# Time Alignment Phase Calculation Spreadsheet



## ecbmxer (Dec 1, 2010)

I wanted to share a little spreadsheet I made for calculating time alignment to make sure things remain in phase at the crossover points. I have it attached here as a .txt file since it wouldn't let me do an .xlsx file. Just change the extension to .xlsx.

I have it setup for doing up to a 3 way plus sub, including left and right side drivers.

Basically, it will take into account the polarity or phase of each driver, the distance from each driver to your listening position, the phase shift induced by the crossover (you can use 6,12,18,24), and then whatever digital time delay you will add from your processor. After setting things up for your given setup, you can play with the time delays until the total phase shift for any two drivers that are crossed together matches up, in addition to matching up between the L and R sides.

NOTE: Right now the calculations are only setup for speakers that are crossed over at a common crossover frequency. If you are underlapping or overlapping, I can't say what the results will be like. Basically, in that situation you will be trying to make one driver in phase with another driver, but at two different frequencies. I could probably add some rows to calculate the phase shift for BOTH drivers at BOTH frequencies and then you could choose a delay that gets it close in between or at both. But for now I didn't do that.










INPUTS:

All of the columns you need to provide inputs to are colored gray.

First, you want to measure the distance to each of your drivers in your car. Get it as close to accurate as you can, but it doesn't need to be perfect. Put these values into the "Distance" column under the "Speaker" category. I have dotted boxes around the HP/LP rows common to a given speaker that is being bandpassed. Make sure the values are equal. Not like you can have the HP portion of your midbass at a different distance than the LP. 

Next setup your crossover frequencies, slopes, and if you have any speakers wired with reverse polarity or with some phase adjustment (for example if you have a 0 or 180 setting on sub phase).

ADDING DELAY:

Now you are ready to stat adding additional time delay to each driver. Typically you will want to keep the furthest driver at 0" and work from there, delaying all of the others accordingly. 

Start with the L/R midbasses and try to make the "Total Phase Shift" in the far right column equal what your sub is. Or if one of your midbasses is farther away, you can start with it at zero and delay the sub and other midbass so the total delay matches between all three. 

Continue doing this for each driver pair that are crossed over together and between L and R.

For my car, using the values calculated from this spreadsheet made a huge difference. I have the values I used in there as reference. Just FYI, my tweeters are mounted very near my mids and the two are crossed over passively, so I cannot adjust anything independently for my tweeters.

I'm not saying this will definitely work for everyone's car or sound better than those of you who can tune it by ear using the doppler effect technique, etc found on this site. But I wanted to share. 

Let me know if you have any questions, it works/doesn't work or if you find any bugs or issues with the calculations. I certainly have not tried all the possible combinations to be sure they all sound coherent/centered/in phase in the car. But in my car, for the default values I have in the spreadsheet, it works.

-Clint


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## minbari (Mar 3, 2011)

sorry, but this whole thing is kinda pointless.

you cant compensate for phase changes with time alignment. for one thing, different freqs are gonna have different time differences. you cant make a T/A adjsutment for only certain freq.

The other big reason this wont work is the phase change in a crossover is only near the xover point. The phase change goes away the farthur away you get from the xover point. you cant make an adjustment like that with T/A. Add in the phase changes over the freq spectrum that a speaker makes and it is even more hopeless.


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## SkizeR (Apr 19, 2011)

also wouldnt this only be good for your car and driver placement?


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## strakele (Mar 2, 2009)

Phase doesn't matter when it's only one speaker playing a given frequency. It DOES matter at the crossover point where you have two speakers playing the same frequencies, which I believe is what this spreadsheet aims to align, which is of great benefit.

Try it.


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## Miguel mac (Sep 28, 2009)

Suscrito 




Enviado desde mi HUAWEI U9508 usando Tapatalk 2


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## ecbmxer (Dec 1, 2010)

minbari said:


> sorry, but this whole thing is kinda pointless.
> 
> you cant compensate for phase changes with time alignment. for one thing, different freqs are gonna have different time differences. you cant make a T/A adjsutment for only certain freq.
> 
> The other big reason this wont work is the phase change in a crossover is only near the xover point. The phase change goes away the farthur away you get from the xover point. you cant make an adjustment like that with T/A. Add in the phase changes over the freq spectrum that a speaker makes and it is even more hopeless.


Right, that is what this is. It calculates the phase shift between two drivers at the crossover point. I should also mention the time alignment numbers here are setup for an Alpine processor/source unit, where you enter "0" for the farthest driver and a large value for one closer to you.


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## ecbmxer (Dec 1, 2010)

SkizeR said:


> also wouldnt this only be good for your car and driver placement?


It will work for any car and driver placement. You can go in and specify the distance from each driver to your listening position. This takes it one step farther by considering the phase shift due to the crossover as well as the frequency at which the drivers are crossed over.

I'll add some more to this at some point to try and handle over/underlapping crossover point. Either by calculating the phase shift for both drivers at both points and letting the user decide what delay is the best compromise, or maybe splitting the difference between the crossover points.


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## seafish (Aug 1, 2012)

subscribed


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## minbari (Mar 3, 2011)

ecbmxer said:


> Right, that is what this is. It calculates the phase shift between two drivers at the crossover point. I should also mention the time alignment numbers here are setup for an Alpine processor/source unit, where you enter "0" for the farthest driver and a large value for one closer to you.


do you take the phase plot of the driver as measured in the installed location for the specified xover freq? if not, then your calculations would be very far off.

just dont see this working and even if it did, I dont see it being very audible.


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## Woosey (Feb 2, 2011)

minbari said:


> *do you take the phase plot of the driver as measured in the installed location for the specified xover freq*? if not, then your calculations would be very far off.
> 
> just dont see this working and even if it did, I dont see it being very audible.


this..


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## Kevin K (Feb 11, 2013)

Thanks for sharing. I'll be doing some studies....and listening....


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## ecbmxer (Dec 1, 2010)

minbari said:


> do you take the phase plot of the driver as measured in the installed location for the specified xover freq? if not, then your calculations would be very far off.
> 
> just dont see this working and even if it did, I dont see it being very audible.


I do not have that information. This isn't meant to try and determine the absolute phase of a given driver, but the relative phase between two drivers due to their distance from the listener and the crossover. 

I do understand that each driver's absolute phase will change with frequency and that it is also affected by the enclosure. I'm just not sure how significant the difference in phase between two drivers at their crossover point is due to the drivers/enclosure compared to that caused by the crossover and differences in arrival time. It could be that I just got lucky in my setup where the absolute phase of the drivers are fairly close to each other where I have them crossed.

I'll look into it a bit more. Thanks for the feedback.


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## minbari (Mar 3, 2011)

ecbmxer said:


> I do not have that information. This isn't meant to try and determine the absolute phase of a given driver, but the relative phase between two drivers due to their distance from the listener and the crossover.
> 
> I do understand that each driver's absolute phase will change with frequency and that it is also affected by the enclosure. I'm just not sure how significant the difference in phase between two drivers at their crossover point is due to the drivers/enclosure compared to that caused by the crossover and differences in arrival time. It could be that I just got lucky in my setup where the absolute phase of the drivers are fairly close to each other where I have them crossed.
> 
> I'll look into it a bit more. Thanks for the feedback.


reason I brought it up is because if you are trying to align drivers phase with time delay, but you have an unknown phase angle, how are you aligning them? Let say you have a 50 degree angle on your woofer and a 20 degree angle on the tweeter at the xover point you are using. Then simply measuring the distance and calculating all the electrical phase differences wont really matter. you might be 30 or 70 degrees off (depending on if it is leading or lagging). The point is if you did nothing, it would be 45 degree off and with your changes it is 70 degrees or 20 or 10. you still cant fix the phase response of the driver.

it is posible you got lucky and the phase of the two drivers was close to each other, or it might just not matter. 

My belief is that trying to critically align phase that close in a car is a hopeless journey. even if you got it perfect, reflections, standing waves, cancelation, etc will have a greater effect on how it sounds and those things cant be fixed most of the time.


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## 14642 (May 19, 2008)

Nope. Unless you are calculating the phase angle using the acoustic output of the speakers filtered by the crossover or measuring phase, and then it will only work if you are phase aligning the drivers to achieve the target sum defined by the alignment. The only way to do that is to EQ each speaker separately so that the acoustic output matches a particular response alignment and then aligning the phase so the output of the two speakers, that now precisely match a target alignment sum according to the alignment. To do that, you need separate parametric EQ on each driver. 

If you are calculating phase shift based on the electrical slopes you are using, then this is just another piece of evidence that super high resolution adjustment of delay isn't necessary. 

Why isn't the simple method good enough? Time alignment should simply be used to align the acoustic centers of the drivers. Using a tape measure from the dustcap to the center of your forehead is sufficient. If you want to make it more complicated, then learn to make some impulse response measurements. Then, you'll be able to decide whether to use the initial impulse from the speaker or a reflection from an adjacent boundary as the apparent source of the sound. In my estimation, this is the only useful reason for setting time alignment by ear.


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## ecbmxer (Dec 1, 2010)

minbari said:


> reason I brought it up is because if you are trying to align drivers phase with time delay, but you have an unknown phase angle, how are you aligning them? Let say you have a 50 degree angle on your woofer and a 20 degree angle on the tweeter at the xover point you are using. Then simply measuring the distance and calculating all the electrical phase differences wont really matter. you might be 30 or 70 degrees off (depending on if it is leading or lagging). The point is if you did nothing, it would be 45 degree off and with your changes it is 70 degrees or 20 or 10. you still cant fix the phase response of the driver.
> 
> it is posible you got lucky and the phase of the two drivers was close to each other, or it might just not matter.
> 
> My belief is that trying to critically align phase that close in a car is a hopeless journey. even if you got it perfect, reflections, standing waves, cancelation, etc will have a greater effect on how it sounds and those things cant be fixed most of the time.


Yea for sure. I'm pretty much assuming that the phase differences aren't as significant as the differences due to varying path lengths and crossovers. But that might be a bad assumption. And any reflections, etc could just totally ruin any potential alignment. When I first tried this, I didn't really expect it to work, but it was a big improvement from what I had before just measuring the distances using a tape measure. 



Andy Wehmeyer said:


> Nope. Unless you are calculating the phase angle using the acoustic output of the speakers filtered by the crossover or measuring phase, and then it will only work if you are phase aligning the drivers to achieve the target sum defined by the alignment. The only way to do that is to EQ each speaker separately so that the acoustic output matches a particular response alignment and then aligning the phase so the output of the two speakers, that now precisely match a target alignment sum according to the alignment. To do that, you need separate parametric EQ on each driver.
> 
> If you are calculating phase shift based on the electrical slopes you are using, then this is just another piece of evidence that super high resolution adjustment of delay isn't necessary.
> 
> Why isn't the simple method good enough? Time alignment should simply be used to align the acoustic centers of the drivers. Using a tape measure from the dustcap to the center of your forehead is sufficient. If you want to make it more complicated, then learn to make some impulse response measurements. Then, you'll be able to decide whether to use the initial impulse from the speaker or a reflection from an adjacent boundary as the apparent source of the sound. In my estimation, this is the only useful reason for setting time alignment by ear.



And yes, I just calculated the phase shift based on the electrical slopes I was using. I have yet to try and do the impulse measurement...although I do have the setup to do it now. 

Andy, since you seem very knowledgeable about this type of thing, how significantly does the door or (god forbid) a-pillar "enclosure" affect the phase of a driver? For example, if one had a phase vs frequency plot from the manufacturer, how different would it measure in the "enclosure"?

Thanks for the input!


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## 14642 (May 19, 2008)

This whole phase discussion as it's related to car audio is almost useless. I say almost because there are two performance issues related to phase that DO matter. The first and most obvious and easiest to affect is the reproduction of a phantom center (and phantom images between the left and right front speakers that aren't in the center). This requires that the ACOUSTIC output that arrives at your ears of left and right are well matched. They should also arrive at about the same time. Above 1k Hz, we don't use phase very well to determine the location of sounds. Above about 3.5k, we don't use them at all. From 1-3.5, we just aren't very good at it and a combination of ITD and IID are helpful, but we just basically suck at it in that region. 

Below about 60 Hz, we don't use either very well because the distance between our ears is really short relative to the length of the wave. What that means is that if you place a 7" space along a sine wave that's 20 feet long, you'll measure only a few degrees of phase difference between one end of the 7" and the other. Above 3.5k or thereabouts, TOO MANY sine waves fit in the space between our ears for them to be effective in determining the location using phase. 

IN any reflective environment (and all environments are reflective), there are several apparent locations for sound: the location of the speaker that emits the sound and the location of all of the surfaces that reflect the original sound toward our ears. For now, think of each reflection as another speaker playing the same sound as the original. Just as in a stereo speaker system, a phantom image appears between the two sources, a phantom image will appear between the speaker and a reflective surface. 

OK..with me?

Next, since the distance from the speaker to the reflecting surface and then to our ears is longer than the distance from the speaker to our ears, the sound of the reflection is attenuated. The inverse square law states that the sound is attenuated by 6dB for every DOUBLING of distance. If the sound of the reflection and the sound of the speaker are the same, that would push the phantom image toward the reflection if the distance was double. If the distance is less than double, the phantom image is pushed farther toward the reflecting surface. 

OK...next...

Comb filtering occurs when two sources play the same thing when they are at different distances from the listening position and the left most dip in the "comb" happens where the difference in distance corresponds to a half wavelength. The dip is repeated at multiples of that first dip. Each successive dip becomes less deep and less wide. The first dip is the only one we really need to be concerned about because it's big. 

If we have two speakers in an anechoic environment located different distances away, (as in a stereo system), then we can use delay on one channel to "fix" the arrival (distance) problem. Fixing the distance problem does two things. It eliminates the ENTIRE comb and centers the image at all frequencies. IF we use a phase EQ (all pass filter) to fix the phase according to the distance at the first dip in the comb, we fix it, but we leave the others intact. IF we reverse the polarity of one speaker to fix the comb, we simply shift the comb to different multiples. When you reverse the phase of one midrange in your car to fix the center vocal, you just screw it up somewhere else.

OK, so time alignment (delay) can "fix" this for two speakers playing the same stuff from different distances. What about for a speaker and a sound reflected from an adjacent boundary? Since the sound of the speaker CAUSES the reflection and we can't violate the law of causality, we CANNOT fix this with delay. We cannot delay the reflection. 

Now, let's consider a 4" midrange. Sound radiates from a speaker in ALL directions below the frequency with a wavelength that's as long as the diameter of the cone. Read that again. That means that below about 4k, your 4" speaker is OMNIDIRECTIONAL. That means that at those frequencies, it doesn't matter which direction you point the speaker--the sound goes basically everywhere in every direction--to the front, to the sides, and even to the back.

Now, remember that what you hear is the sum of the sound from the speaker and the sound from the reflections. The difference in distance to your ears creates a comb filter for each reflection and a phantom image for each reflection because each reflection is out of phase from the fundamental according to that difference in distance. Since every reflection is a different distance to your ears, the frequency response of the sum of the original sound and the reflection includes a slightly different comb filter, and so on and so forth. This condition causes the image to wander around a little bit at different frequencies. It's a complete mess and it can't be fixed.

When you hang your midrange from the a-pillar, you place an "omnidirectional" signal source some distance away from the maximum number of reflecting surfaces in the car. This causes the maximum number of apparent sources and the maximum number of frequency response aberrations and the maximum amount of crosstalk between the channels which narrows the stage. It does, however, place that narrow stage high in the car.

But wait, we don't have ears on the top and bottom of our heads, so we've placed at the top of our list of important criteria designing for a condition that we are not able to distinguish very well. We've painted a beautifully accurate picture (in terms of tone and hue) for someone who is color blind, but all of the shapes are distorted. 

So, what can we do? Can we treat the car for reflections? Hmmm...what are you going to put on the windshield to reduce reflections?

We can eliminate reflecting surfaces by moving the speakers. We can reduce the influence of the reflections by moving the source of the sound AWAY from the reflecting surfaces. We can raise the the frequency of the first null in the comb by reducing the distance between the source and the reflecting surfaces.

If we put the speakers in one of the surfaces rather than in front of it, we basically eliminate the comb filter from it. If we put the speakers in the kick panels, we dramatically reduce the effect of the reflections from the windshield and the top of the windshield. If we put them in the doors, we reduce the effect some and we increase the angle from our ears to the initial source--whiich widens the stage. If we put them at the apex of the dash, side and windshield (the the corner) then we minimize the distance between the source and the reflections.

None of these are perfect because there's a reflection at EVERY ANGLE, but the pillars exacerbate this problem the most. 

Can you make a car sound good with speakers in the pillars? Sure. 

So, the question about the enclosure...the phase of the speaker at various frequencies is derived from the frequency response. So, yes. putting the speaker in box that alters its frequency response alters the phase. It's a small problem and given the BIG problem above, it's not one to focus on. 

One additional way to improve the imaging performance of the system is to INCREASE the number of speakers and DECREASE the dependence on acoustic sums of their output to create images. This is the best argument for a center channel and signal steering.


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## minbari (Mar 3, 2011)

always enjoy reading what you have to say. very well put.


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## cajunner (Apr 13, 2007)

now there's some post goodness..


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## ecbmxer (Dec 1, 2010)

Andy Wehmeyer said:


> This whole phase discussion as it's related to car audio is almost useless. I say almost because there are two performance issues related to phase that DO matter. The first and most obvious and easiest to affect is the reproduction of a phantom center (and phantom images between the left and right front speakers that aren't in the center). This requires that the ACOUSTIC output that arrives at your ears of left and right are well matched. They should also arrive at about the same time. Above 1k Hz, we don't use phase very well to determine the location of sounds. Above about 3.5k, we don't use them at all. From 1-3.5, we just aren't very good at it and a combination of ITD and IID are helpful, but we just basically suck at it in that region.
> 
> Below about 60 Hz, we don't use either very well because the distance between our ears is really short relative to the length of the wave. What that means is that if you place a 7" space along a sine wave that's 20 feet long, you'll measure only a few degrees of phase difference between one end of the 7" and the other. Above 3.5k or thereabouts, TOO MANY sine waves fit in the space between our ears for them to be effective in determining the location using phase.
> 
> ...


I shouldn't have mentioned the a-pillars....LOL 

But seriously, thanks for the input. I totally get what you're saying in terms of reflections (and the resulting delay/phase shift), comb filtering, beaming, etc. None of that cannot be accounted for in a simple calculation like I did here. 

But I think this helps (for me at least) to understand how the time delay and crossover phase shift work for different frequencies, slopes, etc. It could be that it just so happens to work out for my setup, but I have the best center image, up front bass, and (seemingly) sub to midbass integration so far in my car using the values I calculated.

If one were so inclined to know what the phase shift of their speaker looks like based on the manufacturer provided frequency response (SPL and impedance), is that something that could be calculated relatively easily? It is minimum at Fs, right? (max impedance)

Thanks again Andy.


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## minbari (Mar 3, 2011)

ecbmxer said:


> I shouldn't have mentioned the a-pillars....LOL
> 
> But seriously, thanks for the input. I totally get what you're saying in terms of reflections (and the resulting delay/phase shift), comb filtering, beaming, etc. None of that cannot be accounted for in a simple calculation like I did here.
> 
> ...


impedance and phase are two different things and dont really correlate to each other.

impedance would indicate how loud a speaker would be at a given freq.
Phase is a shift in time relative to a reference for a given freq.


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## ShaneInMN (Sep 27, 2013)

I love reading threads like this. Thanks for bringing it up and I've learned a lot even though I dont understand it all yet.


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## Hanatsu (Nov 9, 2010)

ecbmxer said:


> If one were so inclined to know what the phase shift of their speaker looks like based on the manufacturer provided frequency response (SPL and impedance), is that something that could be calculated relatively easily? It is minimum at Fs, right? (max impedance)


Are you referring to the impedance phase that Dayton (among others) post in their FR plots in the spec sheets?

Furthermore, Fs is -free air- resonance. It's highly unlikely you can pull off a true IB setup in a car, Fsc will therefore move upwards and the impedance phase shift along with it. May I ask why you want to calculate this? The phase data that matters is what you measure acoustically. I've tried several methods to measure T/A and the ones that work is; Aligning by -impulse response- with the soundcard in loopback mode. 2nd, observing -group delay- plot with one side playing + sub. The delay will show up between different set of frequency ranges. The last one is useful in aligning mid/sub and it's simply done by -observing the FR plot-. Where the mids/sub(s) sum the most is the 'best' phase relationship.


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## 14642 (May 19, 2008)

Impedance isn't really an indication of how loudly a speaker will play. That leaves out the transformation from electrical energy to acoustic energy. It is an indication of how much current the speaker will draw from the amp.


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## minbari (Mar 3, 2011)

Andy Wehmeyer said:


> Impedance isn't really an indication of how loudly a speaker will play. That leaves out the transformation from electrical energy to acoustic energy. It is an indication of how much current the speaker will draw from the amp.


I didnt mean from an actual DB scale potential. If you have a nominal impedance of 4 ohms on a 6.5" and you have a spike of 20 ohms at a particular freq, it is going to be playing that freq more quietly because it is drawing less current and generating less acoustical energy.

This is for free air.


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## 14642 (May 19, 2008)

except at resonance, where the reduction in output is not proportionate to the risein impedance.


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## minbari (Mar 3, 2011)

agreed


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## kaigoss69 (Apr 2, 2008)

Thank for the great read Andy! :bowdown:


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## ultimatemj (Jan 15, 2009)

> One additional way to improve the imaging performance of the system is to INCREASE the number of speakers and DECREASE the dependence on acoustic sums of their output to create images.


Andy, I was with you until this statement. 

How do more speakers reduce the dependency?

I (think I) get the "spread the chaos around" bit, but I'd expect more combing challenges from more speakers...no? Essentially the x-over points still need to be phase matched...no? Or are you taking this to a theoretical extreme, where enough chaos (sources and reflections) would sum to the point combing is a non-issue?


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## ecbmxer (Dec 1, 2010)

Hanatsu said:


> Are you referring to the impedance phase that Dayton (among others) post in their FR plots in the spec sheets?
> 
> Furthermore, Fs is -free air- resonance. It's highly unlikely you can pull off a true IB setup in a car, Fsc will therefore move upwards and the impedance phase shift along with it. May I ask why you want to calculate this? The phase data that matters is what you measure acoustically. I've tried several methods to measure T/A and the ones that work is; Aligning by -impulse response- with the soundcard in loopback mode. 2nd, observing -group delay- plot with one side playing + sub. The delay will show up between different set of frequency ranges. The last one is useful in aligning mid/sub and it's simply done by -observing the FR plot-. Where the mids/sub(s) sum the most is the 'best' phase relationship.


Yea, exactly. I just saw some driver spec sheets that showed the phase change of the driver and wondered what it looked like for my drivers. I guess just for my own curiosity. I got to reading about parallel resonant circuits that had info on how phase angle related to impedance at different frequencies.



minbari said:


> impedance and phase are two different things and dont really correlate to each other.
> 
> impedance would indicate how loud a speaker would be at a given freq.
> Phase is a shift in time relative to a reference for a given freq.


I guess my assumption was that you could model the speaker as a parallel resonant RLC circuit, which has minimum phase at the resonant frequency along with maximum impedance. Perhaps that is a bad assumption. I don't really know. I'm an ME not an EE, so this is new to me.

I did read another bit of info that said "the phase is roughly proportional to the derivative of the magnitude" (of the impedance I assume). And actually that would make total sense since it would be approximately zero at the maximum impedance at Fs and have a high and low value at the steepest portions of the slope of the impedance curve just above and below Fs. This in no way translates to anything in the car, I guess I've gotten way of track at this point...


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## minbari (Mar 3, 2011)

ecbmxer said:


> I guess my assumption was that you could model the speaker as a parallel resonant RLC circuit, which has minimum phase at the resonant frequency along with maximum impedance. Perhaps that is a bad assumption. I don't really know. I'm an ME not an EE, so this is new to me.


you are not wrong, but this is not an all electronic problem. Some of the phase issues are related to the physical movement of the cone of the driver.


> I did read another bit of info that said "the phase is roughly proportional to the derivative of the magnitude" (of the impedance I assume). And actually that would make total sense since it would be approximately zero at the maximum impedance at Fs and have a high and low value at the steepest portions of the slope of the impedance curve just above and below Fs. This in no way translates to anything in the car, I guess I've gotten way of track at this point...


at FS, yes.


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## Hanatsu (Nov 9, 2010)

ecbmxer said:


> Yea, exactly. I just saw some driver spec sheets that showed the phase change of the driver and wondered what it looked like for my drivers. I guess just for my own curiosity. I got to reading about parallel resonant circuits that had info on how phase angle related to impedance at different frequencies.


Here's a measurement over a 15" sub I've made (open baffle):



Note that the impedance phase and the acoustic phase you measure (derived from FR) are two different things. The impedance phase is what the amplifier is "seeing". Amplifiers generally don't like steep impedance phase angles together with a low impedance load.


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## seafish (Aug 1, 2012)

ultimatemj said:


> Andy, I was with you until this statement.
> 
> How do more speakers reduce the dependency?


X2???


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## Hanatsu (Nov 9, 2010)

ecbmxer said:


> Yea, exactly. I just saw some driver spec sheets that showed the phase change of the driver and wondered what it looked like for my drivers. I guess just for my own curiosity. I got to reading about parallel resonant circuits that had info on how phase angle related to impedance at different frequencies.


... and this is a measurement over a widebander on a flat baffle. It has a Fs of 185Hz. Even if the impedance phase looks similar to the other picture I posted the acoustical phase have no dramatic change in this area. That's to be expected considering how the measurement software derives the phase data.


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## ecbmxer (Dec 1, 2010)

Cool stuff. Did you say you used HOLMImpulse for those measurements?


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## SPLEclipse (Aug 17, 2012)

ultimatemj said:


> Andy, I was with you until this statement.
> 
> How do more speakers reduce the dependency?
> 
> I (think I) get the "spread the chaos around" bit, but I'd expect more combing challenges from more speakers...no? Essentially the x-over points still need to be phase matched...no? Or are you taking this to a theoretical extreme, where enough chaos (sources and reflections) would sum to the point combing is a non-issue?





seafish said:


> X2???


I think he's saying a physical center speaker is better at giving you a (surprise) solid center image because it alleviates the necessity of creating an acoustically perfect alignment between L/R to create a phantom center, which is virtually impossible.

edit: OP: I'm going to give the spreadsheet numbers a shot tomorrow if it warms up a bit. While I think there are some technical flaws, it's only going to take a few minutes and who know, maybe I'll come across something good?


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## ErinH (Feb 14, 2007)

Wow. I was about a second away from going full on ballistic up in here. Then I realized this sheet isn't a rip off of the one I'd been circulating around (some guys here have seen it). I'll admit I haven't read the thread entirely but I did see Andy's reply and believe from that brief overview that I'm on the same page with him. 

I'll try to d/l this tomorrow and compare to a sheet that I made. I'm curious to see if the numbers are similar for the tweeters, though I know for the lower frequency drivers they won't be. I employ a different method altogether and must admit that I've even been cynical of my own method from time to time which is the reason I have yet to post it (and why I was about to fly off the handle when I first saw this thread).


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## ErinH (Feb 14, 2007)

ecbmxer said:


> Cool stuff. Did you say you used HOLMImpulse for those measurements?


He's using aRTA for the measurement he posted.


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## SPLEclipse (Aug 17, 2012)

Hmm...getting beaucoup #NUM errors when I input my data. I know my PLDs aren't great but that's just rude. 

Erin: I'd love to give your spreadsheet a tryout as well. I understand if you want to keep it secret though.


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## seafish (Aug 1, 2012)

SPLEclipse said:


> I think he's saying a physical center speaker is better at giving you a (surprise) solid center image because it alleviates the necessity of creating an acoustically perfect alignment between L/R to create a phantom center, which is virtually impossible.


OK... now THAT makes sense, though it seemed as if Andy was talking about using more speakers in general, not just adding a single center for imaging.


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## ecbmxer (Dec 1, 2010)

bikinpunk said:


> Wow. I was about a second away from going full on ballistic up in here. Then I realized this sheet isn't a rip off of the one I'd been circulating around (some guys here have seen it). I'll admit I haven't read the thread entirely but I did see Andy's reply and believe from that brief overview that I'm on the same page with him.
> 
> I'll try to d/l this tomorrow and compare to a sheet that I made. I'm curious to see if the numbers are similar for the tweeters, though I know for the lower frequency drivers they won't be. I employ a different method altogether and must admit that I've even been cynical of my own method from time to time which is the reason I have yet to post it (and why I was about to fly off the handle when I first saw this thread).


Yea I never saw your calculations. But I do remember you saying something about having a spreadsheet you used for tuning. I was hesitant to post this as well. I pretty much got the responses I expected regarding all of the other phasing issues that may trump whatever alignment I can calculate like this. To be honest I didn't think it would even work, but the results were much better than what I had before with only the measured distances to each speaker, so I decided to share. I'd be interested to see your calcs as well. 



bikinpunk said:


> He's using aRTA for the measurement he posted.


Ah OK. I don't think I've tried that one. I have TrueRTA and HOLMImpulse. I'll check out aRTA too.



SPLEclipse said:


> Hmm...getting beaucoup #NUM errors when I input my data. I know my PLDs aren't great but that's just rude.
> 
> Erin: I'd love to give your spreadsheet a tryout as well. I understand if you want to keep it secret though.


Huh, thats strange. So you just input your phasing in the first column (probably just 0 for all unless you have your sub at 180 or something), crossover frequency and slope for each driver (two drivers being crossed should share the same frequency as well as L/R pairs), and your actual distances in the distance column under "speaker"? (again, make sure to have the same distance for the HP and LP part of a single driver). Then you can just play with the numbers in the "delay distance" column (again, making sure to have the same distance for the HP and LP part of a single driver) to try and get the "total phase shift" to match up between two drivers that are crossed over as well as between L/R driver pairs. What are your measured distances and crossover freq/slope that didn't work?


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## 14642 (May 19, 2008)

SPLEclipse said:


> I think he's saying a physical center speaker is better at giving you a (surprise) solid center image because it alleviates the necessity of creating an acoustically perfect alignment between L/R to create a phantom center, which is virtually impossible.
> 
> edit: OP: I'm going to give the spreadsheet numbers a shot tomorrow if it warms up a bit. While I think there are some technical flaws, it's only going to take a few minutes and who know, maybe I'll come across something good?


Right. Putting speakers in between the center and the right and left would be a next step, but you'd need processing to extract those signals. For now, just adding a center is helpful.

The same thing works to a certain extent for rear speakers. If you want ambience that sounds bigger than the car, put speakers in the back and play the ambient sounds through them. We'll never completely escape the car and be transported into a room that sounds the same as the recording space, but we can make the experience a bit more believable.


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## 14642 (May 19, 2008)

FWIW, measuring the frequency response in the car, smoothing it and generating a minimum phase response is useful for some things, (Equalization, maybe), but it isn't a picture of what's happening in the car. There is no MINIMUM acoustic phase in real audio systems in rooms.


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> FWIW, measuring the frequency response in the car, smoothing it and generating a minimum phase response is useful for some things, (Equalization, maybe), but it isn't a picture of what's happening in the car. There is no MINIMUM acoustic phase in real audio systems in rooms.


right, because the only way to achieve a reflection free measurement in the car is to gate it to something very, very low like 1ms, which has a resolution of 1khz and a response only above 1khz. To put that in perspective, for typical RTA use, you're using windows of at least 1000ms.


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## ErinH (Feb 14, 2007)

And, here goes...

At the risk of total regret, I'm providing my own quirky time alignment sheet. In a nutshell, I'm just taking the wavelength at the crossover and applying that to the standard time alignment values you get from a tape measure and calculator. I've got multiple caveats with this method... but I've captured those internally (via numerous revisions). This has been 'vetted' in some ways by a few guys here. Some stuff worked. Some stuff didn't. Mostly, this method produces worthwhile results for long wavelength; not so much for high wavelength... and I've therefore adjusted my methods in the sheet to account for this. If you're good with excel math then you'll easily figure this out (hint: look at the equations). If you're not, don't bang your head on a wall trying to. It's pretty much as simple as my first sentence in this paragraph with some tweaks here and there.

Read the directions at the top of the sheet. It explains all you need to know. All you need to provide are distance measurements and crossover frequency. Doesn't matter if they're the same between drivers. 

Regarding phase, this essentially assumes that the relative phase of the system is good. It won't fix that. It's just math. It can't hear your car for you. 

This isn’t guaranteed to work. It's been an ongoing thing for almost a year now. But, it may be worth a shot to check out. If you have any issues, don’t blame me. 

Here's the link:
http://bit.ly/1aTIE9J

- Erin


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## 14642 (May 19, 2008)

CAN SOMEONE PLEASE EXPLAIN WHY YOU THINK CALCULATING YOUR DELAYS SHOULD INCLUDE ANY INFORMATION ABOUT THE ELECTRICAL FILTERS YOU USE AS CROSSOVERS?

EVEN A SIMPLE READ OF ANY LOUDSPEAKER DESIGN BOOK WILL ILLUSTRATE THAT THE ELECTRICAL FILTERS ARE MEANINGLESS AND WHAT IS IMPORTANT IS THE ACOUSTIC SUM OF THE FINAL RESPONSES OF THE DRIVERS.


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## minbari (Mar 3, 2011)

Andy Wehmeyer said:


> CAN SOMEONE PLEASE EXPLAIN WHY YOU THINK CALCULATING YOUR DELAYS SHOULD INCLUDE ANY INFORMATION ABOUT THE ELECTRICAL FILTERS YOU USE AS CROSSOVERS?
> 
> EVEN A SIMPLE READ OF ANY LOUDSPEAKER DESIGN BOOK WILL ILLUSTRATE THAT THE ELECTRICAL FILTERS ARE MEANINGLESS AND WHAT IS IMPORTANT IS THE ACOUSTIC SUM OF THE FINAL RESPONSES OF THE DRIVERS.


lol, which is pretty much what I said in the first place, lol.

This method might yield results, but they are haphazard at best.


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> CAN SOMEONE PLEASE EXPLAIN WHY YOU THINK CALCULATING YOUR DELAYS SHOULD INCLUDE ANY INFORMATION ABOUT THE ELECTRICAL FILTERS YOU USE AS CROSSOVERS?
> 
> EVEN A SIMPLE READ OF ANY LOUDSPEAKER DESIGN BOOK WILL ILLUSTRATE THAT THE ELECTRICAL FILTERS ARE MEANINGLESS AND WHAT IS IMPORTANT IS THE ACOUSTIC SUM OF THE FINAL RESPONSES OF THE DRIVERS.


The slope or the frequency? Mine is based on frequency due to its relation with wavelength. Slope would only come in to play as a further means of calculating the median frequency. 

I understand that arrival time is arrival time. But the result has a dramatic difference when you measure drivers WITH a crossover on them. Full range impulse is obviously different than band passed impulse. Measured subwoofer arrival time of first peak is outrageous compared to tweeters; 10-20+ ms ranges compared to as small as 1-2ms ranges. And opening up the LPF of the sub permits full first impulse spike sooner because of the wavelength. And that's where my sheet takes its cue from. It's an exercise; not an authority. 

I do nothing with phase angle. I only work with a band passed notion of crossover frequency to help add or remove delay per wavelength. I've found great results with low end response using this method. Not so much with HF response.

As a side note, I've found some pretty dramatic differences in tweeter impulse from left to right side with only a simple 4khz crossover. IIRC, the results were 7ms apart. And setting the TA values to that was insane. Yes, that was the initial impulse time. I'm sure the aiming had something to do with it as well.


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## 14642 (May 19, 2008)

^^ OK. Now I understand and I also understand the misconception about the impulse response.

In the impulse response, the slope of the line that comprises the initial peak IS high frequency. We use the peak for a full range measurement because the slope is so steep. Along the horizontal axis, there is almost no difference between the beginning of the rise and the peak. In this case, picking the peak is simple and it's close enough.

Now, if we measure a low passed channel, the slope of the line isn't so steep because there's no high frequency. In this measurement, picking the peak is NOT close enough. 

So...here's the scoop. Picking the peak is NOT accurate. The first arrival is NOT the peak, it's the point at which the line BEGINS to rise. The problem that you are attempting to fix by delaying everything to the subwoofer's peak is not a problem. The problem is in interpreting the measurements you are making and applying that mistake to all low passed measurements by ASSUMING that your interpretation of the measurements is correct. 

Try this. IN most of these programs, the impulse response measurements are normalized. That means that the peak is set to the same value in all of the measurements. Find the peak. Then back up to the left until the level falls by some fixed decibel level. 24dB is good. 12 is sufficient for this exercise. You'll notice that you move much further to the left in the low passed measurement than you do in the full range measurement in which the line is steeper. This will get you much closer to the real arrival of sound in BOTH measurements because the arrival is when the line BEGINS to rise, not when it has completed the rise.


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## Woosey (Feb 2, 2011)

bikinpunk said:


> The slope or the frequency? Mine is based on frequency due to its relation with wavelength. Slope would only come in to play as a further means of calculating the median frequency.
> 
> I understand that arrival time is arrival time. But the result has a dramatic difference when you measure drivers WITH a crossover on them. Full range impulse is obviously different than band passed impulse. Measured subwoofer arrival time of first peak is outrageous compared to tweeters; 10-20+ ms ranges compared to as small as 1-2ms ranges. And opening up the LPF of the sub permits full first impulse spike sooner because of the wavelength. And that's where my sheet takes its cue from. It's an exercise; not an authority.
> 
> ...


This is also what I always thought... but I also know this is with sinewaves... with music it could be not that rigorous.. But that's what I learned from reading at the sengpiel website..


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## ecbmxer (Dec 1, 2010)

Found an error guys. Sorry. I misused the "floor" function in excel thinking it would round down when I converted the total phase shift to the shift in a single period (basically removing the extra 2*PI shifts). Let me see if I can fix it.


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## ErinH (Feb 14, 2007)

Woosey said:


> This is also what I always thought... but I also know this is with sinewaves... with music it could be not that rigorous.. But that's what I learned from reading at the sengpiel website..


But, music IS sine waves. All frequencies you hear are that frequency due to its wavelength and the wave is a sine wave. A single chord is a sine wave of some frequency (and it's harmonics). Distortion is still a sine wave... Just with an altered waveform. 

Andy, thanks for the reply. I'm coming back to that. I just need to get near a real keyboard.


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## ecbmxer (Dec 1, 2010)

bikinpunk said:


> And, here goes...
> 
> At the risk of total regret, I'm providing my own quirky time alignment sheet. In a nutshell, I'm just taking the wavelength at the crossover and applying that to the standard time alignment values you get from a tape measure and calculator. I've got multiple caveats with this method... but I've captured those internally (via numerous revisions). This has been 'vetted' in some ways by a few guys here. Some stuff worked. Some stuff didn't. Mostly, this method produces worthwhile results for long wavelength; not so much for high wavelength... and I've therefore adjusted my methods in the sheet to account for this. If you're good with excel math then you'll easily figure this out (hint: look at the equations). If you're not, don't bang your head on a wall trying to. It's pretty much as simple as my first sentence in this paragraph with some tweaks here and there.
> 
> ...


Cool! Thanks for posting Erin! I think it's fun to do these types of simple calculations and see how they may or may not work out in the car. I don't guarantee mine to work either, nor do I think it works out for higher wavelengths where very small delays produce many periods of phase shift. Probably doesn't matter there anyway.


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## ecbmxer (Dec 1, 2010)

OK, I fixed the issue I think. If anyone still wants to try it out, use this version. I think that floor function was crapping out with some distance values. It should have been truncate instead. It still gives me the same values that I found to work well in the car.

I should probably give better instructions for mine....


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## sqnut (Dec 24, 2009)

I'm glad I do the TA by ear. I've set it by ear and then measured via impulse response a few times. In all cases when I had the mids and sub well connected and sounding right, the mids were aligned much closer to the subs initial response than to the hump. 

I think that working on the eq and the TA go hand in hand in determining the overall response that works best. So one can work the eq a bit and then come back and tweak ta on the drivers in a very narrow range ~ +/- 0.04-0.06ms to see what gives the best overall response. When you tweak the TA in small increments, you hear the change in response. Sometimes TA helps you correct a response issue that you are unable to address at the eq. 

For me, TA is the last response tweak that gets rid of the slightly stretched sound and something that makes your sound complete and free flowing.


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## Kevin K (Feb 11, 2013)

Geez Erin I've been holding this like a prized egg and now it's out....
This sheet has really helped me a great deal in understanding a little more about what's going on in the car and why. (Thanks Erin for sharing!)



bikinpunk said:


> And, here goes...
> 
> At the risk of total regret, I'm providing my own quirky time alignment sheet. In a nutshell, I'm just taking the wavelength at the crossover and applying that to the standard time alignment values you get from a tape measure and calculator. I've got multiple caveats with this method... but I've captured those internally (via numerous revisions). This has been 'vetted' in some ways by a few guys here. Some stuff worked. Some stuff didn't. Mostly, this method produces worthwhile results for long wavelength; not so much for high wavelength... and I've therefore adjusted my methods in the sheet to account for this. If you're good with excel math then you'll easily figure this out (hint: look at the equations). If you're not, don't bang your head on a wall trying to. It's pretty much as simple as my first sentence in this paragraph with some tweaks here and there.
> 
> ...


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> The problem is in interpreting the measurements you are making and applying that mistake to all low passed measurements by ASSUMING that your interpretation of the measurements is correct.


To be fair, I hadn't actually built my calculations on real impulse measurements. I simply added the wavelength in time at a given frequency to the stock T/A equation, which was intended to more closely integrate the wavefronts. I also included 1/2 and 1/4 wave values in the sheet for those who want to toy around with it. I would like to use real measurements and see if there's a strong correlation but that would require more time than I have to donate to the task right now. I don't want people to make more of it than it is. Of course, the sheet wasn't exactly easy to throw together when I had to set conditions (that pioneer method sheet is a total PITA since it's not straight forward time based on distance). But, the premise is pretty easy: longer wavelengths are harder to integrate over minute time so longer/shorter delay relative to the subwoofer (and midbass) need larger delays in value to even hear a difference. I'm talking 3ms chunks. Not baby slices. 
That's what my sheet boils down to. I've found, if nothing else, it shows the impact that long wavelengths have on time alignment. It's actually worked quite well for some, based on recent feedback. 



Andy Wehmeyer said:


> Try this. IN most of these programs, the impulse response measurements are normalized. That means that the peak is set to the same value in all of the measurements. Find the peak. Then back up to the left until the level falls by some fixed decibel level. 24dB is good. 12 is sufficient for this exercise. You'll notice that you move much further to the left in the low passed measurement than you do in the full range measurement in which the line is steeper. This will get you much closer to the real arrival of sound in BOTH measurements because the arrival is when the line BEGINS to rise, not when it has completed the rise.


Every now and again, I mean to ask you how you guys determined T/A for subs in the MS-8. I think you may have just given me my answer. 

Most impulse response is given in (x)V on the y-scale rather than dB but I'd imagine there's a pretty simple conversion for this. This may solve it, just have to be mindful of the units: dB dBu dBFS dBV to volts audio conversion digital - calculator volt volts to dBu and dBV dB mW SPL dB decibels - convert dB volt normal decibels relatioship relation analog audio absolute level true rms convertor converter decibel to dbfs converter c

I'm curious how/why you guys came to this value, though. Was this based on experimentation or just some simple math. I also wonder how well it corelates to my notion of using 1/2 wave or 1/4 wave for the subwoofer.


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## ErinH (Feb 14, 2007)

Kevin K said:


> Geez Erin I've been holding this like a prized egg and now it's out....
> This sheet has really helped me a great deal in understanding a little more about what's going on in the car and why. (Thanks Erin for sharing!)


LOL.

Well, the reason I didn't want it floating around was because I didn't want people 'taking the ball and running with it', thinking that it was the end all method to get time alignment. Because it's not. And that's why I didn't start a new thread. It'll probably get lost in here in a day or so. 

but, at least it got some gears turning. So, maybe I can build it in to something a bit more robust and useful. Andy has given me some ideas.


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## Hanatsu (Nov 9, 2010)

bikinpunk said:


> Every now and again, I mean to ask you how you guys determined T/A for subs in the MS-8. I think you may have just given me my answer.


I thought MS8 didn't T/A the sub channels...?

Btw, thanks for the excel sheet, I'll try it out later


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## SPLEclipse (Aug 17, 2012)

ecbmxer said:


> OK, I fixed the issue I think. If anyone still wants to try it out, use this version. I think that floor function was crapping out with some distance values. It should have been truncate instead. It still gives me the same values that I found to work well in the car.
> 
> I should probably give better instructions for mine....


It's working! I've got Erin's settings loaded up (well, as best as I could given my maximum delay of 7.5ms) and I'm going to take a quick drive to the post office. When I get back I'll play around with your spreadsheet a little more and hopefully give it a shot tonight.


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## Kevin K (Feb 11, 2013)

Yes it does work. Will dig a little deeper after work with it, thanks for working on it and providing the tool.

Appreciate you and Erin along with others for sharing your thoughts and time and making the car audio environment better than ever.



ecbmxer said:


> OK, I fixed the issue I think. If anyone still wants to try it out, use this version. I think that floor function was crapping out with some distance values. It should have been truncate instead. It still gives me the same values that I found to work well in the car.
> 
> I should probably give better instructions for mine....


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## ErinH (Feb 14, 2007)

Hanatsu said:


> I thought MS8 didn't T/A the sub channels...?
> 
> Btw, thanks for the excel sheet, I'll try it out later


I think you're right, now that you mention it. I think Andy's point of contention was that the alignment didn't matter; it was the level matching that was important.


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## 14642 (May 19, 2008)

MS-8 doesn't TA the sub. 

It does, however, TA all the signals on the UN-EQ, including the sub and this is the method it uses. The reason that there's no real reason for super precision is that the frequency response aberration increases in frequency as the error gets smaller. All that is needed is to push the error to a frequency where the subwoofer signal is so low that it doesn't affect the sum. Thankfully this is self adjusting. If the subwoofer includes more high frequency, the hump is sharper and moves to the left (toward zero). That increases accuracy in cases where more accuracy is helpful.

For signal summing, whether electrical or acoustic, the sum is the result of combining the frequency response and the phase response. For a low pass filter to sum with a high pass filter according to some prescribed alignment, the frequency response AND the phase response have to follow the alignment. When you adjust delay, you change the phase response. When you EQ, you change the phase response. When you mount the speakers on a baffle at some distance, you change the phase response. This is why time aligning according to the electrical crossover makes absolutely no sense whatsoever. Electrical filters can be minimum phase. The acoustic signals you measure in the car are not. Home speakers and their networks are usually designed in an anechoic room. In a car, the room cannot be divorced from the speaker design because ALL of the reflections are near field reflections. 

Here's another wrench...what about EQ? What if you boost or cut near the crossover frequency. That changes the shape of your electrical filter. How do you calculate that? This is why all of this has to be done according to the acoustic signal. 

Don't get me wrong...I like all of the experimentation and the dedication to learning. My objective here is simply to steer you guys in a direction that will provide useful results rather than hypotheses based on misunderstandings and "proof" that's nothing more than subjective assessment.


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## Hanatsu (Nov 9, 2010)

Andy Wehmeyer said:


> If we have two speakers in an anechoic environment located different distances away, (as in a stereo system), then we can use delay on one channel to "fix" the arrival (distance) problem. Fixing the distance problem does two things. It eliminates the ENTIRE comb and centers the image at all frequencies. IF we use a phase EQ (all pass filter) to fix the phase according to the distance at the first dip in the comb, we fix it, but we leave the others intact. IF we reverse the polarity of one speaker to fix the comb, we simply shift the comb to different multiples. When you reverse the phase of one midrange in your car to fix the center vocal, you just screw it up somewhere else.


Andy, is there a way to build an active allpass filter that act like parametric filter to "fix" the biggest null of the comb? The allpass filters I've seen acts like a phase LP/HP filters with a slope. Guess I'm looking for a P-EQ without affecting magnitude of the filter or something. Can you explain this further? 

... this might be a really stupid question when I think about it =/


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## ErinH (Feb 14, 2007)

^ get a Lake DSP. :/


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## 14642 (May 19, 2008)

Hanatsu said:


> Andy, is there a way to build an active allpass filter that act like parametric filter to "fix" the biggest null of the comb? The allpass filters I've seen acts like a phase LP/HP filters with a slope. Guess I'm looking for a P-EQ without affecting magnitude of the filter or something. Can you explain this further?
> 
> ... this might be a really stupid question when I think about it =/


Sure, sum a 2nd order HP and a 2nd order LP. That's what we did in Infnity's Basslink4SC amplifier. Worked great. Didn't sell.


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## Woosey (Feb 2, 2011)

bikinpunk said:


> But, music IS sine waves. All frequencies you hear are that frequency due to its wavelength and the wave is a sine wave. A single chord is a sine wave of some frequency (and it's harmonics). Distortion is still a sine wave... Just with an altered waveform.
> 
> Andy, thanks for the reply. I'm coming back to that. I just need to get near a real keyboard.


I get your point, but what I mean is: Music is a very complex sum of all the sinewaves.. so can you treat it like a single sinewave? How about synth music with saw-tooth waves block waves you name it... 

AND

This only applies to the crossover region... before or after that it is what is always was... no lead or lag... right?


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## 14642 (May 19, 2008)

^^Right. And yes, for the purposes of EQ, Crossover, etc, you can treat it like a sine wave. It's just an alternating current signal.


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## Woosey (Feb 2, 2011)

Andy Wehmeyer said:


> ^^Right. And yes, for the purposes of EQ, Crossover, etc, you can treat it like a sine wave. It's just an alternating current signal.


Ok, thank you!

So it would be like a shallow crossover you do not create a big mess of the xo region but you make it "longer" ( shallow slope = bigger xo region )
And with a steep crossover you create a bigger mess of the xo region but "smaller" ? ( steep slope = smaller xo region )

and then comes the question of how do you like it; a bit messy for a long time or very messy in a shorter period of time.. ? 

I know I shouldn't use time in that sentence but I hope you know what I mean..


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## Woosey (Feb 2, 2011)

Andy Wehmeyer said:


> ^^ OK. Now I understand and I also understand the misconception about the impulse response.
> 
> In the impulse response, the slope of the line that comprises the initial peak IS high frequency. We use the peak for a full range measurement because the slope is so steep. Along the horizontal axis, there is almost no difference between the beginning of the rise and the peak. In this case, picking the peak is simple and it's close enough.
> 
> ...


And thank you very much for clearing this up! Make a lot of sense now I think about it..

Question:

Some impulses start with a small spike and then go larger into the large spike... Should we use the very beginning of the first small one or the beginning of the large spike? 

Like here :


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## ErinH (Feb 14, 2007)

Assuming that's a single drive unit, that looks like pre ringing. I'd say go with the largest spike. And in that case, it's (absolute) polarity is wrong.


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## Woosey (Feb 2, 2011)

bikinpunk said:


> Assuming that's a single drive unit, that looks like pre ringing. I'd say go with the largest spike. And in that case, it's (absolute) polarity is wrong.


Just a pic I stole from the web.. not my own measurement.. 

I just used it as an example..

I have my laptop with my measurements at home..


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## 14642 (May 19, 2008)

Woosey said:


> Ok, thank you!
> 
> So it would be like a shallow crossover you do not create a big mess of the xo region but you make it "longer" ( shallow slope = bigger xo region )
> And with a steep crossover you create a bigger mess of the xo region but "smaller" ? ( steep slope = smaller xo region )
> ...


THe assumption that a steeper slope makes a bigger mess isn't accurate. I prefer the steeper slopes and smaller xover region in cars.


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## 14642 (May 19, 2008)

Woosey said:


> And thank you very much for clearing this up! Make a lot of sense now I think about it..
> 
> Question:
> 
> ...


Well, in cases like this, it's necessary to think a bit about what's going on. The small humps to the left look like pre-ringing to me, but because the initial peak happens at 0, Erin's suggestion might also be correct. This doesn't look like an acoustic measurement because of that. I have seen acoustic measurements that look similar, but everything is offset to the right (flight time of sound through the air). IN those cases, the two big peaks could be many things. The first spike might be the initial sound. The next one could be the sum of the initial sound and an adjacent reflection (constructive interference), which would explain the greater magnitude. If that's the case, then it may be helpful to determine which one is the apparent source of the sound and time align accordingly. You could check this by measuring from the dustcap to the microphone and determining whether that peak corresponds to that distance. Then with your tape measure and a calculator, you could go looking for the source of the reflection.

It's useful to note that if the initial spike is narrower than the second one, it means that there is less high frequency content in the second spike and that would support my suggestion that this is caused by the sum of the original and a reflection (at a lower frequency). THe corrected response (the rede line) looks like a real world measurement of a nearly perfect impulse response (with pre and post ringing).


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## Woosey (Feb 2, 2011)

Andy Wehmeyer said:


> THe assumption that a steeper slope makes a bigger mess isn't accurate. I prefer the steeper slopes and smaller xover region in cars.


Ok I assumed due to the electric phase shift of 45* ( 6db/oct ) vs 180*(24db/oct )degrees there would be more lag or lead in the xo region in the steeper sloped version..


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## ErinH (Feb 14, 2007)

Andy Wehmeyer said:


> Well, in cases like this, it's necessary to think a bit about what's going on. The small humps to the left look like pre-ringing to me, but because the initial peak happens at 0, Erin's suggestion might also be correct. This doesn't look like an acoustic measurement because of that. I have seen acoustic measurements that look similar, but everything is offset to the right (flight time of sound through the air). IN those cases, the two big peaks could be many things. The first spike might be the initial sound. The next one could be the sum of the initial sound and an adjacent reflection (constructive interference), which would explain the greater magnitude. If that's the case, then it may be helpful to determine which one is the apparent source of the sound and time align accordingly. You could check this by measuring from the dustcap to the microphone and determining whether that peak corresponds to that distance. Then with your tape measure and a calculator, you could go looking for the source of the reflection.
> 
> It's useful to note that if the initial spike is narrower than the second one, it means that there is less high frequency content in the second spike and that would support my suggestion that this is caused by the sum of the original and a reflection (at a lower frequency). THe corrected response (the rede line) looks like a real world measurement of a nearly perfect impulse response (with pre and post ringing).


Throw this out there, too...

The reason why I said "assuming that's a single drive unit" is it could even be a speaker with 2 drivers' initial impulse showing up. Like a mid and tweeter. Possibly.


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## Hanatsu (Nov 9, 2010)

Woosey said:


> Ok I assumed due to the electric phase shift of 45* ( 6db/oct ) vs 180*(24db/oct )degrees there would be more lag or lead in the xo region in the steeper sloped version..


Inaudible imo. A 24dB L-R filter adds 0,48ms group delay. The most sensible part of our hearing (upper midrange) is able to detect 1-2ms at best. 



> It is not unusual for even a vented box to have a group delay of perhaps 20-30ms at the bottom end, and while a tad shy of a day, it's still quite a long time in audio reproduction. By comparison, a 24dB/octave Linkwitz Riley crossover network has a group delay of 480us (see table).


Phase, Time and Distortion in Loudspeakers


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## 14642 (May 19, 2008)

bikinpunk said:


> Throw this out there, too...
> 
> The reason why I said "assuming that's a single drive unit" is it could even be a speaker with 2 drivers' initial impulse showing up. Like a mid and tweeter. Possibly.



Yeah, if it was an acoustic measurement, that could be true too.


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## 14642 (May 19, 2008)

Woosey said:


> Ok I assumed due to the electric phase shift of 45* ( 6db/oct ) vs 180*(24db/oct )degrees there would be more lag or lead in the xo region in the steeper sloped version..


Abandon the assumption that phase shift in the crossover is an unwanted condition. We hear the SUM of the shifts of the adjacent bands. One leads and the other lags, so the sum is 0, in a perfect alignment. This is what 4th order LR filters attempt to do. If the sum is flat in phase and frequency response, then the sum is flat in phase and frequency response and that's what we hear. This is how home speakers are designed in anechoic rooms. IN a car, we can't fix both because we hear the sum of the speakers AND the sum of the speakers and the reflections. You can't fix all of those phase errors in a car because you can't EQ (phase or time) the reflections separately. Just get the frequency response right and attempt to align the first arrival from each of the speakers. 

The all pass phase inverting filter is useful in some cases--mostly when there's a big overlap between one band and another. For fixing the center vocal image, time alignment works better for one seat. The phase filter can improve things for both seats but neither will be correct.


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## sqnut (Dec 24, 2009)

Andy Wehmeyer said:


> Just get the frequency response right and attempt to align the first arrival from each of the speakers.


That is sig material. ^^For everyone who wants to know how to get their cars sounding good. Of course the moot point is that the right frequency response is 'as heard' . 

A significant part of sound is subjective. Not in the 'different sounds for different folks' vein, but in that you tune by picking the difference between 'better' and 'worse' with good accuracy, as you tweak. You'll inch your way towards the right response and know when you're making progress. You can try and measure your way way by starting with a house curve, but you will need to tweak within this to get it sounding right and again you're hearing your way there.


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## Woosey (Feb 2, 2011)

Andy Wehmeyer said:


> Abandon the assumption that phase shift in the crossover is an unwanted condition. We hear the SUM of the shifts of the adjacent bands. One leads and the other lags, so the sum is 0, in a perfect alignment. This is what 4th order LR filters attempt to do. If the sum is flat in phase and frequency response, then the sum is flat in phase and frequency response and that's what we hear. This is how home speakers are designed in anechoic rooms. IN a car, we can't fix both because we hear the sum of the speakers AND the sum of the speakers and the reflections. You can't fix all of those phase errors in a car because you can't EQ (phase or time) the reflections separately. Just get the frequency response right and attempt to align the first arrival from each of the speakers.
> 
> The all pass phase inverting filter is useful in some cases--mostly when there's a big overlap between one band and another. For fixing the center vocal image, time alignment works better for one seat. The phase filter can improve things for both seats but neither will be correct.


Thank you very much sir.. 

Where would we be without someone like you..


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## ultimatemj (Jan 15, 2009)

Interesting thread!

If we go back to the OP's stated intent "_*calculating time alignment to make sure things remain in phase at the crossover points*_"...

And combine a quote from the  link Hanatsu referenced



> the tonal structure of a sound does not rely on the phase integrity of the received sound, only the relative amplitudes of the fundamental and harmonics. So a speaker that has perfectly flat frequency response but is not 100% phase coherent will sound the same as one that is also flat, but totally phase coherent.


With Andy's comment 


> IN a car, we can't fix both because we hear the sum of the speakers AND the sum of the speakers and the reflections. You can't fix all of those phase errors in a car because you can't EQ (phase or time) the reflections separately. *Just get the frequency response right and attempt to align the first arrival from each of the speakers.*


Then it appears there is little reason to put effort into trying to predict/calculate perfect phasing (in excel).
​Toping this with Andy's comment on the essque vs danger 6db slopes thread makes a great car audio Zen saying


> *The only crossover and the only phase shift we hear is the ACOUSTIC response.
> Just get the frequency response right and attempt to align the first arrival from each of the speakers.*


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## Hanatsu (Nov 9, 2010)

ultimatemj said:


> Then it appears there is little reason to put effort into trying to predict/calculate perfect phasing (in excel).


Even so... T/A is important for correct imaging in the lows. "Incorrect T/A" will lead to tonality issues as it affects the comb pattern, even at higher frequencies.


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## ultimatemj (Jan 15, 2009)

Agreed, just saying I now understand why trying to get phase perfect via T/A calculations in excel is not on the top of the priority list 

Regarding combing and phasing, here's another memorable Andy quote from the Flat Response thread.


> The first rule of equalization is that you can't put energy back into a null caused by destructive interference in the acoustic system. You can remove energy from a peak caused by constructve interference.
> 
> The second rule is that high-Q (narrow) peaks and dips are less audible than low-Q peaks and dips.
> 
> ...


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## 14642 (May 19, 2008)

Hanatsu said:


> Even so... T/A is important for correct imaging in the lows. "Incorrect T/A" will lead to tonality issues as it affects the comb pattern, even at higher frequencies.


Only if both speakers are playing the same thing. If we're talking about comb filtering between a 6" and a 4" both crossed at 400Hz with steep slopes. There won't be much of a comb because the 6" will contribute VERY little output at high frequency to be incorrectly summed with the information that comes from the 4".

That's why we use crossovers!


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## jriggs (Jun 14, 2011)

bikinpunk said:


> And, here goes...
> 
> At the risk of total regret, I'm providing my own quirky time alignment sheet. In a nutshell, I'm just taking the wavelength at the crossover and applying that to the standard time alignment values you get from a tape measure and calculator. I've got multiple caveats with this method... but I've captured those internally (via numerous revisions). This has been 'vetted' in some ways by a few guys here. Some stuff worked. Some stuff didn't. Mostly, this method produces worthwhile results for long wavelength; not so much for high wavelength... and I've therefore adjusted my methods in the sheet to account for this. If you're good with excel math then you'll easily figure this out (hint: look at the equations). If you're not, don't bang your head on a wall trying to. It's pretty much as simple as my first sentence in this paragraph with some tweaks here and there.
> 
> ...


Hi Erin,

I have been entering values into you calculator (Pioneer) and when I follow this instruction: *If you are NOT using a 3-way front stage (dedicated midrange) simply type in a 0 (zero) for the Left & Right Midrange values, I get "FALSE" as the Delay Time for the Midbass. 

Any idea what is going on?

Thanks.

-Jason


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## Sulley (Dec 8, 2008)

jriggs said:


> Hi Erin,
> 
> I have been entering values into you calculator (Pioneer) and when I follow this instruction: *If you are NOT using a 3-way front stage (dedicated midrange) simply type in a 0 (zero) for the Left & Right Midrange values, I get "FALSE" as the Delay Time for the Midbass.
> 
> ...


Sub'd for answer, I'm also getting this issue. I manually converted the values from the other spreadsheet but I'm not 100% sure what I did was correct.


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## vetteboy3 (Mar 31, 2012)

Thanks for all the info I'll have to play around with it a bit


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## ErinH (Feb 14, 2007)

stockley.rod said:


> Sub'd for answer, I'm also getting this issue. I manually converted the values from the other spreadsheet but I'm not 100% sure what I did was correct.


I may have goofed in the calculation on that sheet with a conditional. More than likely, I have the condition in the midbass rather than the midrange, where it should be. I'll have to check when I'm near the PC.


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