As I understand it, you need two channels of audio run through the *same* sound card to enable the loopback timing feature in REW. Since you have a USB microphone that does not use any analog sound card channels, you cannot use it with an external sound card. I think what you are asking for cannot be done unless you get a regular microphone like the Dayton EMM-6.

 

OK if i get the EMM6, what sound card would you suggest.? I have tried searching the past threads but it seems they are all over the place on this. It does look like I would need the EMM6 and a pre-amp. I'm just unsure of a good external sound card.

 

So the UMM6 or other USB mics, like the miniDSP Unk1 cannot do time alignment by plugging into a laptop?

 

If using a USB mic, REW will only align the peak of the impulse to time zero. There is no reference for REW to use to determine propagation time, for that you need a 2 channel interface, then you go into preferences, analysis, then check "use loopback as timing reference". Once you've done that, then you'll have the ability to measure actual propagation times. In order to "use loopback as timing reference" you need a loopback, in order to have a "loopback" , you need at least two channels, one for the mic, one for the "loopback". I've used Focusrite Scarlett 2i2, m-audio mobile pre mkII, Focusrite saffire6 USB, Presonus AudioBox USB, Behringer Xenyx 302, TASCAM us1800, AKAI EiE pro, in fact, I still have all of them except the Focusrite Scarlett 2i2, I fried that one. For sure the Focusrite interfaces have ruler flat freq. and phase response. They're my favorite for 2 ch. interfaces. The worst is the little Behringer, I wouldn't recommend that model, the Xenyx 302, it kinda sux for measurement.

 

I just ordered the UMM6 from Cross Spectrum to ship! :-/

I did a bunch of reading about the different mics, and I know about the noisefloor on it, but with the better calibration USB convenience, thought it would be a good mic to cover all bases in tuning stereo systems.

No where did I read that USB is not capable of measuring time alignment. What a MAJOR disadvantage! and they cost more! and have a higher noisefloor!


So do I attempt to cancel my order, get a EMM6 with phantom power?

 

Let me ask ya this, do you own an iPad? (Like an iPad 3 or mini, or iPad Air), because the umm6 works BRILLIANTLY with the CCK and AudioTools for iOS, I'd say, keep it. I have a umm6 from cross spectrum too, and I use it sometimes to reference the FR of other mics/preamps/interface combos. As the FR of the mic is a known, meaning it isn't reliant on having a measurement grade preamp for its cal file to be totally valid, it can serve as a good reference mic too.

 

Over on HTShack there are a few posts about the USB mics having that limitation, no timing reference, but they are much more convenient. I've got a UMIK and another phantom power mic with a usb moble pre that works, but with so many more cables. I prefer the simple one cable of the UMIK.

 

It can't be used with REW to measure relative differences in impulse responses, I DONT THINK, but, if that is what you are wanting to do, there is a pretty neat program for iOS called IOscope, by FaberAcoustical. It has a real time impulse response mode, with it and the CCK/UMM6, you could measure the DIFFERENCE BETWEEN two or more impulse response times...........

 

Hi Niick,

Yes I have a iPad Retina1/older plug, and a Mini with the newer charge plug.

This will be my first time using a RTA and mic. I'm not sure if the 3SIXTY.3 has the ability to do RTA, and take a mic, but I was planning on using a open source sw like REW to adjust my car stereo.

I didn't want to run into some limitation on Time Alignment due to the $120 for the mic I just purchased which I would likely use 1 or 2 times and I planned to give it to my installer.
He just opened up a shop and knows plenty and is certified trained and has experience, but not in all areas, and first time on his own and he can use some gear he hasn't used.
He is also new to tuning although he has some expereince on how things sound.


Here is the 3SIXTY output we are getting and it doesn't sound right.

I'm still listening to speakers in a door, and the timing maybe off as my tweeters are in the dash with mids in doors.....There is lots of tuning work, and we thought that a mic would help us see how the pink noise is shaped. He did so with a iPhone and the built in mic, and used that for a guide.

I've been on the forum for some time, and I had great responses to buying mu gear from folks. Then got advice to not go with big store installers, find a shop that is willing to work with me, and I did that too...

Now that I have it in, I posted my build log, and updated the below link, and Im just a bit stuck as I am learning how to tune with very little guidence. I had a sub issue, which I did get some responses back. And I learned and already should have known to leave that alone until the fronts and rears are done.


http://www.diymobileaudio.com/forum/2660794-post86.html

 

it'll work for taking rta measurements with REW... if you search around on the forums therapture has a 360.3 and used/uses rew to tune his system (and lots of other folks on here as well). The DSP has no RTA function in it, hence you use a seperate program REW or the like to take measurements.
For impulse type response measurements you have to have a a non-usb mic...

 

Or a different program that will do "propagation time" measures/?.

It would have been nice for REW to not force the functions at the same time and you could switch the input to select what it does. Save the reading and then use it in the equation. I haven't even installed the software yet

I'm not even sure I need it, but I do know my tweeters are far in the dash, my mids are low on the doors...While my seat is back my rear door separates(run with the passive xover) are closer to my ears as my seat needs to be pretty far back. Ugh!

 

There are other ways to get your time alignment dialed in that won't require you to buy a bunch of expensive equipment.

Keeping in mind that time alignment is used for finely tuning the phase relationship of two drivers, you can use SPL measurements to optimize your time alignment.

When you have the phase correct the waves will combine constructively, giving you the max SPL possible between the two drivers. Start with physical measurements using a tape measure to get you in the ballpark. Take SPL measurements of each driver and set the levels so that, at the listening position, they are as close in SPL as possible. Get some test tones within the bandwidth of the drivers that you are using and play one at a given SPL, move the mic around and get an average. Do the same with the other driver. Once you have both drivers playing the same SPL you need to play them together. With both drivers playing you are looking for as close to 3dB higher, on average, than the SPL of either single driver. Make slight adjustments until you get the highest SPL you can with the two drivers playing.

At this point you don't need to worry about aligning subs to mids, or mids to tweets, you just need to know the difference between each pair that results in the most constructive phase. Make note of the difference, either in inches, cm, or seconds, depending on the equipment you are using to set delays. This difference will remain the same, even if you add several inches to either pair of drivers.

It may sound time consuming, but it will be time consuming either way, and this way can save you a lot of money, as well as give you the same results.

 

OK I'm still confused. All non USB mikes have the XLR connection but sound cards do not accept this connector. So how would you setup the loop back cables. I'm to much of a newb at this so i pretty much need someone to tell me an inexpensive external sound card that I can use with a EMM6 and explain to me exactly how to hook it up and what cables I will need. If you could do that I will sing at your wedding!!!!!

 

You want an external sound card with an pair of microphone preamps. You will know they are microphone preamps if there is another switch that toggles the "phantom power" on or off. You will also need this feature.

You will also want an external sound card with a pair of outputs. These can be regular RCA connectors. Your external sound card needs to have something called "full duplex" which means you can both output sound and record sound *at the same time*.

There are some sound cards mentioned above, and I have also had great success with the M-Audio Fast Track Pro. It's out of production but can be found on eBay for a reasonable price.

Get a microphone and an XLR microphone cable. Plug that cable into input one, turn on the phantom power option, and set the level to about 75%.

Create a loopback cable from the sound card's right output to the sound card's right input. Then connect the left output to your sound system through a y-cable.

Then look for Hanatsu's threads on this forum about how to setup and use REW.

 

Thanks giJoe. All this when going active sure adds the complexity of doing what you said AFTER the phase correction...Which is?

I'm not sure if you mean "phase" correction is the part where we try and make a nice looking graph. I have a muddy idea of what phase was. I remember the explanation when 2 drivers are "not in phase" and sucj, but dont really know what that defines. Likely something Im familiar with but not knowing the name

We did manage to get a flat graph with minor adjustments after the 3SIXTY3 did its sweep thing analysis and flattening job. So now that its pretty flat, it sound pretty bad.


Martin, Turtle Beach and Creative/SoundBlaster make good cards. Not sure if TurtleBeach is still doing this, but the latter would be what you want. They have multiple inputs. Should be about $150 or less for something nice. You can get something around $40, as long as it has an extra input, which many do.
But what GIJOE is saying maybe all we need ...but not sure when to do this.

So after the speakers are installed and since they are active and we have this DSP the 3sixty3, we of course wanted to play with this thing, and we started, and got the 363 to do the source signal flattening, and then we started to adjust things and setup the speakers for what they are and their MFG recommended cross over points to start with. But thats the left side. Then there is the graoh on right top and the EQ adjusts below it....This area is a ball of wax. We got something to sound decent, but the graph slopes don't look right as there is not proper overlap and fall off.

This is what we have now on one of the presets....
http://www.diymobileaudio.com/forum/2660794-post86.html

 

All of this is pointless, unless you understand phase. For a quick reference look at this photo. The two sine waves are in phase, when they are in phase they combine and create the wave at the bottom.

When we talk about phase, we are talking about how those two sine waves line up, and how the either add to give a final acoustic wave. Phase is relative, there is no such thing as phase unless there are two different waves, mute all but one speaker, switch the polarity of the wires on that speaker, there is not change between the two, the change is only relative to another wave. In the example, the waves are perfectly in phase, so all of the energy combines creating an identical wave that just has more amplitude (same frequency response, only louder).

If you think of the X axis as time, shifting one wave to the right means that it happened later, and the summation of the waves will not be the same as if they are in perfect phase. You can measure this fairly easily with a USB and REW, you just need to understand what you're seeing.

If two speakers are exactly the same distance from the listening position, TA isn't necessary, the waves are already in phase, this is the reason why kick mounted speakers used to be so popular, before TA. If the speakers are already the same distance, you don't need to make any adjustments. TA is all about shifting one graph incrementally to get it to line up with the other, giving the most constructive interference as possible.

 

This is helpful gijoe.

So when we do the RTA/REW, I will have 1 tweeter playing in as close as possible a graph plot as the stock passive xovers would want to plot it. Once that is done, THEN we do the phase graph to match tweeter to tweeter, the mid to mid.
As long as they are in sync at that point we can retune the freq to a "optimal" adjustment.

Is that right?

More info like this and I think the Fibonatcci and golden rule formulas together with this, we can have all the pyramids in sync, and control the sun


I tend to want to set the speakers up for center mic adjustment. Because I think having it for the driver maybe great, but then the front passenger is now even more out of sync. why not compromise a slight off, vs a strong off for passenger?

How much of a difference will a mic set at driver position make vs one set a few inches away from right shoulder(sort of center/center console)?

 

Here is how to connect and use REW on my car:


Sound Card = Behringer FCA610
Mic = Dayton UMM6 XLR from Cross Spectrum Labs


My car system is a little bit complicated because I'm using the Factory head unit with a mobridge preamp. The Audi MMI has a audio connection port called the AMI. There are different plugs for the AMI. One of them is for iPhone. Another one accepts RCA audio/video.

I use the RCA inputs on the AMI for REW. I run a pair of 1/4" TRS jacks out of my sound card and then use a 1/4" to RCA adapter to connect to the AMI plug. One of the TRS jacks out of the sound card is for the audio signal from REW. The other jack outputs the Loopback reference signal.

In my DSP, I use the input matrix to route only the left channel input to all of the outputs. This keeps the audio signal only playing to the speakers. On an extra RCA output from the DSP, I assign the right input so that it only reads the Loopback reference.

Lastly, I run a single RCA to 1/4" TRS cable out of my DSP and back into my sound card. This is for the Loopback reference.



I think it is important to route the Loopback through your same audio signal flow chain as the main signal, so that you are assured the timing differences are due to speed of sound in air and not due to latency of devices.



I will try to draw a picture of the cable connections I make and post it here later.


Sent from my iPad using Tapatalk

 

I'm not suggesting that the method I described earlier is the best way to set time alignment, I was simply pointing out that there is a way to do it that doesn't require purchasing a new soundcard and a new microphone.

I am perfectly satisfied by using a tape measure to get close, then making adjustments with my ears.

TA isn't magic, distance is the key variable here and taking physical measurements with a tape measure will get you very close. It really is that simple to get good results. After I plug in the actual distances I play a a pair of speakers at a time and make slight adjustments until the image is centered. There are several clicks one way or the other that sound very similar, once the actual distance is entered, but you know that if you measured correctly, then you have to be in the ballpark. I usually take some measurements of amplitude first to make sure that each driver is playing at the same amplitude, but even then, moving my head a bit changes things, so I use the RTA to get it right on paper, but I trust my ears to help me center the image by attenuating where I feel it's necessary.

There are people who swear by using your ears, and there are people who swear by using measurements. I say that you need a combination of the two. If the RTA shows that both sides are the exact SPL, but the image is too far to the left, then attenuate the left a bit, it shouldn't take much.

Time alignment isn't that complex, it's simply a matter of the distance the sound travels from the source, which can be measured with very good accuracy with a piece of string and a ruler. Don't over complicate the process. Take your time to measure the distance accurately, then make minor adjustments. If you find yourself making big adjustments, then you've probably done something wrong.

EQ is a different animal altogether, and this is where independent L/R EQ makes the difference, but TA isn't that complicated.

 

I understand this gijoe and I appreciate your help. But now that my OCD has kicked in on doing TA with REW, I will not sleep until I make this happen. I need some meds.

 

This is getting more helpful by the post.

Yes Martin a full duplex card as mentioned I think would be needed for what you want.

subterFUSE, you have the same mic I just ordered, although my order was a Basic+, and not a XLR, which I dont know what that is.

But If you can omit the Audi specific info as you explained it to other Audi folks above, I might be able to understand the setup.

I too kept the stock HU, but then goes to the 3SIXTY.3, and then amps.


Right now I'm reading "Chuck Music's" post that Mr Marv posted on tuning.
http://www.diymobileaudio.com/forum/379331-post1.html

After this read, I'll go out with the laptop and the 3SIXTY3, as it may take a few days more for the shipping of the Cross Spectrum UMM6 mic to come in the mail.

And I'll try and make some improvements.


Wopuld love to see a diagram or any aid to help understand the setup. I looked at Youtube, but not much at all in setting up the hardware. A few useful things about REW.

 

Good to know GIJoe. Makes good sense.
I have no prob dailing in the distance using a tape measure.

 

Phil, this is all very good advice. I'd say, unless you have a real passion for audio, and how it all works, it's unlikely that you're ever gonna get to a level of understanding that will facilitate you the ability to optimize that 360.3 and your other associated equipment.

However, it sounds like you do have a desire to understand this stuff, which is exactly what I'm trying to do also. Every day I read, research, experiment, and when an install comes in that affords me the opportunity to put all that into practice, those are the good days.

I'm still learning, and I think I always will be. Like gijoe said, phase is the key how two different speakers will interact. Really learn the meaning of phase, and it will all make a lot more sense. What I do, is set up experiments outside the cars and take measurements while adjusting things like polarity, distance, axis, stuff like that. It helps you familiarize yourself with the equipment and what different measurements can mean. It's a whole lot easier if you're not trying to figure out your test instruments and how they work when your trying to tune a system.

Sounds to me like you're on the right track,

 

Thanks Niick, yes, I do have a bit under me to unbderstand this stuff. I lived with my dad who was at his home studio for most of my life, and if he wasn't sleeping he was mastering recordings and replaying recordings in multiple formats, speeds, heads, and all kinds of equipment, even reel to reel high speed recordings to expand the sound and remastering to digital....so I grew up with that passion for audio under multiple sets of speakers and audio devices.

But I myself was not taught the science behind sound.

I'm not taking the 2 months of research and effort of this install, and the years of listening to sound detail(in mostly home or studio systems), and the investments I made in the gear I purchased, and the time and money spent on my install to have slightly better than mediocre sound...which is what I have now. I should have the components disappear as I start to listen to real music and get that few seconds, now and then instances, as if the musicians were in the car with me.....

Perhaps hear the singer at the mic had to part his/her lips, or take that one quick breath before singing....to experience that. Or the guitar strings the fingernails ever so slightly caressed as you think the stings were at your ear.

I don't know if my gear is able to do that, I think it is to some degree...but we are far from that at this point

here is the 3SIXTY3 graph now.

http://www.diymobileaudio.com/forum/2660794-post86.html

I will be going to the car tonight or tomorrow and start applying Chuck Music's advice.

 

heres a drawing I snagged off of a thread on here recently...

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I bought a art dual usb pre like this

 

well this would have been helpful before I ordered the USB mic :-/

this looks like it takes the standard mic din input and then from this via USB to a PC?

(which drops the TA as well)

 

I don't think you have to use a loopback to make it work.

Here's what I do:

1) I use HolmImpulse
2) I have Holm "autodetect" the impulse. (Click the "Data Analysis" tab and select "detect time zero.")
3) Once you've let Holm determine where the impulse, lock it by clicking on "Time zero locked (time alignment)"

That's all there is to it! Now that the time zero is locked, you can figure out the distance by simply subtracting one impulse from the other.

Here's an example of what I mean by this:

If you record an impulse of the left speaker, then record the impulse of the right speaker, and the right speaker is one millisecond behind the first speaker, now you know the distance. Because sound travels 34cm in one millisecond.

This is really useful stuff for tuning, because the delay depends on a lot more than location. In particular the low pass filter will introduce a delay which depends on frequency and slope.

 

This is really useful stuff for tuning, because the delay depends on a lot more than location. In particular the low pass filter will introduce a delay which depends on frequency and slope.

Aha!!! So I'm not the only one who thinks that a tape measure isn't the way to set delays!! Thank f**king God!!

just kidding, kinda

 

Good stuff!

 

Thanks Patrick