[time-nuts] Digital Mixing with a BeagleBone Black and D Flip Flop
kb8tq at n1k.org
Mon Oct 13 00:15:03 UTC 2014
The 1/F noise vs beat note “amplification” tradeoff is what pushes me up to 10 Hz rather than staying down around 1 Hz with most setups. It’s also a rational offset to achieve at 10 MHz with common OCXO’s. Once you get past about 20 Hz, your OCXO choices diminish.
On Oct 12, 2014, at 7:57 PM, Magnus Danielson <magnus at rubidium.dyndns.org> wrote:
> Increasing the beat frequency to find a balance between 1/f noise and f/delta-f amplification may be worth doing and have been seen done to find "optimum" performance. If you use hard limiters or audio channels to achieve it is however a little detail.
> The benefit of audio channels is that the A/B channels does not disperse out in time, such that you loose cross-correlation of transfer oscillator noise.
> Some AD inputs may need to be modified to remove DC-blocking cap. Not all ADCs is happy with this. Some boards already have that and do DC-removal in digital filters.
> On 10/12/2014 11:09 PM, Bob Camp wrote:
>> A little more information:
>> If you are doing the ADC thing, you still need to estimate zero crossings. In all likelihood you would be doing bandpass filtering first (say 8 Hz to 12 Hz) on your 10 Hz note. Next you would do some sort of estimator to get the zero cross. A curve fit is one sort of estimator, there are others. A simple straight line fit over 4 or so points might do it. A higher order fit over a few more points is possible. Why does that matter? The fit improves your accuracy quite a bit. It also reduces your vulnerability to odd single sample issues like popcorn noise. Since you are running at a very low frequency 1/f noise can be an issue.
>> On Oct 12, 2014, at 2:37 PM, Hal Murray <hmurray at megapathdsl.net> wrote:
>>> anders.e.e.wallin at gmail.com said:
>>>> Does it matter that the ADC in the sound-card is probably clocked by a
>>>> crystal clock that is 50ppm off and has bad ADEV?
>>> You can calibrate the clock on the ADC.
>>> One way is to feed a known reference frequency in on the other channel.
>>> (That's assuming you have a stereo setup and don't need the second channel
>>> for something else.)
>>> Another way is to compare the sample rate with the PC clock. That will
>>> correct for any long term drift but may not track shorter transients.
>>> These are my opinions. I hate spam.
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