
>
>   

A Wien bridge using a lamp is perhaps the simplest.
I'll create a circuit schematics for this using an OPA2134 (dual lowish
noise JFET opamp).
One opamp for the oscillator one to drive the sound card (attenuation of
the oscillator output will be required for some sound cards and it is
desirable to have a low output impedance driver).

>  
>   
>> Test frequencies:
>>
>> 100Hz
>>
>> 1kHz
>>     
>
> Why no 10 Hz?  (Well, 20 Hz.)
>
>   
No particular reason other than some complications if a lamp stabilised
oscillator is used.
A diode soft (series R) clamped RC oscillator is more flexible in this
regard.
I'll also produce a circuit schematic for one of these oscillators.
>  
>   
>> Sound card sample rate:
>>
>> ~24KSPS
>>     
>
> I assume that this is the lowest rate supported, and certainly is overkill 
> for 1 KHz.
>
>   
It varies with the sound card.
I just suggested that for a starting point in the discussion.

For an AP192 the directly (without sample rate interpolation) available
output sample rates are:

192, 96, 64, 48, 32, 8 KSPS.
>  
>   
>> Test duration:
>>
>> 1000 sec
>>     
>
> At least initially, but we will need longer datasets to see thermal 
> effects clearly.
>
>   

>> File format:
>>
>> Wave file??
>> Resolution 24 bits for 24 bit sound cards, 16 bits for 16bit and lower
>> resolution sound cards, etc.
>>
>> Some refinement of sample rates and test duration is required to keep
>> the data file sizes manageable.
>> With a 24 bit sound card sampling at 96KSPS or 192KSPS for 1000sec can
>> produce file sizes of 1GB or more.
>> Some preprocessing (low pass filter and decimation) may also be 
>>     
> required.
>
> I agree that a simple preprocessor will be needed.  This would be the 
> place to convert from the raw adc 16 or 24 bit format into something 
> universal, perhaps 24 bit or 32 bit (with zero padding as needed).  It 
> probably should be written in C, for speed and portability.  I expect that 
> there are open-source libraries available to read and write wav files, and 
> many analysis programs will accept wav.  However, it would be easy to make 
> the preprocesor able to emit other formats as needed. 
>
>   

Bruce