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.
Cheers,
Magnus
On 10/12/2014 11:09 PM, Bob Camp wrote:
HI
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.
Bob
On Oct 12, 2014, at 2:37 PM, Hal Murray <[email protected]> wrote:
[email protected] 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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