# midbass arrays revisited



## lycan

This is going to be one long thread. Inspired by the interesting work that Patrick has been doing  Lots to cover, for anyone interested. We're going to end up covering : the fundamentals of Blumlein's stereo, vector math, a variety of already-marketed technologies including SRS and audiocontrol ESP2 & 3, and (i hope) ultimately the pros & cons of midbass drivers located _all over_ the vehicle. Maybe a definitive answer to the age-old questions : can i put midbass drivers _behind_ me? How about an array of midbass drivers ... some in _front_, and a few _behind_?

We gotta start with some definitions. And what better place to start than a definition of MIDBASS. For the purposes of this thread, we're gonna use :

1. higher in frequency than the "pressure zone" of the car. This means wavelengths SHORTER than the interior dimensions of a vehicle.

2. lower in frequency than the IID (inter-aural intensity or level difference, aka ILD) range of human hearing. This means wavelengths LONGER than the distance between human ears.

These two constraints boil down to a frequency range above about 80 Hz, and below about 1600 Hz. But i'm going to narrow this down farther, and put an upper limit to midbass at about 320 Hz (i like to think in octaves) ... mostly because the transition from ITD to IID is not abrupt in human hearing, and I want to restrict this discussion to frequencies that are WELL WITHIN the ITD range. So let's avoid _midrange_ frequencies, and stick to _midbass_ strictly (yes, i'm making the STRONG distinction between _midrange_ and _midbass_).

Next up : a quick review of how we localize. Been discussed many times, but a quick review is in order  We'll better explain the terms "ITD" and "IID".


----------



## ncv6coupe

I've been waiting for this thread forever!!! And since me and PB has the same exact model car his measurements and research helps 2 tons! Sub'd. Looking forward to this discussion to come. Thanks Lycan, PB, and whoever else chimes in.


----------



## sqshoestring

Count me in, if Richard Clark can do it....


----------



## lycan

*HOW HUMANS LOCALIZE SOUND SOURCES*

*1. Lateral plane*

Humans have two ears that are displaced _laterally_. Those two ears can be used to localize sources _laterally_, by "measuring" and "comparing" the sound that arrives at each ear. The brain does this automatically, through learned behavior. What characteristics does a sound signal have, that might be _different_ at each ear? There are two : intensity (or level) and time. So we can speak of a difference in acoustic INTENSITY (or level) between the two ears, and we can speak of an acoustic arrival TIME difference between the ears.

*a. Lateral plane, low frequencies*

Let's assume that the sound source is significantly father from our head than the distance between our ears. Let's define "low frequencies" as frequencies with wavelengths substantially _longer_ than the width of our head (which happens to coincide nicely with the distance between our ears). In this case, the _intensity_ difference between our ears will be very small, for any source in the lateral plane. Our head will not "shadow" these long wavelengths, nor will the head cause any significant diffraction. So the brain can't use inter-aural intensity difference to locate these long wavelengths 

Instead, the brain has learned to measure and use inter-aural TIME differences for these long wavelengths. The "arrival time" differences can be thought of as a time difference in the pressure peaks that arrive at each ear.

This frequency range is what we are calling MIDBASS. And here, localization is purely a function of _inter-aural time difference_, or ITD ... nothing more, nothing less.

*b. Lateral plane, higher frequencies*

As frequencies increase, the wavelengths will shorten ... eventually approaching the distance between our ears (about 8 inches, which corresponds to about 1600 Hz). As soon as more than one wavelength can "fit" between our ears, the inter-aural time difference ... think of it as a time difference between pressure peaks ... becomes confusing and ambiguous. IF there's more than one pressure peak between our ears, which one should the brain measure? So ITD becomes useless as a localization cue, as frequencies increase.

Fortunately, at these same wavelengths ... on the order of our head dimension ... the head itself starts to play a role in shaping the amplitude or intensity (or level) difference between our ears. Head shadowing & diffraction start to become noticeable. So the ear/brain can start using inter-aural intensity (or level) differences, or IID, at frequencies higher than the wavelength corresponding to the distance between our ears.

So, in the upper midrange, we see IID as the dominant localization.

At even higher frequencies, into the treble, ANOTHER physical dimension becomes significant : our outer ears  They shape the magnitude as well, contributing to IID.

This whole frequency region ... where IID rules ... is the realm of HRTF (or head related transfer functions). HRTF's shape the magnitude & phase of frequencies high enough to be impacted by head & ear dimensions.

But this frequency region doesn't concern us in this thread. Midbass is ALL about ITD 

*2. Vertical plane*.

Our ears are on the SIDES of our head, remember? The fact that we have two of them, displaced _laterally_ by about 8 inches, is insignificant and useless for vertical localization. So what allows us to localize vertically ... anything? Well, our heads are almost spheres, with holes in the sides ... so there's nothing about _this_ shape that "looks different" as a source "moves" up vertically (for example). What does "look different", to a source up high versus down low, is the shape of our outer ears ... and they pretty much look the same. It's really only our outer ears that spoil the vertical symmetry of our hearing, which allows us to localize vertically. And given the size of our outer ears, it's only frequencies above about 3kHz that can be located vertically.

Bottom line : height cues are treble, and height cues are mono.

This is important as we dive a bit deeper into midbass ITD. Our next goal is to define those points in space that generate the same ITD for a source at any of those points. In other words, we need to define a geometric shape or curve with the following characteristic : the location of a source at ANY point on this curve will be indistinguishable from any OTHER point on this curve.


----------



## otis857

Im in. I'll be the old guy in the back of the class. Plus, Im still trying to get a weigh in on why my friends mid bass drivers mounted in the floor, at the front edge of the seats, angled towards the dash sounded so good. I posted this ? on one of PB's threads and got no responses.


----------



## 2fnloud

So any freq over 3kHz could be mono and we would not be able to tell the difference?


----------



## lycan

2fnloud said:


> So any freq over 3kHz could be mono and we would not be able to tell the difference?


veritcally yes, laterally no

IID (and other HRTF stuff) allows localization through the treble, including freqs above 3kHz, in the LATERAL plane.

In the VERTICAL plane, everthing is mono. And nothing is localizable below about 3kHz.


----------



## 2fnloud

I remember reading that the human ear can not locate the vertical source of sound unless that source is moving up and down while it is making a sound.

Is this true?


----------



## Fast1one

2fnloud said:


> I remember reading that the human ear can not locate the vertical source of sound unless that source is moving up and down while it is making a sound.
> 
> Is this true?


I would imagine this is only true for frequencies below 3khz. That is very complex question that I honestly don't know how to answer. 

Great idea on this thread. I'm still convinced that placing midbass drivers that are on average 6db+ larger in intensity then the "satellite" woofers will work as long as you make some attempt to match the pathlength for each channel. Two midbass drivers per side in the kickpanels along with one satellite per channel placed at the same distance (say the rear deck) will have a 6db differential if ran in parallel or series. You can use an L-pad on the satellite woofer to attenuate it more if needed.


----------



## lycan

2fnloud said:


> I remember reading that the human ear can not locate the vertical source of sound unless that source is moving up and down while it is making a sound.
> 
> Is this true?


no. Easy to imagine an experiment to prove it, or disprove it. My hypothesis is this : height cues exist in the treble only, but don't require a moving source. Can anyone think of a simple experiment to prove or disprove my hypothesis? You'll need a two-way bookshelf loudspeaker, and a blindfold. Maybe handcuffs and whipped cream too ...

However ... we don't want to derail a thread about midbass _too much_ 

Midbass is all about ITD. We need to see what that means next, by identifying those points in space that ALL have the same ITD between our ears. This will be a big stepping stone in the analysis of midbass ARRAYS


----------



## lycan

Fast1one said:


> I would imagine this is only true for frequencies below 3khz. That is very complex question that I honestly don't know how to answer.
> 
> Great idea on this thread. I'm still convinced that placing midbass drivers that are on average 6db+ larger in intensity then the "satellite" woofers will work as long as you make some attempt to match the pathlength for each channel. Two midbass drivers per side in the kickpanels along with one satellite per channel placed at the same distance (say the rear deck) will have a 6db differential if ran in parallel or series. You can use an L-pad on the satellite woofer to attenuate it more if needed.


don't jump the gun yet playboy  we got a few ideas to develop first, before we decide if that's a good idea, or a bad one


----------



## Fast1one

lycan said:


> don't jump the gun yet playboy  we got a few ideas to develop first, before we decide if that's a good idea, or a bad one


I have ZERO patience 

Do continue sir, I am sure you have a good plan in store for us.


----------



## lycan

*THE CONE OF CONFUSION*

To an investor (or accountant), time is _money_. To an acoustician, time is _distance_.

Sound travels at a velocity of about 1130 ft/sec (in air, room temp, average humidity). All frequencies within the range of human hearing travel at this same speed. So a _time delay_, in this sense, is identical to a separation in _distance_. Make sense? Now an acoustic problem becomes a simple geometry problem 

In this post we are interested in determining all points in space that would generate identical ITD's between our ears. An identical question is this :

_Pick a point in space. That point has some distance to the left ear. It has some distance to the right ear, probably (but not necessarily) different. Let's set the difference between these distances = *d*. What other points in space have that same difference in distance between the ears?_

That's what this post is going to answer  *Once we identify that set of points, we know that you can put a midbass driver at ANY of those points, and our ears won't be able to tell one point from the other.* It's ESSENTIAL that this point be understood. Any intelligent midbass array must recognize ... and maybe exploit ... this relationship.

We're going to start in two dimensions (the extrapolation to the third will be trivial). Our 2-D plane will be one the lateral, horizontal plane that intersects both our ears when we are in a sitting or standing position. So pull out some graph paper, and draw three points :

1. Left ear : put it at the fixed x,y coordinates of *(-e/2, 0)*
2. Right ear : put it at the fixed x,y coordinates of *(+e/2, 0)*

The distance between our ears is therefore = *e* 

3. Source location : pick a point in the upper right quadrant, label it *(x,y)*.

The distance from our source at *(x,y)* to our left ear at *(-e/2, 0)* is given by :

*(dl)^2 = (x + e/2)^2 + y^2*

and the distance from our source at *(x,y)* to our right ear at *(+e/2, 0)* is given by :

*(dr)^2 = (x - e/2)^2 + y^2*

If you don't understand how those are derived, you should be ashamed of yourself  Pythagoras could have done it ... twenty-five centuries ago 

The real parameter we're interested in, is the _difference_ between these distances :

*d = dl - dr*

Who wants to write down, simplify and identify the function, or curve, that relates *(x,y)* as a function of the two parameters *e* and *d*? There's some algebra involved, but that's it. No need to argue about what amplifier is better or what sub hits harder ...

I'll give the answer after you guys chew on it for a while. But then, i'm going to ask someone with better internet & software skills to draw it


----------



## lycan

i'm impatient too 

The relationship that emerges is the classic conic section called the _hyperbola_ :

*1 = x^2/a^2 - y^2/b^2*

where, in our case :

*a^2 = d^2/4*
*b^2 = (e^2 - d^2)/4*

and *|d| <= |e|*

If you draw this curve on our x,y graph ... with left ear at *(-e/2,0)* and right ear at *(+e/2, 0)* ... you will have identified all points in a 2-D plane that generate the same ITD's  

*A mibass driver placed ANYWHERE on this curve, will be sonically indistinguishable from a midbass driver anywhere ELSE on the curve.*

This is important. Important enough to draw. Some body draw this for us  Let *e = 8* (since our ears are 8 inches apart, more or less) and draw the hyperbola for a few values of *d* ... say *d = 1, 2, 4, 7*


----------



## justinmreina

lycan,
I am impatient too so: When do we get to reflections??

-Justin


----------



## lycan

justinmreina said:


> lycan,
> I am impatient too so: When do we get to reflections??
> 
> -Justin


good question. the answer is ... later. gotta walk before we can run.

So far, and for the near future, we are dealing with midbass drivers out in free space, far from any reflective surfaces.

Of course, once we understand this scenario, it's easy to understand reflections  We'll just use "image theory" to place phantom drivers _behind_ the reflective surfaces, then imagine the reflective surfaces _gone_. The principle of superposition applies, so the combined response will be the addition of the individual responses: real plus phantom.

But ... one step at a time. Let's understand a SINGLE midbass driver, in free-space, first. Then, we'll apply some vector math to describe the genius of Blumlein's stereo ... that is, the combined response of TWO midbass drivers. At this point we'll have a firm foundation to understand midbass arrays, and how their responses can be manipulated.


----------



## Niebur3

Great thread...looking forward to some fine reading!


----------



## justinmreina

alright fine, well I'm subscribed. And somewhat patient 

-Justin


----------



## Fast1one

lycan said:


> i'm impatient too
> 
> The relationship that emerges is the classic conic section called the _hyperbola_ :
> 
> *1 = x^2/a^2 - y^2/b^2*
> 
> where, in our case :
> 
> *a^2 = d^2/4*
> *b^2 = (e^2 - d^2)/4*
> 
> and *|d| <= |e|*
> 
> If you draw this curve on our x,y graph ... with left ear at *(-e/2,0)* and right ear at *(+e/2, 0)* ... you will have identified all points in a 2-D plane that generate the same ITD's
> 
> *A mibass driver placed ANYWHERE on this curve, will be sonically indistinguishable from a midbass driver anywhere ELSE on the curve.*
> 
> This is important. Important enough to draw. Some body draw this for us  Let *e = 8* (since our ears are 8 inches apart, more or less) and draw the hyperbola for a few values of *d* ... say *d = 1, 2, 4, 7*


I was going to but I need to install matlab on my computer again first. Let me see if I can find a student copy 

Edit: Student version is $99 which I don't have right now. I believe my roommate has a copy. But if he doesn't then I'll do it on campus tomorrow if it's not done


----------



## lycan

Fast1one said:


> I was going to but I need to install matlab on my computer again first. Let me see if I can find a student copy
> 
> Edit: Student version is $99 which I don't have right now. I believe my roommate has a copy. But if he doesn't then I'll do it on campus tomorrow if it's not done


dude you'll be my hero.

I really think a 2-D picture will help visualize what we're talking about. The extension to 3-D will be trivial 

Again ... we're searching for ALL points in space where a midbass driver can be located, and sound identical to any other point. Interesting in itself, don't ya think? This point alone should get some brain cells moving about midbass locations and arrays ...


----------



## justinmreina

You must have gauged how much I am not enjoying my HW right now...

*Curves*









*Path Values on d=4*









*Code...*
MATLAB code if you want to play with it...

-Justin


----------



## lycan

Nice work Justin ... and a big thanks! Don't let that stop anyone else from plotting 

Look at that *d=4* curve (with the highlighted point). Put a midbass driver on that point, and listen. Now put a midbass driver ANYWHERE ELSE on that same curve ... the ear can't tell the midbass was moved (assuming that amplitude, or level, is appropriately adjusted for long distance). Why? All points on that curve generate the SAME ITD  And in the midbass, it's _all_ about ITD for localization.

Assume the front of the head faces upward in this plot. Notice how the curve extends forward, in front of the head ... as well as backward, BEHIND the head. Any comments about that? If you're looking for _clues_ about how a certain (in)famous Buick GN was able to get away with midbass drivers in the REAR quarter panel, you've just found your first BIG one 

Let's extend the curve to 3-D space, as promised. Given the symmetry of our heads, I submit that the 3-D "surface" is easily generated by ROTATING the hyperbola around the x-axis. ANY points on this SURFACE will generate the same ITD's, so midbass drivers ANYWHERE on this surface will be sonically indistinguishable.

What does the resulting surface look like? Simple : a CONE. Hence, the "*cone of confusion*" for midbass drivers. In the past, I've offered a simplified version ... or "section" of this cone to describe midbass "confusion". If you "slice" the cone anywhere parallel to the y-axis, you'll find the "circle of confusion".

Next up: intro to vector math, for describing ITD's. This will alow us to more easily analyze _multiple_ midbass drivers. And we'll start with the smallest number greater than one ... namely, two ... as we reveal the genius of Blumlein stereo.


----------



## Fast1one

justinmreina said:


> You must have gauged how much I am not enjoying my HW right now...
> 
> -Justin


I felt the same way, but I didn't have matlab to do it  Good work Justin.

P.S. I have revised my original proposition, but I won't post it and let Mr. Lycan continue forward


----------



## m3gunner

OK... I took a shot at this using the built-in visualizer in Mac OS X. Friggin' thing uses e as a constant, so I changed it to k.

I suck at this stuff, so I'm hoping this is close...

In any case, I'm trying to follow along on this journey...


----------



## lycan

m3gunner said:


> OK... I took a shot at this using the built-in visualizer in Mac OS X. Friggin' thing uses e as a constant, so I changed it to k.
> 
> I suck at this stuff, so I'm hoping this is close...
> 
> In any case, I'm trying to follow along on this journey...


Looks pretty damn good to me!

We're drawing curves of "constant ITD" for midbass drivers  Seems like you guys are following along just fine! Beats the snot outa arguing about amp sonics, doesn't it?

Rotate those curves around the x-axis, and imagine the resulting "cones of confusion". Midbass drivers ANYWHERE on the surface of each cone are sonically indistinguishable ... at least, until you turn your head  But that's a complication that will be discussed later as well.

Vectors ... or sometimes called phasors ... next.


----------



## quality_sound

This is the best thread ever. Even better than the rack and ass threads.


----------



## Fast1one

quality_sound said:


> This is the best thread ever. Even better than the rack and ass threads.


"You know you're an audio nerd when..."


----------



## quality_sound

I guess the difference is I already know how to use racks and ass properly.


----------



## sqshoestring

Here is a little reading: http://www.icad.org/websiteV2.0/Conferences/ICAD2002/proceedings/81_MasayukiMorimoto.pdf


----------



## ErinH

lycan said:


> And given the size of our outer ears, it's only frequencies above about 3kHz that can be located vertically.


So, for clarification, you are basing this all on entire wavelength rather than 1/4 wavelength? 
I often see the 1/4 wavelength being cited for most audio phenomena (lobing vs. crossover points). I'm just curious why the inconsistencies. 

4.5" = ~ 3khz. 
Is this 4.5" supposed to be the total height of the ear from lobe to top?
My ear isn't quite 4.5" tall.


----------



## lycan

bikinpunk said:


> So, for clarification, you are basing this all on entire wavelength rather than 1/4 wavelength?
> I often see the 1/4 wavelength being cited for most audio phenomena (lobing vs. crossover points). I'm just curious why the inconsistencies.
> 
> 4.5" = ~ 3khz.
> Is this 4.5" supposed to be the total height of the ear from lobe to top?
> My ear isn't quite 4.5" tall.


yeah i wasn't very precise ... but, it's not a "step function" of height recognition either. In other words, it's not like the ear/brain will be completely deaf to height cues at 2.6kHz, but very cognizant of height cues at 2.7kHz.

So pick a frequency between 2kHz and 4kHz, and we'll say that height cues become significant above that  Or, even better, do some simple experiments with a driver of your choice and post _your_ results ... just don't let your _eyes_ influence your _ears_ 

And for the purposes of this thread, we can safely say that there are no height cues in the midbass 

EDIT: the real message is that there are some significant physical dimensions involved, when we're listening to audio in a car. A few of the key ones (not a comprehensive list) are : interior dimensions of the vehicle, distance between our ears, size of outer ears. And "interesting" things happen when the acoustic wavelengths become _approximately_ equal to these dimensions. For our purposes here, midbass frequencies will have wavelengths _less_ than interior vehicle dimensions, but _greater_ than head or ear dimensions.


----------



## ErinH

No, I understood what you meant and I didn't mean to imply that I thought you were talking exact measurements here. I was actually wondering what region of the ears cause the effect... ie: does it have to do with curvature of the upper ear area, or is it basically attributed to the height? Does the canal, or pinnae have a specific influence, or is it very minimal and neglected in regards to the ear?

Also, not sure if you caught my question about the wavelengths, or if your answer above was directed at that. 

Thanks for your thread here. I'm familiar with ITD and ILD, but I really like how you gave examples of the relations they have to the geometry of the head/ear. 
Looking forward to more on this topic and HRTF.


----------



## lycan

I think it's fair to say that various acoustic wavelengths (and their corresponding frequencies, naturally) ONLY have significance in relation to the _physical dimensions_ of whatever is sensing or measuring them.

Cases in point (summary) :

- pressure zone of car transitions when wavelengths about equal to interior car dimensions

- ITD/IID transition when wavelengths about equal to head width (or distance between ears)

- outer ears the only mechanism that can "filter", and thereby "distinguish", height cues

For elephants, the critical frequencies would be different  Can you imagine a car for elephants? I think I saw one at a circus once ...


----------



## ErinH

lycan said:


> Can you imagine a car for elephants? I think I saw one at a circus once ...


Oddly enough, the cars are scales of magnitude smaller for elephants than humans. Very hard for me to base any logical discussion off of this. LOL. 
I kid, I kid...



Revisiting 1/4 vs. Full wavelength... Is there any specific reason why the two are used interchangeably, or am I just missing what you're saying? 
Ie: You're using full wavelength in proportion to our physical dimensions. Talking about combing, etc, I see 1/4 wavelength as the measuring basis. 

I'm trying to stay OT. I swear!


----------



## lycan

bikinpunk said:


> Oddly enough, the cars are scales of magnitude smaller for elephants than humans. Very hard for me to base any logical discussion off of this. LOL.
> I kid, I kid...
> 
> 
> 
> Revisiting 1/4 vs. Full wavelength... Is there any specific reason why the two are used interchangeably, or am I just missing what you're saying?
> Ie: You're using full wavelength in proportion to our physical dimensions. Talking about combing, etc, I see 1/4 wavelength as the measuring basis.
> 
> I'm trying to stay OT. I swear!


nah I'm just not being very specific, that's all. What's a factor of four among friends? LOL It's probably close-to-accurate to suggest that a height-related effect "begins" when 1/4 wavelength equals an outer-ear dimension, and is in full effect when a full wavelength equals an outer-ear dimension. So, if we have to pick a specific point, let's just say 1/2 wavelength 

Now, if you're going to ask _which_ outer-ear dimension : tip-top to bottom-lobe, or top-crease to ear-ring ... well i'm just going to have to ignore you


----------



## loddie

Thread of the decade. Thanks Lycan!

I understand the concepts presented so far in this thread. However, I'm having a hard time understanding width when "wider is better" for midbass placement. Is "wider" measured as a greater angle from the listener to the midbasses?

See this post for my previous inquiries: http://www.diymobileaudio.com/forum/diy-mobile-audio-sq-forum/69060-natural-bass-12.html#post917819


----------



## justinmreina

lycan said:


> I think it's fair to say that various acoustic wavelengths (and their corresponding frequencies, naturally) ONLY have significance in relation to the _physical dimensions_ of whatever is sensing or measuring them.


Fair? I think its mandatory in this context. Its all in the 'ka's! 

-Justin


----------



## Patrick Bateman

loddie said:


> Thread of the decade. Thanks Lycan!
> 
> I understand the concepts presented so far in this thread. However, I'm having a hard time understanding width when "wider is better" for midbass placement. Is "wider" measured as a greater angle from the listener to the midbasses?
> 
> See this post for my previous inquiries: http://www.diymobileaudio.com/forum/diy-mobile-audio-sq-forum/69060-natural-bass-12.html#post917819



Mind if I take a crack at this one?

First, understand that the mighty Bose marketing machine has convinced people that the location of subs doesn't matter. Ironically, those little boxes they sell don't even produce *sub-bass;* they're putting out bass and midbass.

Second, as mentioned on page one, our perception of midbass location is dominated by interaural time differences. IE, contrary to Bose, location is *everything* at these frequencies.







*Here's an example, including the math.*
With a speaker in the stock locations in my doors, the pathlength to the left speaker is approximately 48 inches to my left ear, and 51.3" to my right ear. This is pure pythagorean theorem. Or even easier, just use a tape measure.

Anyways, the main point is that _there's a 0.24ms time delay._ (sound travels 13.5 inches in a millisecond, and 24% of thirteen and a half inches is 3.3 inches.

Does that make sense?

Note that *this isn't a stereo thing; you can easily detect the location of a midbass even if it's in mono. In fact it's easier in mono, because the number of imaging cues are fewer.*

So *that* is the mathematical explanation for why you want your midbasses *as wide as possible.* (Within reason of course; if they're too wide a "hole" appears in the middle of the stage. We can fix that too though. I'll explain in a sec.)

Now that we've covered how and why we localize width, let's talk about depth.

Do this experiment:

Find a quiet room
cover one ear
Throw something on the floor, preferably a carpeted floor
Can you perceive where it landed? Can you tell why?

If you are like me, then YES, you can perceive depth with a single ear. The reason why is that you perceive the *echoes*. The original source is combined with a delay of itself, and the combination of the two is how you perceive depth, even with just one ear.*

This brings us back to DSP - and helps explain why you can't just "dial in" time delay to relocate speakers. We're dealing with mechanisms that have kept us ALIVE for millions of years  If our ears didn't work this well, there would probably be a lot fewer of us roaming the earth!

So there it is - a very abbreviated description of how we perceive width and depth at midbass frequencies. Note that above 1.6khz the mechanisms are different.

Last but not least, I mentioned that spreading the midbasses too far apart wil create a "hole" in the center of the stage. The reason that this happens is simple. Our perception of width is based on the time delay from one ear to the other. As the midbasses move towards the middle, the time delay decreases, hitting zero at the center.

Here's where things get interesting. Above 1.6khz, our perception of location is dominated by interaural level differences. In other words, frequency response is critical, but location can be manipulated, because time delay becomes inaudible at high frequencies.

Seeing where I'm going with this?

You basically end up with a stage that's the complete *opposite* of logical. The midbasses are much wider than you would normally use, so wide that a hole appears in the center. Then the tweeters are placed in the center, filling in the hole in the stage, and leveraging the fact that frequency response is more important than location at high frequencies.

The end result is the best of both worlds - better width than you could get with a pair of full range speakers, and better intelligibility thanks to the physical proximity of the high frequencies. (Midbass has little effect on intelligibility.)

Someone clever might split this into three bands instead of two, and then you'd be on your way to doing this:

Ambiophonic optimal source distribution experiment (part 1: Introduction) - diyAudio

* more info on echoes and localization : http://www.cs.cmu.edu/~lewicki/cp-s08/sound-localization2.pdf


----------



## lycan

before we move on to vectors, i think there's a few solid conclusions that can be drawn so far :

*CONCLUSION ONE : ITD corresponds to a specific cone angle*

As we've seen from our "rotated parabolas", once we know ... or _perceive_ ... a specific ITD, we immediately know ... or _recognize_ ... a midbass "cone angle". By "cone angle", I mean the angle or slope that the hyperbola approaches at large distances. The cone angle completely identifies the particular "cone of confusion" where our midbass resides.

The math for this is pretty simple :

For hyperbolas, the asymtote lines are given by :

*y = (+/- b/a)*x*

In our case, these lines are therefore :

*y = {+/- [sqrt(e^2 -d^2)]/d }*x*

And of course

*e = distance between our ears, about 8 inches*

and the distance *d* is the distance that corresponds to an ITD. Given the speed of sound, 

*d = ITD(in milliseconds)*(13.5 inches per millisecond)*

What all this means is simply this :

*Give me the ITD (in milliseconds), and I'll identify the confusion cone angle. Alternatively, once the ear/brain "measures" the ITD, the brain automatically does this math and identifies the confusion cone angle. And the "cone angle" is as specific as the location can be identified ... the exact location of the midbass on the cone cannot be determined.*

This is important conclusion Number 1.


----------



## ncv6coupe

heres an image that i dug up that describes exactly what you guys are explaining to someone who may be wondering.







[/IMG]

the middle arrow is about the 3.3 inch mark where PB stated earlier


----------



## lycan

*CONCLUSION TWO: The Head-Turning Problem*

I hope we can already appreciate a very basic form of a midbass array: put midbass drivers all over the surface of the cone of confusion ... anywhere you want ... and they will all be perceived as ONE. The ear simply can't distinguish one from the other, if they are all on the same cone of confusion. ITD is all the ear/brain can measure in the midbass, but that only identifies a particular cone ... nothing more!

But let's examine this in our 2-D plane. Put a midbass driver in the _upper_ right hand quadrant, on a particular hyperbola of confusion. Put a second midbass driver in the _lower_ right hand quadrant, on the same hyperbola. So far so good ... the ear has no choice but to recognize these two drivers as "one", since they both contribute the exact same info to the ear/brain.

But happens when the listener turns his head?

The answer is this, and the reader will be encouraged to imagine or draw this same conclusion : each midbass driver now moves to a DIFFERENT hyperbola (or different cone, in 3-D), relative to the NEW location of the ears. And the midbass drivers will NO LONGER be on the SAME hyperbola 

The different hyperbola "in front" of the listener is EXPECTED. The source in front has remained still, non-moving. So the ear/brain will recognize it's "new" angle, in relation to the head that has moved. Make sense?

The "confusing" thing is what happened _behind_ the head at the same time. The midbass driver behind the head has also moved to a different hyperbola/cone (relative to the new position of the turned head) but it now finds itself on a DIFFERENT hyperbola/cone than the midbass driver in front!

How these two drivers "combine" or "conspire" to create a new midbass "signal" to the ear will have to wait awhile. That's what VECTORS are for  They will help us analyze how two or more midbass drivers COMBINE to create a SINGLE inter-aural "signal", and this will come in handy to understand basic stereo as well as midbass arrays. But for now, we can appreciate that something "interesting" happens ... two drivers that started on the same hyperbola/cone, are no longer on the same one ... and it's probably not good for stage stability in the presence of a turning head


----------



## jimbno1

Keeping in mind "There are no stupid questions, only stupid people" I am going to risk asking a question. If ITD rules in midbass, and I am only interested in a single seat setup, what is wrong with mounting the midbasses in the doors and using time delay to fool your mind?


----------



## justinmreina

so to relate that excellent picture to lycan's ITD; path difference to each ear is:

*d = (dL - dR) = a*sin(theta) + a*theta*
*ITD = d*c0
*
_*'a' is earSpacing/2_
_*'c0' is the speed of sound in air_



-Justin


----------



## lycan

jimbno1 said:


> Keeping in mind "There are no stupid questions, only stupid people" I am going to risk asking a question. If ITD rules in midbass, and I am only interested in a single seat setup, what is wrong with mounting the midbasses in the doors and using time delay to fool your mind?


Imagine a single midbass driver, all by itself, playing to a head with two ears.

Add time delay to the midbass driver ... so the signal is delayed by some amount, say 2 milliseconds. How much delay is added to the signal that shows up at the left ear? How much delay is added to the signal that shows up at the right ear? And finally, how has the ITD changed?

When you understand the answer to this question, you'll begin to understand that time delay is NOT the answer to all of our woes


----------



## fish

I have a question, & I apologize if I missed it (I get intimidated when I see letters used in math), when multiple midbass drivers are implemented in the cone of confusion do they have to be EXACTLY the same distance away from our ears? Or is there some sort of leniency in the pathlengths?

Again, sorry if this has been adressed, or if we're not that far along yet.


----------



## justinmreina

He's only referred to a single driver for now, and how we locate it in space. I think we have to have patience to get to two speakers, he'll get there soon.

-Justin


----------



## lycan

fish said:


> I have a question, & I apologize if I missed it (I get intimidated when I see letters used in math), when multiple midbass drivers are implemented in the cone of confusion do they have to be EXACTLY the same distance away from our ears? Or is there some sort of leniency in the pathlengths?
> 
> Again, sorry if this has been adressed, or if we're not that far along yet.


Well, if the distances to our ears are different then the amplitudes (or SPL) at our ears will be different. But the multiple drivers on the cone will simply ADD in amplitude (assuming they are playing the same frequencies, of course, which is pretty fundamental to any array).

Another way to answer your question : put one driver on the cone, pretty close to our ears. Put another driver somewhere else on the same cone, but farther from our ears. If you turn up the volume knob on the one farther away, it will sound identical to the closer one. If the close one and far one are playing together, their outputs will simply add.

EDIT/correction : the addition of two drivers on the same cone will be straighforward, since each one is generating the same ITD. However, the two drivers can still algebraically combine constructively or destructively, depending on distance.


----------



## lycan

justinmreina said:


> He's only referred to a single driver for now, and how we locate it in space. I think we have to have patience to get to two speakers, he'll get there soon.
> 
> -Justin


true ... but the case for two drivers on the SAME cone is pretty simple, as outlined above 

Things get more complicated when we "combine" more than one driver, each with different ITD's (midbass drivers on DIFFERENT cones). That's the case for Blumlein stereo, and it may be the case for multiple drivers in a midbass array.

And that's where vectors come in handy


----------



## trueblue

lycan said:


> Imagine a single midbass driver, all by itself, playing to a head with two ears.
> 
> Add time delay to the midbass driver ... so the signal is delayed by some amount, say 2 milliseconds. How much delay is added to the signal that shows up at the left ear? How much delay is added to the signal that shows up at the right ear? And finally, how has the ITD changed?
> 
> When you understand the answer to this question, you'll begin to understand that time delay is NOT the answer to all of our woes


Basically stating that time delay can't compensate for the two separate distances the a soundwave must travel to reach both ears. Am I following you correctly?


----------



## lycan

trueblue said:


> Basically stating that time delay can't compensate for the two separate distances the a soundwave must travel to reach both ears. Am I following you correctly?


answer the question 

Does time delay, added to a single midbass driver playing all by itself, "move" it's virtual location to a _different_ cone?


----------



## trueblue

lycan said:


> answer the question
> 
> Does time delay, added to a single midbass driver, "move" it's virtual location to a _different_ cone?


no. delay can only alter the time of wave production from a single source which gives the perception of a change in distance to the indivdual driver.


----------



## lycan

trueblue said:


> no. delay can only alter the time of wave production from a single source which gives the perception of a change in distance to the indivdual driver.


Really?

So let's say I have a single driver, playing all by itself. If i add 1 millisecond delay ... will it sound farther away? How about 1 minute delay? How about 1 hour delay?

I'm not trying to be a dickhole. But this is pretty important, as far as understanding what time delay can, and can't, do.

I think we've already established that time delay, operating on a single driver playing all by itself, can't "move" the virtual location of a midbass driver from one cone-of-confusion to another, since time delay for a single driver playing all by itself can't change it's ITD. Also, I think we can agree that adding some time delay to a midbass driver playing all by itself won't change it's amplitude or volume, either.

So what exactly WILL time delay "change" for a single driver playing all by itself?


----------



## ncv6coupe

it would allow the sound to get centered as the wave should be closer to reaching both ears at the same time!


----------



## lycan

ncv6coupe said:


> it would allow the sound to get centered as the wave should be closer to reaching both ears at the same time!


so ... if I hit the "play" button ONE MILLISECOND later for my single driver playing all by itself, this will change it's apparent location somehow?


----------



## rockinridgeline

Nope, it will only help you align the driver with another driver. Phase.


----------



## trueblue

lycan said:


> So what exactly WILL time delay "change" for a single driver playing all by itself?


nothing at all. 

I missed the 'all by itself' part, earlier. :bash:


----------



## ncv6coupe

lycan said:


> so ... if I hit the "play" button ONE MILLISECOND later for my single driver playing all by itself, this will change it's apparent location somehow?


I messed up what i was saying, i forgot that we are talking about midbass only where frequencies are longer than 8 inches rather 13.5 inches. thus the sound should seem to be originating "IN" your head. so the answer is nothing.


----------



## BigRed

with a single driver, time delay does nothing. time delay will alter things when 2 or more drivers are in the equation 

lycan, if you time delayed a single driver for an hour, I guess we could go to lunch and wait for the driver to play the sound 60 minutes later


----------



## lycan

trueblue said:


> nothing at all.
> 
> I missed the 'all by itself' part, earlier. :bash:


BINGO  nothing at all. As rockingridgeline said, time delay only impacts the response of a driver _in relation to other drivers_.

Time delay for a _single_ midbass driver playing all by itself can't move it's virtual location 

HOWEVER, once you add ANOTHER driver ... now there's possibilities for virtually "moving" a source to a DIFFERENT cone of confusion 

That's a powerful statement. The first guy who recognized and exploited this was Alan Blumlein, the inventor of stereo. He was a true genius, in every sense of the word ... he also invented the "differential" or "long-tailed" pair that's so pervasive in modern electronics (every opamp has one, as well as just about every power amplifier ever made), and he probably invented feedback before Harold Black (the commonly recognized father of feedback). Tragically, he died at a young age ...

In any case, the "easiest" way to understand how two, or more, drivers combine is ... wait for it ... vectors  Gotta dig into that next.


----------



## lycan

ncv6coupe said:


> I messed up what i was saying, i forgot that we are talking about midbass only where frequencies are longer than 8 inches rather 13.5 inches. thus the sound should seem to be originating "IN" your head. so the answer is nothing.


well, the answer would also be the same for a tweeter.

Adding time delay to a single tweeter, playing all by itself, won't change it's location either.

If you don't believe me, i'll challenge you to devise an experiment to prove me wrong  Easy to to in a 2 or 3 way system with active control, and time alignment for all drivers. Just get ONE tweet playing all by itself (shut the other drivers off), add in varying amounts of time delay, and listen as that tweet moves all over the dash ... or, not 

You can also try the same experiment with a SINGLE midrange, playing all by itself ... and tell us what you observe


----------



## durwood

So NOW you are ready to trade in path length for azimuth angle?


----------



## durwood

I'm also enjoying all these references to monophonic examples lately as if somehow naturally occuring sound events ARE monophonic.


----------



## lycan

*VECTORS*

Sinusoids (aka sinewaves, or single-frequency tones) play a very important role in ALL of electronics, acoustics and signal processing. There are mathematical reasons that I won't go into, but suffice it to say that whenever somebody asks "who listens to test tones?", the answer is that we ALL do, ever time we turn on the stereo.

The mathematical form is pretty simple:

*A*sin(w*t)

where *A* is amplitude, "w" is the radian frequency, and "t" is time. You can plot this, with time typically being the x-axis.

We can generalize the expression to include another parameter, _phase_:

*A*sin(w*t + *phi*)

where *phi* is a fixed "phase angle".

When we are dealing with a fixed frequency, say we pick a frequency right in the midbass region like 160 Hz, the only "interesting" parameters for _any_ sinusoid are AMPLITUDE *A* and PHASE *phi*. So, for that particular frequency, we can "ignore" time & frequency, and "draw" our sinewave on a new "plane" or graph that ONLY shows amplitude and phase. The new graph or plane will ignore time & frequency.

This new plane is sometimes called a "phase plane", and we represent our sinewave as an ARROW, with a center at the origin. The LENGTH of the arrow is drawn proportionally to AMPLITUDE, and the angle .. from 0 degrees to 360 degrees, or to be consistent, 0 radians to 2pi radians ... is the phase (zero degrees, or zero radians, typically being the positive x-axis). The arrow is called a VECTOR 

*Why bother? Well, because it turns out that additions & subtractions of sinusoids, if they are at the same freqeuncy BUT have different amplitudes & phases, can be easily "drawn" as a simple geometry problem involving VECTORS. So, adding or sustracting sinusoids (of the same frequency) is easily done graphically by adding VECTORS. It's a visual tool, that aids in the understanding of how sinusoids COMBINE.*

Important point : this new plane, the "phase plane", is not a PHYSICAL plane. It's representative of amplitudes and phase of sinewaves that might have real physical meanings & locations, but the phase plane is NOT a physical plane. It's a mathematical abstraction, designed to simplify calculations.

Are you with me so far?

So how do we add vectors? Let's do a simple example, shall we 

Lets say I'm interested in adding two sinewaves (of the same frequency, it's required) that have different phases. But, for simplicity of this first example, we'll assign the same amplitude :

*A*sin(wt) + *A*sin(wt + *pi/2*)

We _could_ use trigonometric identities to simplify ... or we can use VECTORS 

The first vector, representing the first term, will be an arrow from the origin extending right-ward along the positive x-axis, since it's phase term is zero. The second term will be an arrow that extends from the origin straight up-ward, along the positive y-aixs, since it's phase term is 90 degrees (= pi/2 radians).

Now, how the hell do we "add" arrows? Simple : translate one arrow, let's say the second one, without changing the direction it's pointed, until it's _starting_ point is coincident with the _ending_ point of the other arrow  Now, draw a new arrow ... this new vector will be the SUM ... that starts at the starting point of the first arrow, and ends at the NEW ending point of the second arrow. It sounds more complicated than it really is ...

You should have a new vector that points up & to the right at 45 degrees, or pi/4 radians. I'll let someone tell us what our old friend Pythagoras has to say about the resulting amplitude


----------



## Patrick Bateman

This is kinda "out there" but consider this:

A speaker in my door has an interaural time delay of 0.24ms, an an echo at 1.8ms later, off the firewall. _My perception of width is due to time delay, and depth is due to echoes._ (Yes, that's a simplification, and ignores some subtle cues. But it covers the most important parts.)

At home I used to have a setup which used eight subwoofers, and found that it made an audible improvement in frequency response, and eliminated my ability to localize the subs.

Geddes and Toole demonstrated that the use of multiple subs could smooth frequency response. Perhaps my inability to localize the subs was due to so many cues, the brain is no longer able to isolate them.

This solves the "head turning" problem nicely.

Basically the tweeters are optimized for left/right frequency response, the midrange are placed very carefully, and then an array of midbasses is used to smooth low frequency response, and swamp all the cues that help us localize the midbass.

The reason that this trick works is that the wavelengths are fairly long; at high frequencies you wouldn't want to do this, because comb filtering would screw up your frequency response. (And that's the last thing you want at high frequency; at low frequency, it's not a huge issue.)


----------



## lycan

Patrick Bateman said:


> This is kinda "out there" but consider this:
> 
> A speaker in my door has an interaural time delay of 0.24ms, an an echo at 1.8ms later, off the firewall. _My perception of width is due to time delay, and depth is due to echoes._ (Yes, that's a simplification, and ignores some subtle cues. But it covers the most important parts.)
> 
> At home I used to have a setup which used eight subwoofers, and found that it made an audible improvement in frequency response, and eliminated my ability to localize the subs.
> 
> Geddes and Toole demonstrated that the use of multiple subs could smooth frequency response. Perhaps my inability to localize the subs was due to so many cues, the brain is no longer able to isolate them.
> 
> This solves the "head turning" problem nicely.
> 
> Basically the tweeters are optimized for left/right frequency response, the midrange are placed very carefully, and then an array of midbasses is used to smooth low frequency response, and swamp all the cues that help us localize the midbass.
> 
> The reason that this trick works is that the wavelengths are fairly long; at high frequencies you wouldn't want to do this, because comb filtering would screw up your frequency response. (And that's the last thing you want at high frequency; at low frequency, it's not a huge issue.)


all you guys, jumping the gun ...

At the end of this I'm going to be recommending a midbass array (with a little bit of electronic manipulation) that WON'T screw up location cues for a forward-facing head, and will (hopefully) minimize the head-turning problems too.

This thread is kinda designed to be a companion thread to your "natural bass" thread, where multiple drivers at multiple locations can be used to smooth amplitude irregularities. It's my "grand plan" to _also_ offer some new info on building midbass arrays that WON'T wreak havoc with midbass location cues


----------



## lycan

OK, i'm still impatient (this trait doesn't improve with age) ...

Hopefully, we can all now appreciate how VECTORS can be used to solve this simple example :

*A*sin(wt) + *A*sin(wt + *pi/2*) = *A*sqrt(2)*sin(wt + *pi/4*)

The real utility of "vector math", though, is for adding sinusoids of arbitrary (and different) amplitudes and phases. The reader is encouraged to play with other, more complicated, examples 

And now, we're going to provide the last piece of the puzzle we need to describe stereo, and other midbass arrays  Let's add some _time delay_ to our original sinusoid, call it *Td*:

*A*sin[w(t - *Td*)]

In order to use our _phase plane_ to describe this new parameter, *Td*, let's re-write that expression :

*A*sin[w(t - *Td*)] = *A*sin(wt - w*Td*) = *A*sin(wt - *pht*)

where the _phase shift_ associated with the _time delay_ is given by :

*pht = w*Td*

This simple expression should really be all you need to appreciate the relationship between a _phase shift_ and a _time delay_. *Time delay is really nothing more than a phase shift that's proportional to frequency*  And for a SINGLE frequency, it's OK to consider a phase shift to be synonomous with a time delay ... just remember that the phase shift will be _different_ for _different_ frequencies.

So now, we have all we need to describe Blumlein's stereo and _other_ midbass arrays, with vectors on the phase plane


----------



## SSSnake

This is definitely worth repeating (I see it screwed up all of the time):



> And for a SINGLE frequency, it's OK to consider a phase shift to be synonomous with a time delay ... just remember that the phase shift will be different for different frequencies.


Given this fact you can use time delay to implement phase corrections over a driver covering a narrow bandwidth (assuming you don't have access to an all pass filter this could be beneficial) and vice versa. IMO - this is why you hear a LOT of the old school guys suggesting you flip the phase of a driver to effect staging rather than using time delay.

This next one threw me a bit...



> I hope we can already appreciate a very basic form of a midbass array: put midbass drivers all over the surface of the cone of confusion ... anywhere you want ... and they will all be perceived as ONE. The ear simply can't distinguish one from the other, if they are all on the same cone of confusion. ITD is all the ear/brain can measure in the midbass, but that only identifies a particular cone ... nothing more!


Are we assuming that only one speaker is playing at a given time? Otherwise I would think arrival times would be an issue. If you had a speaker near the parabolic maxima and one near the parabolic minima I would think you could resolve the sounds as distinct events if the dimensions were large enough (of course this could be addressed with time delay). Am I missing something?


----------



## ncv6coupe

lycan said:


> I hope we can already appreciate a very basic form of a midbass array: put midbass drivers all over the surface of the cone of confusion ... anywhere you want ... and they will all be perceived as ONE. The ear simply can't distinguish one from the other, if they are all on the same cone of confusion. ITD is all the ear/brain can measure in the midbass, but that only identifies a particular cone ... nothing more!





SSSnake said:


> Are we assuming that only one speaker is playing at a given time? Otherwise I would think arrival times would be an issue. If you had a speaker near the parabolic maxima and one near the parabolic minima I would think you could resolve the sounds as distinct events if the dimensions were large enough (of course this could be addressed with time delay). Am I missing something?


The key is that he pointed out, this only matters for ( lower midbass). i want to believe even 500hz would ruin this effect if all the drivers were within the "cone." of course i may be wrong


----------



## SSSnake

But arrival times can be resolved at any freq. For example if I play a 100 hz tone for 1 sec and then 10 secs later I play the same 100 hz tone for 1 sec I can tell that these occured at different times.


----------



## SSSnake

Are we assuming that the dimensions are small enough that Haas effect will take care of the problem?


----------



## ncv6coupe

justinmreina said:


> You must have gauged how much I am not enjoying my HW right now...
> 
> *Curves*
> 
> 
> 
> 
> 
> 
> 
> 
> 
> *Path Values on d=4*
> 
> 
> 
> 
> 
> 
> 
> 
> 
> *Code...*
> MATLAB code if you want to play with it...
> 
> -Justin


This picture was on page 1 has alot to do with this, and don't forget that sound doesn't stop traveling just because it goes in your left ear and hits your brain, sound going in your left ear goes to your right ear> i hope that doesn't seem dumb. 
and one more pic, i'm a visual type of guy pardon my stick man graphics.


----------



## lycan

SSSnake said:


> This is definitely worth repeating (I see it screwed up all of the time):
> 
> 
> 
> Given this fact you can use time delay to implement phase corrections over a driver covering a narrow bandwidth (assuming you don't have access to an all pass filter this could be beneficial) and vice versa. IMO - this is why you hear a LOT of the old school guys suggesting you flip the phase of a driver to effect staging rather than using time delay.
> 
> This next one threw me a bit...
> 
> 
> 
> Are we assuming that only one speaker is playing at a given time? Otherwise I would think arrival times would be an issue. If you had a speaker near the parabolic maxima and one near the parabolic minima I would think you could resolve the sounds as distinct events if the dimensions were large enough (of course this could be addressed with time delay). Am I missing something?


Good question ... they will be perceived as "one" if the arrival times are less than about 15msec. That's the Haas, or precedence, limit. Audio signals arriving at our ears within this interval are blended as "one", and if that "blending" involves drivers whose different locations can't be separately discerned (because on the same cone-of-confusion), they will be perceived as "one". Now ... you still might get some constructive and/or destructive interference, so the _amplitude_ may not be the simple arithmetic sum, but the ear won't hear more than one source.

Make sense?


----------



## lycan

ncv6coupe said:


> This picture was on page 1 has alot to do with this, and don't forget that sound doesn't stop traveling just because it goes in your left ear and hits your brain, sound going in your left ear goes to your right ear> i hope that doesn't seem dumb.
> and one more pic, i'm a visual type of guy pardon my stick man graphics.


I like your stick-man figures !!! Thank you!

That is, indeed, the cone of confusion. The only correction i'll make, is that the "point" of the cone is really in the center of our heads. The hyperbola deviates from the simple cone very close to our ears, as the hyperbola graphs show. But the "center", or intersection, of the asymptotes of the hyperbola is actually at the _origin_, in our 2D views.


----------



## lycan

SSSnake said:


> Are we assuming that the dimensions are small enough that Haas effect will take care of the problem?


oh ... ummm ... yes, i see you already got it 

Yes, the Haas limit is bout 15msec, which correspponds to about 16 feet.

But howabout I throw a real curve ball at you guys 

If two midbass drivers are playing the same midbass frequency, even with DIFFERENT amplitudes AND even if they reside on DIFFERENT cones-of-confusion, the ear will STILL recognize them as a single source (providing they are located within the Haas limit).

These two sources have no choice but to "combine" at our ears, forming a _single_ amplitude and _single_ ITD. The ear simply has NO WAY to distinguish these two sources from a SINGLE midbass driver, playing on "some" cone-of-confusion. 

The manner in which they combine ... and the "cone" on which the virtual combination can be found ... can be determined with vector analysis on the phase plane


----------



## SSSnake

> If two midbass drivers are playing the same midbass frequency, even with DIFFERENT amplitudes AND even if they reside on DIFFERENT cones-of-confusion, the ear will STILL recognize them as a single source (providing they are located within the Haas limit).


And everything in the car is within the distance so....


----------



## lycan

SSSnake said:


> And everything in the car is within the distance so....


yep 

Up next, the first guy to build a clever midbass array


----------



## ncv6coupe

lycan said:


> Up next, the first guy to build a clever midbass array


Lycan why art thou teaset yee followers? Ok that was terrible but do u mind spitting out what drivers u prefer for said array? I know a few years back you were in love with some aurasounds? I need to gather my schillings to ready an order for these drivers for some R&D. And are you referring to Lycans arc or we going another direction? Impatience is a virtue


----------



## lycan

*BLUMLEIN STEREO*

First thing I need to do, is define a _new vector_. I'm going to take a few liberties for the sake of simplicity, and present a discussion that's complicated enough to get a few key points across ... but no more so 

Let's define an *INTER-AURAL MIDBASS VECTOR*. This vector has the following characteristics:

*1. AMPLITUDE*

The _amplitude_ of the INTER-AURAL MIDBASS VECTOR equals the amplitude at _both_ our ears from a midbass source playing somewhere in space (remember, in the midbass inter-aural intensity differences are nill).

*2. PHASE*

The _phase_ of the INTER-AURAL MIDBASS VECTOR corresponds to the inter-aural time delay, or ITD, _between_ our ears that results from the cone-of-confusion upon which the midbass is playing (remember, we can use phase and time-delay synonomously at any single frequency, as long as we recognize that the the phase will be different for a different frequency).

What this allows us to do, is to simplify a complex problem by indentifying a SINGLE vector for each SINGLE driver playing in space. The strictly accurate version would require TWO vectors for each driver ... one for each ear  Instead, we'll simplify with a single, INTER-AURAL vector for each midbass driver playing in space. It's still accurate enough, i think, to identify some interesting properties of midbass arrays.

Now, we FINALLY have everything we need 

Sometime in the late twenties or early thirties, Alan Blumlein was at a motion-picture theater with his wife. There may have been one, or even a couple, loudspeakers playing ... but what struck the Blumlein's that night was that the voice did not "follow" an actor as he walked across the screen. Blumlein stated, that very night, "I can fix that". I would like to say the rest is history ... but it so happens that history ignored his invention for about twenty years  It's fair to say that Alan Blumlein was literally two decades ahead of his time.

The idea is this : place a _first_ loudpeaker in front of you, thirty degrees to the right. Place a _second_ loudspeaker in front of you, thirty degrees to the left. The first speaker is on a "positive" 30 degree cone-of-confusion, and the second speaker is on a "negative" 30 degree cone-of-confusion.

We immediately need to identify the INTER-AURAL VECTORS associated with these speakers. So draw a _first_ vector a few degrees upward from the positive x-axis. It doesn't need to be 30 degrees ... the "30 degrees" is the _physical_ angle of the speaker, but the _phase_ of our vector corresponds to the inter-aural time delay ITD ... which means it's phase depends on frequency. So, just draw a first vector at, say, plus 20 degrees. And draw the second vector, at minus 20 degrees (we can always pick a frequency, and knowing that the two loudspeakers are on a 30 degree "cone" we can identify the ITD with some math presented earlier on hyperbola asymptotes. And the frequency we choose, together with the ITD associated with a 30 degree cone, can nail down the exact _phase_ of our vector ... exercise left to the reader).

Now, we need to analyze how these two sources ... indeed, how the two corresponding vectors ... combine.

Here's the homework : let's START with vectors that are EQUAL in amplitude. What is the phase angle ... and resulting ITD ... of the COMBINED inter-aural vector? Those following along will know how to add a plus-20 degree vector and minus-20 degree vector, with equal amplitudes 

Where will the listener "hear" a virtual midbass source in physical space?


----------



## lycan

ncv6coupe said:


> Lycan why art thou teaset yee followers? Ok that was terrible but do u mind spitting out what drivers u prefer for said array? I know a few years back you were in love with some aurasounds? I need to gather my schillings to ready an order for these drivers for some R&D. And are you referring to Lycans arc or we going another direction? Impatience is a virtue


all good things to those who wait ...

But no, this isn't about lycan's arc (thanx for remembering), and i don't care what midbass drivers you choose ... that's for a different thread (although I don't recommend Aura whispers for midbass as defined in this thread)  

What we _will_ help determine, i hope, is where you can put them  and what we might do if they are in less-than-desirable places 

Stick with us! Won't be much longer, i promise. Education, if i may be so bold, is rarely an overnight process.


----------



## Fast1one

lycan said:


> *BLUMLEIN STEREO*
> 
> Now, we need to analyze how these two sources ... indeed, how the two corresponding vectors ... combine.
> 
> Here's the homework : let's START with vectors that are EQUAL in amplitude. What is the phase angle ... and resulting ITD ... of the COMBINED inter-aural vector? Those following along will know how to add a plus-20 degree vector and minus-20 degree vector, with equal amplitudes
> 
> Where will the listener "hear" a virtual midbass source in physical space?


I know EXACTLY where this is going. There is a technique called VBEP, or something of the like, which uses the same vector math in it's theory. 

Anyway, for the above question. Summing the two vectors yields a net phase angle of 0 degrees. Hence we get a phantom image directly between two speakers. 

The amplitude is simple trig, the components of the +/- 20 degree vectors in the 0 degree direction: 

2**A**cos(20degrees) or 1.879*A*


----------



## quality_sound

lycan said:


> Really?
> 
> So let's say I have a single driver, playing all by itself. If i add 1 millisecond delay ... will it sound farther away? How about 1 minute delay? How about 1 hour delay?
> 
> I'm not trying to be a dickhole. But this is pretty important, as far as understanding what time delay can, and can't, do.
> 
> I think we've already established that time delay, operating on a single driver playing all by itself, can't "move" the virtual location of a midbass driver from one cone-of-confusion to another, since time delay for a single driver playing all by itself can't change it's ITD. Also, I think we can agree that adding some time delay to a midbass driver playing all by itself won't change it's amplitude or volume, either.
> 
> So what exactly WILL time delay "change" for a single driver playing all by itself?


I know I'm late to the party but time only has meaning in a relative sense. You could delay a driver for a month and it wouldn't matter because you wouldn't know if it was early or late if there was not point of reference i.e., another driver.


----------



## lycan

Fast1one said:


> I know EXACTLY where this is going. There is a technique called VBEP, or something of the like, which uses the same vector math in it's theory.
> 
> Anyway, for the above question. Summing the two vectors yields a net phase angle of 0 degrees. Hence we get a phantom image directly between two speakers.
> 
> The amplitude is simple trig, the components of the +/- 20 degree vectors in the 0 degree direction:
> 
> 2**A**cos(20degrees) or 1.879*A*


well done 

Now let's see the real genius of Blumlein at work 

Let's say I keep each speaker in the same physical location ... so the ITD vectors remain at the same phase angle (for the same frequency). What happens if I adjust the relative AMPLITUDES of the vectors ... say, for example, I reduce the amplitude of the -20 degree vector? No need to be precise ... just tell us what happens to the vector SUM.

Bear this in mind : we are keeping the PHASE angles (corresponding to the ITD's of the speakers) the SAME. Their individual PHASE angles don't change ... we are only adjusting the relative AMPLITUDES, in this question.


----------



## Fast1one

lycan said:


> well done
> 
> Now let's see the real genius of Blumlein at work
> 
> Let's say I keep each speaker in the same physical location ... so the ITD vectors remain at the same phase angle (for the same frequency). What happens if I adjust the relative AMPLITUDES of the vectors ... say, for example, I reduce the amplitude of the -20 degree vector? No need to be precise ... just tell us what happens to the vector SUM.
> 
> Bear this in mind : we are keeping the PHASE angles (corresponding to the ITD's of the speakers) the SAME. Their individual PHASE angles don't change ... we are only adjusting the relative AMPLITUDES, in this question.


First let's define *deltaA* as the change of amplitude for the -20degree direction. 


The amplitude changes by an amount of* -deltaA*cos(20degrees)* (decreasing) in the 0 degree direction. 

However, the 90 degree direction (horizontal component if you will) will experience a net increase in amplitude of magnitude *deltaAsin*(20degrees). *

A change in amplitude effectively rotates the summing vector, correct? Towards the transducer with the LARGEST amplitude.


----------



## quality_sound

Isn't going to be similar to making delay changes because of the precedence of amplitude (I think that's the term) but without the phase changes you'd get with actually making delay changes?


----------



## Fast1one

quality_sound said:


> Isn't going to be similar to making delay changes because of the precedence of amplitude (I think that's the term) but without the phase changes you'd get with actually making delay changes?


Could you reword that? I am having trouble deciphering what you are actually asking 

Edit: NVM I got it. Yes I think that is where this is going


----------



## quality_sound

Sorry, it's still really early over here and I didn't get a lot of sleep.  I'm sure I left oua comma here and there that woudl have helped greatly.


----------



## lycan

Fast1one said:


> A change in amplitude effectively rotates the summing vector, correct? Towards the transducer with the LARGEST amplitude.


Yes.

Here's the magic : *Adjusting the RELATIVE AMPLITUDES of our two vectors will adjust the PHASE of the resultant sum. Since phase is directly related to ITD ... we've created a virtual source on a cone-of-confusion where no physical driver existed in the first place* 

Let that sink in ...

To make sure that the "effect" works over a reasonably wide range of frequencies, the amplitude adjustment of the two sources needs to result in a phase (for the summation vector) that's proportional to frequency ... thereby translating the _inter-aural vector phase_ into a true _inter-aural time delay_. But even in the thirties, there were electronic components ... inductors & capacitors ... whose impedance (or conductance) was/is proportional to frequency, allowing some pretty simple circuitry to electronically realize the required frequency-dependency 

We're getting so close to a really interesting conclusion ...


----------



## solacedagony

Just a quick clarification of phasors...


The first example resembles Lycan's first example where the two midbasses are of equal amplitude (length of the line) and of equal but opposite angle (0 deg. is horizontal right, 90 deg. is vertically up).

In the second example, the midbasses are of equal and opposite angle, but are of differing amplitudes. Thus, we see that the resultant wave connecting the starting and ending points ends up angled as well... All from a difference in amplitude.


----------



## lycan

solacedagony said:


> Just a quick clarification of phasors...
> 
> 
> The first example resembles Lycan's first example where the two midbasses are of equal amplitude (length of the line) and of equal but opposite angle (0 deg. is horizontal right, 90 deg. is vertically up).
> 
> In the second example, the midbasses are of equal and opposite angle, but are of differing amplitudes. Thus, we see that the resultant wave connecting the starting and ending points ends up angled as well... All from a difference in amplitude.


Thank you !!!

To summarize Blumlein stereo :

1. Two speakers at a fixed angle in physical space
2. Adjust the relative AMPLITUDES, to change the PHASE of the resultant SUM
3. Make sure the resultant PHASE is proportional to frequency (through some simple electronic manipulation), and you've "created" a "virtual ITD" over a wide frequency range ... thereby putting a virtual source on a cone-of-confusion, where a real one never existed

Pretty clever, huh? Turns out he also developed recording techniques to easily realize the required "amplitude-panning" law (that frequency-dependent amplitude adjustment needed to make the phase of the sum-vector proportional to frequency), as well as the +/- 45 degree stylus to record two independent channels on a phonograph record.

Couple other thing to note :

1. Each speaker must create an inter-aural vector for the technique to work. In other words, _each_ speaker must be heard by _both_ ears. This is not a "failing" of stereo, it's a _design requirement_. Over the decades, there have been _other_ techniques (some versions of binaural come to mind) that depend on eliminating "crosstalk" between the ears ... but basic stereo isn't one of them.

2. I think it's easy to visualize how a source can be located (via "virtual ITD") just about anywhere _between_ the speakers : simply adjust the amplitudes of the two vectors to realize a resultant summation at _any_ phase angle in between.

New question : Is the vector sum limited to angles BETWEEN the original two vectors? This is where things get REALLY interesting


----------



## Fast1one

lycan said:


> Yes.
> 
> Here's the magic : *Adjusting the RELATIVE AMPLITUDES of our two vectors will adjust the PHASE of the resultant sum. Since phase is directly related to ITD ... we've created a virtual source on a cone-of-confusion where no physical driver existed in the first place*
> 
> Let that sink in ...
> 
> To make sure that the "effect" works over a reasonably wide range of frequencies, the amplitude adjustment of the two sources needs to result in a phase (for the summation vector) that's proportional to frequency ... thereby translating the _inter-aural vector phase_ into a true _inter-aural time delay_. *But even in the thirties, there were electronic components ... inductors & capacitors ... whose impedance (or conductance) was/is proportional to frequency, allowing some pretty simple circuitry to electronically realize the required frequency-dependency*
> 
> We're getting so close to a really interesting conclusion ...


I like where you're going with this


----------



## Fast1one

lycan said:


> Thank you !!!
> 
> To summarize Blumlein stereo :
> 
> 1. Two speakers at a fixed angle in physical space
> 2. Adjust the relative AMPLITUDES, to change the PHASE of the resultant SUM
> 3. Make sure the resultant PHASE is proportional to frequency (through some simple electronic manipulation), and you've "created" a "virtual ITD" over a wide frequency range ... thereby putting a virtual source on a cone-of-confusion, where a real one never existed
> 
> Pretty clever, huh? Turns out he also developed recording techniques to easily realize the required "amplitude-panning" law (that frequency-dependent amplitude adjustment needed to make the phase of the sum-vector proportional to frequency), as well as the +/- 45 degree stylus to record two independent channels on a phonograph record.
> 
> Couple other thing to note :
> 
> 1. Each speaker must create an inter-aural vector for the technique to work. In other words, _each_ speaker must be heard by _both_ ears. This is not a "failing" of stereo, it's a _design requirement_. Over the decades, there have been _other_ techniques (some versions of binaural come to mind) that depend on eliminating "crosstalk" between the ears ... but basic stereo isn't one of them.
> 
> 2. I think it's easy to visualize how a source can be located (via "virtual ITD") just about anywhere _between_ the speakers : simply adjust the amplitudes of the two vectors to realize a resultant summation at _any_ phase angle in between.
> 
> New question : Is the vector sum limited to angles BETWEEN the original two vectors? This is where things get REALLY interesting


I would say no, but I am not sure exactly why. Let me see if I understand your question. Are you asking whether the sum phase angle must lie on the plane created by the two original vectors? If so, mathematically yes unless we start throwing cross products into the mix


----------



## lycan

Fast1one said:


> I would say no, but I am not sure exactly why. Let me see if I understand your question. Are you asking whether the sum phase angle must lie on the plane created by the two original vectors? If so, mathematically yes unless we start throwing cross products into the mix


No, I'm asking this :

Original two vectors at +/- 20 degrees. By adjusting relative amplitudes, must the resultant sum always have a phase that's between plus 20 degrees and minus 20 degrees?

Alternatively, is there anything we can do to the amplitudes of the original two vectors ... without changing their phase (cuz the phase would appear to be fixed by their location in physical space) ... that will allow the resultant sum to have a phase angle _outside_ the +/- 20 degree angle?

Still another way : Is there any adjustment we can make to the 'amplitudes'  of the original two speakers to create a virtual ITD source that's _outside_ the boundaries of the original two speakers?


----------



## Fast1one

lycan said:


> No, I'm asking this :
> 
> Original two vectors at +/- 20 degrees. By adjusting relative amplitudes, must the resultant sum always have a phase that's between plus 20 degrees and minus 20 degrees?
> 
> Alternatively, is there anything we can do to the amplitudes of the original two vectors ... without changing their phase (cuz the phase would appear to be fixed by their location in physical space) ... that will allow the resultant sum to have a phase angle _outside_ the +/- 20 degree angle?
> 
> Still another way : Is there any adjustment we can make to the 'amplitudes'  of the original two speakers to create a virtual ITD source that's _outside_ the boundaries of the original two speakers?


It may be late, but I can't seem to find a solution. Need to think outside the box


----------



## SSSnake

A negative amplitude for one of the vectors would push the resultant phase angle outside of the original but.... isn't that the same as 180 phase shift?


----------



## lycan

SSSnake said:


> A negative amplitude for one of the vectors would push the resultant phase angle outside of the original but.... isn't that the same as 180 phase shift?


BINGO !!!!!     

So OK, it was a trick question (sort of)  

YES. If one of the vectors is allowed to "go negative", the vector summation will yield a resultant "outside" the original vector "window" (+/- twenty degrees in our example). And YES, "going negative" is identical to a 180 degree phase shift ... so it's really not an "amplitude adjustment" per se.

For those following along, imagine one of the vectors ... say, the one at -20 degrees, gradually "reducing" in amplitude, until it reaches zero. The SUM vector will be "rotating" toward the vector of fixed amplitude at +20 degrees. When the negative one hits zero amplitude, the SUM vector is identical to the +20 degree vector. But let's not stop there! Allow the reducing vector to shift polarity, as it gradually emerges in the upper _left_ quadrant, and watch what happens to the SUM  The vector that we're changing amplitude never leaves the "straight line" on which it resides .... we're just allowing that straight line to "extend" into the upper left quadrant 

This happens routinely in regular stereo recordings, by the way. It's the fundamental mechanism that allows stereo to stage & image beyond the width of the speakers. For anyone that has heard a REALLY well set up stereo (long discussion about what's required, not for this thread), the speakers WILL literally disappear as a full stage appears before you. Well placed, well focused images in that stage ... and the stage easily extends beyond the boundaries of the speakers.

This "polarity reversal" mechanism is the absolute KEY for what we're going to discuss next : how to electronically manipulate a pair of midbass speakers to "move" them, virtually, to a NEW cone of confusion. A handy trick for anyone contemplating midbass arrays 

But for now, we can recognize that even basic, Blumlein stereo has the "capacity" to place virtual midbass sources on cones-of-confusion where no real driver is located.


----------



## Fast1one

SSSnake said:


> A negative amplitude for one of the vectors would push the resultant phase angle outside of the original but.... isn't that the same as 180 phase shift?


That's why I was hesitant to make that suggestion. Trick question!


----------



## lycan

Now for some fun  (well, i think so anyway)

For a few decades ... i'm sure i don't know the first documented use of this technique ... it's been known that we could "widen" the apparent stage of a stereo configuration by the following :

1. Mix (that is, add) into the LEFT channel an attenuated version of the *(L - R)* signal.

2. Mix (that is, add) into the RIGHT channel an attenuated version of the *(R - L)* signal.

Yes, this is our old friend ... the stereo *DIFFERENCE* signal !! But in this case, we don't add any time delay.

For those following along in this thread, we now know the REASON why it works. Consider a full, "LEFT ONLY" signal. The left speaker gets *"L"*. The right speaker, however, is not silent (as it would be, if we weren't employing this technique) ... the right speaker will now get a little *"-L"* (that is, "minus L"). What happens when the "other" vector in the phase plane "goes a little negative" ???  

The same thing is true, of course, for the right speaker (so if you're thinking ahead, that this technique might work for BOTH front seat passengers at the same time, well ... that's where you would be right).

The problem is, that this technique causes some un-natural "phasiness" in the _midrange_ ... where IID starts to matter, in addition to ITD.

So ... in the late eighties, a clever guy named Klayman, who was working at Hughes Aircraft (of all places), had the simple idea (simple, in retrospect, as most good inventions are) to :

1. Form the *(L - R)* difference signal.
2. EQUALIZE or filter the signal to be _strong_ in the midbass, but _attenuated_ in the midrange.
3. Mix the _equalized_ difference signal back into the original *(L,R)* pair.

And a new company ... called SRS Technologies (or maybe, SRS Labs) was formed  (Remember, in the late 80's, active crossvers were not that common, at least in the home ... so you had to offer the market techniques that would work with "full range" speakers, or multiple-driver boxes with passives inside). I can give a couple patent references for SRS Labs for anyone interested ...

By now, the faithful reader will begin to appreciate where this long, wandering thread is leading. Remember ... we are dealing with _midbass_ arrays 

Let's say you've got a midbass directly in front of you, with it's stereo partner directly in front of the passenger. Maybe, in the floor boards or firewall ramp. Maybe even on the rear shelf ... for a non-turning head, the rear shelf is identical to the firewall. But perhaps you're a little disappointed with stage width in the midbass ... because your drivers are on a "shallower" cone of confusion than the midrange.

Well here's a simple technique that can WIDEN the MIDBASS stage  and "virtually" place the drivers on _steeper_ cones of confusion. Works great in the midbass, much less so in the midrange.

It seems that all my useful contributions to diyma could be lumped into the category : "clever ways to use the (L-R) stereo difference signal" 

Has any company ever offered a similar thing for car stereo? Why, yes they have  (actually, SRS offered .. maybe still do ... some OEM stuff) But howabout an aftermarket box?


----------



## mosconiac

http://www.srslabs.com/products.aspx?id=470

Here's their demonstration of the SRS Experience.

http://www.srslabs.com/experience/HomeEntertainment.aspx

Thank you, Lycan for opening up an entirely new realm of understanding for me. I'm still trying to tie it all together, but you've been an excellent teacher (not so easy to do in the sterile world of the internet where interpersonal communication is rather difficult).


----------



## justinmreina

lycan, apparently your PMs are disabled. 

I would format this into a white paper after the discussion is completed. Is this ok with you? You would get all of the credit, of course.

-Justin


----------



## buchaja

My how things come full x^2 + y^2 + r^2 (starting at the origin, of course)!

a) Threads like this are _the reason_ I joined diyma. Thanks Lycan.
b) I left Hughes in '87. No regrets.
c) I am enjoying SRS in my truck via a new JVC hu I installed about two weeks ago. And, *THANKS TO THE DIYMA COMMUNITY* that little sucker cost me about $30! (I also scored a couple of polk db's for cheap too -- all together, my system cost about $100.

For the purists our there, I ain't done: I have a DRZ sitting in the wings.  But, I have to say, that surround sound is impressive.

Cheers everybody!


----------



## lycan

mosconiac said:


> http://www.srslabs.com/products.aspx?id=470
> 
> Here's their demonstration of the SRS Experience.
> 
> http://www.srslabs.com/experience/HomeEntertainment.aspx
> 
> Thank you, Lycan for opening up an entirely new realm of understanding for me. I'm still trying to tie it all together, but you've been an excellent teacher (not so easy to do in the sterile world of the internet where interpersonal communication is rather difficult).


My pleasure! I've always participated in audio message boards to teach & learn ... I really want this thread to maybe be a companion to Patrick's thread on midbass arrays. He's a much better balance of theoretical & practical than me ...


----------



## lycan

justinmreina said:


> lycan, apparently your PMs are disabled.
> 
> I would format this into a white paper after the discussion is completed. Is this ok with you? You would get all of the credit, of course.
> 
> -Justin


yeah, no PM's for me 

White paper is fine. I'll help proof-read ... but don't worry about personal credit for me. The credit belongs to Patrick for inspiration, and the whole diyma community


----------



## justinmreina

sweet  Now, on with the discussion...


-Justin


----------



## rockinridgeline

Polk Audio SDA SRS speakers from the mid 80's, back when I used to sell them.

There was a cable that ran from one to the other to get a L-R (I think) signal that played through part of the array to create a phantom image far outside of the speakers.


----------



## mosconiac

Those speakers were my first introduction to quality loudspeakers...I was 18 & thought my 6x9 triaxials in my first car were pretty good. LOL. I immediately bought Polk speakers for my car, 6.5" coaxs.

I drooled over pics like this...


----------



## lycan

buchaja said:


> My how things come full x^2 + y^2 + r^2 (starting at the origin, of course)!
> 
> a) Threads like this are _the reason_ I joined diyma. Thanks Lycan.
> b) I left Hughes in '87. No regrets.
> c) I am enjoying SRS in my truck via a new JVC hu I installed about two weeks ago. And, *THANKS TO THE DIYMA COMMUNITY* that little sucker cost me about $30! (I also scored a couple of polk db's for cheap too -- all together, my system cost about $100.
> 
> For the purists our there, I ain't done: I have a DRZ sitting in the wings.  But, I have to say, that surround sound is impressive.
> 
> Cheers everybody!


Howabout that! Nice to e-meet you 

For those interested, here's the two "fundamental" patents on the technology of SRS Labs (well, i know they've expanded their technology a lot, and i haven't kept current ... but these two are some good reading) :

#4,748,669 Stereo Enhancement System (issued May 1988)
#5,892,830 Stereo Enhancement System (issued Apr 1999)

both by Arnold I. Klayman

sadly, I recently heard that Arnold Klayman has passed away 

The more recent patent is shorter, and probably more read-able. Clearly describes the *(L-R)* formulation, equalization, and re-mixing.

For our purposes, we recognize that the technique is VERY well suited to midbass arrays. Here's an example of an array that might be intriguing :

Put some midbasses on the rear shelf. I've argued _against_ this in the past, because of the stage collapse ... they would not reside on a very wide (or steep) cone of confusion. HOWEVER, i was forgetting about the old *(L-R)* mixing technique that's VERY effective in the midbass. They really can be "electronically" moved to new virtual locations, on _wider_ cones-of-confusion, to better align with midrange placement in doors or kicks, for example.

Now ... you've still got the "head turning problem". But that's where some midbass drivers _up front_ can help, i think. Wide as possible, but if not ... mix in a little bit of our new *(L-R)* friend. I think the additional drivers up front will _help_ to de-sensitize the head-turning problem. It won't solve it altogether, but it may be a pretty darn good compromise !!

Or, let's say you've good a midbass under your break pedal, and one in a corresponding spot on the passenger side. Or, maybe under the seats. You did it, to get a better enclosure (maybe even aperiodic, vented outside) than you can get from a door. But the midbass "width" is a little too narrow. This can help, methinx!

By the way, the company that did a similar *(L-R)* technique for the aftermarket car audio market was Audio Control, with their ESP2 & ESP3 products. I'm pretty sure ... but don't quote me ... that they allow the *(L-R)* signal to be equalized by the user before re-mixing (knobs on top of the units). But if memory serves, they recommend an EQ curve pretty similar to what SRS uses : strong in the midbass (where ITD's rule), but attenuated in the midrange. The patent that describes the Audio Control ESP3 is (this took some digging):

#5,113,447 Method and System for Optimizing Audio Imaging in an Automotive Listening Environment (issued May 1992)

by Brian J. Hatley and Richard A. Chinn

A key point for us, if we've got active control over midbass, is that we don't have to worry about EQing the difference signal. It will already be restricted to the midbass, because they're the only drivers we're playing with !! 

Class dismissed


----------



## mosconiac

So...are we speculating that this is the core technology RC was using in his magic black boxes be bragged about? He said those boxes leveraged the Hass effect, so....

I'm curious how he made the rear-mounted drivers work into the higher frequencies required to mate up with the horns...maybe 500-700 Hz. Or...is this further evidence that he may have employed clandestine drivers up front?


----------



## lycan

mosconiac said:


> So...are we speculating that this is the core technology RC was using in his magic black boxes be bragged about? He said those boxes leveraged the Hass effect, so....
> 
> I'm curious how he made the rear-mounted drivers work into the higher frequencies required to mate up with the horns...maybe 500-700 Hz. Or...is this further evidence that he may have employed clandestine drivers up front?


At this point, you guys know everything I know about that (in)famous GN ... and the principles that could make it work


----------



## thehatedguy

I'd guess with the 2" drivers that were on the horns, he could get to 500 hz, and lower with reduced power handling and probably less pattern control.

Jeffe, I owe you a phone call.


----------



## lycan

thehatedguy said:


> Jeffe, I owe you a phone call.


a rare turn of events !!!


----------



## durwood

lycan said:


> a rare turn of events


:laugh: this seems all too familiar...deja-vu almost.

When you simplify and exploit the holes in the human hearing you might realize it can be done with a single pair...


----------



## Se7en

lycan said:


> yeah, no PM's for me


Now I don't feel so bad about not hearing back on that PM I sent you 3 weeks ago...


----------



## ncv6coupe

durwood said:


> :laugh: this seems all too familiar...deja-vu almost.
> 
> When you simplify and exploit the holes in the human hearing you might realize it can be done with a single pair...


durwood do you mind digging a little deeper in this please,


----------



## Patrick Bateman

lycan said:


> My pleasure! I've always participated in audio message boards to teach & learn ... I really want this thread to maybe be a companion to Patrick's thread on midbass arrays. He's a much better balance of theoretical & practical than me ...



Thanks! I should be starting on the midbass arrays in my Accord this weekend, if everything goes according to plan. The new waveguides are ugly, but functional. They're on the center of the dash, not the corners.


----------



## Patrick Bateman

mosconiac said:


> So...are we speculating that this is the core technology RC was using in his magic black boxes be bragged about? He said those boxes leveraged the Hass effect, so....
> 
> I'm curious how he made the rear-mounted drivers work into the higher frequencies required to mate up with the horns...maybe 500-700 Hz. Or...is this further evidence that he may have employed clandestine drivers up front?



Here's my "best guess" as to the design of the midbasses in the GN:


There was a computer controlled servo used to adjust the gains on the amplifiers. Basically at low volumes the rear midbasses were at a very low level, and as the volume increased, the volume was increased further and further. There is no doubt in my mind that there were midbasses under the dash, likely a JBL 5", since those were some of the best you could use at the time. Even now they're VERY competitive.
This takes advantage of the Haas effect, because the rear-mounted midbasses are too close. Clark is "trading" volume to compensate for the fact that they're too close. So even though the front midbasses are much smaller, they're playing louder (At least at moderate SPLs.)
As far as the upper limit of the midbasses, I believe the twelves that he was using are good to 1-2khz... Off-axis, they're probably falling off at 1400hz or so, but that's no big deal since the horns take over at 300 or 500hz.
In audiogroupforum Clark said that he was using delay on the various drivers. I have no idea how he did this.


----------



## ncv6coupe

Patrick Bateman said:


> Thanks! I should be starting on the midbass arrays in my Accord this weekend, if everything goes according to plan. The new waveguides are ugly, but functional. They're on the center of the dash, not the corners.


PB are you going to document here or in the natural bass thread? I'm following you closely on your developments, I know your bread and butter midbass is the bc 8" in the rear quarter but what other speakers you plan on running, and do you have a pic of the center mounted wG's?


----------



## Patrick Bateman

lycan said:


> So ... in the late eighties, a clever guy named Klayman, who was working at Hughes Aircraft (of all places), had the simple idea (simple, in retrospect, as most good inventions are) to :
> 
> 1. Form the *(L - R)* difference signal.
> 2. EQUALIZE or filter the signal to be _strong_ in the midbass, but _attenuated_ in the midrange.
> 3. Mix the _equalized_ difference signal back into the original *(L,R)* pair.
> 
> And a new company ... called SRS Technologies (or maybe, SRS Labs) was formed  (Remember, in the late 80's, active crossvers were not that common, at least in the home ... so you had to offer the market techniques that would work with "full range" speakers, or multiple-driver boxes with passives inside). I can give a couple patent references for SRS Labs for anyone interested ...
> 
> By now, the faithful reader will begin to appreciate where this long, wandering thread is leading. Remember ... we are dealing with _midbass_ arrays
> 
> Let's say you've got a midbass directly in front of you, with it's stereo partner directly in front of the passenger. Maybe, in the floor boards or firewall ramp. Maybe even on the rear shelf ... for a non-turning head, the rear shelf is identical to the firewall. But perhaps you're a little disappointed with stage width in the midbass ... because your drivers are on a "shallower" cone of confusion than the midrange.
> 
> Well here's a simple technique that can WIDEN the MIDBASS stage  and "virtually" place the drivers on _steeper_ cones of confusion. Works great in the midbass, much less so in the midrange.
> 
> It seems that all my useful contributions to diyma could be lumped into the category : "clever ways to use the (L-R) stereo difference signal"
> 
> Has any company ever offered a similar thing for car stereo? Why, yes they have  (actually, SRS offered .. maybe still do ... some OEM stuff) But howabout an aftermarket box?



If anyone wants to give this a try, here's how to do it without buying any hardware:


Get a copy of Foobar2000, or any media player that allows you to run DSP plugins
Get the SRS software - I think this will do it, but read the instructions first to be sure : http://www.srslabs.com/store/store/comersus_viewItem.asp?idProduct=6
Change the output device to your hard drive, instead of speakers. This will create wav files with the processing intact
Copy the wav files over to your Ipod/Zune/USB drive/whatever


----------



## mosconiac

I have a simple question at this point. The premise behind this whole thread has been the implementation of an array of midbasses. It has been shown that the same manipulation accomplished by a physical array could be carried out through electronic means. Hence, we could continue running a stereo pair of midbasses with an associated black box. These midbasses would be mounted deep into the footwell area for minimized PLDs.

Did I miss something PB...in other words, why are you building an array?

BTW, I am assuming that in a typical system, the footwell-mounted midbasses would also serve midrange duties. We would have a "midbass"-sized driver providing all information from subwoofer to tweeter crossover points with electronic manipulation in the range of 80-320 Hz.

Maybe the goal is to allow ourselves to mount several (true) midbass drivers in non-optimum locations & use small midranges (easier to install) in optimum locations.


----------



## jimbno1

I have been trying to follow this thread but some of it has been over my head. Can someone simplify for the slow kid in class?

To my little brain I think I understand some things. Doors bad for midbass, Kick panels better, rear quarter panels maybe even better. Multiple sources on the same cone of confusion can smooth the response and be percieved as a single source. 

Introducing L - R and R - L to the right and left speakers respectively can fool our minds in perceiving that the midbasses are further away which can widen the sound stage. Could someone show the vectors to help me understand this?

Please slap me and call me Sally if I have misunderstood after all your fine work.

I still do not understand how a louder midbass source matters if the drivers are in the same cone of confusion and are being perceived as one driver. Doe it matter if the loudest source is in front or behind us?

It seems like SRS will provide the L-R and R -L signalling. I also think I read in another thread that the RF surround processor would do the same. Are there any other off board devices which can provide this? 

I would prefer to use my P9 combo due to other features. 

What options are out there to implement the SRS concepts? And are the effects controllable or do you simply engage the SRS function and sit back and enjoy sonic nirvana?


----------



## ncv6coupe

Patrick Bateman said:


> There is no doubt in my mind that there were midbasses under the dash, likely a JBL 5", since those were some of the best you could use at the time. Even now they're VERY competitive.
> [*]This takes advantage of the Haas effect, because the rear-mounted midbasses are too close. Clark is "trading" volume to compensate for the fact that they're too close. So even though the front midbasses are much smaller, they're playing louder (At least at moderate SPLs.)
> [*]In audiogroupforum Clark said that he was using delay on the various drivers. I have no idea how he did this.
> [/list]


PB, considering lycan just exploded a can of beans with the vector midbass theory, do you think it is possible that richard created 2 phantom midbass sources most likely most dominant from the fron underdash midbass which would put that source DEEP out on the hood and the 12"s deep out on the trunk, but due to haas and frequency we just could not decipher what was going onbecause of his illusive tongue twisting methods. And last I read somewhere can't remember but the person had close ties with the holdaways and knew a little about the GNB setup that the driver side rear midbass were delayed the most even beyond the front passenger side which was the key. Maybe they were blowing smoke. Hmm


----------



## lycan

mosconiac said:


> I have a simple question at this point. The premise behind this whole thread has been the implementation of an array of midbasses. It has been shown that the same manipulation accomplished by a physical array could be carried out through electronic means. Hence, we could continue running a stereo pair of midbasses with an associated black box. These midbasses would be mounted deep into the footwell area for minimized PLDs.
> 
> Did I miss something PB...in other words, why are you building an array?


The motivation behind an array is to use multiple midbass locations in the vehicle to "smooth" the irregularities in the _amplitude_ response (vs. frequency). That can be hard to do with a single pair ... particularly where cancellations, and the associated response nulls, are manifesting. Through "spatial randomizing", the nulls _in aggregate_ can be minimized.

The counter-argument to arrays is that, since we _can_ localize in the midbass, the midbass imaging cues can get confused or collapsed. In other words ... we can't only pay attention to midbass _amplitude_ response, we must also pay attention to midbass _phase_ response. So this thread offered an analysis of midbass _phase_ ... as ITD ... and offered a simple technique, or "control knob" (an extra degree of freedom, if you will) to help address the phase issue.

EDIT : A single pair of midbasses, even with some clever electronic manipulation to expand them to _wider_ cones-of-confusion, may _still_ cause amplitude irregularities (vs. frequency). So no matter what you do, you may have an amplitude vs. frequency response for a _single_ pair that looks ugly (response nulls due to cancellation). So we very well might be motivated to employ some "spatial randomizing" by mounting midbass drivers at _various_ locations in the vehicle. BUT ... some of those additional drivers might be on shallow cones-of-confusion  The analysis and technique in this thread can help, perhaps, by making sure all midbass drivers ... even at various points in the vehicle ... can all "virtually" reside on the same cone of confusion.


----------



## sqshoestring

I'm thinking this can keep the midbass 'stage' and direction proper while adding more drivers someplace else for additional output, and the array can reduce the head turning directionality issue.

On a side note, I've listened to so many systems that manipulate sound including SRS. I find that they sound great, for a while. I seem to get fatigued from them listening to music, yet on a HT where the processed sounds are occasional or abrupt and not continuous....it works great. Unless they were not setup just so, but some were factory systems. However if you used this only on MB that would be a different story. I don't even have the T/A on in my car now because the battery died one day (my fault, but lost all the settings. It remembered the EQ and xover though, curious) and I was turning it on/off/fiddling before that. It could be due to some tuning issues I have not sure, but I could not get the T/A to sound that great. The other processing features on there I don't care for either. I'll go through all that again when I swap drivers.


----------



## lycan

sqshoestring said:


> I'm thinking this can keep the midbass 'stage' and direction proper while adding more drivers someplace else for additional output, and the array can reduce the head turning directionality issue.
> 
> On a side note, I've listened to so many systems that manipulate sound including SRS. I find that they sound great, for a while. I seem to get fatigued from them listening to music, yet on a HT where the processed sounds are occasional or abrupt and not continuous....it works great. Unless they were not setup just so, but some were factory systems. However if you used this only on MB that would be a different story. I don't even have the T/A on in my car now because the battery died one day (my fault, but lost all the settings. It remembered the EQ and xover though, curious) and I was turning it on/off/fiddling before that. It could be due to some tuning issues I have not sure, but I could not get the T/A to sound that great. The other processing features on there I don't care for either. I'll go through all that again when I swap drivers.


well said  By the way, the original SRS (maybe changed, by now) also did some *(L-R)* manipulation in the _treble_. So yes ... I'm thinking that, by purely & absolutely restricting any *(L-R)* manipulation to the _midbass_, we might be on to something. The precious midrange & treble (vocals, for example) will absolutely be un-effected 

Plus, we're not trying to "artifically" extend the "entire" stereo image. What we're doing here, is really just trying to "align" the _midbass_ with the un-touched midrange & treble.


----------



## Patrick Bateman

ncv6coupe said:


> PB are you going to document here or in the natural bass thread? I'm following you closely on your developments, I know your bread and butter midbass is the bc 8" in the rear quarter but what other speakers you plan on running, and do you have a pic of the center mounted wG's?


Here's some new pics

My Mind's Playing Tricks on Me - More Fun with Waveguides and Psychoacoustics - Page 2 - diyAudio

I'll be documenting the build over there. There are more people interested in ambiophonics on that forum, so it seems appropriate


----------



## Patrick Bateman

jimbno1 said:


> I still do not understand how a louder midbass source matters if the drivers are in the same cone of confusion and are being perceived as one driver. Doe it matter if the loudest source is in front or behind us?



You can't get away from the Haas effect, so YES, location and amplitude is very important.

In a car, two speakers, or four speakers, or ten speakers will be perceived as a single source. The *location* of that source depends on a number of factors, as noted in this thread.

There is a threshold where it's possible to seperate two sound sources; IIRC it's ten milliseconds, which is equivalent to 13.5 ft. Which exceeds the dimensions of our cabin by quite a lot.


----------



## Patrick Bateman

ncv6coupe said:


> PB, considering lycan just exploded a can of beans with the vector midbass theory, do you think it is possible that richard created 2 phantom midbass sources most likely most dominant from the fron underdash midbass which would put that source DEEP out on the hood and the 12"s deep out on the trunk, but due to haas and frequency we just could not decipher what was going onbecause of his illusive tongue twisting methods. And last I read somewhere can't remember but the person had close ties with the holdaways and knew a little about the GNB setup that the driver side rear midbass were delayed the most even beyond the front passenger side which was the key. Maybe they were blowing smoke. Hmm



I dunno. Clark was known to use DSP, but he also used a lot of FUD to confuse competitors.

It's possible to "trick" the brain into thinking that the stage is as deep as the dash of the car, by putting speakers waaaaaaaaay under the dash. It would be particularly convincing if you treated the area near the midbasses to attenuate reflections, because a great deal of our depth perception is based on echoes. That's why you can perceive depth with one ear - you hear the echoes, and your brain does the math.

As noted by Lycan, creating the illusion of width is trickier.


----------



## sqshoestring

lycan said:


> well said  By the way, the original SRS (maybe changed, by now) also did some *(L-R)* manipulation in the _treble_. So yes ... I'm thinking that, by purely & absolutely restricting any *(L-R)* manipulation to the _midbass_, we might be on to something. The precious midrange & treble (vocals, for example) will absolutely be un-effected
> 
> Plus, we're not trying to "artifically" extend the "entire" stereo image. What we're doing here, is really just trying to "align" the _midbass_ with the un-touched midrange & treble.


Yes there is a reason why none of the processing things are very popular for music, not that it can't work for X person or X install. But this is a different use. IMHO most of the processing tries to do too much, and it gets all gimmicky sounding. Then you are back to....the reverb, lol!

You and Patrick and others have some great threads going here. Looking at the GN thread I so agree that midbass weakness is a big issue. So now we can attempt to address it two ways...using huge pro drivers like in the GN (and maybe under dash ones/etc) or an array of more manageable sized MB but with proper imagining if we can implement some processing. Because I am thinking even four 6.5 are not going to do it, well maybe expensive ones but not common ones. But add to that I like to run them lower say near 50Hz. Hmm, now what drivers to buy.....?

One last thing, if the GN (lets say) had those huge pro midbass in the back, and then what 4" under the dash....is there no issue of using two very different drivers to create the same frequency range? I think it can be done because I've done it a lot, but some people are rather against that and with subs in particular. Or did he just work out/EQ the driver responses to negate that? Is it less an issue in this 'array' if that is what he did? I know if you have different peaks in response, in the midrange you can get a lot of movement in the image as the sound hits the different responses...it can be cool...or not...but that is with near the same output on both sets and no processing. Of course we are talking a small midbass range here, he was running up to much higher frequencies. I'm just thinking if you want to run various sizes of midbass to fit your install could it be an issue.


----------



## mitchyz250f

JBL 2206H SPL vs angle chart.


----------



## Patrick Bateman

mitchyz250f said:


> JBL 2206H SPL vs angle chart.


So at 640hz it's omni... Just as I would expect, based on the dimensions.

And a 5" driver is guaranteed to be omni in that frequency.

For the most part, the reason that a 12" driver sounds different than a 5" driver is due to directivity and the size of the radiator.

At 640hz there *is no directivity* so that takes care of problem A.
And listening to the twelve off axis obscures the actual size of the piston. IE, it would sound "bigger" if you were listening on axis, where the ITDs would be more pronounced.

When you turn it on it's side, the ITDs change.


----------



## SSSnake

> I still do not understand how a louder midbass source matters if the drivers are in the same cone of confusion and are being perceived as one driver. Doe it matter if the loudest source is in front or behind us?


Theoretically, it does NOT matter.



> You can't get away from the Haas effect, so YES, location and amplitude is very important.


Haas effect would dictate that you hear the *first* instance rather than the *loudest* (within reasonable limits - roughly 10db).

However, in practice it DOES matter. Even if you put a steep electronic filter on your midbasses they will create sounds at higher freq than the electronic filter would indicate (stemming from harmonic distortion, suspension noises, sympathetic vibrations in the environment, etc). These higher freq noises WILL give away location. For this reason I believe acoustic filters to be necessary to pull off the midbass matrix.


----------



## Patrick Bateman

SSSnake said:


> Theoretically, it does NOT matter.
> 
> 
> 
> Haas effect would dictate that you hear the *first* instance rather than the *loudest* (within reasonable limits - roughly 10db).
> 
> However, in practice it DOES matter. Even if you put a steep electronic filter on your midbasses they will create sounds at higher freq than the electronic filter would indicate (stemming from harmonic distortion, suspension noises, sympathetic vibrations in the environment, etc). These higher freq noises WILL give away location. For this reason I believe acoustic filters to be necessary to pull off the midbass matrix.


Ah, good point.

It's the "balance knob dilemna." Below 1600hz we're sensitive to interaural time delay, so trading intensity (volume) for ITD (location) is futile.

Basically the balance knob can move the highs, but not the lows.

Here's a dutch page with some info on this, translated via Google:

Google Translate

_The fact that lateralization could be based on both interaural time differences and intensity differences has led to experiments that were designed to investigate whether a lateralization caused by a time difference could be reversed by an intensity difference. This is called in English literature "Time Intensity Trading", ie the exchange of intensity against time. The results show that trading is possible. However, there is disagreement in the literature about the extent to which an example by trading in the middle of the head positioned sound image is identical, ie subjectively indistinguishable from a central image created by identical signals to the left and right ear. For precise introspection can for signals contain frequencies below 1500 Hz, in many cases two sound vision, namely a 'time-image' and an 'image intensity. The first is sensitive to time differences, the second shows the trading behavior [5]._


----------



## lycan

It's interesting to remember that Blumlein stereo works by _amplitude_ (or intensity) panning. Relative _amplitude_ adjustments between the 2 source vectors can change the resultant _phase_ of the sum vector. HOWEVER, that's not the whole story ... the _phase_ adjustment in the sum vector must be made proportional to _frequency_, in order for amplitude adjustments to translate into an ITD over a wide frequency range. This requires a specific "panning law" in the amplitude adjustments, which can be accomplished during recording either electronically, or with specific microphones.

In other words, the balance knob ain't gonna cut it, in the frequency range where ITD rules.


----------



## SSSnake

I'm not sure that I am following the Dutch translation but if they are discussing freqs up to 1500Hz then yes intensity does play a role. However, for our definition of midbass in this thread and for drivers on the same cone of confusion it does not (again theoretically). And again within approx 10db.

Jeff's comments, I believe, refer to the two channels used for stereo imaging. My comments were limited to one channel and the same cone of confusion.


----------



## sqshoestring

SSSnake said:


> Theoretically, it does NOT matter.
> 
> 
> 
> Haas effect would dictate that you hear the *first* instance rather than the *loudest* (within reasonable limits - roughly 10db).
> 
> However, in practice it DOES matter. Even if you put a steep electronic filter on your midbasses they will create sounds at higher freq than the electronic filter would indicate (stemming from harmonic distortion, suspension noises, sympathetic vibrations in the environment, etc). These higher freq noises WILL give away location. For this reason I believe* acoustic filters to be necessary *to pull off the midbass matrix.


Not to get OT but....So you are saying once I pull my quad 12s out from behind my rear foam seat, I _can_ install midbass with the new 15s....behind the foam seat, which does a nice job of taking out higher frequencies from the trunk but does not affect sub bass. I tried it in and out, no difference up to the ~60Hz it plays. I would need enclosures or risk VBA... 

Man I could use *10"* midbass D), but then I'm taking up room again with a pair of 15s_ and_ pair of 10s. I need room for an amp rack I'll have to measure. I have the perfect cheap efficient paper 10s for that.





mitchyz250f said:


> JBL 2206H SPL vs angle chart.


I see, it has a very smooth response from 640 down to 100Hz. Sorry I'm not used to that, lol.
http://www.jblpro.com/pub/components/2206.pdff


----------



## Patrick Bateman

sqshoestring said:


> Not to get OT but....So you are saying once I pull my quad 12s out from behind my rear foam seat, I _can_ install midbass with the new 15s....behind the foam seat, which does a nice job of taking out higher frequencies from the trunk but does not affect sub bass. I tried it in and out, no difference up to the ~60Hz it plays. I would need enclosures or risk VBA...
> 
> Man I could use *10"* midbass D), but then I'm taking up room again with a pair of 15s_ and_ pair of 10s. I need room for an amp rack I'll have to measure. I have the perfect cheap efficient paper 10s for that.



The problem with putting midbasses in the rear deck is that the perceived location will be the center of the cone. In a typical sedan, this means that the edge of your soundstage will seem to be about a foot shy of the cabin walls.

That's one of the reasons that the midbasses in the GN are in the quarter panel, instead of the wall between the trunk and the cabin.

The stock locations for rear speakers have the same problem.

One idea I've had is to use bandpass loading and just redirect the vents to the very edge, which would increase the perception of width, since the sound is radiated from the vent.

A simpler solution is to do what the GN did, and put a second set of midbasses up front.


----------



## rawdawg

I'm not convinced that the GN had any "midbasses" upfront. I've heard the GN when the Holdaways owned it with only 6 drivers and by my recollection, the stage was way up front. In fact, at the time, it sounded too far away from the car. I can't imagine, to what purpose, adding additional midbasses would improve the sound.

But, in any case...


----------



## western47

I do love this thread. Thanks for all whom have been the leaders with the info.

Know that we have a decent idea on the midbass can I make a recommendation to where we start a similar dicussion on the midrange?


----------



## mosconiac

western47 said:


> I do love this thread. Thanks for all whom have been the leaders with the info.
> 
> Know that we have a decent idea on the midbass can I make a recommendation to where we start a similar dicussion on the midrange?


I hope there are some discussions upcoming on the midrange. We have nice ideas for tweeters from PB (3kHz on up) and midbass from lycan (80-320Hz), so that leaves a potentially critical "hole" between 320-3kHz.


----------



## thehatedguy

According to who you talk to, the GN may or may not have had speakers in the front of it. Last time I judged a show with Harry Kimura I asked him about the speakers in the front of the GN. He told me that he and Eric Holdaway at the last Perry Master's event the GN was at went over the car with a fine tooth comb and there were NO hidden Bose 4s under the dash of the car. Now Mark Elridge would tell you the same story Clark gives- there were hidden Bose 4s under the dash that were gated off of the dBX expander.

Who do you believe is telling the truth?

But to build a car around principles used in the GN is a slippery slope. The fact is there are 2 stories about the speakers in the front of the car...and god knows how many more stories about everything else that the car did or did not do.

You would probably have more success trying to figure out how to stay dry when you are pissing into the wind than trying to recreate the GN on pictures and story alone.


----------



## Patrick Bateman

mosconiac said:


> I hope there are some discussions upcoming on the midrange. We have nice ideas for tweeters from PB (3kHz on up) and midbass from lycan (80-320Hz), so that leaves a potentially critical "hole" between 320-3kHz.




Here's the challenge that I've had with the midranges in my setup.

As discussed in the two threads, we're not particularly sensitive to interaural time delay above 1600hz. Above 3.2khz it's basically a non-issue.

So that's why I can get away with putting the tweeters in a very oddball location, in the center of the dash.

But this means that I have to run the midranges up to 3.2khz. My JBL 400GTIs can't do it; there's a peak in the response. My B&C 8NDL51s have nice smooth response, but if you mount them in the kick panels you get all kinds of reflections off the driver, steering wheel etc... (They're located in the kicks right now, but I'm well aware that there are better places for a midrange. The kick panels are a great location for midbass, but above 500hz or 1000hz you start getting serious and audible reflections.)

Kind of a catch-22 huh?

Here's what I'm (tentatively) considering:


Get a Peerless 830970. A single driver is good for about 94dB with 10 watts. Go any higher and you'll run into issues, there's only so much power you can dump into a tiny 1" voice coil. (The voice coil on my _tweeters_ is nearly four times bigger!)
Add a second one in an array. That gets you to 100dB. With a center-to-center spacing of 2", you can run them up to 3375hz*. Due to comb filtering, you have to use VERY small drivers if you want to get up to 3 or 6khz.
Change it from a monopole to a dipole. That gets you to 106dB
Add a 2nd set on the other side of the car. That gets you to 112dB

112dB isn't a whole lot, but keep in mind that they won't be breaking a sweat. Each woofer is only getting 10 watts.

At the midrange frequencies, both interaural time delay AND interaural intensity differences are important. Which is a fancy way of saying, "you better locate those midranges very carefully." At low frequencies and at high frequencies we have a lot more leeway to "cheat" when it comes to location.


_*(speed of sound / CTC spacing / 2)
Due to beaming and comb filtering, you want to keep the crossover within one-half wavelength. 1/4 wavelength would be even better, but that would require a 1" woofer!_


----------



## lycan

Patrick Bateman said:


> Here's the challenge that I've had with the midranges in my setup.
> 
> As discussed in the two threads, we're not particularly sensitive to interaural time delay above 1600hz. Above 3.2khz it's basically a non-issue.
> 
> So that's why I can get away with putting the tweeters in a very oddball location, in the center of the dash.
> 
> But this means that I have to run the midranges up to 3.2khz. My JBL 400GTIs can't do it; there's a peak in the response. My B&C 8NDL51s have nice smooth response, but if you mount them in the kick panels you get all kinds of reflections off the driver, steering wheel etc... (They're located in the kicks right now, but I'm well aware that there are better places for a midrange. The kick panels are a great location for midbass, but above 500hz or 1000hz you start getting serious and audible reflections.)
> 
> Kind of a catch-22 huh?
> 
> Here's what I'm (tentatively) considering:
> 
> 
> Get a Peerless 830970. A single driver is good for about 94dB with 10 watts. Go any higher and you'll run into issues, there's only so much power you can dump into a tiny 1" voice coil. (The voice coil on my _tweeters_ is nearly four times bigger!)
> Add a second one in an array. That gets you to 100dB. With a center-to-center spacing of 2", you can run them up to 3375hz*. Due to comb filtering, you have to use VERY small drivers if you want to get up to 3 or 6khz.
> Change it from a monopole to a dipole. That gets you to 106dB
> Add a 2nd set on the other side of the car. That gets you to 112dB
> 
> 112dB isn't a whole lot, but keep in mind that they won't be breaking a sweat. Each woofer is only getting 10 watts.
> 
> At the midrange frequencies, both interaural time delay AND interaural intensity differences are important. Which is a fancy way of saying, "you better locate those midranges very carefully." At low frequencies and at high frequencies we have a lot more leeway to "cheat" when it comes to location.
> 
> 
> _*(speed of sound / CTC spacing / 2)
> Due to beaming and comb filtering, you want to keep the crossover within one-half wavelength. 1/4 wavelength would be even better, but that would require a 1" woofer!_


I agree, the _midrange_ is a more difficult proposition than the _midbass_ ... certainly if you're contemplating arrays.

But here's something worth considering : make sure EACH small driver in the array is the SAME distance to your ear. That will, at least, make the primary lobe in the radiation pattern pointed where it should be. The reflections will still be messy, so be sure to align the _long_ axis of the array in the direction where you want to _minimize_ reflections.

Depending on where the array is located, you could probably even come close to making sure that each driver is the same distance to your ears, even as you _turn_ your head 

Also, it's not _that_ hard to do a "0.5" type crossover, where you cross over to a _single_ driver in the array in the upper midrange  Remember that volume displacement requirements are a function of frequency _squared_ ... so at 2kHz you only need 1/4 of the volume displacement required at 1kHz (for example). This can help side-step ctc spacing rules for large arrays.


----------



## jp88

Patrick Bateman said:


> Here's the challenge that I've had with the midranges in my setup.
> 
> As discussed in the two threads, we're not particularly sensitive to interaural time delay above 1600hz. Above 3.2khz it's basically a non-issue.
> 
> So that's why I can get away with putting the tweeters in a very oddball location, in the center of the dash.
> 
> But this means that I have to run the midranges up to 3.2khz. My JBL 400GTIs can't do it; there's a peak in the response. My B&C 8NDL51s have nice smooth response, but if you mount them in the kick panels you get all kinds of reflections off the driver, steering wheel etc... (They're located in the kicks right now, but I'm well aware that there are better places for a midrange. The kick panels are a great location for midbass, but above 500hz or 1000hz you start getting serious and audible reflections.)
> 
> Kind of a catch-22 huh?
> 
> Here's what I'm (tentatively) considering:
> 
> 
> Get a Peerless 830970. A single driver is good for about 94dB with 10 watts. Go any higher and you'll run into issues, there's only so much power you can dump into a tiny 1" voice coil. (The voice coil on my _tweeters_ is nearly four times bigger!)
> *Add a second one in an array.* That gets you to 100dB. With a center-to-center spacing of 2", you can run them up to 3375hz*. Due to comb filtering, you have to use VERY small drivers if you want to get up to 3 or 6khz.
> Change it from a monopole to a dipole. That gets you to 106dB
> Add a 2nd set on the other side of the car. That gets you to 112dB
> 
> 112dB isn't a whole lot, but keep in mind that they won't be breaking a sweat. Each woofer is only getting 10 watts.
> 
> At the midrange frequencies, both interaural time delay AND interaural intensity differences are important. Which is a fancy way of saying, "you better locate those midranges very carefully." At low frequencies and at high frequencies we have a lot more leeway to "cheat" when it comes to location.
> 
> 
> _*(speed of sound / CTC spacing / 2)
> Due to beaming and comb filtering, you want to keep the crossover within one-half wavelength. 1/4 wavelength would be even better, but that would require a 1" woofer!_


You are moving towards the dark side Patrick


----------



## stoeszilla

A quick thanks for all the info posted in this. I've had to read it at least 3 or 4 times just to attempt to figure out what is being said, all the while resisting the urge to ask "so where the heck do I put the mid-bass already!?!"  

Lycan, does this mean the mid-bass array will integrate into the Lycan arc? Would the peerless 2" driver Patrick has found work for this sort of idea, utilizing the 0.5 crossover? They are super cheap (or were, haven't looked recently) at partsexpress...sorry, I'm trying to sprint before I've learned to crawl (or even sit up straight yet).

Chris


----------



## lycan

stoeszilla said:


> A quick thanks for all the info posted in this. I've had to read it at least 3 or 4 times just to attempt to figure out what is being said, all the while resisting the urge to ask "so where the heck do I put the mid-bass already!?!"
> 
> Lycan, does this mean the mid-bass array will integrate into the Lycan arc? Would the peerless 2" driver Patrick has found work for this sort of idea, utilizing the 0.5 crossover? They are super cheap (or were, haven't looked recently) at partsexpress...sorry, I'm trying to sprint before I've learned to crawl (or even sit up straight yet).
> 
> Chris


lycan's arc was a separate thought, for a _midrange_ line/arc array


----------



## stoeszilla

That's what I was thinking, thanks for clarifying. 
I'm not ready for that tutorial yet...


----------



## lycan

thehatedguy said:


> You would probably have more success trying to figure out how to stay dry when you are pissing into the wind than trying to recreate the GN on pictures and story alone.


all ya gotta do is run backwards _really_ fast. of course that introduces yet another "head turning" problem ... gimme a little while and i'll find a solution


----------



## traceywatts

subscribed


----------



## mosconiac

Is this the blackbox we would need to accomplish the goals presented in this thread given some may want to run only a single pair of midbasses located within the kickpanels (as far & wide adwe can get them)?

We blend the "left" signal to the "right" after filtering it, inverting it, & adjusting its amplitude for R - nL. The opposite would be true for the "right" signal.


----------



## lycan

mosconiac said:


> Is this the blackbox we would need to accomplish the goals presented in this thread given some may want to run only a single pair of midbasses located within the kickpanels (as far & wide adwe can get them)?
> 
> We blend the "left" signal to the "right" after filtering it, inverting it, & adjusting its amplitude for R - nL. The opposite would be true for the "right" signal.


Well done  after all, this IS a DIY forum, right? 

Focus on your MIDBASS speakers. Yes, these are ... and MUST be ... different from your MIDRANGE speakers. Then do this :

1. Form the *L-R* difference signal
2. Mix, or add, a variable-attenuated version of *L-R* into the LEFT channel. Mix, or add, a variable-attenuated version of *R-L* into the RIGHT channel.
Alternatively, just add a little bit of "*-R*" (that is, "*minus R*") into the LEFT channel; and add a little bit of "*-L*" (that is, "*minus L*") into the RIGHT channel. That's what shown in the quoted post. How much is a "little bit"? That's what the variable attenuation is for 

Seriously guys, by DIY standards, this is EASY  You don't even have to "frequency shape" the *L-R* difference signal, if you ONLY apply this to your midbass speakers. So the bandpass filter in the attached diagram is unnecessary, if you have separate, active control of midbass speakers that are separate from midrange speakers.


----------



## Fast1one

lycan said:


> Well done  after all, this IS a DIY forum, right?
> 
> Focus on your MIDBASS speakers. Yes, these are ... and MUST be ... different from your MIDRANGE speakers. Then do this :
> 
> 1. Form the *L-R* difference signal
> 2. Mix, or add, a variable-attenuated version of *L-R* into the LEFT channel. Mix, or add, a variable-attenuated version of *R-L* into the RIGHT channel.
> Alternatively, just add a little bit of "*-R*" (that is, "*minus R*") into the LEFT channel; and add a little bit of "*-L*" (that is, "*minus L*") into the RIGHT channel. That's what shown in the quoted post. How much is a "little bit"? That's what the variable attenuation is for
> 
> Seriously guys, by DIY standards, this is EASY  You don't even have to "frequency shape" the *L-R* difference signal, if you ONLY apply this to your midbass speakers. So the bandpass filter in the attached diagram is unnecessary, if you have separate, active control of midbass speakers that are separate from midrange speakers.


Wouldn't you want to do this for the midrange speakers as well though? 

Referencing this image:










I feel like it would be beneficial to do this for the midrange as well. 

Link: Matrix Surround for Music


----------



## lycan

Fast1one said:


> Wouldn't you want to do this for the midrange speakers as well though?
> 
> Referencing this image:
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> I feel like it would be beneficial to do this for the midrange as well.
> 
> Link: Matrix Surround for Music


somebody not reading the whole thread


----------



## Fast1one

lycan said:


> somebody not reading the whole thread


Confusing concepts with others. I definitely got it now


----------



## Niebur3

lycan said:


> Well done  after all, this IS a DIY forum, right?
> 
> Focus on your MIDBASS speakers. Yes, these are ... and MUST be ... different from your MIDRANGE speakers. Then do this :
> 
> 1. Form the *L-R* difference signal
> 2. Mix, or add, a variable-attenuated version of *L-R* into the LEFT channel. Mix, or add, a variable-attenuated version of *R-L* into the RIGHT channel.
> Alternatively, just add a little bit of "*-R*" (that is, "*minus R*") into the LEFT channel; and add a little bit of "*-L*" (that is, "*minus L*") into the RIGHT channel. That's what shown in the quoted post. How much is a "little bit"? That's what the variable attenuation is for
> 
> Seriously guys, by DIY standards, this is EASY  You don't even have to "frequency shape" the *L-R* difference signal, if you ONLY apply this to your midbass speakers. So the bandpass filter in the attached diagram is unnecessary, if you have separate, active control of midbass speakers that are separate from midrange speakers.


I sent sent a PM but though I'd post on here for anyone else thinking the same thing I am.

I have a P9 combo with the mid bass playing from 63-250. I am assuming we are talking about the Processor doing it's job and the amp doing it's job, and this "black box" is going between the amp and the speakers themselves. If that is the case, what exactly do I buy to do all the mixing and adding a little bit of this and a little bit of that? Links on where ever these items get bought would be awesome since I'm not sure exactly what I am looking for?


----------



## Fast1one

Niebur3 said:


> I sent sent a PM but though I'd post on here for anyone else thinking the same thing I am.
> 
> I have a P9 combo with the mid bass playing from 63-250. I am assuming we are talking about the Processor doing it's job and the amp doing it's job, and this "black box" is going between the amp and the speakers themselves. If that is the case, what exactly do I buy to do all the mixing and adding a little bit of this and a little bit of that? Links on where ever these items get bought would be awesome since I'm not sure exactly what I am looking for?


A circuit would need to be designed to go in between your processor and amp. It's not as simple as buying a few generic parts. The circuit will be relatively simple to construct.

A while ago I built a cross-feed circuit for headphones to create a virtual sound stage as opposed to the "Super stereo" that you experience with headphones.

Headphone xfeed

The premise is essentially the same. You wouldn't need to band -limit the cross-feed because you are using your processor to do that. All you would need to do is flip the phase on the cross-feed signal for both channels. These channels would then feed to an adjustable attenuating circuit (a simple stereo potentiometer may be enough) and then fed to two summing amplifiers, using opamps. The L would be summed with the attenuated -R and R with the attenuated -L

I took a circuits class a couple quarters ago. I can probably throw a circuit together for testing purposes, but I currently don't have my book to do so. I don't know enough about the industry to suggest a good opamp and what not. We were using the LM741 for pretty much everything in the lab, but I don't know how good or bad the sonic characteristics are.


----------



## Niebur3

Fast1one said:


> A circuit would need to be designed to go in between your processor and amp. It's not as simple as buying a few generic parts. The circuit will be relatively simple to construct.
> 
> A while ago I built a cross-feed circuit for headphones to create a virtual sound stage as opposed to the "Super stereo" that you experience with headphones.
> 
> Headphone xfeed
> 
> The premise is essentially the same. You wouldn't need to band -limit the cross-feed because you are using your processor to do that. All you would need to do is flip the phase on the cross-feed signal for both channels. These channels would then feed to an adjustable attenuating circuit (a simple stereo potentiometer may be enough) and then fed to two summing amplifiers, using opamps. The L would be summed with the attenuated -R and R with the attenuated -L
> 
> I took a circuits class a couple quarters ago. I can probably throw a circuit together for testing purposes, but I currently don't have my book to do so. I don't know enough about the industry to suggest a good opamp and what not. We were using the LM741 for pretty much everything in the lab, but I don't know how good or bad the sonic characteristics are.


I like the Burr-Brown 2604's. If this something you could design and sell? Money making opportunity .


----------



## Fast1one

Niebur3 said:


> I like the Burr-Brown 2604's. If this something you could design and sell? Money making opportunity .


I don't trust myself enough to design something well  I'm an aspiring mechanical engineer, not an EE. lol...

Ideally I would want a PCB made and I can just distribute that for DIY. I don't really have the time to build a full circuit unfortunately.

Let me ask a couple of my EE friends when I get back and see if they are up to the challenge. Not really interested in making a huge profit or anything. Maybe make a single run of 50-100PCBs and slowly distribute them.


----------



## Niebur3

Fast1one said:


> I don't trust myself enough to design something well  I'm an aspiring mechanical engineer, not an EE. lol...
> 
> Ideally I would want a PCB made and I can just distribute that for DIY. I don't really have the time to build a full circuit unfortunately.
> 
> Let me ask a couple of my EE friends when I get back and see if they are up to the challenge. Not really interested in making a huge profit or anything. Maybe make a single run of 50-100PCBs and slowly distribute them.


Cool...keep me on your list!


----------



## Fast1one

Niebur3 said:


> Cool...keep me on your list!


I will probably make a prototype and distribute it for testing. Since you were the first to show interest you will be first on the list to test it. We can distribute it to multiple users and if it's a success, I'll proceed. 

The kit actually wouldn't be that hard to make. Just need to design a suitable power supply for the op amps. If it's easy, I may just break out the soldering iron and make a small run to sell. Would be a fun project


----------



## Niebur3

Fast1one said:


> I will probably make a prototype and distribute it for testing. Since you were the first to show interest you will be first on the list to test it. We can distribute it to multiple users and if it's a success, I'll proceed.
> 
> The kit actually wouldn't be that hard to make. Just need to design a suitable power supply for the op amps. If it's easy, I may just break out the soldering iron and make a small run to sell. Would be a fun project


Sweet, count me in.


----------



## Fast1one

*To Lycan and mosconiac*,

Thank you mosconiac for providing us with the diagram on post #149. I am thinking of contacting my colleagues and making a prototype for testing and possible distribution. Before I proceed, it would be great if I got your permission to do so. I am not looking to churn a huge profit or anything, just looking to help out the community. We are college students willing to work for minimum wage 

The details are yet to be determined, but the idea is to replicate the diagram provided by mosconiac. An attenuating circuit for the -R/-L will have individual adjustments for both channels. Not sure if individual attenuators are needed but why not?

I am considering using the Burr Brown opa2604 opamps (thanks Niebur3!) as a summing amplifier. The power supply will be regulated with an operating voltage from 9-15V. The input voltage will be designed for up to 8V with an adjustable output voltage with a maximum of 4-6V. 

If I have the go ahead to proceed, I will contact my friends and start a thread in the near future outlining the details and distributing a prototype for testing. Thank you for an excellent thread with great information


----------



## lycan

Permission? Hell it's not really even my idea! Mixing in a little (*L-R*) has been around for a while. All we're really recommending is a simple, brute force method of "frequency shaping" the (*L-R*) ... SRS idea ... by simply applying it to the midbass speakers _only_ 

All ya really gotta do is mix in a variable amount of "*-L*" (minus L) into the right midbass channel , and a variable amount of "*-R* (minus R) " into the left midbass channel.

A few words of caution : take heed!

You might build one, and think the idea sucks because of too much noise ... when in fact, the problem was that you didn't use a properly isolated power supply for the opamps, and you've nasty ground loops in your system.

Or, you might be tempted to use fancy-pants megabuck opamps, and get nasty oscillations from sloppy compensation.

Also, don't expect miracles ... all we're trying to do is "align" narrower midbass drivers (rear-shelf, under-seat, or in-floor) with wider midrange drivers (kicks or doors). The idea is certainly NOT the panacea for all stage woes ... and if you "push " the concept too far, you could end up with a "midbass hole" in the center 

What i'm saying is this : good ideas can "look" bad, due to bad execution!


----------



## Fast1one

You initiated the discussion! It would be wrong for me to NOT ask for permission. Thats my opinion of course 

I completely agree with you on execution. I am by no means committed to building such a circuit. I have a lot on my plate with school as is. My plan of attack would be to design, prototype and the community would need to do the majority of the testing. I simply don't have the time to do a full scale project (a simple PCB would be ideal). 

If anything, I just want to do it for the experience. I'm sure my friends would love the idea as well, especially if we formalized it and created a paper on it 

I'll contact them next week and see if they are interested in designing such a circuit. We can test it in my car; if that passes the noise test, ANYTHING will 



lycan said:


> Permission? Hell it's not really even my idea! Mixing in a little (*L-R*) has been around for a while. All we're really recommending is a simple, brute force method of "frequency shaping" the (*L-R*) ... SRS idea ... by simply applying it to the midbass speakers _only_
> 
> All ya really gotta do is mix in a variable amount of "*-L*" (minus L) into the right midbass channel , and a variable amount of "*-R* (minus R) " into the left midbass channel.
> 
> A few words of caution : take heed!
> 
> You might build one, and think the idea sucks because of too much noise ... when in fact, the problem was that you didn't use a properly isolated power supply for the opamps, and you've nasty ground loops in your system.
> 
> Or, you might be tempted to use fancy-pants megabuck opamps, and get nasty oscillations from sloppy compensation.
> 
> Also, don't expect miracles ... all we're trying to do is "align" narrower midbass drivers (rear-shelf, under-seat, or in-floor) with wider midrange drivers (kicks or doors). The idea is certainly NOT the panacea for all stage woes ... and if you "push " the concept too far, you could end up with a "midbass hole" in the center
> 
> What i'm saying is this : good ideas can "look" bad, due to bad execution!


----------



## mosconiac

I wish I could help you design the physical circuits & populate a home-brew PCB, but it's been FAR too long since I cracked vendors app sheets. I used to scour those integrated circuit guides religiously (when I was an MSEE candidate).

I designed a 4-way. analog active crossover for mobile use back then but never implemented it. Everything was ready to build (even had silk-screen ideas for the faceplate), but I graduated & found other things to worry about. Today, its obsolete and my DIY journeys don't venture beyond the occasional pair of speakers.


----------



## MarkZ

I haven't read the entire thread in depth yet, so forgive me if this question has been addressed.

If a single midbass driver is located behind us, perhaps in the center of the backseat, would it be useful to install a midrange in the center channel within the same "cone of confusion" (or whatever it was you called it...) to try to pull the perception of the bass forward and center? My thought is that the directional cues that accompany the upper midrange (~1kHz) will help draw the localization of the lower midrange to the same location since it's coming from the same driver -- and then the crossover point between the lower midrange and the midbass driver (installed in the rear, sharing the same "cone of confusion") will help transition the bass to the front location.

I guess what I'm really asking is if high frequency cues aid us in lower frequency localization by virtue of being part of the same physical speaker. And then I'm asking if we can use the crossover point between that driver and the midbass driver to help pull the midbass driver to the same location perceptually, IF it shares the same cone of confusion.


----------



## lycan

MarkZ said:


> I haven't read the entire thread in depth yet, so forgive me if this question has been addressed.
> 
> If a single midbass driver is located behind us, perhaps in the center of the backseat, would it be useful to install a midrange in the center channel within the same "cone of confusion" (or whatever it was you called it...) to try to pull the perception of the bass forward and center? My thought is that the directional cues that accompany the upper midrange (~1kHz) will help draw the localization of the lower midrange to the same location since it's coming from the same driver -- and then the crossover point between the lower midrange and the midbass driver (installed in the rear, sharing the same "cone of confusion") will help transition the bass to the front location.
> 
> I guess what I'm really asking is if high frequency cues aid us in lower frequency localization by virtue of being part of the same physical speaker. And then I'm asking if we can use the crossover point between that driver and the midbass driver to help pull the midbass driver to the same location perceptually, IF it shares the same cone of confusion.


first, i didn't coin the term "cone of confusion". Second, a driver placed "front & center" would _not_ be on the same cone as a driver right behind you ... if by "center" you mean centrally located between both front seat passengers. Finally, you don't want _either_ single midbass driver to be localized to the center. You want each midbass driver, individually, localized as far left, and as far right, as the midrange. In other words, you want the midbass on the _same_, typically steep-as-possible cone-of-confusion, as the midrange.

Sometimes, ENCLOSURE concerns prevent us from putting each midbass as "steeply" left & right as the midrange. That's where the *L-R* trick can help 

Example : midrange placed on "steep" cones in the kicks. But for best enclosure, we put midbass on a "shallower" cone : under seats, in floor directly in front of seats, on rear shelf directly behind our heads ... these would all be on cones "shallower" than the midrange cones. What we would like, is an electronic trick or "EQ" that could _localize_ the midbass drivers to the _steeper_ cones, the ones where the midrange drivers are. And we'd like that trick to work for _both_ front seat passengers _at the same time_, thank you  That's where a little *L-R* mixing can help.

In any case, a "centrally" located midbass "sound" will appear to be "centrally" located, by _both_ midbass drivers playing the same midbass amplitude ... even though there is no midbass driver that is located in the center. This is a simple virtue of Blumlein Stereo. The "image" is certainly skewed for a non-central listener, which is a related, but somewhat "orthogonal" problem, to getting the midbass drivers as wide as possible ... if not physically, then electronically


----------



## MarkZ

There's a point at which stereo information present in the recording begins to emerge, and that tends to occur somewhere in the "midbass" range (depending on the recording). LFE encoding is typically...what...120Hz? So, clearly, in this range, wide left and wide right doesn't seem to be on the forefront of people's minds at low frequencies in a number of applications. I'm curious how high we can extend this mono channel before we dilute stereo too much. 150? 180?

But I'm not clear on how a center front and center rear can't be placed in the same cone. I'll review the thread to see if I can figure out where you're coming from. If the center front is roughly the same distance and angle from your head as the center rear, it should share the same cone, right?


----------



## stills

ok i know this is oversimplified, i'm just pondering at work.

how about mono midbass up front.


----------



## lycan

MarkZ said:


> There's a point at which stereo information present in the recording begins to emerge, and that tends to occur somewhere in the "midbass" range (depending on the recording). LFE encoding is typically...what...120Hz? So, clearly, in this range, wide left and wide right doesn't seem to be on the forefront of people's minds at low frequencies in a number of applications. I'm curious how high we can extend this mono channel before we dilute stereo too much. 150? 180?
> 
> But I'm not clear on how a center front and center rear can't be placed in the same cone. I'll review the thread to see if I can figure out where you're coming from. If the center front is roughly the same distance and angle from your head as the center rear, it should share the same cone, right?


maybe i misunderstood your question ...

If the question is: 

"My _left_ midbass is _physically_ located just fine, and my _right_ midbass is also _physically_ located just fine, but i want to add a CENTER speaker in the dash. Problem is, I have no room for a LARGE center, so can I just put the midrange and tweet on the dash, and put the center midbass speaker on the REAR shelf, in the middle of the rear shelf?"

The answer is yes, and no  YES, it will be on the same cone of confusion, and therefore localize-ably indistinguishable from a midbass coincident with the front midrange  ... until, that is, you turn your head


----------



## MarkZ

No, my question is more like this...

My mono midbass driver is located in the middle of my backseat. My midrange drivers (200Hz+) are located in the left and right doors. Could adding the same midrange driver (200Hz+) to the center of the windshield not only have the typical effects that we expect center channels to have, but also pull the perception of bass forward if it resides in roughly the same "cone of confusion" as the midbass driver in the backseat?

In other words, can we harness the "cone of confusion" in a way that allows us to use the midrange drivers to help steer midbass localization?


----------



## ErinH

Maybe Mark is asking this, but if not, I am...

At one point does stereo separation for sub/midbass not matter? IE: Do you need to have stereo below 50hz? Does it matter if you do or do not? I'm assuming that at some point the location cues does not matter, so at which point would it be?
Would your need to have stereo mid/subbass depend only on the recording? 


Kind of a confusing subject, I guess. And I'm assuming it depends only on the recording?


----------



## lycan

MarkZ said:


> No, my question is more like this...
> 
> My mono midbass driver is located in the middle of my backseat. My midrange drivers (200Hz+) are located in the left and right doors. Could adding the same midrange driver (200Hz+) to the center of the windshield not only have the typical effects that we expect center channels to have, but also pull the perception of bass forward if it resides in roughly the same "cone of confusion" as the midbass driver in the backseat?
> 
> In other words, can we harness the "cone of confusion" in a way that allows us to use the midrange drivers to help steer midbass localization?


don't know ... but if you have only a mono midbass driver, why do you care where it's located?


----------



## lycan

bikinpunk said:


> Maybe Mark is asking this, but if not, I am...
> 
> At one point does stereo separation for sub/midbass not matter? IE: Do you need to have stereo below 50hz? Does it matter if you do or do not? I'm assuming that at some point the location cues does not matter, so at which point would it be?
> Would your need to have stereo mid/subbass depend only on the recording?
> 
> 
> Kind of a confusing subject, I guess. And I'm assuming it depends only on the recording?


In the pressure zone of the car (frequencies where 1/4 wavelength is longer than any interior dimension), you can't localize.

In free space, there are other low frequency limits to ITD localization also ... but don't know off the top of my head


----------



## ErinH

lycan said:


> In the pressure zone of the car (frequencies where 1/4 wavelength is longer than any interior dimension), you can't localize.
> 
> In free space, there are other low frequency limits to ITD localization also ... but don't know off the top of my head


So, figure a car on average is 6x5' in the cabin (just guesstimating here)...
That means you're around 700-800hz for a 1/4 wavelength.

That seems pretty high to me.


----------



## lycan

bikinpunk said:


> So, figure a car on average is 6x5' in the cabin (just guesstimating here)...
> That means you're around 700-800hz for a 1/4 wavelength.
> 
> That seems pretty high to me.


at 1100 ft/sec, 100 hz has a wavelength of 11 feet.

one quarter wavelength of 100 Hz is 2.75 ft.

One quarter wavelength of 100 Hz is _less_ than a physical interior dimension. Therefore it's _possible_ that 100Hz might be localizable by ITD in a car.

50Hz is really pushing it, though ... not gonna be localizable in a car.


----------



## trebor

Explains the popularity here of ultra low sub xover points.


----------



## ErinH

lycan said:


> at 1100 ft/sec, 100 hz has a wavelength of 11 feet.
> 
> one quarter wavelength of 100 Hz is 2.75 ft.
> 
> One quarter wavelength of 100 Hz is _less_ than a physical interior dimension. Therefore it's _possible_ that 100Hz might be localizable by ITD in a car.
> 
> 50Hz is really pushing it, though ... not gonna be localizable in a car.


Ah, drat!

I was dividing by 4 instead of multiplying, LOL! 
I'm sitting here with a huge spreadsheet open and mathcad up calculating the flow rate of gaseous nitrogen going through an LN2 bath so I can get a proper flow rate going into a test article. So, I plopped 5/4 into my open spreadsheet instead of realizing it should be 5*4. I didn't even remember my associative properties. LOL!

I got it now. Back to the drawing board, cabin min length would be around 5'. So, 5ft*4 = 20'. 55hz = ~20ft. 

Now, _that_ makes more sense.


----------



## lycan

bikinpunk said:


> Ah, drat!
> 
> I was dividing by 4 instead of multiplying, LOL!
> I'm sitting here with a huge spreadsheet open and mathcad up calculating the flow rate of gaseous nitrogen going through an LN2 bath so I can get a proper flow rate going into a test article. So, I plopped 5/4 into my open spreadsheet instead of realizing it should be 5*4. I didn't even remember my associative properties. LOL!
> 
> I got it now. Back to the drawing board, cabin min length would be around 5'. So, 5ft*4 = 20'. 55hz = ~20ft.
> 
> Now, _that_ makes more sense.


no worries  Just remind me to stay away ... in fact, FAR away ... from any gaseous nitrogen lab you're working in


----------



## ErinH

Shoot, me too!


----------



## lycan

I feel compelled to mention, or summarize, the following:

I think there IS some merit to a midbass array in a car. The main virtue, as Patrick has elaborated elsewhere, is to help the rough/irregular amplitude response in the midrange in most vehicles. The response can be rough/irregular (peaks & nulls) because of exciting constructive & desctructive "modes" in the midbass. More "random" placement around the cabin can help "smooth" out these modes. Also, getting strong sub-bass in a car is like falling off a log ... much trickier to get midbass strong enough to "keep up". Multiple drivers can help this integration, also. Finally, some first-choice locations ... like doors ... present less-than-optimal midbass enclosures.

Previously, I had argued against midbass arrays because of possible stage collapse. That is, with no attention paid to ITD's and confusion conses, midbass _localization_ will suffer with an array of randomly placed drivers. I offered this thread, because I think the technique described can help solve that problem  

The single problem that remains, however, is the head-turning problem  Best we can offer so far on that is this : keep some drivers up front, to help reduce sensitivity to head turning.

What this boils down to, for me, is this : one pair of drivers up front, kicks or doors or floors, maybe another pair on that rear shelf (that's just sitting there waiting to get used). Apply *L-R* mixing to any driver pair that is NOT as wide as you would like. Having at least one pair upfront _should_ desensitize the head-turning problem.

On a final note, the 3 pieces of extra signal processing required are :

1. *L-R* difference. Already covered in other threads ... trivial with *balanced* signals.
2. *Attenuation*
3. Mixing, or *summing*, back into main stereo pairs

In addition to the Audio Control equipment already mentioned, I think a halfway-clever guy could readily identify a couple pieces of ZAPCO gear that could already do the trick  Zapco is already *balanced*, they offer nifty little *attenuators* like the SLB-U, and probably an amp channel or two that allow *summation* 

that's all i got

EDIT : oops i already lied, that's not _quite_ all i got 

A _really_ clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction. In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) *L-R* _ambience_ signal (covered to death in those dreaded & controversial "rear fill" threads). In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo *L,R* pair _plus_ and attenuated *L-R* signal to "widen" the midbass confusion cone. Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course. In short, there's no reason why a single pair of speakers can't be employed for _both_ ambience generation, and midbass reinforcement  yeah, let the speakers "sum" those interesting signals!

howz that for clever use of that rear shelf, from which we were all told we need to just remove the speakers ???


----------



## bbfoto

lycan said:


> A _really_ clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction. In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) *L-R* _ambience_ signal (covered to death in those dreaded & controversial "rear fill" threads). In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo *L,R* pair _plus_ and attenuated *L-R* signal to "widen" the midbass confusion cone. Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course. In short, there's no reason why a single pair of speakers can't be employed for _both_ ambience generation, and midbass reinforcement  yeah, let the speakers "sum" those interesting signals!
> 
> howz that for clever use of that rear shelf, from which we were all told we need to just remove the speakers ???


^^^ Behold the new "LycanSound Spacial Processor"! 

Or is that "Spatial"? :/ Engrish seems so strange at times, lol. 

Anyway, Build it and they will come... 

Or did Andy build and finally release a product that achieves these or similar results?

Many thanks to Jeff and Patrick for taking the time (and having the patience) to put together this excellent thread. Kudos.


----------



## sqshoestring

Agreed, fantastic thread. I just wonder what all that would do far as the main reason I often use rears full range with no processing....to make my left go to my left instead of in front of me.


----------



## lycan

bbfoto said:


> ^^^ Behold the new "LycanSound Spacial Processor"!
> 
> Or is that "Spatial"? :/ Engrish seems so strange at times, lol.
> 
> Anyway, Build it and they will come...
> 
> Or did Andy build and finally release a product that achieves these or similar results?
> 
> Many thanks to Jeff and Patrick for taking the time (and having the patience) to put together this excellent thread. Kudos.


i'm just _always_ looking for clever ways to use those big holes in the rear deck


----------



## ib2qwik2c

lycan said:


> I feel compelled to mention, or summarize, the following:
> 
> I think there IS some merit to a midbass array in a car. The main virtue, as Patrick has elaborated elsewhere, is to help the rough/irregular amplitude response in the midrange in most vehicles. The response can be rough/irregular (peaks & nulls) because of exciting constructive & desctructive "modes" in the midbass. More "random" placement around the cabin can help "smooth" out these modes. Also, getting strong sub-bass in a car is like falling off a log ... much trickier to get midbass strong enough to "keep up". Multiple drivers can help this integration, also. Finally, some first-choice locations ... like doors ... present less-than-optimal midbass enclosures.
> 
> Previously, I had argued against midbass arrays because of possible stage collapse. That is, with no attention paid to ITD's and confusion conses, midbass _localization_ will suffer with an array of randomly placed drivers. I offered this thread, because I think the technique described can help solve that problem
> 
> The single problem that remains, however, is the head-turning problem  Best we can offer so far on that is this : keep some drivers up front, to help reduce sensitivity to head turning.
> 
> What this boils down to, for me, is this : one pair of drivers up front, kicks or doors or floors, maybe another pair on that rear shelf (that's just sitting there waiting to get used). Apply *L-R* mixing to any driver pair that is NOT as wide as you would like. Having at least one pair upfront _should_ desensitize the head-turning problem.
> 
> On a final note, the 3 pieces of extra signal processing required are :
> 
> 1. *L-R* difference. Already covered in other threads ... trivial with *balanced* signals.
> 2. *Attenuation*
> 3. Mixing, or *summing*, back into main stereo pairs
> 
> In addition to the Audio Control equipment already mentioned, I think a halfway-clever guy could readily identify a couple pieces of ZAPCO gear that could already do the trick  Zapco is already *balanced*, they offer nifty little *attenuators* like the SLB-U, and probably an amp channel or two that allow *summation*
> 
> that's all i got
> 
> EDIT : oops i already lied, that's not _quite_ all i got
> 
> A _really_ clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction. In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) *L-R* _ambience_ signal (covered to death in those dreaded & controversial "rear fill" threads). In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo *L,R* pair _plus_ and attenuated *L-R* signal to "widen" the midbass confusion cone. Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course. In short, there's no reason why a single pair of speakers can't be employed for _both_ ambience generation, and midbass reinforcement  yeah, let the speakers "sum" those interesting signals!
> 
> howz that for clever use of that rear shelf, from which we were all told we need to just remove the speakers ???


Awesome read! This thread should be a sticky. Took me a while but I think I understand the gist of what's been said. Now if only a really clever guy could start selling processors that would do the above...


----------



## Pad

Lycan, I was wondering if the SDA Polk method (4 speakers, channels with inverted phase) would be a different approach for the same obejctive, the L-R signal.

You can check the white paper here: http://www.polkaudio.com/downloads/whitepapers/SDA_WhitePaper.pdf

I've time alignment (up to 500cm), active crossover and independent gain to try it, but getting the L-R is the problem. Should this method be acceptable, I think those cheap paper cone factory OEM stereo might be suitable for this rear shelf possibility.


----------



## SVOEO

Thanks to all you contributors on this thread- lots to absorb here, even for someone who thinks he knows acoustics pretty damn well.


----------



## diebenkorn

On a final note, the 3 pieces of extra signal processing required are :

1. L-R difference. Already covered in other threads ... trivial with balanced signals.
2. Attenuation
3. Mixing, or summing, back into main stereo pairs

In addition to the Audio Control equipment already mentioned, I think a halfway-clever guy could readily identify a couple pieces of ZAPCO gear that could already do the trick Zapco is already balanced, they offer nifty little attenuators like the SLB-U, and probably an amp channel or two that allow summation

that's all i got

EDIT : oops i already lied, that's not quite all i got

A really clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction. In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) L-R ambience signal (covered to death in those dreaded & controversial "rear fill" threads). In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo L,R pair plus and attenuated L-R signal to "widen" the midbass confusion cone. Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course. In short, there's no reason why a single pair of speakers can't be employed for both ambience generation, and midbass reinforcement yeah, let the speakers "sum" those interesting signals!

howz that for clever use of that rear shelf, from which we were all told we need to just remove the speakers ???


Does anybody know if this is possible with a 701 and if so can you PM with the details? Thanks


----------



## audioanarchist

I'm trying to understand this and thinking about how I can accomplish this in my vehicle. So since I don't use my rear speakers (and of course my rear outputs on my deck)in my car and have an active amp dedicated to my mid basses, couldn't I just fade to the front almost all the way and then combine the faded(rear) L into the Full signal (front)R and vice versa? Then just send that mixed signal to only my mid bass amp? Then I could adjust the strength of the attenuated signal right from my deck using the fader.

I plan on running 2 6 1/2's (anarchy's) under each seat up front (kindof under the knees)so would I only wanna do it to the ones towards the inside of the car (closest to the center council) and leave the outer ones (closer to the doors) to play normally? I have 4 channels on my mid bass amp so I could do it that way. Or would it be best to send the combined signal to all my mid basses?


----------



## sqshoestring

If you have quad midbass then you already have a kind of midbass array. The processed signals to the rears are more for ambiance use than anything else and I don't know how that would work in front. The main idea with a MB array is midbass in other parts of the car can add to the midbass output yet sound like they are all still in the front...if it is implemented properly.


----------



## audioanarchist

I'm mostly worried about the 2 under the seats closest to the center council as I've read that you wanna have them as wide as possible but, these 2 will be anything but wide. I could maybe just run one mid bass under the seats and place them as close to the outside of each seat but, I'm concerned about SPL cause I'm a metal head and like it loud. Mid bass has always been a problem in my systems and I'd like to make sure that that's not the case this time around. The mids will be in the kicks and I'm not even gonna use the door location because I don't wanna mess with it (to much work making my door a good mid bass enclosure). If the L-R difference would help my mid bass array sound better I'd like to try it. I was just wondering if the way I described doing it would work.


----------



## Patrick Bateman

audioanarchist said:


> I'm trying to understand this and thinking about how I can accomplish this in my vehicle. So since I don't use my rear speakers (and of course my rear outputs on my deck)in my car and have an active amp dedicated to my mid basses, couldn't I just fade to the front almost all the way and then combine the faded(rear) L into the Full signal (front)R and vice versa? Then just send that mixed signal to only my mid bass amp? Then I could adjust the strength of the attenuated signal right from my deck using the fader.
> 
> I plan on running 2 6 1/2's (anarchy's) under each seat up front (kindof under the knees)so would I only wanna do it to the ones towards the inside of the car (closest to the center council) and leave the outer ones (closer to the doors) to play normally? I have 4 channels on my mid bass amp so I could do it that way. Or would it be best to send the combined signal to all my mid basses?


Probably the easiest and cheapest ways to mess around with this processing is by processing the music on your computer, then burning the wavs. I had good results with this using winamp and foobar2000. And it's free!


----------



## audioanarchist

Patrick Bateman said:


> Probably the easiest and cheapest ways to mess around with this processing is by processing the music on your computer, then burning the wavs. I had good results with this using winamp and foobar2000. And it's free!


That sounds like a pain in the ass. Whats the easy and expensive way?


----------



## Patrick Bateman

audioanarchist said:


> That sounds like a pain in the ass. Whats the easy and expensive way?












TaCT ambio processor does something similar. It's $2,490 and you'd have to convert it to work in the car.

I'll stick with doing the conversion on my PC 

There are good instructions on how to do it at ambiophonics.org

Again, not the exact same thing as what we're discussing here, but similar. Arguably, ambio is more advanced.

Tact digital amplifiers and preamplifiers


----------



## S3T

Well, a pair of line transformers and pair of linear tapper pots will do the trick of inverse/sum of channels, with variable summation, see the pic.


----------



## audioanarchist

Patrick Bateman said:


> TaCT ambio processor does something similar. It's $2,490 and you'd have to convert it to work in the car.



**** my dirty chili ring. That ain't happening.

So is everything you jam to in your car processed before hand?

For the stuff you do process do you just add Ambiophonic's and call it good? Or do you do more or something different all together?

I wanna learn. I haven't read much on that link you provided yet but, I will be after this post. I would like to experiment with song's that I am familiar with and see what happens. Care to walk me through how you do it?


----------



## audioanarchist

S3T said:


> Well, a pair of line transformers and pair of linear tapper pots will do the trick of inverse/sum of channels, with variable summation, see the pic.



sigh, I may be to stupid to pull this off 

linear tapper transformer pot tricks is making my head hurt


----------



## S3T

The transformers are 1:1 - just for isolation of ground.
When you isolate the ground, you can decide, by connection, which of wires will be a new "ground".

First scheme transfers the audio "as is".
The second scheme reverses + and - (inverts) the signal.

You invert L (get L'inv), and then mix the R and L'inv with resistor. The center tap of the resistor will decide the ratio between L'inv and R signals. The same goes for opposite channel.


----------



## Hernan

Re reading this thread, something cames up my mind.
A roof midbass bar.
Using small midbasses just above our head. Beeing small, they could be placed equidistant to our ears.
The bar should have two pairs, one for driver sit and other for the passenger.
Perhaps headrest drivers could get the same job done.
Crazy? just thinking outside the envelope.


----------



## traceywatts

"Perhaps headrest drivers could get the same job done."

i think Nissan had this in the 80's.


----------



## Patrick Bateman

audioanarchist said:


> **** my dirty chili ring. That ain't happening.
> 
> So is everything you jam to in your car processed before hand?
> 
> For the stuff you do process do you just add Ambiophonic's and call it good? Or do you do more or something different all together?
> 
> I wanna learn. I haven't read much on that link you provided yet but, I will be after this post. I would like to experiment with song's that I am familiar with and see what happens. Care to walk me through how you do it?


When I have an ambio setup, yes, that's correct.

Ideally, you'd buy yourself a MiniAmbio and install it in the car. I haven't done this for a couple reasons:

#1 - I'd heard it was no longer available (turns out this is incorrect)
#2 - I'm impatient. If I'm not mistaken, the MiniAmbio ships from Asia and takes a while to arrive
#3 - You can process your files for free, on the PC
#4 - Every device you add to your stereo can potentially degrade your system, particularly due to noise. I drive Hondas and I've noticed they're extremely prone to noise.

*Due to all this, whenever possible, I process the tracks on the PC instead of doing the processing in the car.*

The processing is pretty simple:


Go get an iPod or an iPhone or an iPad
Take your music, and put it in a folder or folders
Go get Winamp, the ambio plugin from ambiophonics.org, and the VST plugin for Winamp
Load up your entire music library, and have Winamp process every track

The process takes a few hours, just let it run overnight. Storing two copies of your music library is admittedly excessive, but 160gb iPods are only $100 more than a MiniAmbio. You might want to start out with a subset of your music, figure out what settings work best for you, and then convert the whole library.

If you have an iPad, iPhone, or iPod touch, there's an ambio app which will save you all this hassle:

Ambiophonics PC/MAC DIY


----------



## atxtrd

From 1990.

[http://www.polkaudio.com/forums/attachment.php?attachmentid=26687&d=1186346402url]

http://www.polksda.com/pdfs/1991SRSAd.pdf

Carver also did some interesting stuff with the Sonic Holgraphy. I own both the Polks and the Carver pre-amp/processor.
http://thecarversite.com/yetanotherforum/default.aspx?g=posts&t=1730

http://thecarversite.com/yetanotherforum/userfiles/US4218585.pdf


----------



## ultimatemj

:book: Warning: Reading through the vector summation part of this thread may be required to understand why anyone might want to try this. Otherwise you may think :jester:

Earlier in this thread Lycan postulated/titillated:


> A really clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction.
> 
> In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) L-R ambience signal (covered to death in those dreaded & controversial "rear fill" threads).
> 
> In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo L,R pair plus _and attenuated L-R signal_ to "widen" the midbass confusion cone.
> 
> Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course.
> 
> In short, there's no reason why a *single pair of speakers* can't be employed for both ambience generation, and midbass reinforcement
> 
> *yeah, let the speakers "sum" those interesting signals!*


Note: This was posted in a stream of consciousness style and the paragraph forming, bolding and underlining is my emphasis to try and digest it.

Can anyone verify this means simply that "summing by applying 2 different signals to a single driver"? 






​
If so, then to achieve the "_very interesting signal_" it appears you need 4 channels to driver 2 rear deck speakers:

 2 channels processed for midrange "L-R" ambience (320 to 3k, -6 to -9db, delayed 20+ms, 1ch 90deg out of phase, or delayed a few ms more) 
 2 channels processed for midbass "array reinforcement" (80 to 320, -3 to -6db, delayed to achieve time alignment).

Thoughts? Anyone tried this?

The thing I like about it is you can do (and tweak) either, or both fairly easily. The thing I don't like is you need 4 more dsp channels :uhoh2:

BTW, anyone know what is up with the underlined portion in Lycan's postulate? It seems like it was just looking forward to the resulting summation. :inquisitive:


----------



## Wesayso

This never left my mind but have not tried it yet.
Might be over my head to try it .

I do mis Werewolf / Lycan though! I had great fun reading his "lessons" on here...


----------



## Patrick Bateman

@ultimatemj -

You'd sent me an email asking about firing the midbass through a hole in the rear quarter panel.

If you can gather the following info, I can sim it for you:

1) what are the dimensions of the hole? (height, width, and depth of the hole in particular)
2) what driver are you going to use?


----------



## LovesMusic

Great read lycan.

Just thought Id throw this out there...

When summing vectors to get the most accurate phase angle, you should convert from polar to rectangular(complex) form. Add, then convert back to polar.

For smaller angles summing vectors in polar form will be sufficient, however as the angles increase so will the phase angle tolerances if (in polar form).


----------



## Neil_J

LovesMusic said:


> Great read lycan.


Lol.. Lycan hasn't posted on this board in over two years


----------



## LovesMusic

Neil_J said:


> Lol.. Lycan hasn't posted on this board in over two years


I don't get it, so what.....


----------



## quality_sound

You're thanking a member that doesn't post here. That was his point.


----------



## LovesMusic

I accredited the useful information to who purposed the ideas and theories... Dead, alive, prisoned, non poster....

Only to be quoted on three words of a post where I added yet another useful strategy... Ok got the point lol....


----------



## file audio

Ok maybe this is a closed thread but.... 
why everyone claim that Is wrong to put. a pair of image dynamics midbass drivers in the rear doors if I have the focal krx3 on a 3way system I have a helix dsp that has a channel extra for the midbass as 4way I just want to feel the kick in the 80 to 250hz.. but everyone seems to be against it, claiming that the scenario it's going to be ruined! any advice?


----------



## Orion525iT

file audio said:


> Ok maybe this is a closed thread but....
> why everyone claim that Is wrong to put. a pair of image dynamics midbass drivers in the rear doors if I have the focal krx3 on a 3way system I have a helix dsp that has a channel extra for the midbass as 4way I just want to feel the kick in the 80 to 250hz..


Who is "everyone"?


----------



## Patrick Bateman

file audio said:


> Ok maybe this is a closed thread but....
> why everyone claim that Is wrong to put. a pair of image dynamics midbass drivers in the rear doors if I have the focal krx3 on a 3way system I have a helix dsp that has a channel extra for the midbass as 4way I just want to feel the kick in the 80 to 250hz.. but everyone seems to be against it, claiming that the scenario it's going to be ruined! any advice?


Do you have a microphone?

I've always chased after that 'kick' in the bass sound. And generally I've found that the 'lack' of kick can be a masking thing.

For instance, I was recently fiddling with my subs, trying to get that 'kick', because the sound was kinda bass heavy. And when I pulled out the laptop and the mic, I found that my midbasses were much louder in the octave from 100hz to 200hz.

So, ironically, turning up the sub was only making the low bass *consistent* with the midbass, but it wasn't getting to the root of the problem. (The problem wasn't the sub, the problem was the midbass, and I couldn't tell without my microphone.)

This was one of the reasons I started to think long and hard about dipole midbass, as I found that I had too *much* midbass, not too little.

But, again, it's hard to tell without measurements.


----------



## file audio

Patrick Bateman said:


> Do you have a microphone?
> 
> I've always chased after that 'kick' in the bass sound. And generally I've found that the 'lack' of kick can be a masking thing.
> 
> For instance, I was recently fiddling with my subs, trying to get that 'kick', because the sound was kinda bass heavy. And when I pulled out the laptop and the mic, I found that my midbasses were much louder in the octave from 100hz to 200hz.
> 
> So, ironically, turning up the sub was only making the low bass *consistent* with the midbass, but it wasn't getting to the root of the problem. (The problem wasn't the sub, the problem was the midbass, and I couldn't tell without my microphone.)
> 
> This was one of the reasons I started to think long and hard about dipole midbass, as I found that I had too *much* midbass, not too little.
> 
> But, again, it's hard to tell without measurements.



Thanks for the response.. I have found the same in the 90hz to 240hz is the kick... 
i have heard sound systems that sound great even without subwoofers.. but it's just a matter of taste I tried one idmax this month and it's great but since yesterday I installed two idmax 12 bcuz I went to compete today.. in spl, I like how it sound both subs in low gain on amps I get. more kick but as My concern is SQ I'm trying hard everyday to get the kick which I love


----------



## Patrick Bateman

lycan said:


> I feel compelled to mention, or summarize, the following:
> 
> I think there IS some merit to a midbass array in a car. The main virtue, as Patrick has elaborated elsewhere, is to help the rough/irregular amplitude response in the midrange in most vehicles. The response can be rough/irregular (peaks & nulls) because of exciting constructive & desctructive "modes" in the midbass. More "random" placement around the cabin can help "smooth" out these modes. Also, getting strong sub-bass in a car is like falling off a log ... much trickier to get midbass strong enough to "keep up". Multiple drivers can help this integration, also. Finally, some first-choice locations ... like doors ... present less-than-optimal midbass enclosures.
> 
> Previously, I had argued against midbass arrays because of possible stage collapse. That is, with no attention paid to ITD's and confusion conses, midbass _localization_ will suffer with an array of randomly placed drivers. I offered this thread, because I think the technique described can help solve that problem
> 
> The single problem that remains, however, is the head-turning problem  Best we can offer so far on that is this : keep some drivers up front, to help reduce sensitivity to head turning.
> 
> What this boils down to, for me, is this : one pair of drivers up front, kicks or doors or floors, maybe another pair on that rear shelf (that's just sitting there waiting to get used). Apply *L-R* mixing to any driver pair that is NOT as wide as you would like. Having at least one pair upfront _should_ desensitize the head-turning problem.
> 
> On a final note, the 3 pieces of extra signal processing required are :
> 
> 1. *L-R* difference. Already covered in other threads ... trivial with *balanced* signals.
> 2. *Attenuation*
> 3. Mixing, or *summing*, back into main stereo pairs
> 
> In addition to the Audio Control equipment already mentioned, I think a halfway-clever guy could readily identify a couple pieces of ZAPCO gear that could already do the trick  Zapco is already *balanced*, they offer nifty little *attenuators* like the SLB-U, and probably an amp channel or two that allow *summation*
> 
> that's all i got
> 
> EDIT : oops i already lied, that's not _quite_ all i got
> 
> A _really_ clever guy could form a VERY interesting signal to send to a pair of rear-shelf drivers, assuming the rear-shelf drivers are capable of midbass & midrange reproduction. In the midrange, the signal would consist of an attenuated and much-delayed (about 20msec or so) *L-R* _ambience_ signal (covered to death in those dreaded & controversial "rear fill" threads). In the midbass, the signal sent to the rear shelf drivers would be an undelayed, or only slightly delayed (to account for distance-to-listener) stereo *L,R* pair _plus_ and attenuated *L-R* signal to "widen" the midbass confusion cone. Variable attenuation (and delays) would have to be available for midbass & midrange portions of the signal, of course. In short, there's no reason why a single pair of speakers can't be employed for _both_ ambience generation, and midbass reinforcement  yeah, let the speakers "sum" those interesting signals!
> 
> howz that for clever use of that rear shelf, from which we were all told we need to just remove the speakers ???



There's a question that comes up on the forums once every few weeks. The question is:

"What is a good midbass driver?"

Whenever the question is asked, the responses tend to favor a lot of relatively big beefy midbasses. I believe that this perpetuates the idea that if you want good midbass, you really need to go out and get yourself a big ol' eight. Maybe if you're really crazy you'll try to figure out a way to shoehorn a ten in your door. Heck, my home speakers use a FIFTEEN for a midbass.

So the idea seems to be, *"bigger is better."*

Unfortunately, all of the psychoacoustic things we're talking about in this thread aren't practical with big midbasses. If I was willing to tear my car up, I could probably get an eight in my door. But there's no way in hell I'm getting THREE eights in my car without doing some serious surgery on it.

And it's 2013 - I really think that loudspeakers have evolved to a point where we don't need to hack up our cars with a Sawzall to get good sound.

In this post, I'd like to show why I believe that is the case.

In a nutshell, *I'd like to demonstrate that an array of small drivers can do everything that a big beefy eight can do.*









Here's our eight. A Faital 8FE200. Your basic high output, high efficiency prosound woofer.









Here's our small driver. The Dayton ND91, a shameless clone of the old Aurasound 3" woofers, with some improvements requested by Don Keele for his CBT array. (Bigger voice coil, higher power handling, lower inductance.)









Here's a sim comparing the Faital in an open baffle to the Dayton in a very small enclosure. The reason that I opted for this is that the Faital would typically go in a door. If I was going to do a midbass array, like this thread discusses, I would use a very small sealed box and I would distribute them throughout the car. (Under the dash, in the door, along the rear shelf, etc. Basically put 'em wherever you can find a little space for them.)

The Dayton enclosure is just 3.2 liters; to put that in perspective, that's so small you can almost hold it in your hand. 16 liters measures 8" x 8" x 8"; this is five times smaller.

Admittedly, *you still need cone area to make bass, so I had to use a lot of small drivers.* In this case, I used EIGHT of the ND91s. But again, enclosure size is tiny; it's barely big enough to contain the driver.

Bottom line - an array of eight Dayton ND91s is louder at every point in the spectrum from 100hz to 1khz. If you want a high output midbass, a pile of small drivers can exceed one big eight.









Here's the excursion for the two designs. It's basically the same. According to the Dumax measurements of the Dayton, it's xmax exceeds the Faital by about a millimeter. (The spec sheet says 4mm, but the Dumax measurements show that it can do 6mm.)









Here's the impedance of the two. I intentionally used a lower load for the Dayton array, as most car amps are stable to two ohms.

The best reason to use the Faital is definitely cost. Two of the Faitals will cost you $96. Sixteen of the Daytons will cost you $423.68!

But how much value do you shave off your car when you fiberglass the door?
One thing I notice with a lot of these elaborate car stereo installs is that they often make the car virtually impossible to sell. 99% of the people out there want windows that roll down and doors that don't look like a science experiment.

In addition to this, there's significant aural advantages to midbass arrays, particularly if you wrap your brain around the OPSODIS paper.

One thing that we can't simulate easily is thermal compression. The Faital has a bigger voice coil than the Dayton, but EIGHT of the Daytons have a voice coil surface area that's comparable to a 3" woofer. *That's nearly four times the voice coil area as the Faital.* This is also why the Faital is so cheap; in the prosound world a 1.5" voice coil is quite puny. If you look at Dynaudio and Morel spec sheets, you'll see a lot of 2" and 3" voice coils. But EIGHT voice coils spread across the car facilitates some crazy power handling. I wouldn't think twice about putting 250 watts or even 500 watts into eight 1" voice coils.










One last advantage to midbass arrays is push-pull mounting. Push-pull reduces second harmonic distortion, basically makes the drivers sound cleaner. In the pic above, I've illustrated how one can mount four of the Dayton ND91s in a push-pull arrangement. The internal volume of the enclousre is just 1.6 liters. The total footprint measures 4.5" x 13". That's small enough to hide under the dash or possibly even under the seats. (200hz is over five feet long; these sound waves are so long, diffraction from the seat or the dash just isn't an issue.










This configuration is shamelessly stolen from Tymphany's LAT


----------



## Regus

I'm thinking maybe the time has come for the DIYMA community to see how well a midbass array works in practice. I am envisaging an experiment in which an array is installed and measurements made to gauge its real world performance. This would encompass all aspects of the build, such as aesthetics, power handling/SPL, RTA measurements, effects on imaging/sound staging/localisation, etc. as well as the benefit of DSP in aligning drivers and maybe also introducing a L-R signal to drivers with sub-optimal placement, if required.

I'm of the opinion it would probably work best as a system design thread in which the design goals are stated and agreed upon prior to installation and testing followed by a build log in which the array is tested and the results are published for closer scrutiny. I have some ideas as to what these tests should be but I'm sure there are others far more qualified to judge than I am. That said, I'd be happy to start a thread just to get the ball rolling once I've firmed up my ideas, unless someone else beats me to it. Of course, it still needs someone to invest the time, effort and money to do this and those Dayton drivers aren't cheap - if I can find a cheaper alternative I might be willing to (literally) put my money where my mouth is and have a crack at it myself...


----------



## rich20730

Regus said:


> I'm thinking maybe the time has come for the DIYMA community to see how well a midbass array works in practice. I am envisaging an experiment in which an array is installed and measurements made to gauge its real world performance. This would encompass all aspects of the build, such as aesthetics, power handling/SPL, RTA measurements, effects on imaging/sound staging/localisation, etc. as well as the benefit of DSP in aligning drivers and maybe also introducing a L-R signal to drivers with sub-optimal placement, if required.
> 
> I'm of the opinion it would probably work best as a system design thread in which the design goals are stated and agreed upon prior to installation and testing followed by a build log in which the array is tested and the results are published for closer scrutiny. I have some ideas as to what these tests should be but I'm sure there are others far more qualified to judge than I am. That said, I'd be happy to start a thread just to get the ball rolling once I've firmed up my ideas, unless someone else beats me to it. Of course, it still needs someone to invest the time, effort and money to do this and those Dayton drivers aren't cheap - if I can find a cheaper alternative I might be willing to (literally) put my money where my mouth is and have a crack at it myself...


I think that's an excellent idea. This is a concept that I have been interested in for a while now. In home audio, the use of multiple subs is a well-established method to deal with response variations and ringing caused by room modes. It seems logical that one could obtain similar results using midbass arrays to target the higher frequency modes created by the smaller dimensions of a vehicle cabin.

I have already begun experimenting with some of these concepts in my own vehicle, but so far the process has been fairly informal and unscientific. I already have most of the equipment, processing capability, and measurement tools which would be necessary and I am more than happy to take input, implement suggestions, and document my results.

Currently, I have 13 channels of amplification with fully active DSP consisting of:
Zapco DC 1000.4
Zapco DC 1100.1
(2)Zapco DC 350.2
(2)Zapco DC 200.2

For spatial processing, I have an Audiocontrol ESP-3 and also the balanced inputs on the DC amps can be used to create a L-R signal.

For midbass, I have dual Exodus Anarchies mounted on opposite sides of each door and Image dynamics CTX6.5 components in the rear shelf. Additionally, I could probably install a pair of midbasses in the rear doors.

For measurements, I have a laptop, REW, preamp, Dayton EMM-6 and Dayton UMM-6 measurement mics, and iPad with full suite of Audiotools apps.

Regus, feel free to start the thread and set out some designs/ tests/parameters and I will do my best to contribute.


----------



## file audio

I noticed that in the 3way system my woofers alone sounded so deaf and only sound good after hear all the system is sounding complete


----------



## Regus

I've just started a new thread entitled "Implementing a midbass array" in the System Design Sub-forum if anyone else wants to join the party - rich20730, I'll start populating it with my ideas for tests and hopefully so will others.


----------



## Patrick Bateman

rich20730 said:


> I think that's an excellent idea. This is a concept that I have been interested in for a while now. In home audio, the use of multiple subs is a well-established method to deal with response variations and ringing caused by room modes. It seems logical that one could obtain similar results using midbass arrays to target the higher frequency modes created by the smaller dimensions of a vehicle cabin.
> 
> I have already begun experimenting with some of these concepts in my own vehicle, but so far the process has been fairly informal and unscientific. I already have most of the equipment, processing capability, and measurement tools which would be necessary and I am more than happy to take input, implement suggestions, and document my results.
> 
> Currently, I have 13 channels of amplification with fully active DSP consisting of:
> Zapco DC 1000.4
> Zapco DC 1100.1
> (2)Zapco DC 350.2
> (2)Zapco DC 200.2
> 
> For spatial processing, I have an Audiocontrol ESP-3 and also the balanced inputs on the DC amps can be used to create a L-R signal.
> 
> For midbass, I have dual Exodus Anarchies mounted on opposite sides of each door and Image dynamics CTX6.5 components in the rear shelf. Additionally, I could probably install a pair of midbasses in the rear doors.
> 
> For measurements, I have a laptop, REW, preamp, Dayton EMM-6 and Dayton UMM-6 measurement mics, and iPad with full suite of Audiotools apps.
> 
> Regus, feel free to start the thread and set out some designs/ tests/parameters and I will do my best to contribute.


I think I'm going to give it a try.

One thing to note is that these arrays aren't the same as what most people think of when we say "array." When most people talk about arrays, we're usually talking about *line arrays*. And in line arrays, the drivers are tightly spaced.

The arrays we're talking about in this thread can actually be spread about quite a bit, because the frequencies are so long.

For instance, two sound sources that are within one quarter wavelength will basically act like a single unit. And in this implementation, *we're trying to put the drivers close enough that they'll still act like a single unit, but far enough apart that we can spread out the resonances in the cabin.*

For instance, with an upper limit of 500hz, two drivers that are spaced 17cm apart will basically radiate like a single unit.

We could probably even push that two one-third wavelength, which allows for a distance of 23cm or 9".

Right now what I'm leaning towards is two midbass enclosures. I am going to make them super small - just 18cm x 18cm x 8cm, or about the size of a hardback book.

The Opsodis paper seems to indicate that the midbass should emanate from the left and the right. And I heard a car that uses this layout recently and it images really well. (Stock VW Jetta, I posted some info on this in another thread here.)

So what I'm thinking of is two midbass enclosures, one in front of the seat, and one behind the seat. The car seat is about 16" deep, but I think that I should be able to squeeze the midbass enclosure *under* the seat. I may even sandwich it between the seat and the door, if it will fit.

Another cool thing about using an enclosure that's just 7" x 7" x 3" is that I can experiment with various locations. It's small enough that I could even put two of them on the floor and the car would still be drive-able. (Would look ugly though!)

One curveball that I'm going to do, is that I am going to use a transmission line instead of a sealed enclosure. I'm basically doing this because transmission lines and back loaded horns and tapped horns tend to sound 'spacious.' It's interesting, because they don't measure that great. They tend to have dips in the frequency response and their impulse response doesn't look great. *But they sound good.*

My 'hunch' is that these enclosure types sound good because they distribute the sound in the room the same way that an array of monopoles would. Because you basically have two radiators and one of them is delayed about three milliseconds.

I'll post the plans over on my build thread on diyaudio, it's called "28 Days Later."


----------



## Orion525iT

I have been thinking about this on and off for a few months, especially in regards to the Opsodis paper.

For what it is worth, I set up a quasi Opsodis with RS225-4 midbass in the kicks and 2-speaker fullrange arrays slightly inboard of the RS225-4s and then tweeters center mounted. It was pretty unsophisticated arrangement, I high passed the arrays at ~160hz, used passive high pass on the tweets at 2500hz, and ran the midbass off the sub channel. With that said, it was fairly promising and I want to pursue it more once I get proper processing.

That brings me back to the midbass arrays. It absolutely makes sense to spread the nodes out, just as one would with sub-bass in a larger room. But there is a problem here, especially when taking the Opsodis paper into consideration. Midbass, unlike subbass can be located, otherwise the Opsodis paper makes no sense. The question is, where can you place the midbass drivers without narrowing the stage or having a less than optimal stage?

It would seem that you would, in the least, still want to have the majority of the output coming directly from left and right. If you place the midbass under the seats, you will get sound coming from your right, but the left channel originates from directly underneath you. This, to me, almost forces some sort of door mount, either the forward part of the rear door, or the rearward part of the front door. They will be the widest locations. It could be that you only need a small amount of output from the wide left/right locations to get some Opsodis like results, but ultimately, you still should have something there.

Just a thought .


----------



## SPLEclipse

I'm currently using multiple spread midbasses in my car: A single 8" on each corner of the dash facing up and (2) 6.5"s in the doors semi-lateral (opsodicly?) to my ears in the doors. I'll be taking some measurements in REW tomorrow of each side with each component individually and both together. If anyone wants me to measure anything specific post it here.


----------



## Orion525iT

SPLEclipse said:


> I'm currently using multiple spread midbasses in my car: A single 8" on each corner of the dash facing up and (2) 6.5"s in the doors semi-lateral (opsodicly?) to my ears in the doors. I'll be taking some measurements in REW tomorrow of each side with each component individually and both together. If anyone wants me to measure anything specific post it here.


I can't think of anything specifically. But maybe something more subjective regarding perceived stage width. Dash 8" only vs door only vs both together at similar output levels. Maybe try one 6.5" and the eights together or with various attenuation levels to each, ect.

Again, the focus is stage width, perception only exercise, not a subjective sq impression. 

Thanks!


----------



## Regus

Patrick Bateman said:


> The Dayton enclosure is just 3.2 liters; to put that in perspective, that's so small you can almost hold it in your hand. *16 liters measures 8" x 8" x 8"; this is five times smaller.*


Forgive me for pointing this out, but unless I'm very much mistaken, an 8" cube (20cm x 20cm x 20cm) a volume of 8L, and thus the Dayton enclosure is two and a half times smaller than the Faital enclosure. That's still a significant reduction, and the footprint remains very small.


----------



## Regus

Orion525iT said:


> I have been thinking about this on and off for a few months, especially in regards to the Opsodis paper.
> 
> For what it is worth, I set up a quasi Opsodis with RS225-4 midbass in the kicks and 2-speaker fullrange arrays slightly inboard of the RS225-4s and then tweeters center mounted. It was pretty unsophisticated arrangement, I high passed the arrays at ~160hz, used passive high pass on the tweets at 2500hz, and ran the midbass off the sub channel. With that said, it was fairly promising and I want to pursue it more once I get proper processing.
> 
> That brings me back to the midbass arrays. It absolutely makes sense to spread the nodes out, just as one would with sub-bass in a larger room. But there is a problem here, especially when taking the Opsodis paper into consideration. Midbass, unlike subbass can be located, otherwise the Opsodis paper makes no sense. The question is, where can you place the midbass drivers without narrowing the stage or having a less than optimal stage?
> 
> It would seem that you would, in the least, still want to have the majority of the output coming directly from left and right. If you place the midbass under the seats, you will get sound coming from your right, but the left channel originates from directly underneath you. This, to me, almost forces some sort of door mount, either the forward part of the rear door, or the rearward part of the front door. They will be the widest locations. It could be that you only need a small amount of output from the wide left/right locations to get some Opsodis like results, but ultimately, you still should have something there.
> 
> Just a thought .


I too have been giving this some thought and I may have hit upon a partial solution. In the case of a four- or five-door car, why not use the B-pillars? I can envisage two-three (or more?) small drivers mounted vertically in a miniature linear array on either side of the vehicle, which should give us the correct angle for an OPSODIS-based install, give or take a few degrees. It might not work for everyone but it is worth considering as an alternative to door mounting.

I'm also looking at how you would then employ the remaining speakers in the array - I'm currently formulating some ideas for how to combine the "primary" midbass array (in the OPSODIS configuration) with the "secondary" drivers in the array and whether they would need to be fed the L-R signal if they are inboard of the primary drivers.

In the meantime, I found some interesting stuff on the University of Southampton website on OPSODIS, including three-channel alignments - for those of you who would like to know more, here's a taster.

http://www.southampton.ac.uk/images/fee/OPSODIS.ppt


----------



## Orion525iT

Regus said:


> I too have been giving this some thought and I may have hit upon a partial solution. In the case of a four- or five-door car, why not use the B-pillars? I can envisage two-three (or more?) small drivers mounted vertically in a miniature linear array on either side of the vehicle, which should give us the correct angle for an OPSODIS-based install, give or take a few degrees. It might not work for everyone but it is worth considering as an alternative to door mounting.


This would not work in my car. The drivers would need to be tiny, and then you still have to deal with proper enclosure volume. There are other issues too. I do not think you would not want a driver at ear height. Any harmonics would be audible. Get them off angle, and those harmonics would be attenuated somewhat. I am not sure you would find small drivers that could act as proper midbass and also have a sophisticated motor structure. Maybe the drivers Patrick mentioned would fit the bill, but that is a rare breed.

In any case, you would need more than 7 ND91 to equal the cone area of 1 RS225 8", and you would still fall behind in excursion. Cost be damned, I suppose it could be done. I think Patrick has a point regarding cost of massive construction in doors, but you may end up going there anyway.



Regus said:


> I'm also looking at how you would then employ the remaining speakers in the array - I'm currently formulating some ideas for how to combine the "primary" midbass array (in the OPSODIS configuration) with the "secondary" drivers in the array and whether they would need to be fed the L-R signal if they are inboard of the primary drivers.


I think there is a threshold here that needs investigation. It may be that you can get away with using larger drivers in typical locations, such as front of door and kicks that will provide the much desired visceral impact, and use smaller drivers near the sides to get the Opsodis experience and spread the nodes. Or maybe, a single large midbass and then multiples of smaller drivers.

There are many challenges here. Multiple drivers to spread the nodes, enough cone area needed for impactful midbass, and positioning that can achieve the Opsodis epxerience. It may be that one will be sacrificed for the sake of the other.

Or, you could drive a 2-door 4-seat car, that typically has larger storage bins to either side of the rear seats, that sit slightly behind, below, and wide of the drivers ears. Then you can implement the array.


----------



## Regus

Orion525iT said:


> This would not work in my car. The drivers would need to be tiny, and then you still have to deal with proper enclosure volume. There are other issues too. I do not think you would not want a driver at ear height. Any harmonics would be audible. Get them off angle, and those harmonics would be attenuated somewhat. I am not sure you would find small drivers that could act as proper midbass and also have a sophisticated motor structure. Maybe the drivers Patrick mentioned would fit the bill, but that is a rare breed.
> 
> In any case, you would need more than 7 ND91 to equal the cone area of 1 RS225 8", and you would still fall behind in excursion. Cost be damned, I suppose it could be done. I think Patrick has a point regarding cost of massive construction in doors, but you may end up going there anyway.
> 
> 
> 
> I think there is a threshold here that needs investigation. It may be that you can get away with using larger drivers in typical locations, such as front of door and kicks that will provide the much desired visceral impact, and use smaller drivers near the sides to get the Opsodis experience and spread the nodes. Or maybe, a single large midbass and then multiples of smaller drivers.
> 
> There are many challenges here. Multiple drivers to spread the nodes, enough cone area needed for impactful midbass, and positioning that can achieve the Opsodis epxerience. It may be that one will be sacrificed for the sake of the other.
> 
> Or, you could drive a 2-door 4-seat car, that typically has larger storage bins to either side of the rear seats, that sit slightly behind, below, and wide of the drivers ears. Then you can implement the array.


I have though about this some more and have evolved the idea to the following hybrid of OPSODIS and array. You put one pair of midbass speakers to either side of you, for maximum separation, which get the lion's share of the power, then add some smaller drivers to give the natural bass effect that Patrick was originally after, utilising Lycan's cone of confusion to best effect. One thing I am wondering is whether it would be beneficial to use a driver that would play lower than 80Hz, say down to 63Hz (or even 50Hz), as the OPSODIS driver, thereby allowing for a better blend between midbass and subwoofer, whilst restricting the array speakers to frequencies above 80Hz.

In terms of OPSODIS driver selection, it depends on whether you stick with the door or go for an enclosure. In my case I happen to know that there's a large hole in the door of my new car in exactly the right place to mount a large driver, and I could open up the door card to allow sound to pass through the door pocket, but I'm not so sure I want to add another driver to the door - I'd prefer to find something that worked in a sealed or ported enclosure, although the problem here is finding a suitable speaker that doesn't require an unfeasibly large enclosure.

As regards the rest of the speakers, I'm still trying to figure out how to actually install these in a way which is workable - whilst it would be trivial to wire a group of drivers to each channel of an amplifier, I'm not too keen on having a bunch of enclosures just thrown into the car without any thought to aesthetics!


----------



## Patrick Bateman

Orion525iT said:


> This would not work in my car. The drivers would need to be tiny, and then you still have to deal with proper enclosure volume. There are other issues too. I do not think you would not want a driver at ear height. Any harmonics would be audible. Get them off angle, and those harmonics would be attenuated somewhat. I am not sure you would find small drivers that could act as proper midbass and also have a sophisticated motor structure.


A few observations:

1) It's almost impossible to perceive height at low frequencies. This is due to the way that we hear; at low frequency our perception of location is dictated by phase, and due to that, we can hear left and right but not up and down. At higher frequencies we can *definitely* perceive height, but that's because the frequencies are smaller than our ears. (About 4khz and up.)

Long story short - as long as the driver is playing 500hz or lower, the harmonics won't give away it's height. (4khz is the *seventh* harmonic of 500hz, well out of band.)

Moving the driver to a location where you can't see it can help a lot. Our eyes will often override our ears when it comes to location. It's the reason loudspeakers have a better soundstage with the lights off.

2) There are some who believe that harmonic distortion is inoffensive. I am starting to agree with them. For instance, a lot of music has harmonic distortion added to the mix intentionally. Does it sound bad? No. Does it sound different? Yes. Distortion makes things sound louder than they are.

3) At low frequencies the angle of a driver doesn't matter. For instance, 100hz is over three METERS long. The wave is so long, it swamps the driver, and orientation doesn't matter.


----------



## rich20730

Regus said:


> I have though about this some more and have evolved the idea to the following hybrid of OPSODIS and array. You put one pair of midbass speakers to either side of you, for maximum separation, which get the lion's share of the power, then add some smaller drivers to give the natural bass effect that Patrick was originally after, utilising Lycan's cone of confusion to best effect. One thing I am wondering is whether it would be beneficial to use a driver that would play lower than 80Hz, say down to 63Hz (or even 50Hz), as the OPSODIS driver, thereby allowing for a better blend between midbass and subwoofer, whilst restricting the array speakers to frequencies above 80Hz.
> 
> In terms of OPSODIS driver selection, it depends on whether you stick with the door or go for an enclosure. In my case I happen to know that there's a large hole in the door of my new car in exactly the right place to mount a large driver, and I could open up the door card to allow sound to pass through the door pocket, but I'm not so sure I want to add another driver to the door - I'd prefer to find something that worked in a sealed or ported enclosure, although the problem here is finding a suitable speaker that doesn't require an unfeasibly large enclosure.
> 
> As regards the rest of the speakers, I'm still trying to figure out how to actually install these in a way which is workable - whilst it would be trivial to wire a group of drivers to each channel of an amplifier, I'm not too keen on having a bunch of enclosures just thrown into the car without any thought to aesthetics!


I'm glad you're still working on this! I haven't had much time lately, but I'm SLOWLY making progress on getting this implemented. Like you, I also had a difficult time figuring out where to install additional driver's in my car without making it look like complete crap. After weighing my options, I decided to install a pair of coaxials in the rear doors. 

I have one door pretty much done. After getting both speakers in, I'll need to install the additional processor (Alpine PXE-660) and amp before I start tuning. I have enough amp channels and processing to run everything active so tuning should be an adventure.

Here is a diagram of the speaker layout:









Here are a couple pictures of the actual install:

Enclosures for the rear door install:









































Dual anarchies in the front doors:


----------



## thehatedguy

Maybe if we howl at the moon, the wolf will reappear...


----------



## Patrick Bateman

@rich - what kind of car is that?


----------



## danno14

thehatedguy said:


> Maybe if we howl at the moon, the wolf will reappear...


AAawwoooooooooo!!!!!

come back Amarog!


----------



## cajunner

lycan's given us a theory and if there was a way to capitalize on that theory successfully in the commercial sense, somebody producing a black box we can put on the rear decks of sedans that does it all, somebody would have already done it, is my guess.

getting a cheap all-pass for some cheap time delay, adding a little class D amp and doing the circuitry without having to resort to dedicating a miniDSP to the task, it should have been easy, all the steps to making such a product was there, and there are commercial products like Circle Surround and SRS and even Cinema Surround, that maybe already patented the circuits, so any chance at developing this technology is a moot endeavor for production purposes.

but in the spirit of the DIY community, with small-run PCB's for everything under the sun, with arduino and Raspberry Pi and all the other little trinkets available for cheap, a high-performance solution seems doable, within the 100 dollar margin for a kit format...

if it's just a few op-amps and 50 watt class D modules, and maybe a couple of dedicated full range drivers in a slim form enclosure holding us back...


----------



## rich20730

Patrick Bateman said:


> @rich - what kind of car is that?


It's a 2006 Acura TL


----------



## cajunner

cajunner said:


> lycan's given us a theory and if there was a way to capitalize on that theory successfully in the commercial sense, somebody producing a black box we can put on the rear decks of sedans that does it all, somebody would have already done it, is my guess.
> 
> getting a cheap all-pass for some cheap time delay, adding a little class D amp and doing the circuitry without having to resort to dedicating a miniDSP to the task, it should have been easy, all the steps to making such a product was there, and there are commercial products like Circle Surround and SRS and even Cinema Surround, that maybe already patented the circuits, so any chance at developing this technology is a moot endeavor for production purposes.
> 
> but in the spirit of the DIY community, with small-run PCB's for everything under the sun, with arduino and Raspberry Pi and all the other little trinkets available for cheap, a high-performance solution seems doable, within the 100 dollar margin for a kit format...
> 
> if it's just a few op-amps and 50 watt class D modules, and maybe a couple of dedicated full range drivers in a slim form enclosure holding us back...



and this post was a bust, what a brain fart!


mixing the mid-bass array cone of confusion with the decorrelated surround sound for image enhancement/spatial integration of multi-channel upmixers...


I've done that already years back, I wonder why I've got that problem.


maybe it's because I'm waiting for someone to execute either of lycan's acoustic theory suppositions to satisfactory results.


anyways, I think Opsodis approach is more for imaging and less for the bass to mid bass re-distribution processes, or smoothing of resonant peak/nodes but getting those two ideas together is interesting to me.

my understanding of the cone of confusion is that you were sort of limited by installation parameters, you wanted all mid bass drivers as close to equidistant from the listening position as possible, making this idea more of a single-seat approach.


2 seater work would focus the energy into the front passenger section, more or less and be less 'intense' with the exact wavefront accumulation/impulse response.

and that's probably more or less academic because mid bass frequencies have their sinusoidal at intervals measured in feet, so it's not going to have to be quite that precise.

it may be more simple than achieving maximum impact, as much as about controlling the dominant resonant signature unique to each vehicle, with conforming additional node creation interfering with drop-outs, or infinite nulls.

You can DSP equalize a bunch of peaks on averaged response patterns, but you can't push the slider up on a notch, is my presumption and why mid bass arrays are logically delicious...


----------



## canuck

I have installed a midbass array. For pics check my install 
2 center channel. ..front and rear
2 left side front and rear doors
2 right side front and rear doors
6 drivers total

I guess first order would be to DB all the drivers. Then move on to frequency. Any thoughts or advice is welcome


----------



## T3mpest

Orion525iT said:


> This would not work in my car. The drivers would need to be tiny, and then you still have to deal with proper enclosure volume. There are other issues too. I do not think you would not want a driver at ear height. Any harmonics would be audible. Get them off angle, and those harmonics would be attenuated somewhat. I am not sure you would find small drivers that could act as proper midbass and also have a sophisticated motor structure. Maybe the drivers Patrick mentioned would fit the bill, but that is a rare breed.
> 
> In any case, you would need more than 7 ND91 to equal the cone area of 1 RS225 8", and you would still fall behind in excursion. Cost be damned, I suppose it could be done. I think Patrick has a point regarding cost of massive construction in doors, but you may end up going there anyway.
> 
> 
> 
> I think there is a threshold here that needs investigation. * It may be that you can get away with using larger drivers in typical locations, such as front of door and kicks that will provide the much desired visceral impact,* and use smaller drivers near the sides to get the Opsodis experience and spread the nodes.* Or maybe, a single large midbass and then multiples of smaller drivers.*
> 
> There are many challenges here. Multiple drivers to spread the nodes, enough cone area needed for impactful midbass, and positioning that can achieve the Opsodis epxerience. It may be that one will be sacrificed for the sake of the other.
> 
> Or, you could drive a 2-door 4-seat car, that typically has larger storage bins to either side of the rear seats, that sit slightly behind, below, and wide of the drivers ears. Then you can implement the array.


This tends to work well in home audio when using subwoofers. Often you can use different sized subs with different responses to spread the choas around and simply equalize their overall response once your done. In the midbass the car really does change things a lot, putting a beefy set where you can normally fit them and then some smaller ones in your "cone of confusion" zones, seems like it would still help significantly. 

This seems to be the most reasonable avenue and probably the easiest to implement. Most of us WANT to use a big midbass where we can fit it, yet doing multiples like that simply isnt' as feasible. It'd take some testing. You need enough output from the small drivers to fill things in, while still trying to keep your overall staging wherever your "optimal" large midbasses are. Finding that happy medium with the level matching would be the trick. I'm sure they could be quieter and still help, just not sure how many DB down.


----------



## High Resolution Audio

Subbed for reference page 4


----------



## Jscoyne2

Lets see if im understanding this.

Normal 3 way set up. sub in trunk (0-80) Midbass(80-500) in doors. Midrange(500-5khz) in kicks. Tweets(5khz +) on dash. You will probably have a 50-80hz slump between sub and midbass and you will probably have a big dip somewhere between 100-200hz. 

Using back deck mounted midbass drivers as well. , You bandpass them(80-500) or perhaps if they can handle it. 50-500hz. Give them the same amplitude and frequency response (house curve). Time align accordingly and you should fill in the big dips in the sub-front midbass frequency response. 

This method could negatively effect stage width so Adding in some L-r and R-l to the signal would fix that. Thats the part that no one could really come up with a viable method for from what i read. 

am i understanding this all correctly?


----------



## Jscoyne2

^

Sent from my SGH-M919 using Tapatalk


----------



## sqshoestring

Personally I always find the dip around 80hz. I like my subs more at 50 where they are very non directional so I would set up midbass array for 50-100 or so. Any mid can get to 100 usually maybe little higher it depends you could go to even 4" mids though I rather stay larger. And then I really don't have to worry much about stereo issues at ~100 and less. I bought a nice PEQ for midbass because the way my subs act I would certainly need one, but never got to the install and now dumping this ancient car, alas...don't know what is next yet.


----------



## ultimatemj

IMO the method described is about achieving additional width by using -3 to -6db attenuated rear deck midbass, and delayed to align on the cone of confusion. 

The method is not as much about addressing nulls, although it may, but focused on using vector & amplitude to move the perception of stereo midbass location. If you don't attenuate the signal it will likely "pull the stage behind you". 

The part no one has shown they have achieved is the combination processing to put the width (Blumlein like steering) and fill (L-R ambiance) signals into the same amplified channels for a pair of rear deck midbass. 

My feeling is most folks don't have (or want to pay for) the 6 additional channels, on top of the 7 "primary" channels (1Sub, 2Midbass, 2Midrange, 2Tweeter), to (maybe) get 'magic' from rear deck speakers: 
2 channels processed (but not amplified) for rear fill (L-R input, band passed 320 to 3k, attenuated -6 to -9db, & time delayed 20+ms, 1 of the 2 channels 90deg out of phase, or delayed a few ms more)
2 channels processed (but not amplified) for width (stereo input, band passed 80 to 320, attenuated -3 to -6db, & time delayed for alignment)
2 channels to sum the 'rear fill' and 'width' processed channels and then feed to 2 amplifier channels.

Coming up with 2 amp channels isn't that difficult, especially when you don't need the same power/amplitude output, but coming up with the additional processing is not easy/cheap. 

The best options are adding an old Zapco DSP6 or 6ch MiniDSP specifically for this purpose. But don't forget, there is likely to be lots of tuning iteration to get the attenuation and time alignment

If anyone has an old Zapco DSP6 they are looking to sell, PM me and I'll try to do it 



Jscoyne2 said:


> Lets see if im understanding this.
> 
> Normal 3 way set up. sub in trunk (0-80) Midbass(80-500) in doors. Midrange(500-5khz) in kicks. Tweets(5khz +) on dash. You will probably have a 50-80hz slump between sub and midbass and you will probably have a big dip somewhere between 100-200hz.
> 
> Using back deck mounted midbass drivers as well. , You bandpass them(80-500) or perhaps if they can handle it. 50-500hz. Give them the same amplitude and frequency response (house curve). Time align accordingly and you should fill in the big dips in the sub-front midbass frequency response.
> 
> This method could negatively effect stage width so Adding in some L-r and R-l to the signal would fix that. Thats the part that no one could really come up with a viable method for from what i read.
> 
> am i understanding this all correctly?


----------



## pwnt by pat

I know this is a very old thread but I thought of an interesting solution that should be easy to test.

The problem with stereo reproduction is the environment. Everyone will agree to that. But interior reflections aren't the only problem. 

Proper stereo reproduction requires a central listening position located between two speakers positioned at 30° angles from The center of the head. Any non-central listening position will not produce stereo playback. Period. The audio encoding requires it. Not that you can't a left right stage, but it's not stereo.

In a vehicle, the speaker opposite side of the car either in the kicks or doors is pretty close to the correct angle. But the close side speaker, even in it's widest position is going to be at 15° +-. Using the idea of vector addition and abusing the cone of confusion concept, a second driver can be located at an angle equal to +-30°+(30-x) where x is the angle of the near side front driver to centerline of the head. 

The negative option just so happens to line up with where most rear door speakers are mounted. This would be about -45° from listener rear, or 135° from listener forward. This sounds goody at first but remember we can't detect a difference between a speaker on a forward or a reward angle on the COC .

Through vector summing, this would effectively pull the phantom location of the front driver back to the proper 30 degree angle. Then apply significant amplitude reduction and proper time alignment. 

The downsides I see are head turning (but let's be honest, you're not fixing that if you're using additional speakers) and one seat affected. Perhaps if the amplitude is reduced enough then you could use the principle on both sides of the car without significantly affecting the far side listener. I don't think you would need much volume at all to create the phantom speaker.

Thoughts?


----------



## Patrick Bateman

You can definitely get a rock solid center image without being 30 degrees off axis. The tricky part is that you have to use one type of processing about 2000Hz, another type below 500Hz, and a little bit of both in the two octaves between 500Hz and 2000Hz.

Right now I'm mostly tinkering with arrays where the drivers aren't near each other but the pathlengths are equal. For instance, I have some Tymphany TC7s up on the dash and some Aurasound Whispers under the dash.

I'd put them all in one spot if it was practical to mount them vertically but it's hard to find space. I have space to mount them horizontally, but then the array would be asymmetrical and I'm not sure if that's a hot idea. Basically you'd have symmetrical reflections from one side of the car and asymmetrical reflections from the other.

Naturally, you could mount the array horizontally and line up all of the wavefronts with DSP but then I'd need about 10 channels of amplification and DSP.


----------



## durwood

pwnt by pat said:


> I know this is a very old thread but I thought of an interesting solution that should be easy to test.
> 
> The problem with stereo reproduction is the environment. Everyone will agree to that. But interior reflections aren't the only problem.
> 
> Proper stereo reproduction requires a central listening position located between two speakers positioned at 30° angles from The center of the head. Any non-central listening position will not produce stereo playback. Period. The audio encoding requires it. Not that you can't a left right stage, but it's not stereo.
> 
> In a vehicle, the speaker opposite side of the car either in the kicks or doors is pretty close to the correct angle. But the close side speaker, even in it's widest position is going to be at 15° +-. Using the idea of vector addition and abusing the cone of confusion concept, a second driver can be located at an angle equal to +-30°+(30-x) where x is the angle of the near side front driver to centerline of the head.
> 
> The negative option just so happens to line up with where most rear door speakers are mounted. This would be about -45° from listener rear, or 135° from listener forward. This sounds goody at first but remember we can't detect a difference between a speaker on a forward or a reward angle on the COC .
> 
> Through vector summing, this would effectively pull the phantom location of the front driver back to the proper 30 degree angle. Then apply significant amplitude reduction and proper time alignment.
> 
> The downsides I see are head turning (but let's be honest, you're not fixing that if you're using additional speakers) and one seat affected. Perhaps if the amplitude is reduced enough then you could use the principle on both sides of the car without significantly affecting the far side listener. I don't think you would need much volume at all to create the phantom speaker.
> 
> Thoughts?


I suggested this to a friend of mine and he tried it in his golf a while back. With the right level and filtering it definitely filled in the midbass nicely and he was only using some Morels that are not really midbass monsters. With some speakers with more capability I would have been more impressed, but the concept worked.


----------



## High Resolution Audio

Patrick Bateman said:


> A few observations:
> 
> 1) It's almost impossible to perceive height at low frequencies. This is due to the way that we hear; at low frequency our perception of location is dictated by phase, and due to that, we can hear left and right but not up and down. At higher frequencies we can *definitely* perceive height, but that's because the frequencies are smaller than our ears. (About 4khz and up.)
> 
> Long story short - as long as the driver is playing 500hz or lower, the harmonics won't give away it's height. (4khz is the *seventh* harmonic of 500hz, well out of band.)
> 
> Moving the driver to a location where you can't see it can help a lot. Our eyes will often override our ears when it comes to location. It's the reason loudspeakers have a better soundstage with the lights off.
> 
> 2) There are some who believe that harmonic distortion is inoffensive. I am starting to agree with them. For instance, a lot of music has harmonic distortion added to the mix intentionally. Does it sound bad? No. Does it sound different? Yes. Distortion makes things sound louder than they are.
> 
> 3) At low frequencies the angle of a driver doesn't matter. For instance, 100hz is over three METERS long. The wave is so long, it swamps the driver, and orientation doesn't matter.



With regards to 1. and 3.

I disagree with you, because I had my mid-bass drivers located in front of me in the ceiling angled downward at about a 45 degree angle. They were playing 45 - 450 Htz. At a SQ gathering and the next day at a competition, both a listener and a judge said that my bass tracked across the windshield about 6-7" BELOW where the mid-ranges and highs tracked.

I redid my mid-bass enclosures and changed the angle up so that they were aiming parallel to the mid-range and tweeters. (Horizontal)

My mid-ranges and tweeters are in towers in the corners of the cab. The bass now tracks perfectly up high in the same plane as the mid-range and tweeters. 

In my experience, low frequencies are directional, can be located, and aiming is important. 


In my experience, in sound quality competitions and in every day life, and in my profession as an electrician, I have come across some people that can locate direction of sound, and others that cannot locate where sound is coming from ( even high pitched beeps from smoke detectors which are extremely directional)

Perfect example..... the height of my sound-stage. My stage at its highest tracks about 6" above the glass. I have gotten scores for stage height varying from 15 in IASCA to 6 in MECA. ( Highest possible scores from judges that can locate sounds )

To a low of 8 in IASCA up to 11 with the same tune from judges that have difficulty locating sounds. 


With regards to 2:

Any type of distortion is audible and offensive to me. Especially in the low frequencies. 

I believe that we all hear things differently. So blanket statements do not really apply everyone in all circumstances.


----------



## pocket5s

most would probably say that 450hz is well into the lower midrange and at that frequency I would not be surprised if aiming had an effect due to reflections. most would probably consider midbass to be under 200hz-ish, where it generally does not have a height impact, assuming there are no other variables involved (resonance and such). 

however I do agree that different vehicles react differently. for example, my midrange and midbass are below the dash, with midrange playing up to 3k. Tweeters are in the pillars. In my particular case the stage is above the dash all the way across. Turn off the tweeters and it will pull down depending on frequency. 

I have heard two other vehicles with kick panel mids and they did the same as my car. But many others have tried it and it had the opposite effect. Testing, testing...


----------



## Patrick Bateman

I wrote that post four years ago, but I'm not sure what I was thinking. You can *definitely* localize 500Hz, largely because the second and third harmonics are at 1000hz and 2000hz. 

IE, if your midbass is low-passed at 500Hz, it's still going to produce 2nd harmonics at 1000hz and 3rd harmonics at 1500Hz. And though the fundamental may be difficult to locate, the harmonics will give it away. Back in the day, the Speakerworks cars got away with it because there were other drivers that masked the harmonics, and the midbasses were exceptionally low in distortion. (Not just the driver's harmonic distortion, but also distortion produced from vibration generated by the enclosure.)


----------



## High Resolution Audio

Patrick Bateman said:


> I wrote that post four years ago, but I'm not sure what I was thinking. You can *definitely* localize 500Hz, largely because the second and third harmonics are at 1000hz and 2000hz.
> 
> IE, if your midbass is low-passed at 500Hz, it's still going to produce 2nd harmonics at 1000hz and 3rd harmonics at 1500Hz. And though the fundamental may be difficult to locate, the harmonics will give it away. Back in the day, the Speakerworks cars got away with it because there were other drivers that masked the harmonics, and the midbasses were exceptionally low in distortion. (Not just the driver's harmonic distortion, but also distortion produced from vibration generated by the enclosure.)



The Brax matrix 10.1 drivers I'm using for mid-basses are low in distortion.

When designing my mid-bass enclosures, I created a box inside a box separated by rubber. I have almost zero audible cabinet resonances. 1 3/8" thick. ( see photo )

I find it difficult to believe that harmonics were the only reason my bass notes tracked 6-7" lower across my windshield, because re-aiming those drivers corrected said issue. 

It is my belief from my experience, that frequencies well below 500 Htz can be localized. 

Not only from distortion, but most importantly from being out of phase. 

For example, the Magic Bus. The bus has (3) 12" drivers in one large coffin located behind the driver. With the Alpine system he had in there there was only 8 channels, so there was no way to time align all three subs.

The subs were located three different distances as they were positioned front to rear and not side to side.

I was distracted by "sub pull" from the rear when listening to that system.

In conclusion, It is my belief that frequencies below 624 Htz can be localized if out of phase.....but not because of 2nd and 3rd harmonics.


----------



## Patrick Bateman

Multiple subs do a really nice job of making that go away. That's how I run things at home. I haven't had an opportunity to hear it done in a car. I *have* heard The Magic Bus a bunch of times, Jon lives about five miles from me.


----------



## ANDRESVELASCO

impulse said:


> I kind of faked mine using Maxxbass by wave audio. It's some kind of psychoacoustics which makes your 6.5's sound deeper without actually going deeper. I have to say it really works well, I would say it's on par with sounding like an 8 inch sub.
> I used it cause I needed something to bring my subwoofers up front more.
> 
> It's this thing here: http://i102.photobucket.com/albums/m96/bigbubba_82/install020.jpg


(This Quote is from another thread)

I've read almost all this thread, and I'm sort of sorprised how no one talk about phsyco-acoustics technologies and how they really can help with bass and midbass perceptions... some that in Pro Audio is extensively used!

Does any of the experts in this forum have experienced with any MaxxBass device? I.e.:

Earmark Plano, Inc.*::*MARINE*::*EQ's, Line Drivers, X-Overs & Volume Controls*::*EQ's, Line Drivers & X-Overs*::*MaxxBass 103

https://m.ebay.com/itm/Precision-Power-PPI-MaxxBass-Psychoacoustic-Bass-Processor/162724752982

https://www.amazon.com/ORION-MAXXBASS-BASS-ENHANCER-MODULE/dp/B000UQ2H9G



(Most of them are now discontinued because of low demand of these products in car audio world, but not in Pro Audio. Somewhere I read this technology could be the "magic" behind bose tiny speaker systems).


----------



## Patrick Bateman

I've definitely spent a lot of time pondering the technology behind Maxx Bass.

https://www.waves.com/plugins/maxxbass#adding-deep-low-frequencies-with-maxxbass

Maxx Bass creates the illusion of bass by synthesizing harmonics. It is basically the *opposite* of an Audiocontrol Epicenter.

Something that I've noticed with a lot of really high quality installs is that they make the music that I like sound kinda "weird."

Here's an example:

Back in the 90s I was into industrial music, and I listened to it almost daily. In nightclubs and in my car stereo. The nightclub that I went to used Cerwin Vega 18" horn subs. My car used a MTX bandpass box. Both subs have a fair amount of distortion, *particularly since amplifiers were tiny in the 90s and they were being pushed HARD.*

A couple of years ago, I go and see Nine Inch Nails and they're playing on this giant L'Acoustic setup. *It was pretty darn boring TBH.* Though the music was the same, the utter lack of harmonic distortion made everything sound too clean and shiny.

Long story short: I think that distortion can sound good, and the lack of it can make things sound kinda weird. But this is a difficult thing to deal with. If distortion sounds good, then what TYPE of distortions are better than others?

I think the reason that there's a whole subset of people pursuing tube amps and single ended amps is largely because they like the way the distortion sounds.


----------



## ANDRESVELASCO

Patrick Bateman said:


> I've definitely spent a lot of time pondering the technology behind Maxx Bass.
> 
> https://www.waves.com/plugins/maxxbass#adding-deep-low-frequencies-with-maxxbass
> ...


Here some more info:

https://www.google.com.mx/url?sa=t&...8BPcQFghgMAo&usg=AOvVaw0cUhVZhkhgZkUU7NnTC4YK


https://www.google.com.mx/url?sa=t&...8BPcQFghiMAs&usg=AOvVaw0A41pLejdFQewwHK2CQL4d

It's kind of weird see how almost all the information about Maxx Bass for Car Audio applications disapeer at 2006... 

Anyway... Do you know if this tech could be used for mid-ranges freqs? (Would be nice have a solid 80hz perceptions from my 2" uni-q kef's  )

(All the information I have read points that Maxx Bass is only usable for midbass - sub frequencies)


----------

