# What is "use loopback as timing reference " used for?



## acousticignorant (Oct 29, 2011)

What is "use loopback as timing reference " used for? I read up on it in the help page but it didn't quite make sense to me. Do I do another loopback measurement with both L+R out to in and have that box checked before measuring? Is that only for doing impulse responses? If I don't check that box and just do a regular measurement will everything be off by how much in and out latency is cause buy my audio interface? Wow REW is more confusing that I thought I would be. Any help would be greatly appreciated.


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## gotchaforce (Dec 11, 2008)

the "loopback" is the same thing that happens during calibration

left output to left mic in = instantaneous signal, right output to right mic in = signal from your microphone that is delayed because its traveling through the air.

thats the timing reference, just is timing how long it took for the frequency response to come out of the speaker and hit the microphone

not necessary for most people... so i wouldnt worry about it


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## jianhua1975 (Sep 13, 2011)

hi, 
just my understanding, if you have this option checked, you need to loopback one in towards on out(R-Out to R-in, L-Out to L-in, or with "Y"split adaptor with one out to one in) while using another pair of In&out for measurement. this is used to have the time delay caused by Internal Soundcard&PC signal transfer time as reference. 

that's what i understand! hope it's right!


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## acousticignorant (Oct 29, 2011)

Hey thanks for your help. I think I understand a little better now. My focusrite 2i2 Usb audio interface has 4ms in/4ms out latency when at 44,100 samples a second and 3ms in/4ms out at 96,000 samples a second. So if I want the correct measurements I should get a Y adapter and run the split audio to my m-audio bx5a's (bi amped monitors) , loopback the other line out channel and plug in the ECM8000 into the open mic in channel? Is it ok to have the phantom power turned on my audio interface for the mic while the loopback cable is plugged in as well? I hope I understand this a little more now...


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## SAC (Dec 3, 2009)

gotchaforce said:


> the "loopback" is the same thing that happens during calibration
> 
> left output to left mic in = instantaneous signal, right output to right mic in = signal from your microphone that is delayed because its traveling through the air.
> 
> ...


No.

Using a loopback for timing reference is NOT the same thing as calibrating the frequency response of the soundcard.

REW can also be used for time domain measurements.

If you are intending to use REW in order to evaluate specular reflections, the arrival times of the direct signal and the indirect signals are important.

The problem arises in that the test signal does is not processed in zero time - there is some hardware propagation delay inherent in the hardware setup.

In order to eliminate this, a hardware loopback is used in the 2nd channel (typically the Left channel to connect Left out with Left in) to provide a reference for the delay in signal propagation that is then 'subtracted' from the timing of the actual test signal data received from the mic.

In this way the actual "time of flight" of the test signal from speaker to mic can be determined. 

If all you are doing is making frequency response measurements (rather a waste of the capabilities of the platform), you do not need this feature. But if you are intending to move beyond the simple generation of frequency responses, you will need to know what the characteristics of the various direct and indirect signals are that interact (superpose) and that cause destructive interference in the frequency response. 

Above the modal frequencies (actually above ~80 Hz), the frequency response is comprised of the interaction of non-minimum phase interaction of various direct and indirect energy arrivals, and EQ is NOT a valid method to correct for frequency domain anomalies, and this behavior must be viewed and treated in the time domain.

And for this, the ETC response is required. And for accurate time of flight values, the hardware loopback correction must be used in order to remove and compensate for the hardware propagation delay that is added to the measured time. (Note: In Preferences-> Analysis: "Use Loopback as timing reference" MUST be checked; and also "Set IR peak to T=0" must NOT be checked - as it is of no value anyway as it deletes the initial Time of Flight of the direct signal and ignores the hardware loopback correction!)


Thus the best advice I can give is to use the loopback correction by default. Then, when and if you do desire to convolve and view the time domain aspects of your measurements, the time values will be valid in the time domain. 

There is much a larger very significant world of acoustic behavior than that which exist below 80 Hz!





acousticignorant said:


> Hey thanks for your help. I think I understand a little better now. My focusrite 2i2 Usb audio interface has 4ms in/4ms out latency when at 44,100 samples a second and 3ms in/4ms out at 96,000 samples a second. So if I want the correct measurements I should get a Y adapter and run the split audio to my m-audio bx5a's (bi amped monitors) , loopback the other line out channel and plug in the ECM8000 into the open mic in channel? Is it ok to have the phantom power turned on my audio interface for the mic while the loopback cable is plugged in as well? I hope I understand this a little more now...


Huh?

With all due respect, based on this and the other recent soundcard calibration thread I don't think anyone is understanding either concept well.

Soundcard calibration is done separately from the hard ware loopback and also has nothing to do with the mic or phantom power! Nor do you need any input from a mic (and hence any phantom power!) to calibrate the sound card!

Correction for the mic's frequency response is provided separately by a calibration file specifically generated for that purpose by the manufacturer or aftermarket test facility.

The hardware loopback and phantom power has nothing to do with your speakers (or any other external connections for that matter.)! And you cannot determine the propagation delay in the manner you propose.!
The phantom power is used to power the microphone! Period! Hardware loopback is use to provide a reference time baseline to compensate for propagation delay. All of this has nothing to do with your speakers or anything else!

Folks need to actually read and follow the process step by step, as it appears that folks are assuming they understand the concept and are jumping ahead and trying to combine several separate functions into one.


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## acousticignorant (Oct 29, 2011)

I have been reading the help in the REW program and I started with the first page and worked my way on. I don't fully understand everything that is written in it, otherwise why would I be posting questions. Maybe I should learn more about acoustic programs such as REW before I even try to use them. I know that phantom power is for mics only. I know more now about calibrating a soundcard (that was like the first step that I did with no problems). I sent my ECM8000 to cross-spectrum labs to get the .cal files (90° and 45° off axis ) waiting to get it back now. I don't understand what you said about propagation delay (what is that?) or ETC responses (again what is that?). What I want to do with REW is to see where there are errors in my room. So I probably need to google those both to figure out what they mean. Maybe it was my mistake to think I could figure out how to use a program like REW to find out my room errors without understanding all the jargon . I make music and i just want to see what my room errors are so I don't over or under compensate frequencies in my mixing. I'm going to try reading the help files more and hope it will all make more sense. I wish the help files were written more for people like me who don't understand all the jargon within this software. How about REW help pages for idiots like me?


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## acousticignorant (Oct 29, 2011)

I just finished reading more help about REW. This time I read what was posted in http://www.hometheatershack.com/roomeq/wizardhelpv5/help_en-GB/html/primer.html#top and still had a hard time making sense of it all. Maybe I should just sell my ECM8000 once it gets back to me (I'm keeping my RS SPL meter regardless ) and buy JBL's MSC1 or KRK's Erco room correction hardware and a good pair if studio reference headphones . All I want to do is understand what are the problem areas of my room are so I won't under or over compensate frequencies during my mixing (like I've done for years). I have spent a fair amount if money on acoustic treatments and know that I probably didn't fix all the issues in the mixing room. I have looked on so many websites for info before I made/put up my DIY broadband absorption panels. I focused on the early refection points on the side, ceiling and rear walls and put up some bass traps. Now if could just like to learn how to use REW to find my room errors before I throw in the towel and fork up 300+$. Well I'm going to keep reading more help files and hope it all starts to make some sense soon. Hoping for a revelation.


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