# 5.1 HT system measurement/calibration



## Nerys64 (Mar 17, 2013)

I have some (presumably) humble goals and desires.

I have a 5.1 system. currently the subwoofer is hooked up as a L/R pass-through, I have no crossover adjustment, and all in all my receiver is breaking so I am replacing it. I will be getting an Outlaw 975 processor and a 5 channel amp (outlaw 7125 or 7025, depending on funds).

I would like to take some detailed measurements of my room, each speaker individually, all speakers together, and all speakers near-field (mic close range) using Room EQ wizard. I wanted to do this both before and after the new processor is placed. *Q1: Is Room EQ wizard the right software for these needs?* 

The processor lets me assign a crossover on a per-speaker basis. IE if my rears can go down to 80 but my fronts can go down to 40, I can have the crossover for the rears be at 80 and the crossover for the fronts be at 40, and let the subwoofer handle all the rest of the signal. I can furthermore equalize the subwoofer output within the processor. To do this, I will need to measure using sweeps to each individual channel. *Q2: Can room EQ wizard do multichannel sweeps?*

My plan is this: sweep individual channels and measure them. Then set the crossover at that poipnt in the processor. Measure the subwoofer with this crossover, and then equalize it so that it has a flat response. Do this first for Front Left/Right, then for Center, then for Rear Left/Right.

I am hoping I dont have to constly keep disconnecting and reconnecting things in order to make Room EQ wizard play to my rear channels and such. Right now, I've got the crossover set by ear using the program from the link below. It doesnt work for multichannel because my ASIO driver doesnt support multichannel.

My connection is optical from PC to receiver, so I need to output using primary windows driver. That way, the sound card can package the sound using DTS-Connect, send it to the receiver, and have it decode as DTS. To make it a bit clearer, for a 5.1 channel FLAC file, I set VLC's output device to 5.1 (intead of a52 over s/pdif), set the primary device to my sound card speakers, and enable DTS-Interactive/DTS-Connect.

My sound card is an Asus Xonar DSX.
This is the measurement equipment I will be getting.
This is the Tone generator I used for Front Left/Right only. ASIO doesnt work with DTS-connect, so no output to my rears/center/sub channel.


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## Phillips (Aug 12, 2011)

Nerys64 said:


> I have some (presumably) humble goals and desires.
> 
> I have a 5.1 system. currently the subwoofer is hooked up as a L/R pass-through, I have no crossover adjustment, and all in all my receiver is breaking so I am replacing it. I will be getting an Outlaw 975 processor and a 5 channel amp (outlaw 7125 or 7025, depending on funds).
> 
> ...



1. I think that REW will fit your needs.

2. If you intend any EQ etc i would highly reccomend a CALIBRATED MIC (ECM8000 if you want). You can post/contract Cross - Spectrum (Herb) for this. I am not familiar with your Pre/Soundcard.

3. Another option is USB mics (Dayton UMM-6 from Cross - Spectrum and MiniDSP UMIK-1 form MiniDSP) that are portable. Both of these are calibrated. 

4. Multichannel sweeps can be done with setting the receiver, i do this often with my receiver.

Hope this helps a little


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## Nerys64 (Mar 17, 2013)

I was originally planning on getting the uncalibrated behringer mic. Last night, I ran a the calibration. I tried going from sound card out -> reciever -> headphone out -> mic in on sound card. Then, I tried sound card direct analog out -> mic in on sound card.


















I'm not sure if these are good or bad results, but I'm assuming it doesnt matter because these are the results that would be used to bias/calibrate whatever input the microphone is giving in order to simulate a perfectly flat response after the inadequacies of the sound card are found.

Either way, the behringer ECM8000 uncalibrated would cost close to 100 bucks after the phantom power and cables necessary, so i'll probably just pick up the UMM-6 or UMIK and be done with it.

I am a little worried about the initial measurements of my "before" setup. I have set windows to a 5.1 configuration but my sound card works a little oddly because I am limited to an s/pdif connection at the moment. As you guys know, the only 2 ways to send 5.1 data over S/Pdif is by encoding/compressing it with DTS or Dolby Digital. There is no way to send bitstream or PCM 5.1 channel over s/pdif.

So what I have to do is set everything to default, enable DTS-connect on the sound card, and make the application force output of 5.1 sound. For video games, this just works. For VLC it automatically sets it to a52/spdif or 5.1 (which I think means 5.1 individual streams). I dont know how REW will do this though.

Am I just going to create 6 seperate mono measurements and overlay them? Or will I be missing out on some functionality if I do this? The first and foremost goal is to properly EQ the subwoofer and set up my crossovers properly. After that, its just information gathering and data mining, like what is the capabilities of each channel of my speaker.

For this, do I make REW play some pre-ordained tone or does it have its own built in thing? Or do i download a DVD iso or flac test track and do it that way? Or do I use like a tone generator or function generator? I've been looking through the documents but my specific questions arent being answered, rather just my general questions.

From what I understand, REW will test the loop back and use that as a calibration. This way, regardless of how bad the input of the sound card is (and presumably the output), it is already accounted for. Then there is a mic calibration file, which lets the program know to further account for the inadequacies of the microphone. Finally, the measurement takes place and at this point I usually stop reading because that is more information than I need for the moment. I'm mostly concerned with the testing setup and limitations.


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## Phillips (Aug 12, 2011)

> Am I just going to create 6 seperate mono measurements and overlay them? Or will I be missing out on some functionality if I do this? The first and foremost goal is to properly EQ the subwoofer and set up my crossovers properly. After that, its just information gathering and data mining, like what is the capabilities of each channel of my speaker.


Get the mains and subwoofer crossover sorted out first, then do the rest of the speakers



> For this, do I make REW play some pre-ordained tone or does it have its own built in thing


All the tones are built in, including tone generator for RTA



> From what I understand, REW will test the loop back and use that as a calibration. This way, regardless of how bad the input of the sound card is (and presumably the output), it is already accounted for. Then there is a mic calibration file, which lets the program know to further account for the inadequacies of the microphone. Finally, the measurement takes place and at this point I usually stop reading because that is more information than I need for the moment. I'm mostly concerned with the testing setup and limitations.


There is no way to do a accurate soundcard (computer) calibration. There would be no loopback.
USB mics can't do distance settings (as far as i know) due to not been able to do the loopback. I am not sure if John will be able to over come this.

If you go down the road of USB mics you don't need soundcard calibration file, just import (REW will ask for it when opened) and load the mic calibration.

Cable from the laptop to the receiver e.g. headphone out. Then you are good to go.

Hope this helps a bit


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## Nerys64 (Mar 17, 2013)

It helps a little but I'm still kind of confused. Is it a good idea to go with a USB mic like the UMM-6 or DSP1? I understand that i wont get any "distance setting" but what does loopback have to do with this? And wouldnt distance be an in-software type feature?

I thought loop back on a soundcard would be going output to input. according to another user, "The loopback (aka soundcard calibration) “tells” REW what the frequency response of the sound card is, so it can add corrections to assure flat response." So it is obviously a janky solution at best, but going headphone out or at least sound card OUT to soundcard IN should yield the loopback scenario.

I dont know, perhaps it is better to go the UMM1 option, use the provided calibration file, and then simply test that way, as my reciever is hardly flat in its headphone out response to begin with, and i doubt the headphone output mirrors the speaker output. If I change output devices to something I know has a flat analog output, it will mess up the calibration, I think.

I'll try one more thing: I'll go straight from my USB DAC to the sound card input. I'll keep all sample rates at 44.1 because that is the only input my sound card takes. I'll see if that produces another similar output. If so, then I think I'm ok to measure regardless of soundcard input or USB mic. If it is different, then my input/calibration is being tainted with the output characteristics of the devices themselves, but I dont think that is the case.

Either way, I keep seeing a pre-calibrated mic as the best option as far as computerized measurements go.

My next issue is how to get a multichannel measurement. Specifically, where to source the signals from.


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## Phillips (Aug 12, 2011)

> Is it a good idea to go with a USB mic like the UMM-6 or DSP1? I understand that i wont get any "distance setting" but what does loopback have to do with this? And wouldnt distance be an in-software type feature?


Going with the USB mic is totally up to you.
No it isn't as easy as hitting a button, but you can measure with the Frequency Response graph and get the flattest/best response. 
Also Recievers of today are pretty good setting these, so you could fine tune via above suggestion.
Also phase can be set via Frequency Response graph by getting the flattest response through the crossover.



> I thought loop back on a soundcard would be going output to input. according to another user, "The loopback (aka soundcard calibration) “tells” REW what the frequency response of the sound card is, so it can add corrections to assure flat response." So it is obviously a janky solution at best, but going headphone out or at least sound card OUT to soundcard IN should yield the loopback scenario.


That is right if you are using e.g. Behringer ECM8000 and Tascam Pre/Soundcard. 
The soundcard calibration file basically does what the mic calibration file does, when loaded into REW the program takes this into the measurement to give a accurate mesurement. 
If you are going to get a USB mic there is no way of (accurately) measureing your internal laptop soundcard. This has been confirmed by the creator of REW. 

I dont know, perhaps it is better to go the UMM1 option, use the provided calibration file, and then simply test that way, as my reciever is hardly flat in its headphone out response to begin with, and i doubt the headphone output mirrors the speaker output. If I change output devices to something I know has a flat analog output, it will mess up the calibration, I think.

I don't understand why you are going to use the headphone out from your reciever for REW?
If you are talking about (USB mics) plug the mic via a USB port, to get test tones from your laptop to your receiver, *laptop headphone out > reciever input (spare e.g. CD input) * thats it.


As i said in a earlier post i am not familiar with your soundcard, hopefully someone else will.


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## Nerys64 (Mar 17, 2013)

I see. Firstly, I am on a desktop computer, not a laptop. I have a full multi-channel ANALOG output capability, but this capability is not that great. My method of purchase for the sound card was: what is the cheapest thing with which I can do 5.1 gaming?

This sound card, as I've stated, can compress live playing audio into DTS so that I can do proper 5.1 over the s/pdif connection. That is the only way to get a decent signal out of it. The analog outputs are fairly terrible and kind of noisy also. The mic input is probably awful, if those above measurements are any indication. I did repeat the test, as I had done it incorrectly the first time. I did straight sound card analog "front LR / headphone" output going directly to the microphone input. This is the shortest path the signal can take, and presumably measures the sound card well. It doesnt take into consideration the fact that my analog output from the sound card is pretty terrible.

Then it goes directly to the microphone input, which we are already assuming is terrible and hence are attempting to calibrate.

The purpose of going through my reciever headphone out is that I have far greater confidence in the receiver's output than I do in the analog jack. Furthermore, I wanted to make sure that the measurement matched fairly closely, and was expecting a less noisy/"better looking" result from the mic input coming straight out of the receiver's headphone jack, as all the DAC and noise of the computer itself is bypassed and only the receiver's gear is used.

What I will try now is to go straight from my USB dac into the microphone input. This will give me an "incorrect calibration" technically, but will give me the most linear and acceptable input to the actual mic port that I can manage. Any further issues after this calibration is done (IE any errors in measurement) will be 100% in the signal path of S/PDIF -> receiver DAC -> receiver amp. That is a whole lot of variables to account for, but they wont make much of a difference except for high and low frequency roll off.

And here it is:









Now, compare that nice, pretty, linear output going to my mic input to the actual analog output of my sound card, when everything is set up properly. Both from the receiver headphone out (should be higher quality) and from analog sound card out (should be lower quality)

http://imgur.com/a/XjgwG#0

They are not "regulation screenshots" so I have only posted the link here. It looks pretty identical, and seems that the mic has some sort of horrendous high frequency issue that I cant even begin to fathom how it will correct.

And it is because of this sort of anomoly that I feel like the pre-calibrated UMM-6 or UMIK-1 would be a better choice to do these measurements. If I can bypass my purposely sub-par chosen hardware (i literally just wanted an optical port with DTS-connect), I should be able to get fairly accurate measurements. Now, they may not be ENTIRELY great sounding if there is a problem with calibration in the mic, but considering they are calibrated for a fairly broad frequency range, I think I can live as long as 30-16khz is fine. Not to mention, there is the house curve to accomodate for stuff that I hear that I dont like, even if it does measure well.

I've saved the entire measurement configuration so i can post more screenshots of phase and spectrogram and whatnot. I think this one is more or less the winner.


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## JohnM (Apr 11, 2006)

There is certainly something drastically wrong with those loopback measurements. Are you sure there is no feedback loop? No 'Listen to this device' selected in recording properties for the input, or line in being mixed back into the output?


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## Nerys64 (Mar 17, 2013)

Absolutely certain. The first measurement was definitely flawed: I had windows recording device set to Microphone instead of Line In. My Line In measures at 44.1 while my sound card was outputting at 48.

The latest measurement that you see is the most accurate one yet, and those 3 are giving all the same-ish results. One is from REW Signal -> DAC (very linear output) -> Amp (very linear output) -> Sound card input, configured to Line In, 44.1k for everything's sample rate.

The other two you see are my sound card out (analog, jack labeled Front L/R) -> Mic input on the sound card, everything configured to line in, 44.1k. And finally, REW->S/Pdif -> Receiver dac -> receiver's headphone amp -> microphone input.

I should mention, when I do the receiver version and create the loop back, a huge ground loop issue occurs and my subwoofer starts to hum. Not loudly, but suddenly noticeably.

I can attempt to run it again but I'm certain I measured correctly this time since all 3 outputs are matching up. Either something is screwy with the actual line in port/hardware or something is screwy with my output on the sound card. I'm betting its the line in, because 3 different output sources wouldnt yield such a similar measurement.

Line in is configured as: monitoring disabled, level: 100, 2 channel 16 bit 44.1khz, exclusive control allowed, applications allowed to take exclusive priority. The loop back I connect to (mic in) is directly on the sound card, not via computer front panel.

EDIT: I have figured it out. I had one of my previous terrible measurements loaded into the MIC CAL box. I reset all preferences, re-did the test using just AUX, and now it seems to be outputting correctly. I'll post the results in just a tic.


Here are the images of the new measurements. Much to my chagrin, my soundcard provided seemingly the best measurement. I dont know if this means I need practically 0 calibration to the mic input (and should get the behringer ECM8000 uncalibrated after all?) or what. I cant really make too much sense out of these, like what it means that the low frequency is out of phase for 2 of those tests but not the sound card test. It does make me happy that i invested in a proper dac/amp for my HD650's though.


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## JohnM (Apr 11, 2006)

Nerys64 said:


> EDIT: I have figured it out. I had one of my previous terrible measurements loaded into the MIC CAL box. I reset all preferences, re-did the test using just AUX, and now it seems to be outputting correctly.


Well done, though I'm a bit annoyed at myself for not thinking to ask about that. Worth posting an mdat file as well as screen shots when things are not behaving as expected, becomes much easier to spot that kind of thing.


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## Nerys64 (Mar 17, 2013)

I went ahead and ordered the UMM-6. This calibration is pretty nasty, but what it does tell me is that my mic input is pretty linear and that my receiver has some weird problems at 8khz and 15/18 khz for some reason.

I'll plug in the UMM-6 with its own cal file once I get it, so at least that will be a fairly accurate measurement, if not a precise measurement. My next issue is how to do a multichannel output. My asus soundcard ASIO drivers do not support multichannel PCM for obvious reasons: If it did, the multichannel signal would bypass the DTS-connect software DSP, and only output PCM front left/right to the receiver.

So I have to figure out how to measure individual speakers. Obviously I can measure front left/right standalone. I cannot measure subwoofer standalone without physically disconnecting the front left/right, as I have no dedicated subwoofer pre-out. So that isnt terribly difficult. I think i'll head to the manual again once I get my mic and am ready to start measurements.

I just wish there was a way to send only a single signal to the speaker of my choice rather than unplugging, because so far I cannot find anything except HDMI/ASIO drivers that will allow me to do that, and that is not an option with this receiver even though I have the proper output capabilities.


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## Nerys64 (Mar 17, 2013)

I ran the first measurement. I am very confused on this SPL/Levels thing with the UMM6. I do the SPL meter, set it to A weight, and hit calibrate. Then i adjust the box until the recorded SPL from my microphone is the same value in the box (78-80db). Then, when I hit Measure and Check Levels, it shows that I have too little headroom. But when I start the test, RoomEQwizard tells me there was clipping.

The phase response is VERY jagged in my room. The frequency response is fairly flat (more or less, since there are tons of jagged lines in the measurement) but bloated in the bass region from 30-200hz. My subwoofer is crossed over at 80hz right now, so it tells me that my speakers are bassy by default and that if nothing else, I could stand to turn my subwoofer down a tiny bit, or even up to even out that curve at the bottom. It would look more like a house curve then. Either way, i do like the way things sound right now and I'm hesitant to change anything unless I know that everything is calibrated properly and that i'm taking the measurement correctly.

How do I calibrate a UMM6's levels and SPL?


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## JohnM (Apr 11, 2006)

If REW is warning that headroom is too low you need to turn down the input level from the UMM-6 in the Windows volume control. Re-do the SPL calibration afterwards.

Use C weighting on the SPL meter (both in REW and the external meter you are using for calibration).


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## Nerys64 (Mar 17, 2013)

I dont have a properly calibrated SPL meter, or any true SPL meter. i only have my smartphone. Is this a problem?

Correct me if I am wrong, but I need to take my SPL meter app on the phone and whatever DB level it is telling me, I put into the calibration box. So SPL calibration plays a certain signal, whatever the SPL is that is measured on the phone tells REW that at my receiver volume, even though REQ might be trying to play at -80, the actual volume I am picking up is -70 (or whatever it may be) and it adjusts the USB mic input accordingly.

Then I click "measurement", and hit "check levels". It is at this point that I am confused. I cannot get the levels correct. The signal being played is -12db. The signal being recorded should be at what? somewhere between 24 and 18 db?


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## JohnM (Apr 11, 2006)

Nerys64 said:


> I dont have a properly calibrated SPL meter, or any true SPL meter. i only have my smartphone. Is this a problem?


That's fine.



> Correct me if I am wrong, but I need to take my SPL meter app on the phone and whatever DB level it is telling me, I put into the calibration box. So SPL calibration plays a certain signal, whatever the SPL is that is measured on the phone tells REW that at my receiver volume, even though REQ might be trying to play at -80, the actual volume I am picking up is -70 (or whatever it may be) and it adjusts the USB mic input accordingly.


Right.



> Then I click "measurement", and hit "check levels". It is at this point that I am confused. I cannot get the levels correct. The signal being played is -12db. The signal being recorded should be at what? somewhere between 24 and 18 db?


Around -24 dB FS to -12 dB FS would be ideal, but it is not too critical. Just go ahead and make a measurement and see how it comes out.


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## Nerys64 (Mar 17, 2013)

Looks like my measurements are OK as long as REW notices the proper headroom. This one was when I said "Screw it" and just took the measurement at a level I knew was loud enough. Much louder than I normally play but not clipping the mic.

Now my biggest problem is figuring out how to make each speaker play a tone and getting roomEQwizard to measure it. I can only do front left/right + subwoofer, as subwoofer is connected to front left/right pre-out. I have no other way to connected except through the fronts and send it a full range signal.

I need Room EQ wizard to give me the option of which channel to play through, and then send a 5.1 measurement signal to that channel. So all channels would be blank except the channel doing the measurement. Then my sound card can use its DTS encoder to play it on the correct channel, albeit with somewhat compressed characteristics (which is fine, as that is a limitation of my system that I want to measure). Normally, I can take a 5.1 LPCM track and send it to my sound card, which then does the DTS encoding.

I cant use ASIO drivers. The only other option I have is to completely unhook my entire downstairs HT and use the analog to pre-in cables, but that would also screw up all of my settings and be a gargantuan pain. Any ideas on how to get a 5.1 measurement? What you see below is plain L/R+sub.


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