Klippel Near Field Scanner on a Shoestring

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b_force

b_force

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oneworldconcepts.com/
Kravchenko_Audio said:
I do not have the engineering experience to give a useful answer. I basically tossed this out for anyone that has high speed signal experience and in how the potential difference in clocking can effect the computations. Few mentioned a bit back a few pages that a USB mic has timing problems. But I have never read anything else regarding this potential problem. And nor have I ever considered it to look!

Conceptually a physically measured distance and a chirp should be automatically correcting each other.

Again conceptually a timing difference in microseconds should have little detriment to phase measurements. But I have no real experience in these matters.
Click to expand...
It was also more like a rhetorical question, maybe I shouldn't use the question mark next time ;) :D :D

So yes, with a 2nd loopback we can very easily compensate the system for any time/delay/latency/phase issues :)
 
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mbrennwa

mbrennwa

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Kravchenko_Audio said:
I do not have the engineering experience to give a useful answer. I basically tossed this out for anyone that has high speed signal experience and in how the potential difference in clocking can effect the computations. Few mentioned a bit back a few pages that a USB mic has timing problems. But I have never read anything else regarding this potential problem. And nor have I ever considered it to look!
Click to expand...
Yes, a dual channel measurement with the speaker / microphone in one channel and the loopback in the other will remove the uncertainty of the latency in the audio processing. USB microphones will not work because they only have a one-channel audio interface (the microphone), so there is no second channel available for loopback.
 
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b_force

b_force

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gedlee said:
When I was doing this I used HolmImpulse and it worked great. The t=0 point is irrelevant as long as it doesn't vary from point to point. Holm could lock the synch so that this always happened (well 90% of the time.) The fitting algorithms will find the correct delay.
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That sounds an awful lot like looking for a solution for a non existing problem.

With a loopback you know you're always accurate.

Okay that assumes that the hardware is accurate enough.
Which is not difficult to do at all.
(although some audio interface manufacturers still seem to have a talent to mess that up).

I would only use Holm's solution in two ways.

  • you know people are just gonna use USB mics.
  • a 2nd loopback channel is super complicated and expensive to implement

Since the majority of audio interface these days have two channel input anyway, I don't see any reason.
 
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Kravchenko_Audio

Kravchenko_Audio

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gedlee said:
And since a single channel HolmImpulse works just fine, I don't see any reason for two channels.

I would never use a USB mic.
Click to expand...
It's always the devil you know that matters when measuring. It is never absolute, it is an indication of what we hear. And it is as much an art as the rest of this business to get good measurements.

I'm not looking forward to trying new measurement software either. But, if this is going to work out, I will need to learn a few new tricks.
 
T

Tom Kamphuys

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Joined 2020
Tom Kamphuys said:
loopback is not yet working
Click to expand...

I use my Aiyima A07 amplifier that I have lying around. However when I connect the loopback the amp goes into some kind of safe mode and I have to remove the power to get it to work again. I just measured some voltages in order to find out what is going on. I found out that there is 16V between the input and output of the amplifier. Normally that isn't a problem, but when I complete the circle DAC -> amplifier -> (speaker&) ADC this gets troublesome. DAC/ADC is a Behringer U-Phoria UMC204HD

Is this normal? How to get around it? Capacitor somewhere? 'lift'ground?

Or this, the Behringer HD400, which seems to have some transformers inside:

IMG_2606.jpeg
 
Kravchenko_Audio

Kravchenko_Audio

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Tom Kamphuys said:
Or this, the Behringer HD400, which seems to have some transformers inside:
Click to expand...
You generally don't want a transformer in the signal path if you want a wide bandwidth measurement. I know that there are exceptions to this. But as a normal way of setting up a measurement I would stay away from a transformer that you do not know the characteristics of.
 
F

fluid

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Joined 2009
ARTA describes two types of dual channel measurements

What they call dual channel takes the output from the amplifier with a voltage probe.
If the amplifier is bridged there is no ground at the output terminals and it is problematic.

Dual Channel.png


https://www.htguide.com/forum/articles/do-it-yourself-diy/927384-dual-channel-measurement-jig

What many people call dual channel, ARTA classifies as semi dual channel, where the loopback is just the line level output from the soundcard.
This causes no problems.

Semi Dual Channel.png
 
mbrennwa

mbrennwa

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Tom Kamphuys said:
I found out that there is 16V between the input and output of the amplifier
Click to expand...
You could use a DC blocking cap to avoid the DC in the loopback. However, you may not need to include the power amp in the loop back because it does not affect thd latency of the audio processing. To get the timing right, just connect the loopback at the computer audio interface.
 
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