# Sticky  My new custom made multi-element mic array



## Niick

Hello everybody, 

It's been a long time since I posted on the forum. Every now and then I read some of the new posts however, and a common theme that pops up over and over is questions about system tuning. As this is, after all, primarily geared toward DIY, I want to fully disclose that i do this for a living, and my views on the subject of sound system tuning might be quite different from that many enthusiasts. 

That being said, as I have learned invaluable information from many people on this forum, I feel that its only right that I give back if possible. The methods and equipment that i use are but one of many ways to arrive at a largely subjective end result. 

So onto the pics.....

I decided that my previous method of simultaneously averaging together 3 mics was not enough for me to have full confidence that i wasn't still, on occasion, largely "correcting for" position dependent anomalies, and not the overall trends that was my goal. And, for the record, let me state at the onset that what i am doing with these multiple microphones is not summing them together thru a mixer, as many (not on this forum) have assumed. That would not work. What is happening here, is much like a miltiplexer feeding the single input of an RTA, the software I use has the ability to mathematically average the multiple mic channels in real time. Unlike a miltiplexer/RTA combo however, with this setup I can, in real time, measure impulse response and phase/group delay. I have recently switched from using primarily SysTune Pro as my main software to using Smaart v8. I still run SysTune simultaneously along side Smaart when tuning, as SysTune still holds the edge when it comes to real time impulse responses, which can be saved and overlayed, making setting left to right delay times REALLY fast and easy. Smaart however holds the edge in making multi-mic measurements, target generation, and acoustic phase. Smaart's smoothing algorithms (magitude and phase being seperately adjustable) are absolutely brilliant, for it is the only program I have yet seen that can so easily resolve the acoustic phase, in a car, all the way up to 20k. 

As far as what I refer to as "target generation", I have a software DSP setup in such a way as to be permanently part of my on screen measuremnts, so that I can, at all times generate the phase/magnitude of any crossover slope/type/frequency that I want, as well as the sum of said filters. These freely adjustable targets are being shown on the screen live in real time simultaneously along side my mic measurements, thus allowing me to quickly and easily optimize system crossover, equalization, and reference channel delay settings for accurate phase data. Its a complicated thing to try and explain with words, and I hope to someday make some videos detailing the process. 

The interface I am using is a Behringer x18, which is fully multi-client, allowing for simultaneous access from multiple programs. 

I also run REW and ARTA for impedance measurements. By the way, if anyone has a Dayton Audio DATS or WT3, these work brilliantly with ARTA's "LIMP" program, allowing you to make real time impedance mesurements and incorporate things like varying FFT size, averaging, and user selectable excitation signals.


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## rton20s

I would be interested in seeing a video explaining how you implement the various programs and your tuning procedure based on the data you gather through the mic array. You know, just something you can do in your spare time.


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## oabeieo

Sick rack dood ! Dammm . 

Glad to see you back


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## rob feature

badass!


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## oabeieo

Niick, 
Question: 
How do you adress excess phase and that many mics? 
Thx in advance


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## Justin Zazzi

Welcome back Niick! I can't wait to learn more about this.


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## benny z

what if you put a foam head in the middle? would it simulate head shadowing?


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## rton20s

benny z said:


> what if you put a foam head in the middle? would it simulate head shadowing?


Foam? 

Ballistics Gel Head


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## Niick

Im at home with my 2 year old at the moment, and we're about to go on a walk. Ill address each question to the best of my abilities as time permits. The question of excess phase is a good one, as is the question about a simulated head. I will address these and any other questions. Please allow me some time, as these aren't exactly small subjects


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## TOOSTUBBORN2FAIL

Sub'd.


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## thehatedguy

Been wondering where you've been hiding.


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## SkizeR

cool. also, question. i know two others who use a setup like this. why do you want/need the mics in the middle instead of all around where your ears are?


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## Niick

SkizeR said:


> cool. also, question. i know two others who use a setup like this. why do you want/need the mics in the middle instead of all around where your ears are?


The middle microphone, in addition to contributing to the spatial average, is used for all timing related adjustments. Remember, real time IR is a measurement that is continuously updating and on the screen at all times. Therefore, when setting delay times for my left midbass as compared to my right midbass, for example, all i have to do is capture the measurement of the right midbass, then mute it. Now, by unmuting my left midbass, i can see instantly how much earlier in time it is from the right. Then all i do is add delay time to the left and watch its IR walk across the screen in real time until it overlays right on top of the saved IR of the right midbass. This takes less than 30 seconds to accomplish, left to right delay setting. So long as the mic youre using for this part of the process is your center microphone (for me thats mic #4, the red one) and you've marked the height and forward/rearward position of the listeners head before setting up the array (my center mic ends up somewhere between the listeners ears and the tip of their nose, somewhere in that range) then the delay times you achieve will be "correct"


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## Niick

oabeieo said:


> Niick,
> Question:
> How do you adress excess phase and that many mics?
> Thx in advance


The quick answer to this question.... Smaart v8. 

Its its absolutely unbelievably powerful for this type of system tuning. Cars that is. 

Smaart v8 employs a smoothing algorithm that is EXACTLY like other programs frequency dependent windowing. 

The best way i could describe this is to show you a measurement of a tweeter (since this is the worst case scenario for measuring acoustic phase in cars) measured quasi-anechoic (time gated, mounted on very tall pole) then overlaying that measurment with one made of the same tweeter, but this time inside the car. With Smaart's phase smoothing set to none, or 1\48th, the phase trace is a barely discernible explosion of lines, as you would expect. 

Now, simply set the phase smoothing down 1/3 or 1/1, and all of a sudden the trace overlays almost exactly on top of the quasi anechoic measurment. 

now, unless i show you this in a video or in person, its very difficult for me to explain. Curios as to how this worked, ive compared smaarts phase smoothing to SysTune's windowing, and sure enough they are the same. The difference is, changing the smoothing in smaart is WAY faster and non destructive where as using windowing (frequency dependent windowing that is, which is not the same thing as regular, sigle time windows) in other programs is tedious and time consuming. The other massive benefit is that Smaart allows averaging, in real time. of phase data as well as magnitude. So, for tweeter to midrange crossover's for example, i can sey my reference delays in smaart to accurately measure phase on each individual mic channel, then average the phase data, then work with that OR I can choose to work with just the center mic, OR i cam choose to work with both of them simultaneously, all while simultaneously looking at the minimum phase version of the acoustic responses (via my target generator traces) in the time and frequency domain, overlayed live, on the same screen.

This was probably a pretty poor explaination, there are so many variables that the only way for me to really convey this concept effectively would be thru a video.


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## Niick

*Re: My new custom made multi-element mic ggineersarray*



benny z said:


> what if you put a foam head in the middle? would it simulate head shadowing?


I suppose it would, however, my current thoughts on using a head (ive pondered this alot) is like this.....

Ive been listening to the world around me with a head and a pair of shoulders and torso attached to me my entire life. Its not like the mixing and mastering engineers in the studio were just etheric beings who mixed the album without the encumbrance of a physical body. No, they too were solid, 3 dimensional humans. 

To me, "calibrating out" or otherwise accounting for the presence of the human body only makes sense if this body was something "extra" that i "put on" or that otherwise "got in the way of" my ability to hear the recording the way it was mixed. Like if the recording was mixed by and intended to be heard by beings that were bodyLESS. 

Now, on the topic of binaural recordings, thats a different topic. 

Anyways, this is just my current line of thinking when it comes to the use of dummy heads for calibrating music reproduction systems, car or otherwise. 

I may very well be way off base here, as Ive never actually went forward with using a dummy head, so until then, this is just my speculation.


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## oabeieo

Niick said:


> The quick answer to this question.... Smaart v8.
> 
> .
> 
> This was probably a pretty poor explaination, there are so many variables that the only way for me to really convey this concept effectively would be thru a video.


Yeah I wasn't exactly speaking about Hilbert Tranceform directly, but that too makes me wonder how it's done in an array. It made me curious how, Seems it would have to know in 3D space where the mics would be. IDK tho  seems to be some pretty cool stuff out there. 

I'll go back to waving my microphone, and having my neighbors wonder why I'm sitting in my car for 4hrs straight listening to sweeps and tripping out on FFT blocks swearing 64k PN noise says "I can see you" between FFT cycles, oh and I'm sweating in a hot car with windows up so I look like I'm on crack.  #jealos


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## Niick

oabeieo said:


> Yeah I wasn't exactly speaking about Hilbert Tranceform directly, but that too makes me wonder how it's done in an array. It made me curious how, Seems it would have to know in 3D space where the mics would be. IDK tho  seems to be some pretty cool stuff out there.
> 
> I'll go back to waving my microphone, and having my neighbors wonder why I'm sitting in my car for 4hrs straight listening to sweeps and tripping out on FFT blocks swearing 64k PN noise says "I can see you" between FFT cycles, oh and I'm sweating in a hot car with windows up so I look like I'm on crack.  #jealos


So, technically speaking (during system tuning) i dont actually us the hilbert transform for anything. The way that i derive the minimum phase of a given passband is by generating, in the electrical domain, a bandpass, highpass, or lowpass filter using a permanently connected DSP. I set the DSP (the one permanetly connected to my measurment rig, not the DSP in the car) to show, in Smaart, the desired acoustic response that I ultimately want the drive units that I am working with to have. It is by virtue of the fact that I am measuring, in the electrical domain, the TF of an IIR DSP that i am able to see what would be the "idealized min. phase behavior" of a given passband. 

By then adjisting the reference delay in Smaart until my live IR trace and my phase trace are more or less the same as the min. phase version, I am able to tell where the ideal reference delay setting should be. 

I know this might sound very complicated and convoluted and time consuming, but I assure you it is not. On the contrary, I am able to very quickly, and very confidently, dial in a system to a highly refined state. 

There are two programs that I highly recommend anyone interested in acoustic analysis and/or audio test and measurement download. Well 3 programs really, including REW.

the other two are EKIO and Voicemeeter. (im talking about Windows here, for Mac i believe soundflower and crossover unit are very similar)

Both EKIO and Voicemeeter can be downloaded for $0.00

With these two programs, and REW, you can quickly and easily measure any filter slope, type, and frequency, as well as measure the sum. Well, not ANY type, just BW, LR, & Bessel.

Anyways, it has been very enlightening for me to get a better understnding of how these differenct filters interact, and how they behave in both the time and frequency domain. 

Speaking of which, a while ago there was a discussion about the "nulling" method not working for LR4 filters, because, it was said, that the phase response of said filter was such that inverting one passband wouldnt change the frequency domain summation in the magnitude. I then posted a short video detailing that LR4 filters do indeed have a null at crossover if one passband is inverted, thus, nulling would, in a pinch, or as a sanity check, work just fine.

Well, turns out that I now know that there is indeed a filter type that nulling doesnt work for. Its odd order butterworth. 

Starting with 3rd order.

Try it for yourself. no matter which way you flip the polarity of an odd order butterworth filter (excluding 1st order) the resulting magnitude summation will look the same. flat thru crossover.

The phase response, however, will of course change. Its my current understanding that the resulting TF of an odd order butterworth filter exhibits a type of all-pass behavior, depending on which way the polarity of the relative passbands is oriented. 

I still ave alot to learn in this regard, but it wasnt until i started experimenting with simply measuring different filter types that many obvious things that have alluded me came to light.


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## jnorman5

So sub'd!! I have been using Smaart V8 for the past several months and have been pleased with the results. However, I am only using a single mic and am very interested in setting up a mic array. A video series similar to Kyle Ragsdale's approach would be awesome for the gang. I realize it would be quite a time commitment though. Thanks for sharing and "welcome back"!!


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## Niick

jnorman5 said:


> So sub'd!! I have been using Smaart V8 for the past several months and have been pleased with the results. However, I am only using a single mic and am very interested in setting up a mic array. A video series similar to Kyle Ragsdale's approach would be awesome for the gang. I realize it would be quite a time commitment though. Thanks for sharing and "welcome back"!!


Oh yeah man, youve GOT to go multi-mic, ESPECIALLY with Smaart. Smaart is, in my (and many others) opinion, hands down absolutely unparalleled when it comes to multi-mic setups. Its the only software that i am aware of that can measure multiple transfer functions simultaneously, all real time, in both the frequency and time domain, all simultaneously. Limited only by your imagination and your hardware. 

Are you on the Rational Acoustics Smaart forum? Have you been running any of the beta's? I love the new data handling thats coming with 8.1


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## oabeieo

Very interesting....

So if smaart can do all that could it; 
1.export biquads? 
2. Convolute ? 

What would be very awesome for me would be able to work out a correction filter for both sides simultaneously, . But not just one global filter but 7 total filters for a 4way. 

I would love to be able to measure each speaker , and build correction filters in IIR and FIR all from one UI and be able to work Left and Right channels for each speaker pair side by side or overlay so when I work the filter I can see how magnitude and phase overlay between left and right. If that was a possibility it sure would make working a correction between channels a lot better. Also re-measuring between corrections to validate acoustical error correction would be nice without loosing your palace and starting all over again 

Seems that the more symmetrical a correction is between L&R the better the system sounds, the more different a correction is between speaker pairs are the worce it sounds.
Also could dial phase so much easier as far as time domain to see both IRs real time. Which you said it does.


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## hot9dog

I saw the picture of the cutting wheel next to the mic in the vice and I instantly fell in love with this thread! I love your setup with the multiple screens, your tablet, oscilloscope and wires draped at the ends!! Very cool!


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## Niick

oabeieo said:


> Very interesting....
> 
> So if smaart can do all that could it;
> 1.export biquads?
> 2. Convolute ?
> 
> What would be very awesome for me would be able to work out a correction filter for both sides simultaneously, . But not just one global filter but 7 total filters for a 4way.
> 
> I would love to be able to measure each speaker , and build correction filters in IIR and FIR all from one UI and be able to work Left and Right channels for each speaker pair side by side or overlay so when I work the filter I can see how magnitude and phase overlay between left and right. If that was a possibility it sure would make working a correction between channels a lot better. Also re-measuring between corrections to validate acoustical error correction would be nice without loosing your palace and starting all over again
> 
> Seems that the more symmetrical a correction is between L&R the better the system sounds, the more different a correction is between speaker pairs are the worce it sounds.
> Also could dial phase so much easier as far as time domain to see both IRs real time. Which you said it does.


Smaart does not have a convolution engine, no. Jriver does, im pretty sure. Smaart is purely an audio measurement application. 

As far as setting your DSP values all from one application, this can be done with Smaart and SysTune both with a number of different pro sound DSPs. Apex and Lab Gruppen come to mind. The DSP GUI's can access the TF and spectrum data from the analysis software overlayed against the parametric EQ and Xover controls. It's pretty cool stuff.

SysTune can export IIR filters created right from withtin the software's Virtual EQ feature, however it does not generate FIR filters. AFMG, the makers of SysTune have a software called FIRmaker. 

There is an analysis software that does alot of this stuff youre asking about, I THINK. Its from a company called Wave Capture WaveCapture

They have "FIR Capture" "Room Capture" "Tuning Capture" and a few others. Ive never used any of these softwares, so I cannot comment on them.


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## Niick

Here's some links to some videos I've made. 

https://www.youtube.com/watch?v=qJ9ajOaeUmc
https://www.youtube.com/watch?v=wl5doTBN1-c
https://www.youtube.com/watch?v=5vRoRvIdIvs
https://www.youtube.com/watch?v=HoIY3glPuWA
https://www.youtube.com/watch?v=XT0sZXBAzu8
https://www.youtube.com/watch?v=AMU1vXq07ug
https://www.youtube.com/watch?v=yLIHQ6owzCc
https://www.youtube.com/watch?v=utQtxyNgxak


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## Niick

Here are a couple videos I made yesterday at work, showing a useful scenario for using real time impedance measurements using ARTA and DATS v2

The post befor this one has some videos of the use of my tuning setup. I'll probably leave these links up for a short time, then take them down and replace them with new stuff periodically. 

If anybody has any questions relating to the use of acoustic analysis software and/or hardware, just ask, I'll do my beast to answer so long as the questions are general enough. 

My goal is to help lift the veil of mystery regarding successful sound system tuning 

https://m.youtube.com/watch?v=AMU1vXq07ug
https://m.youtube.com/watch?v=yLIHQ6owzCc


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## jnorman5

Thanks sooooo much!! Just got thru the sub/ mid bass alignment and while I am on the right path with Smaart V8, you taught this ol' dog a few tricks!! Please keep these up awhile... My build is in a Nissan 370Z and she is about two weeks away from going in a "car bag" for Winter So no tuning for me until Spring.

Again, thank you! I am sure I'll learn a lot from the other vids too!!


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## Kevmoso

Thank you for taking the time to post all of this Nick! That is an amazing testing rig. Do you offer preinstalled tuning sessions or just tunes on work you install?


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## Frogsrule

Pretty cool, Nick.


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## strohw

What do you use voicemeter for in a standard single mic setup?


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## captainobvious

Awesome Nick. I've slowly learning using the SysTune product after some instruction from both yourself and John K. 

I'm digging the "modified" mics. I would love to do something like that to make positioning easier and reduce the overall size/footprint of the mic array I'm using. Neat idea.


-Steve


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## LumbermanSVO

Here is another piece of FIR software: https://eclipseaudio.com.au/fir-designer/

I played with the demo for a bit, but since it'll be awhile before I fire up my system again I deleted it. The developer is very responsive to feedback.


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## rton20s

I've watched some of Nick's videos, but haven't had opportunity to play with any of the software or anything yet. 

I am curious about the mic array though. It appears that the mics you modified for use in your seven mic array began life as Dayton EMM-6 XLR mics. You're looking at an investment of $350 in mics. 

Would it be possible to use the Dayton IMM-6 instead? Possibly without the need to modify? Investment there would be under $115 for mics. My understanding is that they both use the same 6mm electret condenser. Is the TRRS 3.5mm connection any sort of hindrance?

What would be the pros and cons of using the less expensive IMM-6 vs the EMM-6?


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## Justin Zazzi

rton20s said:


> Would it be possible to use the Dayton IMM-6 instead? Possibly without the need to modify? Investment there would be under $115 for mics. My understanding is that they both use the same 6mm electret condenser. Is the TRRS 3.5mm connection any sort of hindrance?
> 
> What would be the pros and cons of using the less expensive IMM-6 vs the EMM-6?


I know the EMM-6 requires phantom power. I'm pretty certain the IMM-6 does not since that is not a feature I've heard mentioned about 4-pin headphone jacks on phones and tablets.

I don't know a ton about microphones, but that leads me to believe they are not the same product. However, if you can get a calibration curve for each of the IMM-6 mics you want to use, that might be a viable solution?


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## Frogsrule

1063 Adafruit Industries LLC | Programmers, Development Systems | DigiKey

Here's a lower cost alternative. Similar capsule. The capsules for these things are basically interchangeable, so you could also replace it with a flatter one if this turned out not to be flat to 20k.


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## ErinH

regarding the mics...

The behringer ecm8000 used to use the Panasonic WM-60 electret. That was discontinued and replaced with the Panasonic WM-61A electret. That was discontinued as well. However, both can be found on eBay pretty easily. There's an auction there now for 8 @ $17. 

If you google either of the above you'll find a lot of various discussion on possible replacements. You can also find information on Linkwitz's mods for the electrets which enable higher SPL at lower distortion and a lower noisefloor:
System Test

You can build your own little mic array out of those electrets and save yourself a considerable amount of money with a little DIY effort.

Google is your friend here, though, because there is a lot of info on the Panasonic electrets. FWIW, those same electrets are used in numerous recording mics (even my binaurals use them). 
https://www.google.com/#q=Panasonic+WM-61a

Hope it helps!


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## Niick

Kevmoso said:


> Thank you for taking the time to post all of this Nick! That is an amazing testing rig. Do you offer preinstalled tuning sessions or just tunes on work you install?


I will tune ANY system, installed by me or not. In fact, just this morning, I did some remote tuning work for a vehicle headed to finals in KY (I think). I'm in Oregon and the vehicle this morning was on the East Coast. So, yeah, anything.


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## Niick

strohw said:


> What do you use voicemeter for in a standard single mic setup?


So, to be clear, I don't actually use voicemeeter on my tuning rig. I use it on my Windows tablet and on a couple laptops as a way to have, basically, an 8 ch audio interface (ASIO) that can be used to connect EKIO (software DSP) to REW/SysTune/Smaart/ARTA/ whatever. This way, with only a computer, nothing else, no hardware required, one can measure and study exactly how different filters behave in the time and frequency domain, how the effects of different slopes, different delays, etc. affect the summation, and it allows me insight into the min. Phase behavior of a given passband. Quickly and easily adjustable on the fly. 

It is these measurements of the electrical filters engaged with EKIO that I use as targets for individual passbands on my tuning rig (it's just connected via my Behringer x18 instead of Voicemeeter).


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## Niick

Frogsrule said:


> 1063 Adafruit Industries LLC | Programmers, Development Systems | DigiKey
> 
> Here's a lower cost alternative. Similar capsule. The capsules for these things are basically interchangeable, so you could also replace it with a flatter one if this turned out not to be flat to 20k.


I have a bag ful of these at work. I think I bought 16 of them, they were so cheap. This was originally what I was going to use, these, with wm61a capsules. Problem was, while I was successful at getting the phase and mag traces to be DAMN near identical to my reference CSL EMM6, but the IR shape was different, so, when I played a single sine tone thru a speaker, the cheap little guys had horrendous distortion. Huge harmonic peaks poking up damn near as high as the fundamental. An at relativelyLow volume too. So, I thought, let me see what happens when I put the better capsule. Same problem, which led me to believe it was a problem of the tiny, TINY surface mount preamp. (Not that there is anything inherently wrong with surface mount, of course not, don't get me wrong there) but anyways, I decided to use semi-"real" mics as I've found that a very important part of what I do is being able to have faith in your equipment and your data, and I couldn't achieve that level of confidence that allows me to properly do my job with those little fellas. 

BUT, they still have potential value to me, as I may deploy them for other purposes. Purposes that don't require faithfully accurate measurements. Like maybe certain reverberation, reflection investigation I'm thinking, something like that maybe. I can deploy these things through a vehicle and gather all kinds of RELATIVELY accurate data (by "relatively" I mean one little mic as compared to another) ya know, to see how the acoustic energy had changed from one point or another. That type of thing maybe. 

Lastly, if you look close at the pics, the array is actually made up of a combination of EMM6s and ECM8000s. I think the red cables are all EMM6 and the grey cables are all ECM8000, or it's the other way around, but either way, there is both. Red cables are one type, grey cables the other. 

Which brings me to this next point, the EMM6 IS NOT the same mic as the ECM8000. After cutting into a bunch of both types, removing the preamps, and building this array, I can tell you, that regardless of what I've read or what other people have said, I PERSONALLY will not be buying any more EMM6s. The ECM8000 is unquestionably better built. I realize that in the CSL website a while back Herb had written something about the ECM8000 no longer being available as a result of quality problems, and I have no doubt that he did in fact experience this. 

But, my own experience, and for my own purpose, I can feel much more confident that If I have a working ECM8000 today, it will still be working next month, and probably next year. I DO NOT have that same confidence about the EMM6. The aluminum tube that goes from the mic body to the capsule is thinner, the capsule tip is glued on and easily pulls off, the wires soldered to the back of the capsule pull free of the pad with the tiniest movement/stress, the preamp is very insubstantial and made obviously much cheaper with much fewer surface mount parts compared to the ECM8000 bigger, heavier, bulkier thru-hole preamp enclosed in white heat shrink. The wires that go to the ECM8000 capsule are heavier gauge and do not break right off at or with the pads. 

Anyways, an eye opening experience to be sure. Of course, I keep my latest CSL EMM6 in its case with the desiccant pack for use as a calibration reference, and it will remain my reference until I can afford to purchase a better, real lab grade mic to use as a reference. But as far as further mics for arrays/general purpose use, ECM8000s for me please. I make my own calibration files for each mic after the array is constructed, so I have no need for a purchased cal file. 

I am also tempted to try out a couple Rational Acoustics RTA420 mics, to see how they're built. 

I guess the moral of the story is, just because a mic appears to look EXTREMELY similar, like in the case of the ECM8000 and the EMM6, does not necessarily mean it just the same thing rebranded. In this case, it most certainly is not.


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## Niick

rton20s said:


> I've watched some of Nick's videos, but haven't had opportunity to play with any of the software or anything yet.
> 
> I am curious about the mic array though. It appears that the mics you modified for use in your seven mic array began life as Dayton EMM-6 XLR mics. You're looking at an investment of $350 in mics.
> 
> Would it be possible to use the Dayton IMM-6 instead? Possibly without the need to modify? Investment there would be under $115 for mics. My understanding is that they both use the same 6mm electret condenser. Is the TRRS 3.5mm connection any sort of hindrance?
> 
> What would be the pros and cons of using the less expensive IMM-6 vs the EMM-6?


Here's my personal feeling on this topic:

The capsule may very well be the same, I have no idea, kinda doubt it, but who knows, maybe. Anyways, in my opinion, the difference is found in the way these two mics are intended to be used. The IMM6 can get away with not having a preamp which takes the tiny voltage from the capsule and converts it to balance audio signal capable of driving 10's of feet of cable thru potentially electrically noisy environments. The IMM6 is made to plug right in to its input device, putting the actual capsule within a coup,e inches at most of the input circuitry. 

With a "real" measurement mic however, there is more to it than just the capsule. The phantom powered preamp contained within the body of the mic is there so that a person can connect a long cable to it, use the mic some real distance away from the actual input device/audio interface, yet successfully get a relatively noise free signal into the audio interface, despite the fact that the signal from the mic to the interface might be traversing 20 feet or more (very often a lot more) crossing other electrical wires, and running up and over God knows what. So the importance of common mode noise rejection inherent in balance audio circuitry becomes apparent. 

You cannot get this type of successful signal to noise ratio at the input of your audio interface if you do not have a preamp located close to the mic capsule. I was worried about the 3 feet of added length that making the array required, but it turned out to work just fine. I'd be very concerned about going much longer however.


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## jackk

Thanks for making the effort to show us what u r doing Nick. VERY educational. Appreciate it.


Sent from my iPhone using Tapatalk


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## Niick

Here are a couple quick videos i made showing Audio Tools for iOS, a VERY useful set of softwares that I dont hear too many people using that often, but they have the power to be very useful.

https://www.youtube.com/watch?v=E9A70Hpiuwc
https://www.youtube.com/watch?v=jlHFXyyB7xU


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## danno14

Niick said:


> I will tune ANY system, installed by me or not. In fact, just this morning, I did some remote tuning work for a vehicle headed to finals in KY (I think). I'm in Oregon and the vehicle this morning was on the East Coast. So, yeah, anything.


Woo-hoo! Lucky me!!!

I'm only a few hours away, up near Tacoma


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## bugsplat

Holy crap!!!!!! 

My head's hurts. I need to lie down after reading all that what i can only assume is witchcraft. I need to study more.


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## Babs

Fun stuff! Nice vids and work Niick.
Heck I can't even figure out getting a reference loop wired on an old mobilepre. It's sad. LOL! 
Gotta do a custom 1/4" TS to 1/8" TRS and 1/4" TS Y-cable I think unless I'm just doing it wrong.
But anyway, this does make me want to try to throw something together with the two IMM-6's I have.


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## Babs

Kinda figured I'd be in the forums and eventually drool over a rack or two in some threads, but this is not quite what I anticipated. That rack is so.. So.. Epic! For lack of a better word.


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## oabeieo

You are the **** man. ****ing nerdy as hell but gosh dam dood your doing some sick stuff. 
Man I would love to throw a FIR car at you and see what magic you could do. I bet that **** would be dialed!


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## Babs

What's even nerdier is we all go "FIR... ooooooohhh!" Hehehe 


Sent from my iPhone using Tapatalk Pro


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## Niick

Hey Guys,

i figured I'd post a new video. Again, this is really short, but it details a scenario exemplifying the need for FULLY VARIABLE all pass filters to be incorporated into car DSPs. 

If I'm not mistaken, I think Andy W. has talked about this a time or two in the past as well. 

Word has it that a popular new DSP (cant say which one) is going to be getting fully variable all pass filters in a future update. By fully variable, I'm meaning that the corner freq. isn't fixed to the crossover frequency. 

I've also heard that the new Helix DSP Pro is this way too. 

Anyways, I THINK that this video shows a good example of just one way in which all pass filters can be beneficial. Of course, I did NOT have an all pass filter to use, but I would have liked to. I have yet to actually work with them though, so I'm not claiming to be an expert in this area. 

https://www.youtube.com/watch?v=tWiWxXMXOlk


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## oabeieo

Sick vid niick

Heya can you get me a good deal on smaart v8 or v7

I want that real time phase and fr for two curves simultaneously.
And the auto mic delay removal is sick. Really speeds things up I bet.
Do u think I can get a copy of either for around 300$ max?


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## Niick

Smaart v8 (and 7) can measure any number of simultaneous transfer functions, limited only by your imagination and your hardware. When I'm working I am running no less that 11 simultaneous TF's. 

The auto mic delay removal DOESN'T WORK in cars the vast majority of the time. There is almost always a reflection (arriving after the initial impulse of course) that is of higher amplitude than the direct arrival, therefore, using the "find delay" or "track" button in Smaart is good to take away the propogation delay.....BUT, you almost always have to adjust the final reference delay time by hand, which is absolutely no problem at all, thats what the "Live IR" display at the top of the screen is for. 

I dont know of anywhere that sells Smaart that cheap. But remeber, for the $900 or so dollars that the program costs (the hard copy version is just over $1000) you get TWO licenses. So, if you can find a buddy to go in on it with ya, you can each get a copy for just under $500. 

I highly recommend using a multi channel soundcard with Smaart, and running a target generator using EKIO. I have a video detailing that as well. 

I also figured out a way to make highly accurate, full range impedance measurements using music as the test signal and leaving the sub or whatever speaker connected to the actual amplifier in the car. This way, problems that might only manifest at higher volumes can be investigated. I'm just waiting for that video to finish uploading. Its super short too


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## oabeieo

Niick said:


> Smaart v8 (and 7) can measure any number of simultaneous transfer functions, limited only by your imagination and your hardware. When I'm working I am running no less that 11 simultaneous TF's.
> 
> The auto mic delay removal DOESN'T WORK in cars the vast majority of the time. There is almost always a reflection (arriving after the initial impulse of course) that is of higher amplitude than the direct arrival, therefore, using the "find delay" or "track" button in Smaart is good to take away the propogation delay.....BUT, you almost always have to adjust the final reference delay time by hand, which is absolutely no problem at all, thats what the "Live IR" display at the top of the screen is for.
> 
> I dont know of anywhere that sells Smaart that cheap. But remeber, for the $900 or so dollars that the program costs (the hard copy version is just over $1000) you get TWO licenses. So, if you can find a buddy to go in on it with ya, you can each get a copy for just under $500.
> 
> I highly recommend using a multi channel soundcard with Smaart, and running a target generator using EKIO. I have a video detailing that as well.
> 
> I also figured out a way to make highly accurate, full range impedance measurements using music as the test signal and leaving the sub or whatever speaker connected to the actual amplifier in the car. This way, problems that might only manifest at higher volumes can be investigated. I'm just waiting for that video to finish uploading. Its super short too


yeah gating every measurement in rew is so tedious and than the loopback of course . This just looks so much better for aligning left and right drivers . 

Capture is nice on rew but it's a picture not real time. 

For example tuning with IRs averaged takes FOREVER and sometimes (most of the time). I just use PN noise and a RTA . It goes 1000000x faster and gets better tune than the IR method. However that phase data is missing. Ugh 

I will try find someone that wants to go in halves . 

So the track doesn't work in car (I shoulda guessed) but once you gate it manually you can still track real time two speakers in car right? Sounds like you implied that just wanna be clear. That's the main reason I would pony up the cash. I'm just tired of taking hours to do what you can do in 5min


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## Niick

So I'm not quite sure what you mean when you say "Track two speakers at the same time" . If you play 2 (or more) speakers at the same time, the software isn't going to somehow differentiate the two, in other words, you won't get a reading of two individual speakers as if each one was playing by itself. What you'll get is the summed response of both speakers. Of course, there are certainly ways, just by observing the measurement data alone, that one could deduce that indeed two speakers are playing. Depending on the timing relationship of these two speakers, the IR will show two distinct arrivals, however, it takes practice to get used to looking at and making sense of time domain data acquired in-car. 

In that last video, what was shown was one trace was a saved trace, the other a live trace. I played a single mid bass, captured a trace of it, then muted that mid bass and played the other. This second mid bass was the live trace in the video. It wasn't that both mid bass drivers were playing simultaneously and the software was somehow able to keep them separate.


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## oabeieo

Niick said:


> So I'm not quite sure what you mean when you say "Track two speakers at the same time" . If you play 2 (or more) speakers at the same time, the software isn't going to somehow differentiate the two, in other words, you won't get a reading of two individual speakers as if each one was playing by itself. What you'll get is the summed response of both speakers. Of course, there are certainly ways, just by observing the measurement data alone, that one could deduce that indeed two speakers are playing. Depending on the timing relationship of these two speakers, the IR will show two distinct arrivals, however, it takes practice to get used to looking at and making sense of time domain data acquired in-car.
> 
> In that last video, what was shown was one trace was a saved trace, the other a live trace. I played a single mid bass, captured a trace of it, then muted that mid bass and played the other. This second mid bass was the live trace in the video. It wasn't that both mid bass drivers were playing simultaneously and the software was somehow able to keep them separate.



Watch the video on smaart website for v8

They take two speakers one mic and you can see both phase traces . 
Maybe they did do a screen shot like you now that I think about it .
Regardless this can't be done in rew  at least without a whole lot of sweeps and averages . Over and over and over again


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## Niick

https://youtu.be/HHxRtHTGZs8

Here is another video. This one really short unfortunately, but it shows me making an impedance measurement of a subwoofer in operation, in the car connected to the amplifier that its always connected to (usually impedance measurements require that you disconnect the DUT from the system in order to test it) and I'm using music as the test signal. 

Im doing this by using a 2 channel differential probe that measures both voltage and current. This probe is connected to SysTune, and a TF is being run, with the magnitude data scaled linearly in ohms. Sine dual FFT systems like SysTune work by mathematically dividing one channels by another, if you calibrate one channel as voltage and the other as current, then....... E/IR baby!!! you have impedance  

WAAYYYY nerdy! Just like I like it!!

I have about 4 more videos on the way that I made just today, detailing the difference between types of pink and white noise, random vs. periodic. As well, i go into fractionally octave banded analyzers (like RTA's) vs. raw FFT spectrum analyzer displays, and why pink noise looks horizontal on one and sloping downward on the other. Ive noticed a bit of confusion lately with people making single channel spectrum measurements in the electrical domain with fractional octave analyzers, and they're using white noise as the test stimulus.

I'll post the links to these videos (they're not so short and useless!!) as they finish uploading


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## Niick

Here are 4 videos I made detailing some of the differences between Pink and White Noise, raw FFT data and fractionally octave banded (RTA) data, and a note on Audio Tools for iOS FFT module, which displays data differently than almost all other programs, and unless you are aware of this, it can be confusing. 

Ive noticed lately alot of people using single channel spectrum measurements (hardware/software RTAs) for electrical testing of the frequency response of factory radios. Which I also do on a very regular basis. However, if I'm doing a single channel measurement as a quick check of, say, whether or not a speaker signal is full range, or whether or not it's EQ'd (as almost ALL new vehicles are), I'll use PINK Noise. NOT White Noise. Again, I'm referring to single channel measurements made on fractionally octave banded analyzers. Using White Noise on a 1/3 octave RTA for example, will show you a response that APPEARS be to rising as frequency increases, compared to the actual response of the DUT you're measuring. It makes absolutely no difference whether you're making an electrical measurement or an acoustical measurement. This IS NOT the deciding factor of whether or not you should use white or pink noise. 

https://www.youtube.com/watch?v=pAvZspUCzTo
https://www.youtube.com/watch?v=gnhrrKQIGg8
https://www.youtube.com/watch?v=ghk79hsLZfU
https://www.youtube.com/watch?v=tmF7tU51cuY


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## Elgrosso

Real cool videos thanks! You have amazing gears, it's crazy 
How is the focusrite dock? i thought about using one as source long time ago, just for the knob and the easy swap ipad/iphone. But I think it's only analog right?


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## Niick

Yes, the Focusrite iTrack Dock has only analog outputs. 2 balanced TRS outs and one unbalanced mini-TRS out (headphone). It also samples at 96kHz 24bit, and it uses 12VDC for power. It charges the iPad while connected, and is a great companion to Audio Tools for iOS. With phantom power for measurement (condenser) mics as well. 

OK, here is one last video for tonight, this one is showing how to use completely free programs to generate crossover targets for REW. Generating targets like this allows you to not only have the magnitude, but the phase and IR as well, so you can easily see what the ideal timing relationship should be between passbands. 

https://youtu.be/qYWrjnkDgaA


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## Niick

https://youtu.be/QzV4nuX75PA

Here is an improved method of creating targets in REW, doing it this way removes the latency of EKIO from the phase trace, so manual removal of the broadband delay induced by using a pre-DSP reference channel isn't necessary after every sweep jsut to get useful phase data. For this video I used a screen capture program, so the audio sucks and I apologize for the loud "pop" sounds of me hitting the keyboard at 2:28 and 5:38.


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## lizardking

Niick said:


> Here are 4 videos I made detailing some of the differences between Pink and White Noise, raw FFT data and fractionally octave banded (RTA) data, and a note on Audio Tools for iOS FFT module, which displays data differently than almost all other programs, and unless you are aware of this, it can be confusing.
> 
> Ive noticed lately alot of people using single channel spectrum measurements (hardware/software RTAs) for electrical testing of the frequency response of factory radios. Which I also do on a very regular basis. However, if I'm doing a single channel measurement as a quick check of, say, whether or not a speaker signal is full range, or whether or not it's EQ'd (as almost ALL new vehicles are), I'll use PINK Noise. NOT White Noise. Again, I'm referring to single channel measurements made on fractionally octave banded analyzers. Using White Noise on a 1/3 octave RTA for example, will show you a response that APPEARS be to rising as frequency increases, compared to the actual response of the DUT you're measuring. It makes absolutely no difference whether you're making an electrical measurement or an acoustical measurement. This IS NOT the deciding factor of whether or not you should use white or pink noise.
> 
> https://www.youtube.com/watch?v=pAvZspUCzTo
> https://www.youtube.com/watch?v=gnhrrKQIGg8
> https://www.youtube.com/watch?v=ghk79hsLZfU
> https://www.youtube.com/watch?v=tmF7tU51cuY



Interesting comparison with RTA versus Spectrum. So with using the RTA mode would we still try for a downward house curve when using Pink PN?


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## Niick

lizardking said:


> Interesting comparison with RTA versus Spectrum. So with using the RTA mode would we still try for a downward house curve when using Pink PN?


yes, you absolutely do. The whole point I was trying to make was that it's important to know what you're measuring and why. And how the analyzer you're using displays data, as they're not all the same. It's easy to determine these things by simply inputting a signal with a known spectrum into the analyzer, and see how it displays a signal that you know to have a certain spectral content, like pink or white noise. 

When using RTA mode you would definitely want to tune the system with a downward sloping trend from left to right. Maybe I'll make another video detailing this a bit. I simply wouldn't recommend using spectrum mode for system tuning. Not for eq, tonal balance, level setting, etc. This is best done with the analyzer set up as fractional octave banding, in other words, RTA. Well, actually, this is best done with a transfer function measurement of some kind, as without time domain information I cannot imagine how somebody could successfully optimize a fully active dsp controlled system. 

But yeah, if I had to choose between the two, spectrum mode or RTA mode, I'd choose RTA mode. 

Now, I don't have direct experience with this, but I would think that if you took a "sweep" using REW's main measurement mode, then compared the magnitude results with the results of the RTA mode set to, say, 1/24th octave acquired using PPN, the two SHOULD overlay more or less right on top of each other. However, comparing REW's "sweep" to RTA mode set to "Spectrum" SHOULD result in the data NOT overlaying nearly as perfectly. The "spectrum" data should appear to have a reduced high end. This is to be expected. If I remember correctly, I think I can recall this very phenomenon confusing people in the past, as I think some people didn't realize the differences between sweep, RTA, and Spectrum. 

Whether you use random pink noise or periodic pink noise won't make a difference as to whether or not you will want to use a house curve. The PPN simply allows you to acquire more stable data with little to no averaging more quickly.


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## lizardking

Good stuff. Would love to see an actual tuning video using REW and perhaps a DSP most of us are familiar with - Helix. However, the tuning video would be nice Helix or not. It would be nice to see a series of actual tuning videos with REW from you.


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## oabeieo

Niick, 

How come when you play Pink PN and you move the mic while le it's on it sounds like the responce changes coming out of the speaker. Or when the mic is moving the FFt blocks sound to go faster? 

I always wondered what that is exactly 

Cheers


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## Niick

oabeieo said:


> Niick,
> 
> How come when you play Pink PN and you move the mic while le it's on it sounds like the responce changes coming out of the speaker. Or when the mic is moving the FFt blocks sound to go faster?
> 
> I always wondered what that is exactly
> 
> Cheers


I don't think I'm following what you're asking. Describe the scenario. You're physically in the vehicle, and holding the microphone, moving it side to side? And while doing so you're playing periodic noise....of what sequence length, and at what sample rate? I'm gonna a assume 48k...i only ask about the sequence length of your periodic noise because shorter noise because the shorter the sequence, the more audible the repetitions of the noise waveforms become. 

Anyways, so, you're saying that physically moving the mic around changes the sound you hear? Are you playing only a single speaker when this happens? What would happen if instead of a mic you held nothing in your hand? Do you still hear a change with movement? If so I'd guess that the physical position of your moving body is affecting the acoustic energy that's propagating thru the vehicle on its way to your ears. A coworker of mine told me that his imaging changes as he puts his hands on the steering wheel. This is totally possible le in the small confines of an automobile. 

As far as the FFT blocks going faster..... You mean they seem to rise and fall at a faster rate, as if there was suddenly less averaging selected?


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## oabeieo

Niick said:


> I don't think I'm following what you're asking. Describe the scenario. You're physically in the vehicle, and holding the microphone, moving it side to side? And while doing so you're playing periodic noise....of what sequence length, and at what sample rate? I'm gonna a assume 48k...i only ask about the sequence length of your periodic noise because shorter noise because the shorter the sequence, the more audible the repetitions of the noise waveforms become.
> 
> Anyways, so, you're saying that physically moving the mic around changes the sound you hear? Are you playing only a single speaker when this happens? What would happen if instead of a mic you held nothing in your hand? Do you still hear a change with movement? If so I'd guess that the physical position of your moving body is affecting the acoustic energy that's propagating thru the vehicle on its way to your ears. A coworker of mine told me that his imaging changes as he puts his hands on the steering wheel. This is totally possible le in the small confines of an automobile.
> 
> As far as the FFT blocks going faster..... You mean they seem to rise and fall at a faster rate, as if there was suddenly less averaging selected?



Dude maybe that's all it is. My big fat butt movin around and my head moving. 
Even tho I think I'm sittin still....or I'm going bonkers ...but I swear it does it : 

I'll try to make a video of it doing it, but first I'm try moving mic but not be holding it and see if makes difference


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## captainobvious

Movements of the mic, or movements in the environment (your body, vehicle panels, etc) are going to change the way it sounds when reaching your ears (and the mic) so this sounds pretty normal. You'll probably notice it more with higher frequencies.


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## oabeieo

This thread must be read again so TTT

This is filled with knowledge by Niick Ames (creator of the JL Max)

enjoy ….. man that guy is becoming my idol….

mods this should be a sticky…. Please… this thread answers so much especially for anyone wanting to build a serious tuning rig and Niick explains things so good…. I’ve never asked for a sticky, this would be one that should be up for everyone to read (at least for awhile). And the nostalgia of where the MAX was conceived


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## saltyone

Done. 🫡


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