Loopback in REW is it important and how do do it?

Trdat

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This whole loopback thing on REW is confusing me. Do I need it to get an accurate measurement?

And the most obvious question how to do it? The whole thing sounds complicated for me. I have a presonus itwo soundcard and a standard NADC510 DAC what do I connect what with what. In laymens terms would be appreciated. T
 

John Mulcahy

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It depends what you mean by "loopback thing". To make a loopback connection, in layman's or anyone else's terms, you would connect an output of the DAC to an input of the soundcard. If you wanted to do that to provide a timing reference it wouldn't work because the input and output device are not the same, so the timing would vary from one measurement to the next. If you wanted to do it to compensate for the frequency response roll-offs of the DAC output and soundcard input then you would make a measurement and save it as a calibration file.
 

Trdat

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I am guessing timing reference is what I would prefer it for. I thought timing reference was crucial and thought that's what everybody does somehow connect the soundcard to the DAC. When you say it won't work you mean for my set up?

Essentially I need the input and output device to be the same. How can I achieve this?
 

John Mulcahy

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If you have separate devices for input and output REW's acoustic timing reference can be used. If you want to use an electrical loopback the output and input need to come from the same interface.
 

Trdat

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Yes, I think I got it. I can use the timing reference if the input and output are different which is what it is in my case, output NAD DAC and input Pre Sonus. So technically I don't need a loopback.
 

LCRLive

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This loopback stuff is still going past my head.
Could anybody please reiterate in which timing reference situations this method is needed? Because it's still unclear to me.

So if one is not using any separate DAC and only one sound card interface doing all the conversion of both a to d and d to a then I assume this loopback stuff is not needed.

Aside from this I've been taking some measurements and I assume I have been doing it the right way to get what I've been needing in room measurements. But for Speaker testing, I have a feeling this loopback method will somehow need to be taken into account due to aligning stuff a bit more properly in the time domain if I'm not mistaken? But why not just move the peak of the IR to zero? Isn't that the same thing if it's not lined up?
 
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