[time-nuts] Digital Mixing with a BeagleBone Black and D Flip Flop
magnus at rubidium.dyndns.org
Sun Oct 12 19:57:42 EDT 2014
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
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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