# Xover phase shift



## Niick (Jun 3, 2015)

I'm posting this pic to illustrate the amount of phase shift induced by a common crossover. All filters will induce phase shift to varying degrees.

This particular crossover was the passive network from an older Diamond Audio component set. 

Active crossovers do this too.

Notice how the LP and HP are 180 degrees apart at the frequency where the two cross. The ELECRTICAL summation would result in a very deep dip at crossover, which is why one driver often needs to be wired in reverse polarity. 

Just thought some might find this interesting, I know I do. 

The top graph is magnitude, the bottom phase. I placed markers at the crossover freq. to better illuminate the phase difference.


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## Patrick Bateman (Sep 11, 2006)

And people keep telling me phase doesn't matter.
Despite the fact that some of the best sounding and most popular loudspeakers put phase as a top priority. (Dynaudio, Vandersteen, Thiel, Dunlavy, etc)


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## SkizeR (Apr 19, 2011)

the pic is kinda hard to make out. would you be able to get a screen shot of it?


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## thehatedguy (May 4, 2007)

And what about the the phase of the reflected energy?

As long as the signal is in phase at the listening point, it doesn't matter what is happening at the speaker.


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## Patrick Bateman (Sep 11, 2006)

thehatedguy said:


> And what about the the phase of the reflected energy?
> 
> As long as the signal is in phase at the listening point, it doesn't matter what is happening at the speaker.


I have generally found that it's totally impossible to get the phase correct at the listening point if it's not correct at the speaker.

Here's an example:









I have a bunch of these Monsoon planars. I was expecting that they would have wonderful phase response, because they have wide bandwidth and no crossover.

When measuring them, *I couldn't get a phase measurement at all.* The measurement wasn't just bad; it was garbage.

This was a big "Eureka" moment for me, because it "clicked" why we only want one tweeter. (If you care about phase.)

Basically the sound from one edge of the diaphragm is out-of-phase with the sound from the center of the diaphragm. This is because the *center* of the diaphragm is slightly farther away than the edge.

And when I say "slightly", I'm talking about a fraction of a centimeter. But at 20khz, *a fraction of a centimeter makes a difference.*









I keep re-using this pic, which was actually intended for something else, but up in the right corner you can see what a great phase measurement looks like. Basically there's less than 90 degrees of rotation per octave.



On a side note, I've noticed that what a lot of people are calling "phase response" is actually just lining up the loudspeakers in time. I've seen a few people do this; they use DSP to get all the loudspeakers to arrive at the mic simultaneously.

That's a step in the right direction, but it doesn't fix your phase response. Actually, it would if you had something like one speaker per octave  But if you have two or three speakers, you need to look at the entire phase response curve. And you can't even get a clean measurement of that if you have more than one tweeter, or if you have some serious early reflections. You basically have to do this:

1) Use one tweeter
2) You can have as many mids as you want but they need to be within 1/4wl at the xover point from tweeter to midrange
3) You need to eliminate early reflections by pulling the midranges away from a barrier, or put the midrange so close that the reflections are within 1/4wl. Or use a waveguide.


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## gijoe (Mar 25, 2008)

Patrick, we've talked a bit about the post you made years ago regarding using a single tweeter, it came up during the acoustic lens discussion.

How much stereo information is there (typically) above say 5kHz? If I wanted to try experimenting with a single tweeter setup, what is the best way to process the signal? Is it simply a matter of doing L+R?


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## Patrick Bateman (Sep 11, 2006)

Here's a reply, but I moved it to the other thread :

http://www.diymobileaudio.com/forum...n/72891-anyone-tried-using-one-tweeter-9.html


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## thehatedguy (May 4, 2007)

Sometimes you can have one side in phase with each other, but that doesn't automatically translate into being in phase with each other at the listening position due to the asymmetric seating position in the car. 

The same can also be said of drivers in different locations like mids in the kicks and tweeters in the pillars.

So to me it makes more sense to worry about things at the listening position rather than at the speaker location.


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## Niick (Jun 3, 2015)

So my original intention was to show a specific person the pic of a measurement I made of a passive Xover, but I couldn't figure out how to post a pic in a PM using my iPad. 

So I figured, what the hell, let's see what people make of this. 

Umm......regarding phase shift induced by speaker location differences.......I believe that this is precisely where fully active systems with adjustable delay per driver have an advantage. As long as you can measure the final, acoustic phase of the two drivers thru crossover, then you can set your delay time very accurately. Much more accurately than just measuring the distance with a tape measure. 

This is where some definitely disagree, but I personally think that using delay to get your drivers in phase thru crossover is a wonderful idea. And it works. The key is understanding how to measure phase in a highly reflective space like a car. It can be done, and the results are worth it.......

I do need to figure out how to do screenshots


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## Niick (Jun 3, 2015)

Here's a couple more, these are done using AudioTools from Studio Six Digital, these are of a 2 way active setup, here it's easy to see the difference between the trace with the tweeter in phase and the trace with the tweeters polarity flipped by 180 degrees. The big discontinuity in phase at crossover is a dead giveaway that this system needed work thru the crossover region.

Ok, I can't figure out how to post more than 1 pic at a time, so this one is of 3 separate overlays, dark blue: woofer
Light blue: tweeter 180 deg.
Yellow: tweeter 0 deg.


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## Niick (Jun 3, 2015)

Now this one is two overlays showing the summation of the tweeter and woofer
Yellow: woofer + tweeter 180 deg
Blue: woofer + tweeter 0deg

These were old KEF point source components

Anyways, often times when you see a large phase discontinuity like in the yellow trace, it's usually a problem with the acoustic phase thru crossover, which can be addressed a few different ways depending on the system and the reason for the discontinuity


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## Niick (Jun 3, 2015)

Patrick Bateman said:


> And people keep telling me phase doesn't matter.
> Despite the fact that some of the best sounding and most popular loudspeakers put phase as a top priority. (Dynaudio, Vandersteen, Thiel, Dunlavy, etc)


I agree with you 100%, although I think the jury is still out on wether or not ABSOLUTE phase is audible, relative phase between drive units at crossover is obviously extremely important.


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## 14642 (May 19, 2008)

Niick said:


> I agree with you 100%, although I think the jury is still out on wether or not ABSOLUTE phase is audible, relative phase between drive units at crossover is obviously extremely important.


Phase at the crossover point matters, but the phase of the electrical filter only matters because it modifies the acoustic response and the acoustic phase response. If your acoustic response follows the target alignment and the speakers are the same distance from the listener or the mic or you use delay to correct for that distance, then the acoustic sum will follow the acoustic sum for the target.

This is all quite simple for home speakers measured in an anechoic chamber. It's less simple in a car where we have to EQ the speaker and the reflections in one process.


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## subterFUSE (Sep 21, 2009)

Niick said:


> The key is understanding how to measure phase in a highly reflective space like a car. It can be done, and the results are worth it.......



What is the technique for getting good phase measurements in a car?

I have had only small levels of success trying to do it. Most of the time the reflections make a mess of the phase plots.


Most of the time, I just measure the 2 speakers together and choose the alignment that gives the best SPL support through the crossover range.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> This is all quite simple for home speakers measured in an anechoic chamber. It's less simple in a car where we have to EQ the speaker and the reflections in one process.


I think to say it's "less simple" is quite the understatement! ?

I like your technical approach to things, so here is what I have been doing lately as a means of achieving useable data to work with when attempting to optimize a fully active system with processor, in other words, when there is enough flexibility for me to adjust parameters and achieve worthwhile results. 

I have found that using a FREQUENCY DEPENDENT window on the impulse response (then doing the FFT of that) yields useable data that can be used to adjust things like timing, polarity, even EQ. The key is the frequency dependent window, and lots and lots of practice and experimentation. The results I've been able to achieve with this method are both (relatively) quick and sound better than anything I was ever able to achieve before, short of having a customer come back many times and tweak little by little over many weeks/months. 

Have you ever experimented with frequency dependent windowing before, and if so, what did you think. 

For anyone reading who hasn't experimented with it, frequency dependent windowing is EXTREMELY different from conventional windowing, the window time gets (considerably) longer as frequency decreases.


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## Niick (Jun 3, 2015)

subterFUSE said:


> What is the technique for getting good phase measurements in a car?
> 
> I have had only small levels of success trying to do it. Most of the time the reflections make a mess of the phase plots.
> 
> ...


You have to use a program that has some type of frequency dependent windowing, I personallty use SysTune Pro, but Smaart v7 does it too. I take ACOUSTICAL phase response plots all the time in cars, the traces are stable, repeatable, useable!! 

Don't let anyone tell you it can't be done! It can, I do it all the time, and it has helped me tremendously to learn and be a better system tuner.


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## Niick (Jun 3, 2015)

There is a very recent beta version of REW where John has implemented frequency dependent windowing, if you go to the REW forum and find the thread called frequency dependent windowing, look on the last page of that thread, John posts a link to the installer. I haven't tried this version myself, but it sounds promising, as I know a lot of people on here use REW, and if this is anything like SysTune's TFC window, WOW!! and for free!!


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> Phase at the crossover point matters, but the phase of the electrical filter only matters because it modifies the acoustic response and the acoustic phase response. If your acoustic response follows the target alignment and the speakers are the same distance from the listener or the mic or you use delay to correct for that distance, then the acoustic sum will follow the acoustic sum for the target.
> 
> This is all quite simple for home speakers measured in an anechoic chamber. It's less simple in a car where we have to EQ the speaker and the reflections in one process.


So re-reading what I wrote, you're right, what I should have said, and what I meant was that relative ACOUSTIC phase AT THE LISTENING POSITION of different drive units is extremely important.


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## Patrick Bateman (Sep 11, 2006)

subterFUSE said:


> What is the technique for getting good phase measurements in a car?
> 
> I have had only small levels of success trying to do it. Most of the time the reflections make a mess of the phase plots.
> 
> ...


What program are you using?

I'm using HolmImpulse. I can good phase measurements if I choose the gates carefully. I usually end up with a gate that gives me data from 500hz to 20khz.


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## Patrick Bateman (Sep 11, 2006)

Niick said:


> You have to use a program that has some type of frequency dependent windowing, I personallty use SysTune Pro, but Smaart v7 does it too. I take ACOUSTICAL phase response plots all the time in cars, the traces are stable, repeatable, useable!!
> 
> Don't let anyone tell you it can't be done! It can, I do it all the time, and it has helped me tremendously to learn and be a better system tuner.


Are you at the shop on 82nd? I used to live right by there.


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## Niick (Jun 3, 2015)

Patrick Bateman said:


> Are you at the shop on 82nd? I used to live right by there.


I work at the aloha store, I use SysTune Pro


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## 14642 (May 19, 2008)

Hey Niick.
No, I've never used frequency dependent windowing in a car. I'm usually not so concerned about the phase response except to provide a good center image and to make sure crossovers work properly. 

In the car it's a huge crapshoot because of all the reflections, so I typically use a spatial average of several mic locations and a spatial average throws the phase response away. If I'm in a hurry or if the DSP I'm using just has a 1/3rd octave EQ, then I usually use a spectral average--1/3 or 1/6th octave above 500Hz and 1/6 or 1/2 below. The results are similar.


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## 14642 (May 19, 2008)

I just read the thread on HTS and I'm not quite sure what to make of it. I did find the mention of the Revel speaker measurements helpful, since I know the measurement and design methods behind those. 

It appears to me that this is a pretty useful tool for evaluating speakers to be used in rooms--maybe a way to approximate a directivity index without making 360 degrees worth of measurements. I'll have to read some more. I'm not sure how this applies to cars, other than providing a different measurement routine and a different target curve. 

Hmmm...maybe I'm not smart enough to figure this one out.


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## Niick (Jun 3, 2015)

Well, I don't think necessarily that the psychoacoustic smoothing option presented in REW is or isn't helpful for car systems, what I was excited for, was the ability to implement a VARIABLE frequency dependent window, which I find immensely helpful to be able to "see" the phase response of the HF range which is usually obscured by all the early reflections. However, after having a chance to evaluate the beta version with psychoacoustic smoothing, while it does help a little to clear up the phase in the HF, the freq. dependent window that JohnM implemented (in order to achieve said smoothing)happens completely behind the scenes and isn't at all adjustable ar accessible. Maybe in a future version!?


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## Justin Zazzi (May 28, 2012)

Andy Wehmeyer said:


> I just read the thread on HTS and I'm not quite sure what to make of it. I did find the mention of the Revel speaker measurements helpful, since I know the measurement and design methods behind those.
> 
> It appears to me that this is a pretty useful tool for evaluating speakers to be used in rooms--maybe a way to approximate a directivity index without making 360 degrees worth of measurements. I'll have to read some more. I'm not sure how this applies to cars, other than providing a different measurement routine and a different target curve.
> 
> Hmmm...maybe I'm not smart enough to figure this one out.


Can you point us towards this particular thread you're referring to? I would like to read it.


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## Niick (Jun 3, 2015)

Got to the home theatre shack forum, then REW forum, then about 5 threads down is a thread called "feature request-frequency dependent windowing" 

The thread was started by a guy named Bob Katz, who I believe is a quite well known mastering engineer?


Feature Request: Frequency Dependent Windowing - Home Theater Forum and Systems - HomeTheaterShack.com


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## 14642 (May 19, 2008)

I guess my point, Niick, is that as frequency is increased, the frequency response and the attendant phase response is more and more dependent on a particular mic placement. That's less likely in a bigger room and unlikely in an anechoic chamber. 

In my experience, designing a crossover for a home audio speaker in a chamber is a very different exercise that implementing crossover and EQ in a car. I've never found making any kind of an approximated anechoic response in a car to be helpful. Of course, without some kind of adaptive windowing, the range of the frequency band that can be measured anechoically through gating is limited to very high frequencies because the window is too small to provide sufficient resolution at low frequencies. 

I find that phase issues tend to be more problematic in the midrange and midbass, but much of the difficult is due to placement and reflections. In my last job, we developed a phase EQ that provided great measured results, but the benefits were no more audible than simple delay and EQ. 

All of this and lots of tuning time suggests to me that because all of the reflections are VERY early, EQing the direct sound and the reflections at the same time is sufficient. 

I also find that using steep (24dB/octave) slopes everywhere greatly simplifies tuning. Minimizing overlapping frequencies helps to minimize phase issues in the crossover region. 

I've also found that if the measured response of the speakers and their crossovers can be made to track a 24dB/oct LR response, the acoustic sum is close to ideal in frequency response. If you do it that way, then you'll need a parametric EQ per speaker and delays should be set before measuring the acoustic sum. When I've done it this way, I've never had to "fine tune" any delay settings. 

Finally, setting the crossover between midbass and sub the same way and then applying a shelf filter to shape the bass response works much better than just increasing the level of the sub channel, reducing the LPF frequency and screwing with delays.


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## ErinH (Feb 14, 2007)

something about 'gating' response people always overlook is that gating impacts the resolution of the measurement. 

if you gate the response down to 3ms, you're at about a resolution of 333hz. That means for every multiple of 333hz you have a data point. Basically, you've smoothed the response. Then things like 1/24th octave resolution start to become moot...


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## Niick (Jun 3, 2015)

ErinH, that is true of CONVENTIONAL gating, what i was referring to however was NOT conventional gating.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> I've never found making any kind of an approximated anechoic response in a car to be helpful.




Andy, your accomplishments in the field of car audio are not completely lost on me, and it is with the utmost sincerity and respect that I ask, have you ever worked with an analysis system that implements frequency dependent windowing? The reason I ask this is because, after having worked with SysTune in the car environment, I couldn't imagine going back. 

Edit: sorry, somehow missed your post where you already answered this question!

I have found that using the phase trace to determine proper delay times instead of what amounts to an educated guess with a tape measure, or trying to use the impulse response only to realize that ANY impulse response is dominated by the HF, and lining up the peaks is CERTAINLY not the way to achieve a good delay time for drivers covering different freq. ranges, so, yes, for me, phase response is the way to go for proper time delay adjustment, no matter the frequency range. Lo, mid, hi, doesn't matter. The phase trace works with them all. 

And so, yes, in order to see the phase trace and make sense of it in the HF (in a car) I will (frequency dependent) window out the reflections with my top graph, while watching my un-windowed amplitude response thru crossover on my bottom graph. All WHILE LISTENING TO MUSIC as my excitation signal. If anyone reading this wasn't aware, yes, you can and sometimes should use music as your test signal with these types of measurement systems. Music or pink noise, the results are the same. 

So for me, I've discovered a way to absolutely guarantee my customers, who are paying me their hard earned money, that I can and will optimize their systems to the fullest, and so far the results have been better than I could have ever predicted. 

I never would have discovered this method if I hadn't experimented and tried out different things, whether or not someone told me it didn't work. I'm learning new things, expanding my understanding of acoustics and loudspeaker interaction every day. And it is all a direct result of being able to quickly switch back and forth between windowed and un-windowed measurements, to instantly see and simultaneously hear the affects of reflections, phase, delay, etc.



Andy Wehmeyer said:


> I also find that using steep (24dB/octave) slopes everywhere greatly simplifies tuning. Minimizing overlapping frequencies helps to minimize phase issues in the crossover region.
> 
> I've also found that if the measured response of the speakers and their crossovers can be made to track a 24dB/oct LR response, the acoustic sum is close to ideal in frequency response. If you do it that way, then you'll need a parametric EQ per speaker and delays should be set before measuring the acoustic sum. When I've done it this way, I've never had to "fine tune" any delay settings.


I should probably mention that just as often, I use tools like SysTune for troubleshooting as often as for system optimization. And of the times I do use it for optimization, I don't always have the ideal system to work with. Sometimes our crossovers are limited in adjustability, sometimes they're completely passive, and I just want to be sure that (the customer's)tweeter and mid-bass are summing together as good as possible, every now and then, when a tweeter is far from the midbass and at a different angle, maybe the polarity needs to be swapped, with SysTune I can find out which way is optimum in literally under 3 min. It's all about the fact that I'm working on OTHER PEOPLES systems, and I rarely get to choose the components. So SysTune helps me to get the best out of ANY random scenario.


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## Niick (Jun 3, 2015)

Niick said:


> Well, I don't think necessarily that the psychoacoustic smoothing option presented in REW is or isn't helpful for car systems, what I was excited for, was the ability to implement a VARIABLE frequency dependent window, which I find immensely helpful to be able to "see" the phase response of the HF range which is usually obscured by all the early reflections.


I should probably make clear that while I find it helpful to to be able to measure phase, It is NOT the shape of the line drawn for any one drive unit's phase response that is really what's important. It's not like I'm using the phase measurement the same way as one would normally use a freq. response measurement. With the phase trace, All I ultimitely care about is NOT how much or how little it deviates from a preconceived notion of "ideal", what I'm looking for is simply the tweeter's phase trace to overlap the midbass. So you take a trace of just the tweeter, then keep that stored trace on the screen, and then mute the tweeter and take a trace of the midbass. Whichever one needs to be delayed, you adjust it live while keeping the other driver's stored trace on the screen. When the two traces overlap thru the crossover region, that's it. You're there. 

So I'm only wanting to be able to see the phase in order to make sure that the two drive units sharing a crossover are aligned. Because as Andy said, the phase trace will change DRAMATICALLY with mic placement, and also with your reference delay in the software. In fact you change the reference channel delay in the software to PURPOSEFULLY alter the phase trace to make it flattest in the frequency region of interest, thus making this whole process a little easier. 

Again, to clarify, I'm not measuring phase because I want to find out the phase response of a tweeter, or a woofer, or whatever. That WOULD require an anechoic chamber and ALOT better microphones than what I use! 

No, I measure phase response only to be able to see the RELATIVE DIFFERENCE between this driver unit (maybe a tweeter) and that drive unit (maybe a midbass) at this listening position. Which, in a car, is basically fixed in place, since you only have one place to sit. �� 

Edit: yes, yes yes yes yes, I absolutely understand the whole spatial averaging thing, I have multi mic arrays that I use for this purpose.....................just not for setting delays!

And keep in mind, all while doing this, your other graph on the screen can be anything you want, it can be the IR, the two IR's (one live of the driver being adjusted and the other static of the stored trace), the magnitude otherwise called frequency response, whatever. And it can all be done while listening to music instead of noise. 

It's all about correlating what I'm measuring with what I'm hearing. All the more ways I can measure something, all the more perspectives I can see that same data from, the deeper my understanding of how it all interacts will become.

Edit #2: Andy, my buddy Mike with Big Bear Marketing has been telling me about a new processor that you're coming out with, he says it's absolutely awesome. I believe he worded it like "AudioFrog (don't know model number)and the ARC PS8 are the best on the market right now"

We're working on trying to get my employer to carry AudioFrog, even if it's only special order, I haven't got my hands on any AudioFrog processors yet, but I will soon, my buddy sure thinks highly of them, and he knows his **** pretty good. I'm definitely looking forward to checking this unit out.


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## 14642 (May 19, 2008)

Niick said:


> Andy, your accomplishments in the field of car audio are not completely lost on me, and it is with the utmost sincerity and respect that I ask, have you ever worked with an analysis system that implements frequency dependent windowing? The reason I ask this is because, after having worked with SysTune in the car environment, I couldn't imagine going back.
> 
> Edit: sorry, somehow missed your post where you already answered this question!
> 
> ...


Hey Niick, the next time I'm in Portland, would you give me a primer in person in exchange for lunch or dinner?


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## ErinH (Feb 14, 2007)

Niick said:


> Music or pink noise, the results are the same.


Maybe for the context in which you are speaking (the particular manner of measurement) but I do want to note that this isn't true in general. Pink noise fails to excite high-Q modal issues as well as something as simple as a slow swept sine. I can't tell you the number of times I've listened and tuned via pink noise or a particular set of tracks only to find sometime later another song that accentuates a problem the other test tracks didn't. 

What I find works best is a good mix of everything: test tones, pink noise and music. I typically focus on test tones below the Schroeder frequency because above that, as you've somewhat touched on, the mic placement variances of even as little as 1/4" can affect the measured response. In a home this isn't as big of an issue; in a car where comb filtering is prevalent it is. So test tones are less beneficial the higher in frequency you go. That said, a swept sine can still prove useful. But these are all various methods to achieve a desired goal. With experience you learn what works well and what doesn't and better understand the faults of each source and test method.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> Hey Niick, the next time I'm in Portland, would you give me a primer in person in exchange for lunch or dinner?


Absolutely. I'll PM you my contact info.


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## 14642 (May 19, 2008)

Niick, one more question:

When you're using delay to adjust phase at the crossover for the right and left independently, how do you reconcile the need for the sound from right and left to arrive at the mic at the same time if the delay must be changed to fix the crossover? 

Maybe this isn't really a problem. I've never done it this way because I have always used delay to correct for distance and used an all pass filter to adjust phase at the crossover.

You can email me at [email protected]


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## Niick (Jun 3, 2015)

ErinH said:


> Maybe for the context in which you are speaking (the particular manner of measurement) but I do want to note that this isn't true in general. Pink noise fails to excite high-Q modal issues as well as something as simple as a slow swept sine. I can't tell you the number of times I've listened and tuned via pink noise or a particular set of tracks only to find sometime later another song that accentuates a problem the other test tracks didn't.
> 
> What I find works best is a good mix of everything: test tones, pink noise and music. I typically focus on test tones below the Schroeder frequency because above that, as you've somewhat touched on, the mic placement variances of even as little as 1/4" can affect the measured response. In a home this isn't as big of an issue; in a car where comb filtering is prevalent it is. So test tones are less beneficial the higher in frequency you go. That said, a swept sine can still prove useful. But these are all various methods to achieve a desired goal. With experience you learn what works well and what doesn't and better understand the faults of each source and test method.


Oh totally, I agree. Just yesterday in fact my store manager wanted me to listen to his car because he said one of his door speakers sounded like it was blown or something. So we go out to his car, he plays a song, and says "hear that?"

What he was referring to was a buzzing sound that sounded like a cone and voice coil former coming unglued, coupled with door panel resonance, all singing in terrible unison together. But it didn't always happen.

So I said wait here and went a grabbed an old iPhone 4 that I use as a tone generator (AudioTools for iOS). I plugged it into his aux in and swept slowly downward and as we got to 70Hz-80Hz-ish, BZZZZZZZZZZ...........there it was, the offending sound that only occasionally reared its ugly head with music. 

So yeah, I totally agree.


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## 14642 (May 19, 2008)

And for measuring arrivals, the lack of high frequency information does make it difficult. You can get around that by normalizing the impulses, finding the peak (or the top of the hump) and then backing up to the left until you reach some consistent level of attenuation to approximate the arrival. -12dB works pretty well.


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## Niick (Jun 3, 2015)

Andy, I'll send an email to ya when I get to work, literally walking out door right now. Very much looking forward to diving deeper into this with you, be prepared for a lot of questions!! I have an almost unhealthy need to understand the inner working of things, like processors for instance......(hint hint)?


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## Niick (Jun 3, 2015)

Takes me about an hour to get to work


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## Niick (Jun 3, 2015)

Andy, I sent you an email just now. I sent it to [email protected]. 

audiofrog with an "r" correct?


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> Niick, one more question:
> 
> When you're using delay to adjust phase at the crossover for the right and left independently, how do you reconcile the need for the sound from right and left to arrive at the mic at the same time if the delay must be changed to fix the crossover?
> 
> ...


Andy, just got home from work, thought I'd try to take a minute and answer these couple questions. So, if you are wanting to optimize the system for the driver seat only, then you'd first work with one channel at a time. Say the left midbass and tweeter, and for the sake of example, the tweeter is leading. So you'd play both drivers simultaneously, in order to set your reference delay in the analyzer. DO NOT CHANGE THE REFERENCE DELAY IN ANALYZER ONCE THIS IS SET. Then mute the tweeter. Take a trace of just the midbass. Now mute the midbass. Make the captured midbass trace visible on screen, and now unmute the tweeter and adjust the delay until the tweeter's trace overlays the midbass's. Now, note the delay time in the processor that you have now established as necessary for these particular 2 drive units to be phase coherent, thru crossover, at this particular mic location. Now, no matter how much you delay the left channel as a whole, just keep the relative delay between the drivers the same. See where I'm going with this.........?

Now keep in mind that I haven't used a processor yet with adjustable phase. Just delay. When I get my hands on one of those, my methodology will be adjusted for that processor


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## Niick (Jun 3, 2015)

I guess I should have gone a little further in my explaination, ok so, now that you have optimized the crossover region for the left mid-tweet, do the same for the right mid-tweet, then, using the impulse response, you can play the entire left side as a whole (mid and tweet together) and compare it's propagation time to that of the right side, and now enter the necessary delay for the entire left side, keeping the delta time between mid and tweet. Get it?

Edit: seriously, does that make sense? There could very well be a flaw in the methodology here, I'm learning new things all the time, so who knows if one day I'll look back at this method and say "what was I thinking". &#55357;&#56832;


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## 14642 (May 19, 2008)

^^

Yeah, I get it, but I'm not sure I agree with this unless the delay setting necessary for aligning the mids and tweeters are the same on both sides. That's why this method works for home audio speakers--both speakers are exactly the same and then they are placed in a room and room correction is applied. 

In a car, the left and right measured crossover responses may be quite different and if dramatically different delay settings are used in the crossover correction, then that may throw off the left and right settings. 

In a home speaker, delay is used in the same way--aligning the arrivals of the various elements. Using delay for that is fine and that's what it's for. If additional delay is required to align the crossover (which is think is suspect because there is an alignment that makes this unnecessary) it doesn't screw up left and right delays in the room because both speakers are identical--measured in a chamber, outside or with some kind of a gated system.


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## 14642 (May 19, 2008)

I'm quite convinced that this is a misapplication of delay as phase adjustment. We misapply things all the time in audio when we have to choose between the benefits and drawbacks of all of the compromises that we make to optimize systems, components and settings for specific applications.

I think that for the transition from tweeter to mid, this is likely not to be a big problem--right and left delays for tweeters aren't very critical, since we use level to determine azimuth at high frequencies. Between a midrange and a midbass, it's less appropriate. Between a sub and a midbass, it probably works OK, so long as you're adjusting BOTH midbass drivers along with the rest of the front speaker system symmetrically or just adjusting the sub. 

I'm sticking by my hypothesis for now--that in a car, an all pass filter is the way to adjust phase at the crossover (or use LR4 and EQ the responses to match the target) and reserve delay settings for correcting for distance.

So here's my suggestion for a little more sleuthing. After you've EQed. try running your FDW on the entire right (including the sub) and the entire left (including the sub) and compare those impulses and phase graphs.

Of course, if the delay adjustments for crossover optimization are symmetrical for both sides, then this is likely not an issue.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> ^^
> 
> Yeah, I get it, but I'm not sure I agree with this unless the delay setting necessary for aligning the mids and tweeters are the same on both sides. That's why this method works for home audio speakers--both speakers are exactly the same and then they are placed in a room and room correction is applied.
> 
> ...



Ok, so full disclaimer: ya know, to be honest with ya, I am not convinced that always optimizing the system for the drivers seat only is the way to go. In fact, when a customer approaches me about this very question, I usually respond with something like, "hmmm....well.....yeah, you can, but..........", so, of the times I've used this method, the delta delay times for the two sides have been the same. And also, I haven't been using this method for very long, so every time I do it I learn new things and experience yet another car's acoustical environment, so I have by no means done this with every car out there, so yeah......it's an evolving thing, ok, hang on, I'm gonna post this, then go down and read the next post from ya, answer that, then I'll check my email......ok, here goes........(there's so much more that I'm thinking, it's like, "oh yeah, I should have mentioned this, oh yeah I should have mentioned that.....?"


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> I'm quite convinced that this is a misapplication of delay as phase adjustment. We misapply things all the time in audio when we have to choose between the benefits and drawbacks of all of the compromises that we make to optimize systems, components and settings for specific applications.


Ok, so I'm thinking this, now I could be wrong, but here it is-

Let's say a person, who has no means of acoustic analysis at all, installs an Audison Bit one, (I use the Bit One at work). And they have a fully active system. The tweeters are in the dash, at the base of the windshield, the midbass in the doors. Let's say that person was gonna make two presets, one for both seats, one for the drivers seat, now, let's look at the "both seats scenario".

I always thought that the person would use a tape measure to determine the distance from some point between the seats, and each drive unit, then work out the math to determine the delay times necessary for the acoustical energy from each drive unit to now reach that point all at the same time. 

If that is how you would do it, then all I'm proposing is using the phase trace as a means of determining that timing. 

I could be wrong, totally, but I always thought that's what people did? 




Andy Wehmeyer said:


> I'm sticking by my hypothesis for now--that in a car, an all pass filter is the way to adjust phase at the crossover (or use LR4 and EQ the responses to match the target) and reserve delay settings for correcting for distance.


I in no way disagree with you on that. I personally have yet to get my hands on a processor, because we don't sell any, that have adjustable all pass, I looked into it like maybe I missed a very fundamental setting in the 360.3 and the bit one, but I don't think so. Those processors don't have all pass filters that can be implemented IN CONJUNCTION WITH HP or LP do they? Hell maybe they do! So, those are really the only two processors that I have experience with using this method, that and head units with delay. And in that scenario, the phase trace is a HUGE benefit for me, and the customer of course, because if someone has a sub in the trunk, midbass in the doors, and their H.U. has delay, like a lot of the alpine's we sell, then you can at least use delay to get rid of the cancellation induced dip that is almost always there. Even on the steepest slope settings, little overlap, you won't see it on a 1/3 octave RTA, but 1/24, 1/48, or 1/96 octave and it's clear as day, in this scenario the phase trace can be used to determine the delay setting that gets the midbass and sub in alignment thru crossover. And yes, it's a symmetrical setting. And when you get rid of that dip, WOW, the difference it makes.

Ok I'm gonna make me a cup of coffee and check my email............


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> use LR4 and EQ the responses to match the target)


Hang on here, elaborate a little on this, you mean select all pass instead of HP or LP, then set filter type to LR 4th order, then use EQ to "shape" the type of filter you want, whether it be LP or HP?


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## Niick (Jun 3, 2015)

Ok, just checked my email, yeah, totally, same delay times between tweet and mid for left and right

Edit: but now you've got me thinking, maybe I'm missing something inside these processors, maybe it does something I didn't realize it did?


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## Niick (Jun 3, 2015)

Same delay times BETWEEN mid and tweet, so, just like if you had the luxury of physically aligning the two drive units into their optimum places, like in a custom kick.

So now once you have the left SYSTEM (mid and tweet) working together nicely, and the right SYSTEM working together nicely, then ,if you want to center your self in the soundstage, then measure the impulse of those now completed two systems, and delay the closer of the two SYSTEMS as a whole, keeping the delay time DIFFERENCE between the mid and tweet the same. 

But to be honest, I don't necessarily ALWAYS like tuning for the drivers seat only.


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## sqnut (Dec 24, 2009)

Niick said:


> Same delay times BETWEEN mid and tweet, so, just like if you had the luxury of physically aligning the two drive units into their optimum places, like in a custom kick.


In most car installs, with the woofer down low in the door and the tweet on pillar or dash, the PLD between the left and right mid and tweet will not be the same. Unlike with home speakers as Andy has already mentioned.


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## Niick (Jun 3, 2015)

sqnut said:


> In most car installs, with the woofer down low in the door and the tweet on pillar or dash, the PLD between the left and right mid and tweet will not be the same. Unlike with home speakers as Andy has already mentioned.


Right, IF you're optimizing for the drivers seat................also just like Andy said, the best choice of COMPROMISES is what we're all really shooting for.........so that's what I'm trying to do.....


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## Jepalan (Jun 27, 2013)

Niick said:


> Hang on here, elaborate a little on this, you mean select all pass instead of HP or LP, then set filter type to LR 4th order, then use EQ to "shape" the type of filter you want, whether it be LP or HP?


No. I believe Andy is talking about a specific type of filter (one that isn't available in most HUs and amps, but is available in a few processors/DSPs).

(from the interwebz...)
An all-pass filter is a signal processing filter that passes all frequencies equally in gain, but changes the phase relationship between various frequencies.


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## sirbOOm (Jan 24, 2013)

Here I am thinking some pink noise, an RTA, and some time alignment will cover me...

fuuuuuuuuuuukarroooooo hahaha


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## Niick (Jun 3, 2015)

Jepalan said:


> No. I believe Andy is talking about a specific type of filter (one that isn't available in most HUs and amps, but is available in a few processors/DSPs).
> 
> (from the interwebz...)
> An all-pass filter is a signal processing filter that passes all frequencies equally in gain, but changes the phase relationship between various frequencies.


Right, that's what I thought to, but it's totally possible that I've been missing something. I fully get the difference between delay, which will delay all frequencies equally, thus changing the phase a greater and greater amount as frequency increases, and all pass filters, which can shift the phase of just a group of frequencies. 

And when I do get an opportunity to use a processor with these type of features..........hell yeah!


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## Niick (Jun 3, 2015)

sirbOOm said:


> Here I am thinking some pink noise, an RTA, and some time alignment will cover me...
> 
> fuuuuuuuuuuukarroooooo hahaha


Not quite sure how to interpret this response?


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## crazhorse (Mar 9, 2010)

was browsing and stumbled across this

Waveflex Caraudio

looks like it does some interesting stuff phase and eq wise...


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## cajunner (Apr 13, 2007)

yeah, fancy stuff there.


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## subterFUSE (Sep 21, 2009)

Niick said:


> Right, that's what I thought to, but it's totally possible that I've been missing something. I fully get the difference between delay, which will delay all frequencies equally, thus changing the phase a greater and greater amount as frequency increases, and all pass filters, which can shift the phase of just a group of frequencies.
> 
> 
> 
> And when I do get an opportunity to use a processor with these type of features..........hell yeah!



My Helix DSP Pro has phase adjustments in 11.5 degree steps based on 2nd order allpass filters.

I haven't figured out how to use them yet, however.


I would certainly love to know more about them and how to adjust.




For now, I usually just adjust delays by ear using pink noise. I play the midbass speakers together and adjust delay until the pink noise images from the center. Than I play the horns and do the same. Lastly, I play the horns and midbass together and use pink noise that straddles their crossover region. I link the midbass together and horns together and then adjust them as a group to the other set until it sounds "right."


The phase angle stuff seems to complicate things so I have been ignoring it until I learn more about how to use it.


Sent from my iPad using Tapatalk


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## Niick (Jun 3, 2015)

So just today I tuned a system that I installed yesterday. While keeping in mind all that we've discussed on this thread, and all the input I've gotten from Andy W. I made an attempt to see how exactly I would go about tuning for the drivers seat specifically. 

Let me first say that the end results were really, really good. You know you've got a system dialed in right when you're not even using THAT good of speakers and it sounds excellent. Great imaging, spacious, deep, great blending between sub, midbass and tweet. Just really good. The system was an alpine cd deck, Audison bit one, JL XD600.6,Hertz HE 2 way (run fully active), Hertz HE 5x7 rears, Rockford PS-8. (None of this was my choice, equipment wise, my job was to install and tune) it was in a 2004ish Ford F-350 extended cab.

What I did was place my mic array (3mic array, CSL EMM6s) in the middle, between the driver and pass. seats. I first determined the tweeter Fs by dong an impedance sweep, this way I could get a good idea of minimum safe Xover freq. then I set up the xovers for the front 2ways, crossing the tweeters as low as I felt was safe and well above Fs, then bringing the midbass in freq until I got relatively good summation thru Xover, the woofers LP freq. in the processor ended up well above the teeters HP. 

Ok, so I set the Xover for the front left and front right the same. Is that good? Seriously I don't know, like I've said before, I've had to learn all this on my own, there isn't anybody I work with now or have worked with before in 15+years who truly knows about this stuff. 

So, now that I got the front xovers set, I moved the mic array to the drivers location, measured the IR of front left, (mid and tweet together) front right, rear left, and rear right, saving the trace of each. Rear right was the farthest back, (I think? Can't remember for sure,) anyway, I kept the saved IR of the farthest speaker on the screen and one by one adjusted the delay of each remaining corner until they all arrived at the same time. For the sub, I adjusted it so it's IR peak was a little behind the others, but they STARTED to rise from zero about the same time. This would be finely adjusted in a minute.

So now that I had each speaker SYSTEM location arriving at the same time, I focused on perfecting the summation thru Xover of the fully active 2 way fronts with the only tool I had, which was delay. This processor doesn't have all pass filters. For the drivers side, about .20 ms or so with tweeter polarity inverted got me great summation thru Xover. For the pass side, it was as good as it was gonna get already. For the sub I ended up adjusting its delay down about 1.75 ms, and the sub to midbass transition filled in nicely. Before any EQing the entire bandwidth fro 20 to 20k was within a 16 dB window.

A little EQ, and I was done. Keep in mind that every time I do this I learn a little more, and every time I do this I'm working against a the clock, I only have a finite time to accomplish the tuning.

I hope I explained that ok.

Edit: I should mention that I first enter 5-8 ms of delay for every channel. Same delay for all channels, this way, you can adjust either direction, no matter whether you need to go "up" or "down" with your relay times. So for instance, with this system, I first put every channel at 6 ms delay. Then, when I got to the sub, which was on the floorboard, in the middle behind the front seat, I was able to take away about 1.75 ms. Had this been at 0ms already, I wouldn't have been able to add negative delay.


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## Niick (Jun 3, 2015)

One of these days I'm GOING to put together a screen shot sequence of the tuning process that I've developed so far. This way, I'm sure I'll get a lot of good input and ideas, as there are some truly brilliant people on this forum. Thanks to all who have shared knowledge.


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## sqnut (Dec 24, 2009)

Niick said:


> A little EQ, and I was done........


It takes at least a few days or about 10-15 hours to get the sound in the ball park. In every setup I've tuned, about 30% of the time is devoted to setting timing/crossovers and 70% on using the eq. To get the right response, you need to eq a lot. 20% of eq time is setting L/R response by measuring and 80% is setting the combined response by ear. 

With the next car you tune, set it the way you do and then pick one well recorded track and go back and forth between a 2ch and the car. You may need to do this a few times before you start picking the differences. Try and block everything out and just listen to the vocals.

While we are on the topic of timing, how you set L to R determines where the vocalist images up. In front of you, at the rear view or in front of the passenger. For any given setting, you're never going to get similar placement from L&R seat. With the vocalist at the rear view it gives equal space to left and right stage.

The L&R timing is one part the next is to time the sets of drivers. Example, in a two way, hearing your tweets 0.02 ms before the mids, with the mids and 0.02 ms after the mids are three very different sounds. 




Niick said:


> Edit: I should mention that I first enter 5-8 ms of delay for every channel. Same delay for all channels, this way, you can adjust either direction, no matter whether you need to go "up" or "down" with your relay times. So for instance, with this system, I first put every channel at 6 ms delay. Then, when I got to the sub, which was on the floorboard, in the middle behind the front seat, I was able to take away about 1.75 ms. Had this been at 0ms already, I wouldn't have been able to add negative delay.


Relative delay between your sub and the mids is what matters. Achieving it by adding 5ms to all drivers and then reducing delay on subs to 3 ms is the same as leaving your sub on 0 and delaying the mid 2 ms.


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## Niick (Jun 3, 2015)

sqnut said:


> Relative delay between your sub and the mids is what matters. Achieving it by adding 5ms to all drivers and then reducing delay on subs to 3 ms is the same as leaving your sub on 0 and delaying the mid 2 ms.


Dude, I totally get that. What I'm saying is, by delaying all channels the same amount first, you then have the freedom to add, OR take away delay as you go. This is an old trick used by prosound guys doing alignments in the field. Remember, I AM WORKING in a time sensitive environment. I don't have 10 hrs to devote to system optimization. I don't have 5 hours usually. 2-3 is the most I'll usually get. If I could, I'd love to have more, but that isn't the current reality at the shop I work at. 

Therefore, I've been trying to develop an objective means of system optimization that takes my own subjective interpretation out of the equation as much as possible. I find, that If you keep in mind the theory that we, as installers, are a "waveform delivery service", trying to deliver the waveform of the music from the electrical domain to the acoustical domain with as little alteration a s possible, then there really DOES start to manifest a "baseline" concept of "ideal", that isn't open to interpretation, that ISNT a matter of opinion. That thru physics, we can achieve. It's all about making the most of your available time, equip. Etc. I probably look at the concept of system tuning from a different perspective than most people here, because of this time/efficiency equation that plays a MAJOR role in everything I do.


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## cajunner (Apr 13, 2007)

Niick, have you looked at Raimonds device, the APL-1?

it's a DSP unit that may help you find a faster window to turnkey?

looks like it may be better off in the hands of an install shop since the software is the expensive part.

it was made to go in the car but as a diagnostic device it may prove to be even better in cost/performance as long as you can automate the testing scheme with a robotic microphone array that would be left to do it's measurements, sort of a ROOMBA vacuum but for making DSP measurements, haha...


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## Niick (Jun 3, 2015)

Nope, never heard of it.....I'm definitely gonna check it out, if for no other reason than it sounds cool. Here's a pic of my current measurement setup at work......


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## crazhorse (Mar 9, 2010)

Thinking I have bad shifts at my xover phases...........

sub/mid 80hz and mid/tw 3000hz


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## seafish (Aug 1, 2012)

Niick, I am wondering if this article below is partly similar to how you tune--

The Subwoofer DIY Page v1.1 - Time Alignment using HolmImpulse


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## Niick (Jun 3, 2015)

seafish said:


> Niick, I am wondering if this article below is partly similar to how you tune--
> 
> The Subwoofer DIY Page v1.1 - Time Alignment using HolmImpulse


Yeah, kinda, I didn't read it really in depth, but I've read it before. That's pretty much it when it comes to delay times for 200 Hz and up. So, MY personal tuning method is evolving all the time, slight changes here and there as I experience more different scenarios and learn more and more each time. But yes, the basics is time alignment thru impulse response, then EQ right, then left, then bring in the sub. I can get more detailed if you'd like at a later time, as Imjust got home from work and am about to eat dinner. My wife is an excellent cook! Lucky me!

Yesterday I attended a training by Scott Welch and Randal K in Washington state, it was all about system tuning and some fabrication stuff at the end, but mostly focused on tuning. The system Scott was teaching was using Holmimpulse and True RTA, but whatever. (Edit: by "whatever" I mean, it's not important what program u use, so much as you know how to use the program you choose to use, whatever it may be) It really covered the basics, we didn't go to deep "down the rabbit hole" as he called it, but I definitely got some good things out of it, for sure. I also got to hear Scott Welch's Dodge Charger, AWESOME!! (At least I think I was a Dodge Charger, I also got to hear two other cars, a Nissan Maxima I think and an older thunderbird, all were quite awesome, all were IASCA SQ Competitors)


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## Niick (Jun 3, 2015)

Crazhorse, is that left and right that were looking at there?


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## crazhorse (Mar 9, 2010)

Ya it is


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## Niick (Jun 3, 2015)

Ok cool that's what I thought, just makin sure. So, at the training Scott said what he does is leaves the sub delay set to zero, does all his delays using IR with all xovers DISABLED, then, lastly, plays a single tone at the freq. of crossover between the sub and mains, then while tone is plaing adds delay to the sub channel until he hears it go totally out of phase with the mains, then keeps going until he hears the sound from the sub "pop up front and land on the dash". I'm pretty sure he said a single tone, although thinking about it now, he might have said filtered pink noise, I'll have to check my notes. Filtered pink noise is also really helpful to set your center imaging, it can be generated from within Smaart.


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## Niick (Jun 3, 2015)

I haven't actually tried this yet, it seems to me that you might often need to take delay AWAY from the sub channel, not add it. But who knows!!


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## crazhorse (Mar 9, 2010)

I think I have it at 0 ( the delay ) on the sub at the moment.... Haven't got to figure out the preamp usb asio thing yet either. Those measurements were done with apl workshop.... When it takes those measurements it also does a impulse trace as well ( I'll upload those of my laptop tomorrow) not sure how it does it without a loop back but it does use windowing or gating or something like that I think


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## crazhorse (Mar 9, 2010)

Which I am waiting on a reply from support on why the high end is dead in the upper k's measurement wise...don't know if it's a setting in the app or a problem with my mic


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## Niick (Jun 3, 2015)

Does that measurement system you're using there, does it do real time? Or is it sweeps?


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## crazhorse (Mar 9, 2010)

It is sweeps


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## Babs (Jul 6, 2007)

Niick said:


> Ok cool that's what I thought, just makin sure. So, at the training Scott said what he does is leaves the sub delay set to zero, does all his delays using IR with all xovers DISABLED, then, lastly, plays a single tone at the freq. of crossover between the sub and mains, then while tone is plaing adds delay to the sub channel until he hears it go totally out of phase with the mains, then keeps going until he hears the sound from the sub "pop up front and land on the dash". I'm pretty sure he said a single tone, although thinking about it now, he might have said filtered pink noise, I'll have to check my notes. Filtered pink noise is also really helpful to set your center imaging, it can be generated from within Smaart.


Nice to know I'm not too far off then.. That's kinda how I was taught and how I've sort of taught myself for phasing in the subs. ErinH has a great article in here on sub phase, and I had the luxury of seeing him do it in my my car at the NCSQ meet. 

Yep, I guess something like 1/3 octive tones or maybe 1/3 octave warble tones as well maybe. Funny I can kinda make the bass move around the car by phase adjustment or delay as well. I guess it's a matter of alignment of the wave-forms along with dealing with modal properties of the car. I think I just used tones at 80 (crossover freq) then 85, 90, 100hz even. I guess my thinking is being in phase at higher frequencies even than right at crossover point.

Good way of detecting a TA or phase issue between your mids as well if the sub dials in nicely in phase with one side, but isn't on the money for the other side at the crossover freq.

One question I have though is, let's say my sub and mids are in phase at 100 or 120hz, can I assume they will be in phase at the crossover freq of 80hz? Or does the phase shift screw that up at 80hz?

.. fascinating stuff.


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## Niick (Jun 3, 2015)

Babs said:


> One question I have though is, let's say my sub and mids are in phase at 100 or 120hz, can I assume they will be in phase at the crossover freq of 80hz? Or does the phase shift screw that up at 80hz?


Excellent, excellent, excellent question! This is EXACTLY the reason why the phase trace is good to use for aligning subs to fronts. There are articles on this very thing. Of course, they are NOT written by car audio guys, they are written by pro-sound guys....................................

Of course this could just be my perception, but there seems to be a school of thought that says moving coil/cone loudspeakers being driven by amplifiers and w/ various types of electrical filtering ceases to exist in the material world of physical matter, just as soon as you put them into a mythical "car". The magical car changes the physical properties of the transmission medium know as "air" in all places outside the car, once inside the car this transmission medium takes on properties that our scientific instrumentation cannot deal with or even measure. Much like the realm of the afterlife. For because there are early arriving, often very strong reflections, this must therefore mean that all knowledge gained by all others who work with these very same electrical circuits and electromechanical acoustic transducers should be immediately dismissed as they work not within the mythical realm of the car, therefore can't possibly understand the complexity of all that changes when speakers are placed in cars.

I don't personally subscribe to this school of thought. Small room acoustics, VERY small room acoustics is how I like to think of it. 

I'll see if I can't find that article........


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## Niick (Jun 3, 2015)

Well I couldn't find exactly what I was looking for, but here is the only real POTENTIAL downfall with doing it that way, you could be many cycles off, like meaning wayyyyy to much or too little delay, and get that one frequency to have maximum summation, using either nulling, like ErinH describes, or the method I described as told by Scott Welch. (IF indeed Scott's method WAS to use a single tone.......I'm almost certain it was.) you could be many multiples of 360 degrees out of phase and get ONE SINGLE FREQUENCY to sum maximally.

But then, when you turn the Sytem back on, ouch! Something's not right. 

But long story short, yes, as long as you keep in mind the actual physical distance that your adjusting for, and the wavelength range of the frequencies you're affecting, then yes, it would totally work great.


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## Babs (Jul 6, 2007)

Wow. Ok. So at least in this instance best bet is go back to REW (tool used in my case) and measure after each such changes and see what the mic is hearing. Given the knowledge that a wide dip is more audible than a very narrow cutout. Push for minimizing the affects, keeping bass upfront and "by and large" in phase with mids in as much of the cross band as possible. Thus the recommendations of folks like Andy for 24db slopes whenever prudent. Another advantage for 4-way total systems vs 3-way (tweet/mid/sub). 

In short and maybe oversimplification.. Having as much as possible of the cross-bandwidth in phase understanding you can't have it all 100% in phase, narrowing phase cutouts as much as possible and minimizing the quantity of the same. 


Sent from iPhone using Tapatalk


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## Babs (Jul 6, 2007)

I discovered this also with tweeters recently playing with variable test tone app. I could manually "sweep" the tone freq and find points of either direct or reflection-caused cutouts that were obvious phase cuts. Any tweaking of TA at that point would require remeasuring to tell if the tweak went for the better or worse. Either between pairs, OR groups. 


Luckily these were fairly inaudible with music, but certainly by a single tone.. In which case I'd stick with going for dead on image and stage, worrying not so much about phase related issues up that high anyway, but it's an interesting demonstration that just because you're kinda sorta fairly on with TA, you can still be out of phase extremely easy.. Within those .02ms bumps in one direction or another of bliss or chaos.

Sent from iPhone using Tapatalk


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## sqnut (Dec 24, 2009)

Kid,

We all love your enthusiasm and the fact that you want to share your experiences and journey. While you are at it, just drop the attitude.



Niick said:


> Excellent, excellent, excellent question! This is EXACTLY the reason why the phase trace is good to use for aligning subs to fronts. There are articles on this very thing. Of course, they are NOT written by car audio guys, they are written by pro-sound guys....................................


You haven't answered the OP's question. If the sub and mids are in phase @ 100 and 120 hz, are they in phase at 80? We all know there are plenty of articles on the topic, some written by pro audio guys. What did those articles tell you about the question at hand? 



Niick said:


> Of course this could just be my perception, but there seems to be a school of thought that says moving coil/cone loudspeakers being driven by amplifiers and w/ various types of electrical filtering ceases to exist in the material world of physical matter, just as soon as you put them into a mythical "car". The magical car changes the physical properties of the transmission medium know as "air" in all places outside the car, once inside the car this transmission medium takes on properties that our scientific instrumentation cannot deal with or even measure. Much like the realm of the afterlife. For because there are early arriving, often very strong reflections, this must therefore mean that all knowledge gained by all others who work with these very same electrical circuits and electromechanical acoustic transducers should be immediately dismissed as they work not within the mythical realm of the car, therefore can't possibly understand the complexity of all that changes when speakers are placed in cars.
> 
> I don't personally subscribe to this school of thought. Small room acoustics, VERY small room acoustics is how I like to think of it.
> 
> I'll see if I can't find that article........


Having a bad day at work?



Niick said:


> Well I couldn't find exactly what I was looking for, but here is the only real POTENTIAL downfall with doing it that way, you could be many cycles off, like meaning wayyyyy to much or too little delay, and get that one frequency to have maximum summation, using either nulling, like ErinH describes, or the method I described as told by Scott Welch. (IF indeed Scott's method WAS to use a single tone.......I'm almost certain it was.) you could be many multiples of 360 degrees out of phase and get ONE SINGLE FREQUENCY to sum maximally.
> 
> But then, when you turn the Sytem back on, ouch! Something's not right.
> 
> But long story short, yes, as long as you keep in mind the actual physical distance that your adjusting for, and the wavelength range of the frequencies you're affecting, then yes, it would totally work great.


And you still haven't provided a definitive answer. Let's keep this aside for a bit, the thread is about xover phase shift. Can you describe what that sounds like?


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## Niick (Jun 3, 2015)

sqnut said:


> Kid


Kid? I'm not your kid. 



sqnut said:


> You haven't answered the OP's question. If the sub and mids are in phase @ 100 and 120 hz, are they in phase at 80? We all know there are plenty of articles on the topic, some written by pro audio guys. What did those articles tell you about the question at hand?


My bad, I thought I did answer his question, yes, you can get a sub and main to be in phase at exactly ONE freq but be varying degrees out at all other frequencies depending on how far from that one freq you go. That would be, LIKE I SAID BEFORE, what would happen if you delayed the leading driver by many multiples of 360 degrees, at that frequency, get it? That's what those articles told me about the question at hand.




sqnut said:


> Having a bad day at work?


I hadn't actually left for work yet when I made that post, so no. Besides, I VERY RARELY have a bad day at work. I love my job and it gets better every day. 

What made you think I was:

1: at work
2: having a bad day



sqnut said:


> the thread is about xover phase shift. Can you describe what that sounds like?


Being that crossovers operate in the electrical domain, I guess the phase shift induced by these filters doesn't really SOUND like anything, now does it? 

How the phase shift induced by a filter in the audio band will affect the sound we hear depends on many factors, including the speakers themselves, their relation to each other and their relation to the listener, the lobing patterns that result, your position in relation the the horizontal and vertical axis of the driver alignment, etc. 

So no, looks like ya got me there smart guy! You're right, I can't describe what crossover phase shift SOUNDS LIKE!


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## 14642 (May 19, 2008)

Niick said:


> Excellent, excellent, excellent question! This is EXACTLY the reason why the phase trace is good to use for aligning subs to fronts. There are articles on this very thing. Of course, they are NOT written by car audio guys, they are written by pro-sound guys....................................
> 
> Of course this could just be my perception, but there seems to be a school of thought that says moving coil/cone loudspeakers being driven by amplifiers and w/ various types of electrical filtering ceases to exist in the material world of physical matter, just as soon as you put them into a mythical "car". The magical car changes the physical properties of the transmission medium know as "air" in all places outside the car, once inside the car this transmission medium takes on properties that our scientific instrumentation cannot deal with or even measure. Much like the realm of the afterlife. For because there are early arriving, often very strong reflections, this must therefore mean that all knowledge gained by all others who work with these very same electrical circuits and electromechanical acoustic transducers should be immediately dismissed as they work not within the mythical realm of the car, therefore can't possibly understand the complexity of all that changes when speakers are placed in cars.
> 
> ...


Well, I certainly don't subscribe to this school of thought either, but small room (very small) acoustics require different solutions in many cases. 

This particular case doesn't have so much to do with the car, specifically, as it does with the necessity of using asymmetrical EQ and delay because the sound is being optimized for an offset listener. 

Designing a single home audio 3-way speaker is not the same as a 3-way system mounted in the doors or dash of a car. We've already been through that.

Nick is right about the several cycle delay that may happen if you delay the sub to match the phase of the front at the crossover, especially when the sub is the speaker that's furthest away from the listener. 


Crossovers can be designed in such a way (LR4) that the phase of one side leads and the other side lags by the same amount. The result of such a crossover is flat phase and flat magnitude. Now, if that filter is available and if careful tuning of the electrical filters (crossover and EQ) can provide that alignment for the acoustic signal, then the crossover phase problem is fixed. Then, delay can be used to correct for distance which is a requirement for this alignment to work. That setting is also appropriate for optimizing the center image. There. Fixed. Simple.


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## sqnut (Dec 24, 2009)

Niick said:


> Kid? I'm not your kid.


Did I say you were my kid? I just called you a 'Kid' not 'my kid'. See the difference? I called you a kid cause you're behaving like a petulant 5 year old. 





Niick said:


> My bad, I thought I did answer his question, yes, you can get a sub and main to be in phase at exactly ONE freq but be varying degrees out at all other frequencies depending on how far from that one freq you go. That would be, LIKE I SAID BEFORE, what would happen if you delayed the leading driver by many multiples of 360 degrees, at that frequency, get it? That's what those articles told me about the question at hand.


Sure, you did.........just after taking about an hour to Google the answer, while giving vague answers meanwhile.






Niick said:


> I hadn't actually left for work yet when I made that post, so no. Besides, I VERY RARELY have a bad day at work. I love my job and it gets better every day.
> 
> What made you think I was:
> 
> ...


Good for you.





Niick said:


> Being that crossovers operate in the electrical domain, I guess the phase shift induced by these filters doesn't really SOUND like anything, now does it?
> 
> How the phase shift induced by a filter in the audio band will affect the sound we hear depends on many factors, including the speakers themselves, their relation to each other and their relation to the listener, the lobing patterns that result, your position in relation the the horizontal and vertical axis of the driver alignment, etc.
> 
> So no, looks like ya got me there smart guy! You're right, I can't describe what crossover phase shift SOUNDS LIKE!


You're again:

1. Thinking in the pro audio domain.
2. Don't know the answer and are obfuscating. Google is your friend.


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## crackinhedz (May 5, 2013)

sqnut said:


> I called you a kid cause you're behaving like a petulant 5 year old.


as a non-partial bystander (because I dont know either of you guys)...to me it seems you are the one trying to instigate something here. 

Having read through this thread trying to learn something, I think because Niick disagreed with you on 'sub delay', this may have ruffled your feathers and now youre just itching for a schoolyard fight. I didnt interpret Niicks response as being disrespectful torwards you, but it sure seems you took it that way. :worried:


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## sqnut (Dec 24, 2009)

crackinhedz said:


> as a non-partial bystander (because I dont know either of you guys)...to me it seems you are the one trying to instigate something here.
> 
> Having read through this thread trying to learn something, I think because Niick disagreed with you on 'sub delay', this may have ruffled your feathers and now youre just itching for a schoolyard fight. I didnt interpret Niicks response as being disrespectful torwards you, but it sure seems you took it that way. :worried:





Niick said:


> Of course this could just be my perception, but there seems to be a school of thought that says moving coil/cone loudspeakers being driven by amplifiers and w/ various types of electrical filtering ceases to exist in the material world of physical matter, just as soon as you put them into a mythical "car". The magical car changes the physical properties of the transmission medium know as "air" in all places outside the car, once inside the car this transmission medium takes on properties that our scientific instrumentation cannot deal with or even measure. Much like the realm of the afterlife. For because there are early arriving, often very strong reflections, this must therefore mean that all knowledge gained by all others who work with these very same electrical circuits and electromechanical acoustic transducers should be immediately dismissed as they work not within the mythical realm of the car, therefore can't possibly understand the complexity of all that changes when speakers are placed in cars.
> 
> I don't personally subscribe to this school of thought. Small room acoustics, VERY small room acoustics is how I like to think of it.


And a few others.


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## crackinhedz (May 5, 2013)

sqnut said:


> And a few others.




To me, sounded like Niick was trying to convey how difficult it is to measure a response in the car because the environment creates all types of havoc and/or anomalies. (Or was he disagreeing with this?...I dont know) ...but...

You took that personal?


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## sqnut (Dec 24, 2009)

crackinhedz said:


> To me, sounded like Niick was trying to convey how difficult it is to measure a response in the car because the environment creates all types of havoc and/or anomalies. (Or was he disagreeing with this?...I dont know) ...but...
> 
> You took that personal?


C'mon re-read what he is saying, both context and content.


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## Niick (Jun 3, 2015)

Andy, I took another measurement of a passive network I had lying around, is this the type of Xover you were referring to? It does seem to fit the description, I just don't really know how to determine what type of network I might have, beyond the slope and component count. I'm also trying to wrap my head around the whole "lobing" phenomenon. I understand that the crossover type doesn't determine IF there is lobing, but WHERE it occurs. Is this something that can be used to the advantage of someone who is going all out, like an SQ competitor for example?


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## crackinhedz (May 5, 2013)

sqnut said:


> C'mon re-read what he is saying, both context and content.


admittedly, about 95% of what has been discussed in this thread has been over my head. 

But your responding by calling him "kid" and insinuating (disrespectfully) that he simply 'googles' his answers is just a tad bit immature. 

Is he not entitled to an opinion or thought? Disagree like gentlemen and move on already.


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## cajunner (Apr 13, 2007)

Andy said car acoustics is a different discipline, Niick probably doesn't want to accept that all his pro audio tools and tricks are less effective in the confines of a space where everything is subject to rules of delay that are only effectively introduced as an artificial insertion and not a natural room acoustic.

However, the energy storage of various body panels will mimic the delay reverb of a room, so the overlap in the acoustics which is represented by impulse response, isn't totally without merit.

I think I would like to assert the effectiveness of using impulse response to detect crossover anomalies but I think we're doing great if we can just have a combined response that doesn't exhibit nulls, by and large...


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## Niick (Jun 3, 2015)

crackinhedz said:


> To me, sounded like Niick was trying to convey how difficult it is to measure a response in the car because the environment creates all types of havoc and/or anomalies. (Or was he disagreeing with this?...I dont know) ...but...
> 
> You took that personal?


No, I definitely meant no disrespect towards anyone here, what I was getting at was that there seems to me, after having worked as an installer since 1999, that there is DEFINITELY a line of thinking that cars are something almost magical, and that only a "golden ear" can extract high performance results from car audio equip. In fact, it's probably other installers whom I've known and or worked with over the years that have conveyed this point of view the most.

There have been some discussions on this forum that have swam in the waters of this disagreement, but my belief and observations of this mentality have their roots WAY before I ever joined DIYma. So there ya go. That's where that comes from.


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## Niick (Jun 3, 2015)

cajunner said:


> Andy said car acoustics is a different discipline, Niick probably doesn't want to accept that all his pro audio tools and tricks are less effective in the confines of a space where everything is subject to rules of delay that are only effectively introduced as an artificial insertion and not a natural room acoustic.
> 
> However, the energy storage of various body panels will mimic the delay reverb of a room, so the overlap in the acoustics which is represented by impulse response, isn't totally without merit.
> 
> I think I would like to assert the effectiveness of using impulse response to detect crossover anomalies but I think we're doing great if we can just have a combined response that doesn't exhibit nulls, by and large...


Less effective?! What!?


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## cajunner (Apr 13, 2007)

Niick said:


> Less effective?! What!?


just seemed like you had took a shot at Andy right after he explained that your method for indicating crossover phase fixing, was suspect or didn't work so well, (or he needed more information) in the small room, so I thought I'd get a little shot in myself..


you gotta watch out, man.

just when you think you got this internet thing figured out, somebody will send one over the fences...


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## Niick (Jun 3, 2015)

cajunner said:


> just seemed like you had took a shot at Andy right after he explained that your method for indicating crossover phase fixing, was suspect or didn't work so well, (or he needed more information) in the small room, so I thought I'd get a little shot in myself..
> 
> 
> you gotta watch out, man.
> ...


A shot at Andy ? I'm sorry, I must be missing something...I have nothing but admiration for Andy's level of knowledge and expertise.....I would never.......what??


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## Niick (Jun 3, 2015)

sqnut said:


> Did I say you were my kid? I just called you a 'Kid' not 'my kid'. See the difference? I called you a kid cause you're behaving like a petulant 5 year old.
> 
> 
> 
> ...



So I just got home from work and actually took the time to read what sqnut had posted, to which I say the following:

Google!? What the hell are you talking about, if you olny new how ridiculous u sound. I spend a ridiculous amount of time, when I'm not actually working, setting up experiments, measuring loudspeakers IN and OUT of cars, measuring the transfer function of amplifiers, processors, I've measured almost every tweeter we sell, on, 90 and at 45 degree off axis, just so I can better understand how they perform. Audio test and measurement and acoustic analysis is my passion, hobby, joy, I incorporate it into my daily work.

And you f**cking think I just "Google" the information and responses I put on here dude, you are FAR out of touch with reality buddy.


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## sqnut (Dec 24, 2009)

Niick said:


> So I just got home from work and actually took the time to read what sqnut had posted, to which I say the following:
> 
> Google!? What the hell are you talking about, if you olny new how ridiculous u sound. I spend a ridiculous amount of time, when I'm not actually working, setting up experiments, measuring loudspeakers IN and OUT of cars, measuring the transfer function of amplifiers, processors, I've measured almost every tweeter we sell, on, 90 and at 45 degree off axis, just so I can better understand how they perform. Audio test and measurement and acoustic analysis is my passion, hobby, joy, I incorporate it into my daily work.
> 
> And you f**cking think I just "Google" the information and responses I put on here dude, you are FAR out of touch with reality buddy.


...and yet you can't qualify what a out of phase drivers sound like Anyway, carry on. Oh and stop lying while you're at it. You did take an initial shot at Andy and the post I highlighted was sneeringly aimed at me. If you've said something, have the ball's to stand up to it and admit it. See why I called you a kid?:laugh2:


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## SkizeR (Apr 19, 2011)

sqnut said:


> ...and yet you can't qualify what a out of phase drivers sound like Anyway, carry on. Oh and stop lying while you're at it. You did take an initial shot at Andy and the post I highlighted was sneeringly aimed at me. If you've said something, have the ball's to stand up to it and admit it. See why I called you a kid?


Can you describe what out of phase drivers sound like?

Sent from my HTC6525LVW using Tapatalk


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## sqnut (Dec 24, 2009)

SkizeR said:


> Can you describe what out of phase drivers sound like?
> 
> Sent from my HTC6525LVW using Tapatalk


et tu Brutus? I've mentioned it in more than one thread....in detail. I don't feel like typing that much. If you want I can look for it later and PM you. Meanwhile let's see if Niick can qualify the same.


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## seafish (Aug 1, 2012)

sqnut said:


> .. Oh and stop lying while you're at it. You did take an initial shot at Andy If you've said something, have the ball's to stand up to it and admit it.


I didn't catch his sneer at AW at all, even on rereading…I think that you are making WAY too much of what Niick is saying and taking it WAY too personally. AFAICT, you are the on who started shooting bullets with little to no reason at all. 

Just my .02, not that you are gonna care.


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## SkizeR (Apr 19, 2011)

seafish said:


> I didn't catch his sneer at AW at all, even on rereading…I think that you are making WAY too much of what Niick is saying and taking it WAY too personally. AFAICT, you are the on who started shooting bullets with little to no reason at all.
> 
> Just my .02, not that you are gonna care.


X2

Sent from my HTC6525LVW using Tapatalk


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## Jepalan (Jun 27, 2013)

Babs said:


> One question I have though is, let's say my sub and mids are in phase at 100 or 120hz, can I assume they will be in phase at the crossover freq of 80hz? Or does the phase shift screw that up at 80hz?


Maybe. Maybe not. The sub and mids are only playing the same freqs within the narrow transition region around the crossover freq.

Electrically it depends on the combined phase response of the low-pass filter on your sub and the high-pass filter on your mids. Electrical phase changes as a continuous function versus frequency through the transition region and is dependent on the filter type and slope. *If* the two filters combine in a way to cancel each other's phase response out, then the combined response will mean there is no relative electrical phase shift between the signals feeding the drivers *at all frequencies that matter*.

As Andy already pointed out, one way to ensure the sub and mid is eletrically in phase through the transition band is to use 4th-order Linkwitz-Riley (LR4) filters for the LPF and HPF forming the crossover between mid and sub. These filters *will* combine electrically to be flat in gain *and* phase at all freqs through the crossover region. The linkwitz-riley topology ensures phase & gain combine to be flat, using 4th-order (24dB/oct) keeps the transition band narrow. Both are good things.
---

Now, once the crossover is flat in gain and phase *electrically*, you may still have to deal with acoustic nulls or peaks due to: 1) phase distortion in the drivers themselves, and 2) the way direct and reflected energy combine at the listener.


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## Jepalan (Jun 27, 2013)

Niick said:


> Andy, I took another measurement of a passive network I had lying around, is this the type of Xover you were referring to? It does seem to fit the description, I just don't really know how to determine what type of network I might have, beyond the slope and component count.


Niick -

Those filters do seem to be combining into nice flat phase and amplitude responses like the LR4 filters Andy was describing, but it looks like these have second-order roll-offs (12dB/octave) from the amplitude response, so they would not be LR4 per se. They *could* be LR2 filters. (i.e. 2nd Order Linkwitz-Riley).

To be 100% sure, you would have to draw out the circuit with actual component values.


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## subterFUSE (Sep 21, 2009)

If you are using TA then you're only going to get the phase locked in at a very narrow band of freq. Moving either direction away from the freq where the phase was aligned, it's going to drift out of phase.

So if you have a sub and mid that are time aligned so that 120Hz is in phase, then I feel confident you will NOT be in phase at 80Hz. It's going to be off quite a bit.

It doesn't even matter if you are using a LR24 crossover. If TA is used, that's going to cause the phase relationship to change based on frequency. It can't be avoided. That's just the nature of what TA does.

That's my understanding of it, at least. Someone stop me if I'm off-base.


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## crackinhedz (May 5, 2013)

subterFUSE said:


> So if you have a sub and mid that are time aligned so that 120Hz is in phase, then I feel confident you will NOT be in phase at 80Hz. It's going to be off quite a bit.


So if they are in phase at 80hz, would 60hz now be out of phase? Just trying to understand what the goal is for phase alignment...at/near the crossover as much as possible (between two drivers)?


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## subterFUSE (Sep 21, 2009)

crackinhedz said:


> So if they are in phase at 80hz, would 60hz now be out of phase? Just trying to understand what the goal is for phase alignment...at/near the crossover as much as possible (between two drivers)?


Correct. The crossover point where the 2 drivers' response is overlapping is where we are most concerned, because that's where our ears will be hearing them interact the most. When you move above or below the crossover point, one of the drivers is going to take precedence because it is playing much louder than the other. That's not to say phase isn't important there at all, but it's more important where the drivers are equal in loudness.


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## Niick (Jun 3, 2015)

sqnut said:


> ...and yet you can't qualify what a out of phase drivers sound like Anyway, carry on. Oh and stop lying while you're at it. You did take an initial shot at Andy and the post I highlighted was sneeringly aimed at me. If you've said something, have the ball's to stand up to it and admit it. See why I called you a kid?:laugh2:


First of all, You said, "can you describe what crossover phase shift SOUNDS like" to which I replied, NO, no ether can you, neither can any one. 

TWO DRIVERS that are out of phase, that CAN be described, IF THE REASON those drivers are out of phase IS because of a physical mis-alignment, thus ALLOWING the phase shift induced by the crossover to now become an audible phenomenon, then I guess in that regard you could say, kinda, that your hearing the "crossover" , but you're not, you're still hearing the drive units. 



sqnut said:


> yet you can't qualify what a out of phase drivers sound like


Out of phase drivers and the UNAVOIDABLE phase shift induced by crossovers are two TOTALLY DIFFERENT THINGS YOU DUMB BASTARD!!

What you are implying then, is that by using ANY crossover, this will result in audibly out of phase drivers, and that my inability to describe the sound of something that operates in the electrical, not the acoustical, domain, means that I'M the one lacking understanding here?! 

If you would have said, "can you describe the sound of two drive units, like a woofer and a tweeter, sharing a crossover, those drive units are physically aligned in such a way and their wiring polarity is such that the ELECTRICAL paste shift induced by the crossover results in ACOUSTICAL cancellation thru the region of overlap"--something like that......then ok, now you're making sense. 

As for the rest of your comments, what can I really say that won't get me kicked off of here?
im not a liar, if you new anything about me, you DEFINITELY wouldn't sat I ain't got balls, so.....


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## 14642 (May 19, 2008)

I didn't take Nick's post as a sneer at me. We've had several offline conversations about this.
The phase at the crossover is important for only one reason--so that the acoustic sum doesn't include a big null in the frequency response. We don't hear absolute phase, we hear relative phase. A crossover is designed to separate the total band of reproduced frequencies into sub bands so that the drivers for those bands can be optimized for use in that band. The region where the two drivers interact is the only place that the relative phase of the drivers matters because AC signals sum according to magnitude AND phase.

The HUGE misunderstanding EVERYWHERE in car audio is about what matters in designing a crossover. There are two things that matter most and the rest are all small considerations. One, in a passive network is impedance. We have to keep the system impedance above the minimum limit the amp can drive. Second, we listen to speakers, not to electrical filters. The electrical filters simply change the amount of power that is delivered to each of the speakers. A high pass filter (whether passive or active) reduces the power delivered to the speaker at low frequencies. A low pass filter reduces the amount of power delivered at high frequencies. The slope of the crossover determines the rate at which power is attenuated.

Crossovers do this by changing the phase of either the current or the voltage in the AC signal by a time constant. A capacitor stores energy as voltage, so the rise in voltage available at the cap is delayed compared to the current that's applied. When current flows to the cap, it takes a little while for the cap to charge to the same voltage that's applied. That lag changes the phase by 45 degrees and the resulting power over the frequencies where the capacitor "works" (low frequencies) is reduced because power is voltage times current times the cosine of the phase angle of 45 degrees.
An inductor (low pass filter) is similar but opposite. When AC voltage is applied to an inductor, the current lags by 45 degrees. Same thing, but high frequencies are reduced.
So, a capacitor "resists" changes in voltage and an inductor "resists" changes in current.

Active filters do the same thing. Digital filters don't have to be designed this way, but many are. Why? Because it works just fine.

The low frequency roll off of a speaker is a crossover that's built into the speaker. The spider and surround are a mechanical analog of a capacitor. The high frequency roll off of a speaker is also a crossover built into the speaker. That rolloff is controlled by two things-- the mass of the moving assembly (mechanical analog of an inductor) and the inductance of the voice coil at high frequencies.
So, the electrical phase of the speaker changes due to these two built-in crossovers. Anytime and anywhere the frequency response changes direction, the phase also changes direction. This applies to the acoustic output of the speaker, too. When you put a speaker in a room and measure the response, frequency response changes as a result of reflections also changes phase.
When we talk about minimum phase systems, we are talking about a system in which the phase is directly related to the frequency response. That system only exists as a single driver in a nonreflective environment. A second driver, which has a phase response of its own, combined with the first driver is NOT a minimum phase system. A single speaker in a room is not a minimum phase system because each reflection changes the measured acoustic phase independent of the response of the speaker by itself.

Despite this condition, there are regions in a non-minimum phase system that behave like a minimum phase system. What defines that is if inverting the frequency response of the electrical signal sent to the speaker by the same magnitude as the dip or peak fixes the problem. In these regions, equalization is effective. In regions that don't act like minimum phase, equalization won't fix the problem. An acoustic null is a region in which boosting with an EQ doesn't affect the problem by the same magnitude as the boost. We've all experienced that in trying to boost a deep and narrow dip in the response. When the relative ACOUSTIC phase between two drivers is 180 degrees, boosting doesn't help. However, changing the polarity of one of the drivers changes the relative phase by 180 degrees and now we don't have that dip anymore.

When we design a crossover, we try to remove power at high frequencies at the same rate as we remove power at low frequencies from the other driver. reducing the voltage by 6dB cuts power in half. If we have two speakers both playing the same frequency with both having their output level cut in half, then the resulting response is flat. Two halves make a whole.
OK...Now remember that we don't hear the electrical filters. They exist only to modify the acoustic response, which is what we hear. What we hear is result of the application of the electrical filters. It's just like EQ. We use an EQ to modify the acoustic response. This seems obvious to everyone. Crossovers are EXACTLY THE SAME. It doesn't matter what the electrical response (magnitude or phase) is EXCEPT that is should combine with the speaker to produce a measured ACOUSTIC response that matches the Butterworth, Linkwitz-Riley, 2nd order, third order, or what have you. In many cases, additional EQ is required to make the acoustic output match one of those classical alignments. In a passive network, we can change the Q of the filters to boost or cut at the knee. In an active system, we can do the same thing by choosing a different filter shape or Q. Or we can use an equalizer to tweak the response so it matches.

All of these classical alignments assume (and require) that the sound from one speaker arrives at the same time as the sound of the other speaker because the summation of the low pass and high pass acoustic responses depend on magnitude and phase. Phase at the listening or measuring position is ALSO determined by distance. So, we can use delay to align the arrival of the sound from the low pass and the high pass output of the two speakers.

When we design a home audio speaker, we design ONE speaker in an anechoic room so its response is correct. Then, we REPLICATE it for the other speaker in a stereo pair. Then, we sit in between. We are the same distance from the speakers and the speakers are the same. If we sit a little closer to one than the other, then we can delay the ENTIRE speaker to correct for the distance. The responses of the speakers and all of the correction filters that are contained within each one are still the same.

In a car, this is not the case because we use asymmetrical filters to correct the responses of the left and right channels independently. If we set the delays of all the speakers individually so that the sound arrives at our ear at the same time, we do two things. We align the arrivals so we can design crossovers according to the classical alignments AND we correct for the offset listener.
In order to hear a correct image of a stage, the left and right acoustic signal has to arrive at our ears at the same time, in phase and precisely matched in level at all frequencies. So, let's say that we delay the right mid in a three way to fix the crossover, what have we done to the relative phase between that mid and the one on the other side that has to be in phase with the first one in order to provide a correct image? We've changed it. Now, the image will shift over those frequencies toward the speaker from which the sound arrives first.
This is why using delay to fix phase at the crossover is fine for home audio speakers, but isn't OK for car audio systems that use delay to optimize for a single offset listener. Get the crossover right using EQ and set the delays based on measured distance. It's the only correct way.

In systems (like those with upmixers and center speakers), the use of delay to create an image is less important because sound is steered to a real center speaker and we don't rely on the left and right being precisely matched in frequency and phase to create that center image.

So, when you look at the phase curve in the analyzer, try adjusting the EQ or the crossover to achieve flat frequency response and flat phase and leave the delay setting alone.

The reason that i recommend 4th order slopes in cars is because the rate of attenuation is so steep that the acoustic output of the speaker is much more likely to track the electrical response. And the electrical response of a 4th order LR filter sums flat in phase and magnitude. This minimizes the need for EQ. It's much more predictable and much more likely to just work without a bunch of futzing around.


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## Niick (Jun 3, 2015)

Jepalan said:


> Niick -
> 
> Those filters do seem to be combining into nice flat phase and amplitude responses like the LR4 filters Andy was describing, but it looks like these have second-order roll-offs (12dB/octave) from the amplitude response, so they would not be LR4 per se. They *could* be LR2 filters. (i.e. 2nd Order Linkwitz-Riley).
> 
> To be 100% sure, you would have to draw out the circuit with actual component values.


Gotcha, ok....thanks for checking that out. For years I've been reading about how crossovers, or any electrical filter for that matter, changes the phase response of a circuit, but until I've been able to directly measure these things, I've never REALLY understood how it works. Now it's becoming clear to me. 

I suppose that makes sense that it's a 12dB network, huh, cause it was a passive unit, 24dB passive would be huge and expensive. It was just from an old component set. 

I once measured the transfer function of a rockford 360.3, but the computer I did it on crashed, and I lost all the saved traces, I had every different type of filter order and type the processor offered, interesting to say the least. 

One last thing, I discovered a method of setting dealy times using a polarity pulse, an oscilloscope with external trigger, and a freq. response measurement system, and a broken RCA cable! Seriously! I took a video of how it's done to show my buddy mike on the way to the training in Washington last Sunday. It's really cool. It's a short couple, like 2 min. Videos in my iPad, I just don't know how to post something like that. A video I mean. I'm not real computer saavy. I could email them to someone if anyone might be interested in seeing it. I'm serious, it's pretty f**king cool.


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## subterFUSE (Sep 21, 2009)

As I understand it, even a point source coaxial would require delay on the tweeter to compensate for the crossover-induced delay on the midrange.

For example, in an LR24 crossover the lower range driver ends up delayed by 360 degrees of phase rotation. The drivers will be "in-phase" in the sense that the peaks of the waves are aligned, but the bass driver will be starting a full cycle late compared to the tweeter.

So either the tweeter needs to be moved physically further away, or it needs to be delayed so the wavefronts of both drivers are aligned in both time and phase.


This is why the high frequency drivers in a Synergy horn are in the back of the horn, and the bass drivers closer to the front. The drivers are placed according to the wavelegths at the crossover frequency, with the high frequency drivers being moved further away from the listener to compensate for the bass drivers delay.


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## 14642 (May 19, 2008)

Niick said:


> Gotcha, ok....thanks for checking that out. For years I've been reading about how crossovers, or any electrical filter for that matter, changes the phase response of a circuit, but until I've been able to directly measure these things, I've never REALLY understood how it works. Now it's becoming clear to me.
> 
> I suppose that makes sense that it's a 12dB network, huh, cause it was a passive unit, 24dB passive would be huge and expensive. It was just from an old component set.
> 
> ...


That's a pretty good idea. It's the external trigger that's important for a timing reference.


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## 14642 (May 19, 2008)

subterFUSE said:


> As I understand it, even a point source coaxial would require delay on the tweeter to compensate for the crossover-induced delay on the midrange.
> 
> For example, in an LR24 crossover the lower range driver ends up delayed by 360 degrees of phase rotation. The drivers will be "in-phase" in the sense that the peaks of the waves are aligned, but the bass driver will be starting a full cycle late compared to the tweeter.
> 
> ...


Once again...home audio speaker. 

For a car, you could delay the entire system except the sub relative to the sub. If you do this differently on the left vs. the right in a car you'll completely FU the image.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> Crossovers do this by changing the phase of either the current or the voltage in the AC signal by a time constant. A capacitor stores energy as voltage, so the rise in voltage available at the cap is delayed compared to the current that's applied. When current flows to the cap, it takes a little while for the cap to charge to the same voltage that's applied. That lag changes the phase by 45 degrees and the resulting power over the frequencies where the capacitor "works" (low frequencies) is reduced because power is voltage times current times the cosine of the phase angle of 45 degrees.
> An inductor (low pass filter) is similar but opposite. When AC voltage is applied to an inductor, the current lags by 45 degrees. Same thing, but high frequencies are reduced.
> 
> So, a capacitor "resists" changes in voltage and an inductor "resists" changes in current.
> ...


Thank you very much Andy for all your help and knowledge that you share. I remember a lot of what you're talking about from when I studied Electronics and telecommunications at Tulsa Technolgy Center back in 1996-1999. What you are describing is capacitive and inductive reactance right. So, I set up an experiment to measure the phase relationship between the current and the voltage of a woofer, to see if the measured current an voltage phase relationship was the same as that plotted by REW when I ran an impedance sweep, an sure enough, it was. 

For anyone out ther who may want a current probe for their oscilloscope, but can't justify the usually high price of such things, there is a company called QuantAsylum that make a low cost differential probe/current probe in one. It's less tha 100 dollars, and works great. You can measure current an/ or voltage with it, on single-ended input oscilloscopes, non-ground referenced voltage (differential) measurements! It's pretty sweet.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> That's a pretty good idea. It's the external trigger that's important for a timing reference.


Yeah! In fact, I'll email you the videos. Check em out, tell me what you think.


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## Niick (Jun 3, 2015)

So I figure that this can be used for pretty much all alignment work above 200 Hz or so for sure, haven't tried it with subs yet. But I have tried it in car and it works perfectly. In fact, because of the triggered scope method it has a natural noise immunity, even when the compressor kicked on it doesn't matter, doesn't mess up the measurement at all. You can capture these pulses even when you can barely hear them.


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## Babs (Jul 6, 2007)

Niick said:


> One last thing, I discovered a method of setting dealy times using a polarity pulse, an oscilloscope with external trigger, and a freq. response measurement system, and a broken RCA cable! Seriously! I took a video of how it's done to show my buddy mike on the way to the training in Washington last Sunday. It's really cool. It's a short couple, like 2 min. Videos in my iPad, I just don't know how to post something like that. A video I mean. I'm not real computer saavy. I could email them to someone if anyone might be interested in seeing it. I'm serious, it's pretty f**king cool.


Upload that rascal to youtube. Log in w/ your google acct and go from there. That would be interesting. Can upload it directly from your iPad.


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## Niick (Jun 3, 2015)

Babs said:


> Upload that rascal to youtube. Log in w/ your google acct and go from there. That would be interesting. Can upload it directly from your iPad.


Ok, I've never done that before, but I'll try. Let me see what Andy thinks first. I wouldn't want to give away something that I might regret


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## Babs (Jul 6, 2007)

subterFUSE said:


> As I understand it, even a point source coaxial would require delay on the tweeter to compensate for the crossover-induced delay on the midrange.


Yeah makes me wish sometimes I either had money to burn or hadn't shot the wod on other toys. There's a certain AF GS42 coax speaker Andy knows a little something about, I'd love to get my grubby paws on a pair to try something interesting in the A-pillars and test with different axis angles etc.  Alas.. Kids tuition just kicked in. :bigcry:


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## Niick (Jun 3, 2015)

Babs said:


> Upload that rascal to youtube. Log in w/ your google acct and go from there. That would be interesting. Can upload it directly from your iPad.


Ok, so I selected " anyone with a link can view" 

So how do I now provide you with a link?


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## Niick (Jun 3, 2015)

I recently had the chance to check out some of the Audiofrog tweets and mids! Sweet! Awesome mounting schemes too, I mean wayyy cool.


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## seafish (Aug 1, 2012)

Niick said:


> Ok, so I selected " anyone with a link can view"
> 
> So how do I now provide you with a link?




sqnut…I knew it, you see, this PROVES he's NOT a "kid"…ANY kid would know just how to do all this 'puter interwebz stuff by himself..LOL

To Niick, thanks for being curious and coming up with interesting and creative ideas on how to tune a SQ system in a car…too bad much of it seems beyond me…I'm just gonna have to "use my ears"…lol


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## Niick (Jun 3, 2015)

Here's a mid, grill setup is very flexible, a lot of different mounting options


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## Niick (Jun 3, 2015)

Look, it's me and my actual "kid", Sophia!
Hey, we're upside down!!


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## Babs (Jul 6, 2007)

Niick said:


> Ok, so I selected " anyone with a link can view"
> 
> So how do I now provide you with a link?


From iPad, youtube app.. pull up the vid. Tap on the vid.. See the share icon top right between + and 3 vertical dots... Tap that.. Copy Link.


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## Niick (Jun 3, 2015)

Babs said:


> From iPad, youtube app.. pull up the vid. Tap on the vid.. See the share icon top right between + and 3 vertical dots... Tap that.. Copy Link.


K, gimme a sec, let me try that


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## Niick (Jun 3, 2015)

Babs said:


> From iPad, youtube app.. pull up the vid. Tap on the vid.. See the share icon top right between + and 3 vertical dots... Tap that.. Copy Link.


Ok, so I open the YouTube app on my iPad, now how do I find the video that I just uploaded? ****, I don't know, maybe I'll just go back, re post them, and click on "anyone can view."


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## sqnut (Dec 24, 2009)

subterFUSE said:


> As I understand it, even a point source coaxial would require delay on the tweeter to compensate for the crossover-induced delay on the midrange.


Yes I agree. 



Andy Wehmeyer said:


> Once again...home audio speaker.
> 
> For a car, you could delay the entire system except the sub relative to the sub. If you do this differently on the left vs. the right in a car you'll completely FU the image.


In my 2 way + sub, I have my mids and sub timed for same arrival as also left right woofer. The left and right tweets are timed for same arrival but with the xover at 2.5 on 4th order, the tweets are delayed ~ 5" or 0.4 ms from the sub and mids. There is a fair bit of eq work to do after this to get it sounding right. The sound is more open, more like a 2ch than a car.

I ran all drivers for same arrival for many years and actually stumbled on this alignment while fooling around. I have had it about two years this way and its one of those game changers that lets you tune the sound to a new level.


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## Niick (Jun 3, 2015)

I don't know how to do this ****, doesn't seem to be working, I don't know. To tell ya the truth, I f**king hate computers, the ONLY reason I own one is for work, as a tool for audio use. And for that I actually have 2. 

Ok, at the bottom of the screen, when I tried to upload the videos, was a progress bar, it said like 10 sec, then 2,1, then, when the progress bar reached 100%, it just said, "uploading to YouTube", with a blue x to the right of it, I waited and waited and the screen just didn't change, so I finally just clicked the blue x, was I not supposed to do that? Was I supposed to wait for the screen to change from the greyed out screen back to the normal screen on its own?


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## Niick (Jun 3, 2015)

Ok, so I realize that this isn't a "how to upload to YouTube" forum, but if I can get this to wrk, then I think it will be worth it, here is a screenshot of what it says after I press "publish"

It's just stuck on that screen, waited like 10+ min.?


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## Niick (Jun 3, 2015)

Ok, so I figured it out, I think! Here is the link, it's public, it still says "we're processing this video, check back later" but you can try it

http://youtu.be/QVSRPaVQeiA


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## crazhorse (Mar 9, 2010)

Pretty nifty....


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## Niick (Jun 3, 2015)

As soon as I get a chance I'll document this method being applied in car and with as full a range of drivers as possible. We'll see how it works. I've already tested it in car, and it works perfectly as far as being able to see the peaks among the reflections, I think it has to do with the external trigger being used on the scope.


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## Niick (Jun 3, 2015)

when setting time alignment, let's say you have a fully active system, and you're using the IR of each individual drive unit to set your delays, do you first turn off all xovers for the purpose of time alignment, or do you do this process with xovers engaged? 

Discuss....


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## subterFUSE (Sep 21, 2009)

Niick said:


> when setting time alignment, let's say you have a fully active system, and you're using the IR of each individual drive unit to set your delays, do you first turn off all xovers for the purpose of time alignment, or do you do this process with xovers engaged?
> 
> Discuss....


Crossovers should be set before attempting time alignment because the crossovers will introduce delay.


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## Niick (Jun 3, 2015)

subterFUSE said:


> Crossovers should be set before attempting time alignment because the crossovers will introduce delay.


Now see, I would think so too, in fact, I still do set xovers first, however, the method taught by Scott Welch was to DISABLE all xovers first, he also said though that the IR method was good for 200 Hz and above, which I agree. Just thought it might be interesting to see how others were doing it. There is DEFINITELY "more than one way to skin a cat" as they say.


http://youtu.be/S72DdeJ7leo


Purely because I thought someone might find it interesting, I posted on you tube another old video I had of a measurement I was taking of a co-workers car, super simple system, deck power to coax speakers, amp on sub in trunk. This measurement shows how the phase discontinuities through the crossover region of the sub and midbass transition result in not only a dip in the freq. response, but a measured drop in coherence. Coherence is the trace at the top of the upper graph, that you can see dropping thru the 100Hz range. I took this video when I was first learning the software, so it's really not good for much anymore other than as an example of coherence problems thru the crossover region of sub to midbass.


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## Jepalan (Jun 27, 2013)

Andy Wehmeyer said:


> <I trimmed the quote for sanity, see original post for entire quote>
> 
> This is why using delay to fix phase at the crossover is fine for home audio speakers, *but isn't OK for car audio systems* that use delay to optimize for a single offset listener. Get the crossover right using EQ and set the delays based on measured distance. It's the only correct way.
> 
> So, when you look at the phase curve in the analyzer, try adjusting the EQ or the crossover to achieve flat frequency response and flat phase and leave the delay setting alone.


Andy - thanks for posting this. It makes complete sense and I agree 100% with everything you said, but I do have one clarifying question.

As you stated, it is obvious to me that attempting to use delay to correct phase in the crossover region of the stereo drivers (midbass, mid, tweeter) is a bad idea any way you slice it.

However, it does seem like it would still be OK to use delay to address issues with the subwoofer only - yes? In this case one would *only* adjust delay to the sub (or to all other drivers in equal amounts) after having set time-alignment and EQ correctly for the rest of the system.


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## 14642 (May 19, 2008)

Niick said:


> Now see, I would think so too, in fact, I still do set xovers first, however, the method taught by Scott Welch was to DISABLE all xovers first, he also said though that the IR method was good for 200 Hz and above, which I agree. Just thought it might be interesting to see how others were doing it. There is DEFINITELY "more than one way to skin a cat" as they say.
> 
> 
> http://youtu.be/S72DdeJ7leo
> ...


The reason to turn off the crossovers is to that the impulse response will have a peak rather than a hump. The peak, caused by high frequency content is a CLOSER representation of the initial arrival than the hump. 

Even for a signal with lots of high frequency content, the peak is NOT the arrival. The arrival is the point on the impulse at which the signal BEGINS to rise. That point is difficult to find because of pre-ringing.


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## 14642 (May 19, 2008)

BELOW is the magnitude and phase of a perfect LR4 crossover. You can see that the phase sum is zero. If the ACOUSTIC response of your speakers at the crossover matches this closely and the delays are set for DISTANCE (like with a tape measure), the response will be flat and so will the phase. Now, let's look a little more closely at the "delay" in the signal. Since it's a phase shift and not a delay, we can calculate the delay DUE TO THE PHASE SHIFT for any frequency we want. These phase curves are asymptotes. They NEVER reach the axes, but they get damn close. At 1k (the crossover point), the high pass is +180 and the low pass is -180. So, 1132 feet (speed of sound) divided by 1000 and then divided by two (180 degrees is half the sine wave) tells us that at 1k, the HP is ahead by .566 feet and the low pass is behind by .566 feet. 

Of course, this is impossible as delay because it violates causality. The high pass signal cannot arrive before it arrives and this is why phase is not delay. 

Now, the "delay"--let's call it offset from zero (to keep me sane) at 2k, by the same formula is .14 feet. Now, if we look at the low pass side, we'll see that at every frequency, the delay is .566 feet. This is represented in my crappy drawing above (sorry about the placement of the diagrams--I'm out of patience trying to rearrange them) of what the group delay curve would look like for this speaker (sum of LP and HP)

Ah HA! see, we should delay the high pass by .566 feet to account for group delay! 

Wrong. If we do that, then we screw up the sum of the phase of the high pass side and the low pass side. where they combine in the crossover region. This is precisely why group delay doesn't really matter much, so long as the value isn't HUGE. Group delay is ABSOLUTE phase. So long as the low passed speaker is the only speaker playing the band of frequencies that are "delayed" we don't hear it.


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## 14642 (May 19, 2008)

Just to clarify things a bit...I don't have a low passed impulse response measurement handy, so I've drawn one on this same graph. The top of the hump is NOT the arrival. Removing the low pass filter will add some high frequency to create a sharper peak. This is why Scott is proposing turning off the filters. Then, it will look more like the other measurement (which is a midrange driver with a low pass filter at 1k). It'll be easier to pick the peak, which is still an approximation.


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## 14642 (May 19, 2008)

This is another way to do this and this is how MS-8 determines the arrival time. It works no matter the frequency content of the signal.


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## Niick (Jun 3, 2015)

Andy, I'm still not fully grasping what you meant when you've said in the past to use a LR4 and adjust EQ to match target, by target, did you mean to use EQ to match the classic alignment of what a LR4 SHOULD look like acoustically, IF your drivers are not exactly following the classic LR4 alignment ACOUSTICALLY. (Like, meaning, make the acoustic response of each individual driver mimic the response of an LR4 filter measured in the electrical domain)

Or by, "EQ to match target" did you just simply mean, EQ to match whatever target curve you are trying to achieve, AS A WHOLE. maybe I was reading too much into this? No? Yes? I don't think so, but.......hmmmmmm..........I would think you meant the first one, EQ to match the target FILTER response.

Also, it's dawned on me lately that, since I've had to learn ABSOLUTELY EVERYTHING (until last Sunday, anyways) about system tuning and acoustic measurements and doing it in a manner that is reliably repeatable and timely, completely on my own, as there is no one else who I currently now, or have ever worked with that seems to think these things are important, yet they wonder why......well you know, we've discussed this.

Anyways, it's came to my realization that the phenomenon I was experiencing of the amplitude of the summed response of two drivers sharing a crossover having a big notch at crossover, when installed just "wherever" , like factory locations, passive network, whatever.....that WAS THE RESULT OF THE DRIVERS NOT BEING IN CORRECT PHYSICAL (THEREFORE TIME) ALIGNMENT!! Duh!! Ok, so NOW I really get it. When the "classic" crossover alignments are used in a design, the PREDICTED response of the system is supposing that the drivers are equidistant from the listener, correct? 

So that's why when I would change the distance, or delay, while simultaneously measuring phase.....(my head is still warm from the light bulb going off!!!)

See, what I was thinking was a problem with the ACOUSTIC CROSSOVER REGION, was really a problem, yes, but it was there because the drivers weren't in the classic alignment positions of being equidistant in time, correct? 

So by using the phase trace to determine distance, rather than physical measurement or IR, I was POTENTIALLY adding too much, or too little, wait, damn! The lightbulb just shattered! 

I'm gonna have to run some more experiments when I get back to work ?
But thank you again for all the wonderful knowledge! 
I feel almost daily that I get a better and better understanding.......can't wait for tomorrow!


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## Niick (Jun 3, 2015)

Ok, I get it.......

By the way, Scott Welch's SRT 8............HOLY **** THAT'S A BAD ASS SYSTEM,!!!

Edit: Actually, I heard 3 cars while in Washington, ther was Scott's SRT8, a Nissan Maxima owned by a dude, I think his name was Bill, I hope I got that right, and Drew with a Ford Thunderbird, they were all quite awesome systems in their own right. Especially the Nissan, he had it set up where selecting a different preset would give you either a 2 way front stage or a 3'way front stage, pretty sweet. Sounded awesome both ways, the 2 way setup is what won the SQ comp he entered though. I think he won, that or he placed really high.


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## 14642 (May 19, 2008)

Niick said:


> Andy, I'm still not fully grasping what you meant when you've said in the past to use a LR4 and adjust EQ to match target, by target, did you mean to use EQ to match the classic alignment of what a LR4 SHOULD look like acoustically, IF your drivers are not exactly following the classic LR4 alignment ACOUSTICALLY. (Like, meaning, make the acoustic response of each individual driver mimic the response of an LR4 filter measured in the electrical domain)
> 
> Or by, "EQ to match target" did you just simply mean, EQ to match whatever target curve you are trying to achieve, AS A WHOLE. maybe I was reading too much into this? No? Yes? I don't think so, but.......hmmmmmm..........I would think you meant the first one, EQ to match the target FILTER response.
> 
> ...


You may want to read upwards again. I made a few changes for clarity.

You were ABSOLUTELY on the right track until the lightbulb shattered.

When I say "match the target" I mean that the responses that you measure from the tweeter and the midrange, when you measure them separately with the mic, should match the target crossover HP and LP shapes for the alignment you choose (I suggest LR4 as in the diagram above). Then, if you set the delays for distance, the crossover will sum correctly in phase and magnitude. 

Go glue the lightbulb back together. You just got it.


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## Niick (Jun 3, 2015)

Andy Wehmeyer said:


> You may want to read upwards again. I made a few changes for clarity.
> 
> You were ABSOLUTELY on the right track until the lightbulb shattered.
> 
> ...


   yep, uh huh, I get it. Locked and loaded. Thank you sir, one more weapon in my arsenal of knowledge that will help me to conquer the plague that is bad sound in the Portland area! Hahaha

Today Portland.......tomorrow the world!!


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> You may want to read upwards again. I made a few changes for clarity.
> 
> You were ABSOLUTELY on the right track until the lightbulb shattered.
> 
> ...


and this, I believe, is where the MS-8 had some brilliance... Once you adjust the level of the subwoofer you are off of your target. My understanding is the MS-8 essentially corrected for this as you adjusted the output level.

Additionally, Hanatsu has a very informative thread on setting up your crossover points in this same manner described above, detailing the use of REW for this purpose. Quite a good read. That said, I'm still not sure how I feel about altering the polar response of a 'speaker' simply due to it's averaged measured response; how does that affect everything else going on... how does alter the imaging/staging.


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## jpsandberg (Jun 12, 2008)

Jepalan said:


> Maybe. Maybe not. The sub and mids are only playing the same freqs within the narrow transition region around the crossover freq.
> 
> Electrically it depends on the combined phase response of the low-pass filter on your sub and the high-pass filter on your mids. Electrical phase changes as a continuous function versus frequency through the transition region and is dependent on the filter type and slope. *If* the two filters combine in a way to cancel each other's phase response out, then the combined response will mean there is no relative electrical phase shift between the signals feeding the drivers *at all frequencies that matter*.
> 
> ...


This is a very educational discussion! (and somewhat 'entertaining at parts lol)

I don't want to get too far off subject, but I'm considering trying the 4th-order (24db/oct) xovers on my new setup. But I'm not sure if I should set the xover point for my mid bass (SLS 6) and my widebanders (TB bamboo 3") at the same point being that they are pretty steep slopes, or if I should have a gap between the low pass and highpass. I was assuming @350 for a starting point, so could anyone suggest how I should set each?

Thanks in advance!


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## imjustjason (Jun 26, 2006)

cajunner said:


> so I thought I'd get a little shot in myself..


You realize that trolling for the sake of trolling is frowned upon and can lead to banning? I believe you may have experienced some time off for such a thing in the past.


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## cajunner (Apr 13, 2007)

imjustjason said:


> You realize that trolling for the sake of trolling is frowned upon and can lead to banning? I believe you may have experienced some time off for such a thing in the past.


I wouldn't consider my response as trolling.

I was attempting humor, which I suspect may have gone over some heads. My limited understanding of the concepts presented, led me to believe that Niick, was throwing around various measurement ideas using his audio tool box to address crossover phase shift, and one method was placed in doubt by Andy. This approach of presenting a technique that uses specialized software, to me, was interesting but conflicted with what I understood to be more traditional methods to address the problem.


When I posited that Niick's method was less effective, and qualified it by my explanation, I was unable to present the reasons why very well, and Andy came in and did a clinic. Just saying the car audio environment is subject to different criteria, is non-specific and obviously more clarity was needed.


I take note that I am being scrutinized for content, and won't attempt humor again. 

There was a facetious and sarcastic undertone to Niick's post about the mysterious phase of cars, and I did possibly become infected by that capricious comment since Andy had basically stated how cars are different, and saw Niick's response as a measure of incredulity at his words.


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## Niick (Jun 3, 2015)

cajunner said:


> I wouldn't consider my response as trolling.
> 
> I was attempting humor, which I suspect may have gone over some heads. My limited understanding of the concepts presented, led me to believe that Niick, was throwing around various measurement ideas using his audio tool box to address crossover phase shift, and one method was placed in doubt by Andy. This approach of presenting a technique that uses specialized software, to me, was interesting but conflicted with what I understood to be more traditional methods to address the problem.
> 
> ...


Cajunner, I can see how my description of the automotive interior as a mystical acoustic place where only those with magic ears could successfully optimize a stereo music playback system COULD have been read as a dig at Andy. 

I assure you it wasn't. If anything, I was agreeing with Andy when he's EXPLICITLY stated in the past that he firmly believes the proper use of scientific measurement instruments for acoustic analysis is the only logical way to go about trying to reign in all the complexities and variables that come with tuning modern stereo systems, which I think applies to just about any system, in car or not. Andy has stated more than once that to try to tune a car solely by ear is nuts.

I was saying the same thing, ALTHOUGH I believe that there are a VERY different set of criteria and "rules" or "standards" to adhere to as far as methodology is concerned, depending on if you're tuning your own system for the pure enjoyment of it, and for your own listening pleasure vs. charging clients in a professional capacity for a system optimization service. 

If anything, the sarcasm contained in my statements was aimed at those who insist that the only way to tune is by ear, and that even professionals and top place scoring competitors tune solely by ear, because the complexities of the acoustic interior of a car can't possibly be quantified with measurements, and MOST people over the YEARS that I've heard this point of view from, come not from this forum, but from the industry in general. Customers, installers, shop owners.........there may be some here in this forum who share this outlook, and if they're tuning their own personal systems, cool, there's nothing wrong with that. It's when they start to charge the general public for a service that the results WILL depend on their mindset, wether or not they're hungry or tired or mad or hot or cold or whatever! 

That's really what I was sayin.


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## cajunner (Apr 13, 2007)

Niick said:


> Cajunner, I can see how my description of the automotive interior as a mystical acoustic place where only those with magic ears could successfully optimize a stereo music playback system COULD have been read as a dig at Andy.
> 
> I assure you it wasn't. If anything, I was agreeing with Andy when he's EXPLICITLY stated in the past that he firmly believes the proper use of scientific measurement instruments for acoustic analysis is the only logical way to go about trying to reign in all the complexities and variables that come with tuning modern stereo systems, which I think applies to just about any system, in car or not. Andy has stated more than once that to try to tune a car solely by ear is nuts.
> 
> ...


then let me offer an apology as it was my mistake in comprehension.

since I've gone on and on about how Andy has 22 channels of super software available for tuning in his own car, (from statements he made here) and probably access to all the best room-equalization algorithms since before his stint as MS-8 proprietor and maybe even before his hiring at Harman, and if not, then surely during his tenure at Harman, I would suspect that if anyone was capable of judging whether or not the car interior was mappable, he'd be in the short list.

I also remember Andy and lycan going about with virtual drivers and phase, and how both agreed that there were possibly, and probably hundreds of these competing sources of sound before being captured by the ear drum, and that trying to account for each was simply too large a proposition to be done reliably and with precision, and at the time, possibly with limitations in the speed of computerized modeling.

so, as the science progresses and the law governing things like terabytes moves in a forward fashion, some of these long held beliefs about what is and what isn't possible have to be re-examined for thoroughness, or validity, or something else, something thesaurus-ey...

anyways, I'm glad it only appeared that you had taken exception to having your methods questioned, and that you were simply misunderstood by a couple of us "touchy" types who made it their business.

regardless, I didn't expect to have my comments seen as trolling, since my position was simply in agreement with Andy in that I don't fully understand phase correction using your techniques, yet. And I am naturally skeptical of easily produced "new theory" that can be done simply, and hasn't been acknowledged as yet.

probably because something new, strikes as unfamiliar, and what we don't see every day, we suspect...


or is it, We covet what we see.. Hannibal-style..

oh, I'm sorry, my humor muse is tapping out, continue...


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## Niick (Jun 3, 2015)

Please, no need for apologies, I didn't see it that way either.


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## sqnut (Dec 24, 2009)

Niick said:


> If anything, the sarcasm contained in my statements was aimed at those who insist that the only way to tune is by ear, and that even professionals and top place scoring competitors tune solely by ear,


Who said you should tune ONLY by ear? Certainly no one in this thread. You MUST measure to a certain point AFTER that it's tuning by ear.


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## sqnut (Dec 24, 2009)

sqnut said:


> the tweets are delayed ~ 5" or 0.4 ms from the sub and mids. There is a fair bit of eq work to do after this to get it sounding right. The sound is more open, more like a 2ch than a car.
> 
> I ran all drivers for same arrival for many years and actually stumbled on this alignment while fooling around. I have had it about two years this way and its one of those game changers that lets you tune the sound to a new level.


Can't leave wrong information on here. The above is factually wrong.

Thank you Andy for your detailed post and it is your post that prompted me to go back in and recheck. My tweets aren't delayed by 5" or 0.4ms. I checked the distances entered in the bit 10 and then broke out the measuring tape. The input distances are the measured distances. The bit 10 allows you to set distance and then fine tweak it in +/- 0.02 ms. All the drivers were at measured distance and +/- 0.04 ms. 

At some point of tuning by ear I must have knocked the tweets out of kilter, then had this brainwave and added the delay, thereby bringing tweets and mids back to same arrival times and the wrong notion got stuck in my head. 

Just for the heck of it, I delayed the tweets by 0.2 ms and ran like that for a couple of days, to see what happens. Visually the stage splits horizontally and the upper part which are the highs get pushed further away from you and the stage has a weird disjointed feel. It's not extra depth cause only the highs have been pushed back.

Tonally, the sound is duller. It seems like there is more bass but its really dull. You loose a ton of dynanics and vocals sound stretched. You cant correct any of this with the eq. The extra delay on the tweets is messing up the natural delay between the fundamentals and harmonics which is on the recording. We hear that as tonal imperfections and no amt of eq can fix it. 

TLDR: Ignore what I said earlier you definitely want same arrival times.


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## drop1 (Jul 26, 2015)

Cool thread. I was thinking about time alignment the other night. 

Basically was trying to figure out if it would be possible to process time alignment with multiple adjustments per channel. My thoughts are to be able to shift multiple groups of frequencies. Say 4 different alignment per channel. So bass goupings, mid bass , mid and high. 

What is tricky is figuring out whether this is possible mechanically for a speaker without having it cancel it's self out. Fun to think about .


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## Niick (Jun 3, 2015)

drop1 said:


> Cool thread. I was thinking about time alignment the other night.
> 
> Basically was trying to figure out if it would be possible to process time alignment with multiple adjustments per channel. My thoughts are to be able to shift multiple groups of frequencies. Say 4 different alignment per channel. So bass goupings, mid bass , mid and high.
> 
> What is tricky is figuring out whether this is possible mechanically for a speaker without having it cancel it's self out. Fun to think about .


I think what you're talking about is a form of "group delay" but purposefully induced and controlled. I think, but I'm not sure, but I think I once remember Andy talking about this type of thing on a product from the past. One that he had designed? Maybe?

I also think I once saw a product from alpine that claimed to do this exact thing, how well it actually worked though, was another thing. Obviously if it was for real, it didn't make to big of a splash.


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## drop1 (Jul 26, 2015)

Niick said:


> I think what you're talking about is a form of "group delay" but purposefully induced and controlled. I think, but I'm not sure, but I think I once remember Andy talking about this type of thing on a product from the past. One that he had designed? Maybe?
> 
> I also think I once saw a product from alpine that claimed to do this exact thing, how well it actually worked though, was another thing. Obviously if it was for real, it didn't make to big of a splash.


I produce a little electronic music in my spare time and have several toys in my toy bix that can do things lije this and it seems like it would be hard to get right. Even if you nailed it at the drivers head it would likely cause a flange like super fast delay on each channel for everyone else and would be a pain to keep your head steady enough to enjoy it. Really cool in theory but I have a feeling real world application would not yeild the desired results.


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## Niick (Jun 3, 2015)

Yeah, it's hard to imagine a worth while balance between actual performance gains and developement implementation effort/costs.


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## 14642 (May 19, 2008)

drop1 said:


> Cool thread. I was thinking about time alignment the other night.
> 
> Basically was trying to figure out if it would be possible to process time alignment with multiple adjustments per channel. My thoughts are to be able to shift multiple groups of frequencies. Say 4 different alignment per channel. So bass goupings, mid bass , mid and high.
> 
> What is tricky is figuring out whether this is possible mechanically for a speaker without having it cancel it's self out. Fun to think about .


Yes, it's possible. Can be done with phase EQ or by breaking the band of frequencies up into sub bands in a DSP with a crossover, delaying them and then summing the outputs of the crossovers. It's not simple.

At my last job, one of the DSP guys developed a phase EQ like this at the request of one of the car companies. We listened to it in several cars and there was no audible difference. Simple delay and EQ performed just as well.


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## sqnut (Dec 24, 2009)

Andy Wehmeyer said:


> At my last job, one of the DSP guys developed a phase EQ like this at the request of one of the car companies. We listened to it in several cars and there was no audible difference. Simple delay and EQ performed just as well.


Is this because the audibility of phase is restricted to the response and or the time domain?


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## Niick (Jun 3, 2015)

Got another crossover/EQ question here, as Andy has mentioned in the past, I'm EQing to match target, "target" being the electrical crossover slope, so I'm trying to get my acoustical slope to match the electrical. 

My question is, how far down in dB do you go with this? Is 30 dB below reference (the top of the curve) far enough? 

I'm attaching a pic of my progress so far on the right tweeter. I've been able to use EQ to match the target down to about 30 dB below freference. I'm wondering how far down I should chase it?


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## subterFUSE (Sep 21, 2009)

Niick said:


> Got another crossover/EQ question here, as Andy has mentioned in the past, I'm EQing to match target, "target" being the electrical crossover slope, so I'm trying to get my acoustical slope to match the electrical.
> 
> My question is, how far down in dB do you go with this? Is 30 dB below reference (the top of the curve) far enough?
> 
> I'm attaching a pic of my progress so far on the right tweeter. I've been able to use EQ to match the target down to about 30 dB below freference. I'm wondering how far down I should chase it?


I honestly think you have matched it well enough, from the look of the chart.

By the time the signal is 20+ dB down, the other driver is going to be overpowering that tweeter so much that I don't think you're going to hear that hump.

That's just my estimation... and how I have been operating in my tunes so far.


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## subterFUSE (Sep 21, 2009)

Here is a somewhat related question:



How much phase angle rotation does .01ms of delay cause at 800Hz?

(obviously I understand phase angle rotation will depend on frequency, so let's just use 800Hz as an example)


I'm curious to know how much +/- delay should be maximally necessary once we have determined the correct time alignment based on the wavefront arrival. 



Edit: Nevermind, I found the answer. 2.8 degrees per .01ms @ 800Hz.

So the answer is, a lot of delay could be maximally necessary.





So.... to Andy W.... if my DSP offers both time delay and phase angle adjust via 2nd order All Pass filters, does that mean I should align the drivers in time by the wavefront arrival, and then use the all pass filters to adjust the phase angle?


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## Niick (Jun 3, 2015)

Looks like about 2.86 degrees.....wouldn't ya say?

Edit: err......make that maybe about 2.85...not much at all, basically none to be concerned with I wouldn't think

Edit #2: I actually measured a y adapter, and entered .01ms into the reference delay, thus giving me 2.86degrees, but I posted the answer just after you edited your answer, I was trying to help!! Haha


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## subterFUSE (Sep 21, 2009)

Niick said:


> Looks like about 2.86 degrees.....wouldn't ya say?


Yeah. I Googled after asking the question here. Derp...


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## Niick (Jun 3, 2015)

subterFUSE said:


> I honestly think you have matched it well enough, from the look of the chart.
> 
> By the time the signal is 20+ dB down, the other driver is going to be overpowering that tweeter so much that I don't think you're going to hear that hump.
> 
> That's just my estimation... and how I have been operating in my tunes so far.


Thank you for your input on this matter, I think you're right. This by the way is the vehicle with the STEEP crossovers (48dB Butterworth), so far, so good, but I do think that in the future, I'll use shallower slopes.

This particular vehicle was kinda a test bed of sorts, so my decisions on crossover setup were for as much experimentation as anything.


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## sqnut (Dec 24, 2009)

Niick said:


> Got another crossover/EQ question here, as Andy has mentioned in the past, I'm EQing to match target, "target" being the electrical crossover slope, so I'm trying to get my acoustical slope to match the electrical.
> 
> My question is, how far down in dB do you go with this? Is 30 dB below reference (the top of the curve) far enough?
> 
> I'm attaching a pic of my progress so far on the right tweeter. I've been able to use EQ to match the target down to about 30 dB below freference. I'm wondering how far down I should chase it?


Like already mentioned 20-24 db down is sufficient. If the dsp gives an eq per channel, what you can do is to cut everything from 20hz-2khz to the max on each tweeter. The squiggly stuff from 20-400 hz is 30 db down, but its pure distortion. When you use the eq to cut these frequencies you'll notice a difference in measurements but more importantly, the sound will be* much *cleaner...............and you can play it louder without the sound breaking up.


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## Niick (Jun 3, 2015)

sqnut said:


> Like already mentioned 20-24 db down is sufficient. If the dsp gives an eq per channel, what you can do is to cut everything from 20hz-2khz to the max on each tweeter. The squiggly stuff from 20-400 hz is 30 db down, but its pure distortion. When you use the eq to cut these frequencies you'll notice a difference in measurements but more importantly, the sound will be* much *cleaner...............and you can play it louder without the sound breaking up.


That's a good idea, I never thought of using the remaining filters for the tweeter channels like that........it makes a lot of sense too, because often I've been finding that, especially with the 360.3, I'll have about half the available filters left, after EQing to match target. 

Although, I should probably mention that the 20-400 Hz stuff in this particular graph is the noise floor of the shop at the time of measurement. Air compressor in background, people using power tools etc. 

Another kinda cool thing I found out how to do yesterday was, let's say for instance you have a processor, like the RF360.3, and it has parametric EQ with variable Q. And you've EQ'd the pass band of a particular driver to match target as close as you can get it, and it took, let's say 13 filters. 

Now, you still have 17 or so filters left, you can go back and attack a couple of those pesky high Q peaks that your processor's SINGLE filter Q wasn't able to address before, because the Q of a single filter wasn't sharp enough.

By cascading multiple filters, each one of increasing magnitude, you can achieve a sharper Q than any one filter can achieve on its own.


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## subterFUSE (Sep 21, 2009)

Niick said:


> That's a good idea, I never thought of using the remaining filters for the tweeter channels like that........it makes a lot of sense too, because often I've been finding that, especially with the 360.3, I'll have about half the available filters left, after EQing to match target.
> 
> 
> 
> ...



Yeah, the Helix DSP Pro allows cascading of up to 4 bands of parametric EQ, each with -15dB for a total cut of -60 dB.
Combine those cascaded EQ bands with high Q and you basically have a notch filter.


Sent from my iPad using Tapatalk


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## Niick (Jun 3, 2015)

subterFUSE said:


> Yeah, the Helix DSP Pro allows cascading of up to 4 bands of parametric EQ, each with -15dB for a total cut of -60 dB.
> Combine those cascaded EQ bands with high Q and you basically have a notch filter.
> 
> 
> Sent from my iPad using Tapatalk


Right!! I did a test on a FR360.3 yesterday doing exactly this, and by coincidence, I used 4 filters, all set to the same center freq. and Q, but each one a little more attenuated than the previous one, and I achieved a very sharp Q "notch like" filter. I don't think the 360.3 limits how many filters you can cascade, but with only 30 or so to begin with, doing his too much will run you short of filters real fast.


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## sqnut (Dec 24, 2009)

Niick said:


> That's a good idea, I never thought of using the remaining filters for the tweeter channels like that........it makes a lot of sense too, because often I've been finding that, especially with the 360.3, I'll have about half the available filters left, after EQing to match target.


I do it on all drivers. An octave beyond the LP and an octave under a HP everything is cut to the max on the eq. 



Niick said:


> Although, I should probably mention that the 20-400 Hz stuff in this particular graph is the noise floor of the shop at the time of measurement. Air compressor in background, people using power tools etc.


didn't think of that :blush: 



Niick said:


> Another kinda cool thing I found out how to do yesterday was, let's say for instance you have a processor, like the RF360.3, and it has parametric EQ with variable Q. And you've EQ'd the pass band of a particular driver to match target as close as you can get it, and it took, let's say 13 filters.
> 
> Now, you still have 17 or so filters left, you can go back and attack a couple of those pesky high Q peaks that your processor's SINGLE filter Q wasn't able to address before, because the Q of a single filter wasn't sharp enough.
> 
> By cascading multiple filters, each one of increasing magnitude, you can achieve a sharper Q than any one filter can achieve on its own.


6-8khz are typical problem areas right? This region always shows a peak, but our hearing sensitivity is falling off a cliff here. 6-8khz typically requires a lot of cuts to remove the 'shouty' vocals and the chaff in the sound, typically 6-8 db cuts are fairly normal. Sometimes you still have a peak even after cutting. If the vocals aren't shouty, don't bother chasing that 1.5db pesky peak. If you cut too much here you can loose some dynamics in the upper mid range and the sound can get kinda flat.


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## seafish (Aug 1, 2012)

Quick question you guys…are you saying that if I use a 12db xo on the headunit and a 12db xo at exactly the same frequency, I can achieve a 24 db slope?? 

Also, does it matter which type of xo the HU and amp use??


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## Niick (Jun 3, 2015)

seafish said:


> Quick question you guys…are you saying that if I use a 12db xo on the headunit and a 12db xo at exactly the same frequency, I can achieve a 24 db slope??
> 
> Also, does it matter which type of xo the HU and amp use??


That's not exactly what I'm talking about. I'm referring to EQ filters in a DSP processor. 

HOWEVER, if you cascade two Low Pass filters for example, each at 12 dB, then yes, you would end up with a steeper slope tha either one of those filters by themselves.

And for what it's worth, don't forget that (almost) all filters will create phase shift, so there's that too.


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## seafish (Aug 1, 2012)

Niick said:


> That's not exactly what I'm talking about. I'm referring to EQ filters in a DSP processor.
> 
> HOWEVER, if you cascade two Low Pass filters for example, each at 12 dB, then yes, you would end up with a steeper slope tha either one of those filters by themselves.
> 
> And for what it's worth, don't forget that (almost) all filters will create phase shift, so there's that too.


OK…thanks for clarifying that…I'll start another thread about stacking xo filters.


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## Patrick Bateman (Sep 11, 2006)

seafish said:


> Quick question you guys…are you saying that if I use a 12db xo on the headunit and a 12db xo at exactly the same frequency, I can achieve a 24 db slope??





seafish said:


> Also, does it matter which type of xo the HU and amp use??


You can do this to infinity; if you were willing to live with the insane levels of phase shift that would occur with a 96dB octave xover, you could indeed do it by cascading eight 12dB octave xovers.
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## seafish (Aug 1, 2012)

Patrick Bateman said:


> You can do this to infinity; if you were willing to live with the insane levels of phase shift that would occur with a 96dB octave xover, you could indeed do it by cascading eight 12dB octave xovers.


But simply pout, since the steepest slope xo in my DRz9255 is only (18db) (which while maybe steep enough, also results in 180 degree phase shift at some xo frequencies), then I though I might achieve a viable 24 db slope by "cascading" either the 18 db slope on the HU with a 6db slope on the amp, or perhaps doubling up a 12 db slope on each. 

You guys are telling me this is doable and that I can experiment with with zero disadvantages except to the possibility of phase shift at xo points, which is happening anyway, right??

Of course, it is certainly possibly that the 18db slope on the DRZ9255 will be perfect for my install, but since Andy W. has always been saying that a 24db LR is the ideal slope for vehicle installation xo points, I thought that I would want to experiment with this using only my existing equipment as long as there are no sonic or electronic downsides to it.


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## sqnut (Dec 24, 2009)

seafish said:


> But simply pout, since the steepest slope xo in my DRz9255 is only (18db) (which while maybe steep enough, also results in 180 degree phase shift at some xo frequencies), then I though I might achieve a viable 24 db slope by "cascading" either the 18 db slope on the HU with a 6db slope on the amp, or perhaps doubling up a 12 db slope on each.


Theoretically yes you can use cascade two filters, but the practical difficulty in your case would be that, at the HU you would select an exact frequency say 80hz and apply say a 12 db slope but on the amp dial unless its exactly marked, or the amp allows you to input the frequency, 80hz is going to be an approximation at best. So you could end up with a 12 db slope @ 80 on the hu and another 12db @ anywhere from 60-100... cascading works best when you can select the exact frequency at both ends.


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## seafish (Aug 1, 2012)

sqnut said:


> Theoretically yes you can use cascade two filters, but the practical difficulty in your case would be that, at the HU you would select an exact frequency say 80hz and apply say a 12 db slope but on the amp dial unless its exactly marked, or the amp allows you to input the frequency, 80hz is going to be an approximation at best. So you could end up with a 12 db slope @ 80 on the hu and another 12db @ anywhere from 60-100... cascading works best when you can select the exact frequency at both ends.



I figured that might be a problem…I'll bet that there is some way I could use an RTA or some there measurement device to get the xo frequency to overlap exactly??


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## Niick (Jun 3, 2015)

seafish said:


> I figured that might be a problem…I'll bet that there is some way I could use an RTA or some there measurement device to get the xo frequency to overlap exactly??


If you had an RTA with a high enough resolution, like 1/48 or 1/96 th octave, 
then yes, you probably could, but to know the precise frequency and attenuation level, you'd need a measurement system with very fine resolution. In fact, you could use the free program REW for this. You would want to take your measurements in the electrical domain. Use the overlay feature for this.


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## seafish (Aug 1, 2012)

Niick said:


> If you had an RTA with a high enough resolution, like 1/48 or 1/96 th octave,
> then yes, you probably could, but to know the precise frequency and attenuation level, you'd need a measurement system with very fine resolution. In fact, you could use the free program REW for this. You would want to take your measurements in the electrical domain. Use the overlay feature for this.


Niick…I am VERY mechanically inclined, but NOT electronically adept...think I'm just gonna have to plan a trip to OR after I get my upgraded install done this winter and have YOU do it tune my new system, that is!!! LOL


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## Niick (Jun 3, 2015)

Hey, I'd love to have ya come to my shop and let me work on your system. But, I truly believe, that the same skills of logical thought that are required for an individual to become truly mechanically skilled, can also be used to learn electronics/acoustics. 

Well......on second thought, maybe not. Hell, I don't know! 

So here's what ya do, I'll PM ya my email address, shoot me an email and we'll schedule a day and time, seriously, that would be awesome.


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## seafish (Aug 1, 2012)

Niick said:


> Hey, I'd love to have ya come to my shop and let me work on your system. But, I truly believe, that the same skills of logical thought that are required for an individual to become truly mechanically skilled, can also be used to learn electronics/acoustics.
> 
> Well......on second thought, maybe not. Hell, I don't know!
> 
> So here's what ya do, I'll PM ya my email address, shoot me an email and we'll schedule a day and time, seriously, that would be awesome.



pm replied to.


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