It's a very curious case.
Compensation is mandatory in some shape or form.
That doesn't have much to do with being "clever", but that are just the fundamentals of data acquisition.
Even more so when you introduce active filters and/or DSP's.
I don't exactly know what HOLM is doing under the hood, but if you just approach it like a black box, the best they can do is compensate audio just before it's going into the DAC of the soundcard.
Like said above, which can ONLY work when a soundcard is being used with a fixed clock, as well as anything behind it with a fixed clock.
In the old days, I have also done hundreds of measurements with just a single channel.
Which works, but just only when there isn't to much going on in your chain.
There are also ever so slight artifacts.
This method falls on its face when you're introducing different active DSP systems or amplifiers with a DSP front-end (surround receivers etc). Time reference is completely gone at that point.
Or a much better way of saying it; you don't know if the time reference will be good enough or not.
With a fixed loopback wire, this little bit of unknown is fixed.
The issue also seems to be much worse with Win8/10/11.
I had significantly less problems with my old WinXP setup.
Measuring can be pretty hectic at times.
The last thing I want, is yet another variable I have to take care of.
I totally understand that it might work for some people. (it used to work for me as well).
But it wouldn't be my general generic advice, mostly because you just don't know what systems people have.
On the other hand, I also don't want to claim it to be holy (like some do).
Can anyone tell me why I'm not able to offset time zero? I take a measurement and I click the options tab enter a new value in the offset window but zero never moves...I've watched a video on YouTube and followed what he did he enters the new value and zero shifts he enters a different value and it shifts again...I get nothing...what am I doing wrong?