# basic theory behind what we do



## bjs (Jun 12, 2008)

Just wondering...

If we succeed in the impossible and manage to equalize all the peaks and valleys out of our system unbelievable:making it RULER FLAT:unbelievable will that automatically mean the system impulse response becomes perfect too?

Or is it possible that we might succeed in flattening the frequency response only to find that it actually became worse in the time domain in some manner?


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## brucek (Apr 11, 2006)

> If we succeed in the impossible and manage to equalize all the peaks and valleys out of our system (making it RULER FLAT) will that automatically mean the system impulse response becomes perfect too?


Nope, not at all. 

It takes some careful filtering to control the time domain.

And ruler flat is not so nice to listen too either. We don't hear that way. You don't hear a 30Hz and 100Hz tone played at the same SPL at the same level. The 30hz would seem weaker.


brucek


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## JohnM (Apr 11, 2006)

If the system is minimum phase then a flat frequency response will correspond to a perfect impulse response. However, systems with excess phase can have flat frequency responses but non-ideal impulse responses. Rooms are mostly minimum phase at subwoofer frequencies and filtering that flattens the frequency response also improves the impulse response, but obtaining perfect inverse filters is more of a theoretical than a practical possibility.


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## JohnM (Apr 11, 2006)

brucek said:


> And ruler flat is not so nice to listen too either. We don't hear that way. You don't hear a 30Hz and 100Hz tone played at the same SPL at the same level. The 30hz would seem weaker.


How we hear has no bearing on the correct response of the reproduction chain though, if the frequency response of a system (from original input to final output) is perfectly flat it will reproduce recorded signals exactly as they would have been perceived at the original location they were recorded. Altering the balance may be subjectively preferred but it is no longer what was recorded.


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## brucek (Apr 11, 2006)

> How we hear has no bearing on the correct response of the reproduction chain though


Yes, of course. I was trying to ensure that this wasn't some goal that bjs had, trying to create listening nirvana.

brucek


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## bjs (Jun 12, 2008)

Thanks guys.

If I understood your response correctly JohnM, this leads to a supplemental question.

Hypothetically, if one saw a peak at say 60hz in REW and equalized that down to flat *but* in reality the peak was due to perhaps a room mode at 55hz and another at 63 hertz interacting...then presumably the flattening would mess up the impulse response (perhaps worse than before)? Correct impulse response would only be maintained if one realized the 60hz peak was actually due to two separate modes and equalized them instead (ie the 55hz and the 63 hz separately). The latter would be the only approach that would preserve minimum phase behaviour of the equalization. Is this a correct?

PS. Brucek...I've got REW, and my BFD, AND the correct cables...I'm already in listening nirvana...and I haven't even turned on the stereo! :jiggy:


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## thewire (Jun 28, 2007)

JohnM said:


> How we hear has no bearing on the correct response of the reproduction chain though, if the frequency response of a system (from original input to final output) is perfectly flat it will reproduce recorded signals exactly as they would have been perceived at the original location they were recorded. Altering the balance may be subjectively preferred but it is no longer what was recorded.


My room has more bass 70Hz and up or does the signal? Both? 










With receiver and bass management










Without a receiver and bass management (same room setup mostly, same type of sub but more of them, different settings.)

Something seems to be occurring before the signal reaches the subwoofer other than just a +10dB boost. Maybe the LFE crossover? I read that an LFE crossover targets certain frequencies. Is that true? :scratchhead:


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## JohnM (Apr 11, 2006)

bjs said:


> Hypothetically, if one saw a peak at say 60hz in REW and equalized that down to flat *but* in reality the peak was due to perhaps a room mode at 55hz and another at 63 hertz interacting...then presumably the flattening would mess up the impulse response (perhaps worse than before)? Correct impulse response would only be maintained if one realized the 60hz peak was actually due to two separate modes and equalized them instead (ie the 55hz and the 63 hz separately). The latter would be the only approach that would preserve minimum phase behaviour of the equalization. Is this a correct?


The wrong correction in that case wouldn't mess up the IR but it wouldn't improve it much, and the frequency response would not change as predicted (i.e. the measurement of the equalised response would not match the prediction) if the peak being countered was actually a combined effect of two nearby peaks.


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## JohnM (Apr 11, 2006)

thewire said:


> Something seems to be occurring before the signal reaches the subwoofer other than just a +10dB boost. Maybe the LFE crossover? I read that an LFE crossover targets certain frequencies. Is that true? :scratchhead:


The bass management in your receiver applies a low pass filter to the signal, the -3dB point of the filter is at the bass management cutoff frequency. That low pass filtered signal gets sent to the subwoofer, whilst a high pass filtered version gets sent to the main speaker for that channel. If you want to see what the signal looks like completely free of room effects connect the subwoofer output signal from your receiver to the line input on your soundcard and measure that.


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## thewire (Jun 28, 2007)

JohnM said:


> The bass management in your receiver applies a low pass filter to the signal, the -3dB point of the filter is at the bass management cutoff frequency. That low pass filtered signal gets sent to the subwoofer, whilst a high pass filtered version gets sent to the main speaker for that channel. If you want to see what the signal looks like completely free of room effects connect the subwoofer output signal from your receiver to the line input on your soundcard and measure that.


Can you please explain what this quote means. Are they saying that a subwoofer needs to measure louder to be flat?



> Home-theater subwoofers require a boost in output in the 40-80Hz range to make effects seem real and solid. (Why this is not encoded on the DVDs themselves is beyond me. If the DVDs simply had the boost encoded where it is needed, one subwoofer could serve for both music and movies.) Subwoofers for music need linear response from around 20Hz up to 160-200Hz. Surprisingly, movies are not that dependent on response below 40Hz or so. When there is sub-40Hz content in the soundtrack, there seems to be some built-in augmentation to make those frequencies heard and felt, even on subwoofers that have a hard time going that low. DVD soundtracks played back on music subwoofers with linear frequency response sound a bit restrained and lack the impact and serious amounts of air, floor, and wall motion that home-theater subwoofers can generate.


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## Wayne A. Pflughaupt (Apr 13, 2006)

JohnM said:


> How we hear has no bearing on the correct response of the reproduction chain though, if the frequency response of a system (from original input to final output) is perfectly flat it will reproduce recorded signals exactly as they would have been perceived at the original location they were recorded. Altering the balance may be subjectively preferred but it is no longer what was recorded.


No production house or recording studio uses "perfectly flat response" (the movie industry uses the X-curve, for instance). And, unless you're listening in an identical room with identical speakers and electronics as the production house or studio, you're not going to hear "what was recorded" anyway.

Regards,
Wayne


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## thewire (Jun 28, 2007)

Wayne A. Pflughaupt said:


> No production house or recording studio uses "perfectly flat response" (the movie industry uses the X-curve, for instance). And, unless you're listening in an identical room with identical speakers and electronics as the production house or studio, you're not going to hear "what was recorded" anyway.
> 
> Regards,
> Wayne


I have to agree with this. I look at many graphs studying and even the finest studios have a curve. None of the rooms measure perfectly flat. There are so many other great features in REW and a sweep is just one of them. When a person says here is a flat response... :snoring: I would rather look at RT60, room size details (air resonance?, room modes?), what diffusion is there and are the other speakers doing well, whats the decay like? What is the reference level sound good at? That kind of stuff.


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## JohnM (Apr 11, 2006)

Wayne A. Pflughaupt said:


> No production house or recording studio uses "perfectly flat response" (the movie industry uses the X-curve, for instance).


The X curve is primarily to compensate for the perceptual differences of listening in very large spaces and the correction becomes less and less as the size of the space reduces, in domestic sized rooms (extrapolating the data for the larger spaces in the spec) the X curve correction would be flat or very close to it.



Wayne A. Pflughaupt said:


> And, unless you're listening in an identical room with identical speakers and electronics as the production house or studio, you're not going to hear "what was recorded" anyway


What was recorded is what the mic picked up. The mastering engineers may well have applied all kinds of EQ to it afterwards so it sounds how they want it in their production facility, or they may not, but I don't believe they have a consistent bias in the balance they apply that warrants targeting a bass reproduction response that is not flat. Many people like a rising low frequency response, just as many people like ketchup on their food, but both are personal taste and not a correction for some underlying imbalance.


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## JohnM (Apr 11, 2006)

thewire said:


> Can you please explain what this quote means. Are they saying that a subwoofer needs to measure louder to be flat?


In the opinion of the author of that extract DVD soundtracks require boosting between 40 and 80Hz.


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## Guest (Jul 9, 2008)

brucek said:


> And ruler flat is not so nice to listen too either. We don't hear that way. You don't hear a 30Hz and 100Hz tone played at the same SPL at the same level. The 30hz would seem weaker.
> brucek


This is a point that is correct, but one has to be careful of the context in which it is applied. If we are talking about calibrating a system that is not itself intended to correct for perceptual effects, then it is not relevant. If we are trying to get the system to faithfully reproduce what is input, we want flat response. One of the important points about calibration is that it removes variables from the reproduction chain so that we can better appreciate the production. If we then want to equalize to our preference, either for program specific correction or a general preference, there is nothing wrong with doing so. When calibrating a system, or comparing systems, however, IMO it is best to target flat response to start with.

We run into the same issue with video, where I do much more calibration work. People are conditioned to like higher color temperatures. If you calibrate to a higher color temp, however, you bias everything in one direction. Programs that were produced with a bias may look better or worse but those that happen to be off in the same direction are much worse than if you start with a neutral display. There is nothing wrong with changing the settings to one's preference, but given that you have variance in the programming, it is usually useful to start with a system that does as little as possible to alter what is input. Then one can make better judgements about what corrections are appropriate and desired.

The bottom line is, when we talk about calibration of any system, we have to start by understanding what the system is doing to the signal. Only then can you adapt that system to the preferences of the user or if you prefer, the perceptual tendencies of a population.


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## brucek (Apr 11, 2006)

> it is best to target flat response to start with


Yeah, I suppose, but I would think you would need to listen to music at the same SPL level that it was originally mixed at to get the same sound as the engineer heard.

SPL level is quite important in determining the type of emphasis you want to apply to your low frequencies.
The weighting curves certainly have a range they use that changes the emphasis as the level of low and high frequencies change.
See below:

A-weight recommended level use between ~ 20dB - 55dB.
B-weight recommended level use between ~ 55dB - 85dB.
C-weight recommended level use between ~ 85dB - 140dB.

So we can see as the SPL level gets lower, we require more compensation with regard to the filtering needed to make the same frequency 'sound' at the same level to our ears.

Of course we remove any weighting with REW, to produce a flat response, and I sure don't see any other way than that to do it either, as it trys to reproduce the original mixed sound. But I ain't gonna listen to music that way.. especially since I listen at such low levels 

brucek


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## thewire (Jun 28, 2007)

It looks like a Media Director would give us more than a standard X-Curve. Do you see any reasons why this would not work? We would have to start at a flat response? If the director intended a rising response than how does one get that rising response when starting at a flat one? This would require boosting then I would think. If it were flat to 10Hz and the content says there is roll off at 20Hz, this would also mean allot of cutting. I would think then maybe that the best accurate way to reproduce the intended experience would be to get close as possible to the intended frequency response, prior to changes being made. I also wonder if there would be both an intended version for both home, and commercial cinemas. I would think that rather the director would also want this to become a faithful reproduction in both areas, therefore most likely the mix would be intended for commercial theaters, where response is not flat. There would of course be limitations as Marshall discussed with storage and not all may like the idea. 

http://www.thx.com/technologies/mediadirector/how.html


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## DrWho (Sep 27, 2006)

brucek said:


> Yeah, I suppose, but I would think you would need to listen to music at the same SPL level that it was originally mixed at to get the same sound as the engineer heard.


That is a common misconception.

Take for instance a very simple example of a studio engineer recording a cello...single mic placed about 4 feet away with the cello sitting in a nice full sounding studio. And then assume the goal of this studio engineer is to capture exactly how that cello sounds in that room.

If the engineer is monitoring the mix at say 90dB, then the engineer is going to call upon his acoustic memory of what a cello sounds like at 90dB. If instead, he chooses to listen at 70dB, then he will call upon his acoustic memory of what a cello sounds like at 70dB. His acoustic memory is already compensating for the F-M curves.

Trying to make your system respond with the inverse of the F-M curves is not an approach towards more accurate sound. If anything, it's more an approach of hyper detail - kinda like super bright neon colors. Another result is that we can _almost_ (it's not perfect) make things appear louder than they really are because our acoustic memory gets mixed in with the perceived tonal balance.

But regardless, if you're listening at 50dB, then you're hearing the correct tonal balance of the cello as if it were playing at 50dB. I don't think it makes much sense to listen to a 90dB cello at 50dB (when accuracy is the goal).


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## Guest (Jul 24, 2008)

EQ...that term makes me shiver. Here are few drive-by comments...

This is A LARGE can of worms that would be well worth discussing and requiring everyone to check off on, is a thorough understanding of the limits of equalization and the recognition of just what can, and cannot be equalized! As this subject is one of the lest well understood, and subsequently, one of the most abused.


First, the frequency domain is not the domain of primacy. That distinction belongs to the time domain. Simply put, I would suggest that for all practical purposes, except to view the room modes, that you stop looking at the system in the frequency domain. (A gross oversimplification perhaps, but one that would have the practical effects of actually resulting in major improvements without the concomitant errors SO COMMONLY made!) 

If you resolve issues in the time domain, you resolve much of what is displayed as an anomaly in the frequency domain.

A thorough understanding of what can, and cannot, be equalized is required, and EQ is among the last thing to be utilized. Simply put, one cannot equalize a non-minimum phase environment. Without getting into the math or discussing systems where the poles exist only in the negative (left) half of the S plane, or as a system that can release its potential energy in minimum time, (is everyone confused yet? ;-) ) one might think (a bit over-simplistically) of minimum phase as a system (or regions of a system response) that exhibit no destructive superposition of real and virtual signals.

{Edit - add: After addressing room modes...} All the more reason to run for the ETC diagram _with displays and details of the discrete reflections_ within the envelope as a 'first defense' for general acoustical analysis and remediation. :bigsmile:


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## thewire (Jun 28, 2007)

thewire said:


> It looks like a Media Director would give us more than a standard X-Curve. Do you see any reasons why this would not work? We would have to start at a flat response? If the director intended a rising response than how does one get that rising response when starting at a flat one? This would require boosting then I would think. If it were flat to 10Hz and the content says there is roll off at 20Hz, this would also mean allot of cutting. I would think then maybe that the best accurate way to reproduce the intended experience would be to get close as possible to the intended frequency response, prior to changes being made. I also wonder if there would be both an intended version for both home, and commercial cinemas. I would think that rather the director would also want this to become a faithful reproduction in both areas, therefore most likely the mix would be intended for commercial theaters, where response is not flat. There would of course be limitations as Marshall discussed with storage and not all may like the idea.
> 
> http://www.thx.com/technologies/mediadirector/how.html



To correct myself after speaking with my instuctor John Dahl, the Media Director does have a PEQ and everthing else. It will setup the system exactly like the studio had it unless settings are overiden. The reponse we recommend to be flat and within +-2dB of 0dB to start with. The settings come on the disk themselves.


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## Wayne A. Pflughaupt (Apr 13, 2006)

Mark,

What is the extent of your experience using equalizers?

Regards,
Wayne


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## Guest (Jul 26, 2008)

Wayne A. Pflughaupt said:


> What is the extent of your experience using equalizers?



Quite extensive over a near 35 year period.
They work great in a minimum phase environment - notably on source material in recording and to effect the direct signal from a speaker - assuming the various acoustic origins are in alignment (which is usually limited to one physical axis as they are normally stacked and not coincident).

But they are not effective in correcting for the effects of superposition of multiple sources, real and/or virtual. And unfortunately, this is exactly the case that too many try to resolve when one tries to EQ a system in a room - (as only the direct signal Ld can be effectively EQ'd - and the rare regions (passbands) of the response at a particular location where the passband signal may be minimum phase - where a parametric eq can be of use.) 

But I will also stick my neck out and argue vehemently against the all too common use of simply measuring the room's frequency response - either at a location or averaged over the room - and simply inverting the signal and reapplying it as feedback in the thought that we can address room anomalies by tampering with, and flattening the frequency response in the frequency domain. If only acoustics were so simple...;-)

Traditional use of EQ simply imparts small LC induced changes in phase of the direct signal(s) which when recombined with multiple direct signals as well as any virtual sources result in small shifts in the comb filtering and polar anomalies. It was always instructive to understand that the 'walking spectrum analyzers' (he said affectionately) I know who can identify a signal within 3-5 Hz in ringing out a system merely caused the nulls at the FOH mix position caused by overlapping coverage of the various arrays in a large hall to be shifted 6 feet onto the paying customers' seats... and since no one really cared what happened there (and as so few folks have ever walked across a hall and have a clue as to the extent of the affect of the polar anomalies), the problem was "solved".:devil:

This case was further 'brought home' at the SAC seminars where two Auratone speakers were stacked vertically and fed with identical signals and the comb filtering was displayed (at frequencies above the quarter wavelength limit below which they would effectively sum) and to watch and listen as the top speaker was slowly physically shifted back, rendering both the measured phase and resultant frequency domain comb filtering readily apparent in the measured display while the listening experience was akin to that of listening to a high Q air raid siren or Leslie being rotated about the room as the polar anomalies became more greatly exaggerated as the inter-driver spacing varied. 

And if that was not dramatic enough, the fun part came when a cut and boost capable EQ was put into the signal path - at which point the valiant attempts to use the 'classical' approach (meaning, use whatever tools you have on hand!) resulted not in resolution, but rather in fascinating phase wrap anomalies that are hard to describe in words, but where it is painfully obvious the something 'ain't quite right'! Not to mention resulting feedback issues... 

The above demonstration remains one of the simplest and yet most dramatic demonstrations that I have ever encountered in audio.

The issues of which I refer (attempts to EQ non-minimum phase environments) were hot topics during the ~1987-1991 period when principles such as Don Davis - who ironically first introduced the 1/3 octave Equalizer while at Altec and subsequently illustrated this limitation quite convincingly with the assistance of the TEF and folks such as Don Keele and others; and at least in the pro markets this debate has thankfully _ceased_ to be a debate - and the racks of EQ that used to be so common in SR rigs are nor reduced to one or two parametric or third octave 'cut only' units now - with EQ largely limited to the direct mic sources. Unfortunately, the use of EQ seems to continue unabated in the 'audiophile world' where the frequency response perspective unfortunately remains a primary focal point to the exclusion of an effective awareness of the time domain. :bigsmile:


Edit2: 
For more detailed information regarding the limitations of equalization, see _*Sound System Engineering*_, Davis & Patronis, 3rd Ed. , _Chapter 14, Signal Processing_.


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## Wayne A. Pflughaupt (Apr 13, 2006)

I have no idea what any of that means, but this Shack member was very pleased with the results he got using a pair of Rane parametrics.

Regards,
Wayne


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## JohnM (Apr 11, 2006)

mas said:


> Unfortunately, the use of EQ seems to continue unabated in the 'audiophile world' where the frequency response perspective unfortunately remains a primary focal point to the exclusion of an effective awareness of the time domain.


On what basis do you say that? Just about all inhabitants of the "audiophile world" I have come across are strongly opposed to equalisers in general and full range EQ, as you were discussing, in particular.


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## Guest (Jul 26, 2008)

JohnM said:


> On what basis do you say that?


With all due respect, this question genuinely made me laugh.

If _only_ your assumption were the case! To coin a phrase, "Tis a consummation devoutly to be wished"!I _sincerely_ wish that I could agree.

One begins to relate to how Dorothy felt when asking for directions from the Scarecrow in the Wizard of Oz, as the subsequent responses are diametrically opposed. I only wish the response to my post mentioning some of these issues had been met with anything other than a statement that they were not familiar with anything that I said! So, I guess that we could begin there...or simply with the original assumptions posited in this original question for this very thread!:bigsmile:

Not only are the majority still living in the flatland of the frequency domain and far too many totally unaware of time domain aware tools such as TEF, EASERA, SMAART, PRAXIS, et al, simply mentioning the fact that EQ is a technique of limited applicability is almost assured to initiate a _heated_ debate. 

In fact, what I wrote above regarding the inability to EQ non-minimum phase 'environments' - which BTW is correct, was met with anything but acceptance, and I am sure that far more who have not commented have little idea of that which I refer!

This topic is perhaps the simplest way to ascertain the mindset and awareness of an audience. How many possess a sufficient understanding of modern acoustical models such that they can delineate the limitations of equalization? And the fact is that this debate is limited almost exclusively nowadays to the realm of home based 'audiophilia'.

And while there has been a growing understanding of this fact, primarily due to the substantial efforts of Don Davis and the consortium of prominent figures in acoustics via SynAudCon, all one has to do is visit any forum. In fact, few things would please and impress me more than to have this issue cease to be a prime source of misunderstanding, as it would reflect an increasing understanding of the primacy of first addressing (most) issues subject to superposition via techniques such as signal alignment in electronics as well as acoustically within the time domain.

In fact, when you ask why one might suggest this topic as a litmus test of folks awareness, may I refer one to DrWho for his take on the response generated elsewhere when this topic was breached...


As I have said, I wish MORE were aware of this. And I wish comments regarding this topic were met more with "yeah, we already know", or "of course, you are preaching to the choir" rather than the all too common response of dealing with masses of folks showing up in droves with pick forks and torches to burn the witch!:devil:


:reading:
A good introduction to the issue can be found in John Murray's post located at: 

http://www.prosonicsolutions.com/articles/Equalization%20Revisited.pdf
{edit: Note, I do not completely agree with his conclusion regarding the treatment of room modes. Resonant traps and absorption are indeed useful.}


:reading:
And Davis and Patronis address this issue in a bit more technical depth in _*Sound System Engineering*_, Chapter 14.

Take care...


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## Wayne A. Pflughaupt (Apr 13, 2006)

mas said:


> With all due respect, this question genuinely made me laugh.
> 
> If _only_ your assumption were the case!


Hmmm. Well, I can only assume then, that what pro audio people define as "audiophiles" and what home audio people define as "audiophiles" are not the same thing. 

I can assure you that John is correct: The die-hard home audiophile crowd - and I'd define those as the people whose systems cost tens of thousands and include such novelties as interconnects with silver wire - generally revile equalizers (and indeed, even simple tone controls) and would not be caught dead with one in their systems. The fact that you'd send us to ponder some pro audio references to bolster your stance further indicates that we're not on the same page.



> And the fact is that this debate is limited almost exclusively nowadays to the realm of home based 'audiophilia'.


Perhaps you could give us some references. 

Regards,
Wayne


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## Wayne A. Pflughaupt (Apr 13, 2006)

mas said:


> As I have said, I wish MORE were aware of this. And I wish comments regarding this topic were met more with "yeah, we already know", or "of course, you are preaching to the choir" rather than the all too common response of dealing with masses of folks showing up in droves with pick forks and torches to burn the witch!


 You have to keep in mind that this Forum is mainly populated by folks with no pro audio or room-measuring experience, or experience with professional-grade equalizers. One of the objectives of this Forum is to lend them a hand.

As for myself, most of what you've written I've seen before, albeit in a much more readable presenation. I can't say that the "it doesn't work" theory has been my experience, but then, I don't "room equalize"...

Regards,
Wayne


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## Guest (Jul 27, 2008)

I am aware of a great many folks with more money than 'awareness' (how was that for cleaning that up?!?):bigsmile:

And far too many attempt to solve acoustics issues in the small signal arena or by simply buying a new piece of gear with 'magic room correction' capabilities that are _all the rage _now.
...Especially since so many are simply generating a frequency response curve and feeding back an inverted signal to flatten the response!:rofl:

To mention a closely related aspect...

But then so many in this crowd also spend far too much time and money trying to justify magic wires when they could achieve the same effects by simply varying the reactance of their current interconnects with a few capacitors and inductors! Of course, how many have take the time to evaluate the 3Space Nyquist and Heyser spirals of not only the interconnects, but of both the sources and speaker/crossover with which they combine to effect the total load in order to evaluate the total impedance in all of its domains and the relationships exhibited by the various combinations and variations. 

I mean, should anyone be surprised that a source that is, say, capacitive in the low frequencies, might sound a bit different that a source that is inductive in the low frequencies? Hmmm. Is the concept of matching sources to loads that foreign? While I admit our understanding of precise relationships is still in its infancy, we now have the tools to explore this aspect in MUCH greater depth!

But, have you seen one test that attempts to quantify and qualify such variations? Or for that matter, how many reviewers are even aware of that to which I refer? And I haven't even mentioned further implications of aspects such as the relationship of the potential and kinetic energy characteristics of each comp0onent that are also easily evaluated in depth...

And if it is suggested that acoustics extends just a 'bit' beyond the fancy new piece of gear they simply plug in, it is wise to be wearing a brand new pair of running shoes in which the 'fast' has not yet worn off.

In fact, it wasn't until just a few years ago that several of the 'audiophile' mags even became aware that room treatment could work. and the review are quite interesting to read if only to watch the bemused amazement of the authors.

And yet we are still routinely encountering the attitudes as expressed in this thread which begin by focusing upon the frequency domain and equalization rather than upon signal synchronization in the time domain and then addressing the various acoustic anomalies.

For that matter, when technologies such as TEF, EASERA, SMAART, and other industry standard analytical tools are mentioned, the response is "Huh"?

Again, once the can of worms is opened, MANY interrelated issues are released from Pandora's Box. 

And the fact is that I have dealt with MANY 'audiophiles' with systems costing far more than many folk's homes. And even more electronics engineers and technicians in the field where our discussions do not move to the implications and questions that such new paradigms evoke, but rather, like here, to simply breaching the subject and trying to effectively communicate some of the most basic precepts and attempting to communicate - nay, simply counter existing preconceptions - in order to get many to even consider that such new paradigms are valid! 

To perhaps overly simplify in a sardonic manner, it is one thing to be limited to using an RTA and a RS SPL meter due to one's inability to afford some of the more robust tools. It is quite another to employ them because one believes that they are the optimal tools available and to not understand their fundamental limitations, and to possess a desire to obtain the more robust time domain capable tools.

In other words, such an understanding would result in anything BUT one's initial step of simply trying to achieving a "ruler flat frequency response in the room".

But the notion that the majority of 'audiophiles' are conversant with, and understand, the new acoustical paradigms as suggested by Heyser and made practical by by the advent of such tools as TEF and EASRA is wishful thinking at best. Especially as just the mention of these topics, let alone some of the best practices employed in using them, in the past few posts has resulted in queries as to what they even are!

We are unfortunately_ not_ on the downward slope where such an understanding is the norm! That remains a goal! 

Rather the realm of consumer audio remains firmly on the uphill slope fighting both inertia and the marketing of fancy room correcting canned solutions that one simply plugs into the signal chain as well as cable where electrons magically know which conductor to traverse - other wise know as a crossover. ****, I was taken to task for merely stating the fact, also supported by the writings of D'Antonio, Berger, and an entire _plethora_ of others, that a small acoustical space can _easily_ be rendered too dead by the inadvertent use of absorption rather than its surgical use and complement of diffusion! (Infact, that was exactly the initial response to the first construction of the LEDE concept by Chips Davis in 1977!) And many still think a small acoustical space exhibits a sufficient statistically reverberant sound field to render RT60s to be effectively calculated. But its nice(sic) to see 'some' measuring and calculating something that doesn't even effectively exist in a small acoustical space! I am sure that I have stepped on quite a few toes here; but my intent is simply to, at the least, point to more current models that have supplanted those notions that have been heretofore widely accepted and to hopefully supplant them with a few new substantial notions that have been widely examined and which define the modern acoustical models.

And while some audiophiles eschew tone controls preferring to rely upon small signal analysis has not translated into the widespread understanding of acoustical behavior in the small acoustical space (as defined by that Manfred Schroeder guy).

Doubt that?...just visit any of the online 'audiophile' sites and dare to suggest to those who find solutions in boxes and feel no apprehension at spending thousands of dollars for interconnects or a 3 foot power cord, that they would be well served to pay a couple hundred bucks to have a _qualified_ acoustician come and measure their room and follow up any modifications with a proof of performance analysis.

...Been there and done that. At best, the result is a debate over the validity of such analysis and some fascinatingly obscure subjective versus objective debate by folks who have no idea or understanding of the concepts to what they even refer.

Of course, we could simply start by asking folks what minimum phase is...and what it implies, or simply what cannot be EQ'd. A very simple question that will tell one exactly where they are coming from. The fact is, simply being able to afford expensive gear does not render one conversant or understanding of the current acoustical models that have been undergoing a radical transformation over the past 35 years. (And ironically, the practical models which we are currently employ in room tuning still fail to approach the _most current _ acoustical models for very practical reasons - as a trip to Blackbird studios in Nashville will quickly demonstrate!)

And it is a query in high end shops and elsewhere, that _far_ too often returns a response that tells one that a discussion, except on a social basis, will be less than fruitful.

I'm sorry if I write referring to the issue(s) itself rather than spending pages to re-present the new acoustical models from scratch in "more readable forms". But nothing that I have presented should be in any way obscure to one familiar with the concepts. 

And as far as not "room equalizing", then it might have been appropriate for the initial response to this thread to have reflected that approach, as the intent was _definitely not_ to equalize the direct signal Ld!

But then it is a bit surprising if one cannot understand that the the 'pro-audio' references were _making reference to fundamental acoustical principles_ and had little to do with whether one spends big bucks for a system or spends big bucks to attend a concert that further amplifies the problem.

For you see, acoustics doesn't care if you are a pro or an audiophile. But in the future I will try my best to find an acoustical physics reference that appeals to amateurs. I guess that means we should avoid sources such as D'Antonio, Heyser, Keele, Ahnert, Benson, Davis, Patronis, Murray, Brown, Prohs, Janssen, Berger, Peutz, Cox, Mitchell, Hughes, Carey, Andrews, Snow, Schroeder, Fletcher, Armstrong, and so many others.


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## Guest (Jul 27, 2008)

Hmm. Seems like you are long on condescension and short on practical advice. You make lots of assumptions about what people do, know, believe, and understand. Perhaps in the future you can contribute with some useful input on some specific issues or problems, rather than just ranting about what is wrong with the world of consumer audio and how everyone thinks. Here at HTS we focus on being helpful rather than being purely critical. Obviously someone who knows as much as you could be a valuable contributor and play an great role in educating our readers, but you will have to take a more positive approach and get more specific as a participant.


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## bjs (Jun 12, 2008)

Mas....to the point...are there any conditions where a Parametric EQ would be useful in a home music/theatre situation? If yes, what are they? And what measurements does a user require in preparation for EQ'ing?


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## Guest (Jul 27, 2008)

lcaillo said:


> Seems like you are long on condescension and short on practical advice. You make lots of assumptions about what people do, know, believe, and understand.


Really?
I started off by expressing concern about the perceived use of EQ, especially since the thread began with the question that presumed a use for which EQ has traditionally been used and fo which it fails.

Except for sparse regions, the sound field of a room with multiple real and/or virtual superposed signals is not minimum phase and cannot be EQ'd.

{Of course, instead of acoustics principles, you can always choose to discuss me, as if that changes the validity of the concept to which I refer... But I would suggest that that is a rather boring subject as compared to the acoustical physics issue at hand.}

So, one might ask, if they are not familiar, what superposition is, or what real or virtual sources are, or why destructive superposition is not equalizable...but so far no one has. One of the moderators has referred to minimum phase conditions. Do you or others understand what is meant by minimum or non-minimum phase? As the term was used, should I assume that no one understands it, or that this has been covered in depth and that it is part of the assumed baseline knowledge here? And if it is, why are we having this discussion? And since we are, is it valid for me to assume that the term is not widely understood, as its implications for EQ should be understood as well! Hmmm, I understand the term and its use and implications...and I am both chastised for assuming some understand it and also for others not knowing it. LOL!


Then I made a comment to the effect that many folks who _consider themselves_ 'audiophiles' still have the notion that room problems (other than the direct Ld signal) can be equalized and that room issues consist of tuning the frequency response of the room in the frequency domain, (of which DrWho can verify _at least_ one prominent instance of this!) as well as the additional evidence of many of the increasingly common 'room EQ' functions built into many devices - many of which simply measure the frequency response of the mic position and invert and feedback the signal in an attempt to flatten the frequency response - if ONLY such a method were so simple or effective!

Then others who expressed confusion over the limitations I had quickly mentioned stated that others were quite happy with using EQ. As I was concerned with limitations regarding its use for the general use in altering room response other than in the limited situations originally stated (LF modal peaks and minimum phase regions) I am not sure what their point was (as the link did not work for me).

Or you can waste your time debating the semantics of just what constitutes an "audiophile'! If some what to debate a definition of the term 'audiophile', they are more than welcome to do so. If they want to debate whether the aggregate to whom I refer are 'audiophiles', I suggest that this is not only a separate issue, but it is spurious to the basic substance of the acoustical issue stated regarding the valid and invalid uses of equalization which are very limited in addressing full range acoustical room phenomena. 

I also suggested that others could query another member (DrWho) for verification that at least several prominent persons who consider themselves "audiophiles" have expressed quite the opposite opinion. If anyone doubts the facticity of this claim, PM me and I will provide the links!

I then provided several links which address some of the legitimate uses and the very limitations to the use of EQ to which I asserted and was then told that acoustic references from pro-audio sources, because they are considered from 'pro' sources, are somehow invalid in 'amateur' acoustics - a distinction which is curious to me as within the limits of similar atmospheric conditions, large acoustical space and small acoustical space acoustics are the same - be it for pro _or_ amateurs!! A very curious distinction!



As to how one EQ a room other than for LF modal response which has already addressed at length in the forum?

Its simple, yet not so simple. 

*Measurement tools such as the TEF EZ-Tune software or the use of SMAART are able to identify the minimum phase passbands (if any indeed exist) where the response is minimum phase and for which parametric EQ settings are calculated and specified, and can then be applied and the response verified, Other than these methods, I am not aware of a simple efficient manner to to do it. And such attempts, without the aid of specialized equipment are generally more destructive than useful
*

*But even if one has the ability to identify the minimum phase passbands of a room response capable of being EQ'd, one is focusing on the very last step of a long process and missing the larger more productive steps in tuning the room and putting the cart before the horse!

Instead, after addressing room modes, one should be focusing upon the envelope time curve (the ETC response).
Here you need a ETC display capable of displaying the discrete specular reflections in terms of their arrival times and their amplitudes. From this can be surgically resolved the specific paths and points of incidence whereby the surgical (minimum) application of absorption is used to damp the early first order reflections sufficient to define the Initial Signal Delay (ISD) gap - a period of time in which the listening experience is essentially anechoic and the predominate source is the direct signal Ld.

After this defined period where the early arriving specular reflections that interfere with intelligibility (Haas effect), we address the discrete focused specular reflections. Rather than absorbing these reflections (a technique which dominated the period of time prior to Peter D'Antonio's application of Manfred Schroeder's reflective phase gratings and which this technology has largely rendered this technique obsolete since the late 80s), the discrete display of specular reflections via the ETC and diffusive technology allow us to resolve each focused specular reflection with regards to its path and reflective points of incidence. At these points we employ diffusive treatments which not only diffuses the signal reducing the concentration (amplitude) of acoustic energy, but in so doing distributes the energy in time, producing a more diffuse sound field. The overall later arriving specular reflections are adjusted as necessary using diffusive treatments (and only rarely the additional surgical use of absorption for very exceptional issues) until the energy that constitutes the later arriving specular reflections is reduced in amplitude and diffused in time in a manner that it approaches a decreasing logarithmic slope decay. 

The finite available later arriving acoustical energy in the form of specular reflections is not, nor need it be, absorbed. Rather it is diffused in a well behaved manner such that it it is devoid of more intense specular reflections that are detrimental to the perceived intelligibility while at the same time imparting an increased, psycho-acoustically pleasant, sense of space. In doing so, a region dominated by sparse focused specular reflections is transformed into a more greatly diffused semi-reverberant space.*

*Only after this is accomplished is EQ to be attempted!

At this point, if one has the means, minimum phase regions of the passband at a particular listening position are identified and precise parametric equalization applied to the degree possible. Note, that there do not always exist minimum phase regions capable of being equalized. And lacking the ability to effectively identify the minimum phase passbands capable of being equalized, the majority of the most significant tuning which constitutes the MAJOR improvement in intelligibility has already been accomplished. It is the fundamental addressing of the acoustical field with respect to time that resolves the vast majority of issues, including those that may may be observed in the alternative perspective of the frequency domain.*

*In lieu of the ability to identify minimum phase passbands, the application of EQ merely shifts the phase of the direct signal which then recombines with the various additional real and virtual sources (reflections) via superposition and the resultant comb filtering and polar anomalies are shifted slightly with regards to their frequency and orientation about the axis, as well as severe phase wrap (group delay) issues.*

So, I would suggest that there are_ much more fundamental issues to be addressed before EQ is even considered _in regions other than for the mitigation of LF modal peaks.

And to identify minimum phase regions requires a few advanced tools.


And it was only several days ago that I was asked just what such things such as TEF, EASERA, SMAART, and other advanced industry standard tools are. So, should I assume that ll are familiar with them, or should I assume that they are not? 

You are worried about what I assume, when I quite frankly can only assume that many are not familiar with the equipment, or the theory as evidenced by some of the questions and assumptions posited in the nature of the thread itself!! And all you offer is a challenge to what you perceive as my assumption, despite a few here claiming to be familiar with the concept of minimum phase and others not having any idea as to the limitations to which I refer!!! Again we can relate to Dorothy's dilemma! "Which way to OZ, Scarecrow!" :bigsmile:

So, as in logical truth tables, a negative in an either/or case renders the results negative, I have chosen to assume that many here do not understand the limitations of the use of EQ. And such an understanding is important. But in order to understand the full reasons, it requires a broad understanding of acoustical behavior in the time domain as well as superposition. And we are very quickly back at a very large change in the entire perceptual paradigm that has occurred during the last 35 years in acoustics - a veritable quantum leap if you will. So where do you want us to start? Again, I would begin by pointing you to *Sound Sytem Engineering, 3rd Ed.*, by Davis and Patronis for a baseline introduction to the subject. (Do you denote a pattern here? ;-)

I can refer you to a few quality sources that attempt to introduce many of the fundamental changes in the paradigm, but it is _*just a bit*_ beyond the scope for me to present the full explanation of the changes! And how many have bothered to query the sources such as Davis and Patronis' _*Sound System Engineering, 3rd Ed.*_ that attempt to address this issue? How many have read Heysers works which are collected in the AES Anthology of his works? I can make reference to them, but if you are expecting a 25 word or less explanation of them, aside from summary statements regarding what can be done and what cannot, you what what to my knowledge cannot be done. 

If you want to attribute that to a personal limitation of mine, have at it. But I must suggest that it also reflects on those who refuse to seek out the readily available sources published in various texts of D'Antonio and Cox, the AES Journal, ASA Journal , Davis and Patronis, Manfred Schroeder, various SynAudCon publications, ProSoundWeb articles and any number of other sources!

I can make comments and references. Do I have to do all of your reading for you too? The irony is that I have not only read, but been involved with a lot of the research over the years. If someone has a question that they want to ask me, PM me and I will be glad to chat or even call if you live in North America and discuss it and try to narrowly refine whatever you might like to know that I can assist with. But if all you have is an issue with your presumptions of my assumptions, you are going to have to do a bit better than that - as evidently you are not familiar with the topic to which I refer.

And folks are evidently _still_ wondering about the desirability or, even more fundamentally, the possibility to simply EQ a room response to achieve a "ruler flat" response! The fact is, without it being minimum phase, you can't. You are going to have comb filtering and the accompanying polar anomalies.

Needless to say, my assumption is that many are not aware of the fundamental changes in the acoustical paradigm. And they have profound implications for the tuning of rooms, as well as system design! And acoustics does not vary between amateur or audiophile or pro applications! The only variant that one might legitimately make between these groups might, generally but not necessarily, be the nature of the acoustical environment, be it a large or small acoustical space as defined by Schroeder (for which you can read the pertinent chapters in Davis and Patronis' _*Sound System Engineering*_) and perhaps a variation in the purpose for the various spaces being utilized. (I wonder if I should make reference to SSE once again??? ;-)

Bottomline...one cannot EQ a room to resolve issues resulting from the superposition of multiple phase variant (non-minimum phase) signals. If you already understand this, great. For those who do not, may I suggest discovering why by referencing some of the sources listed, and then asking further questions. But if you do not know, and simply want to complain about my assumptions, I would suggest that your eforts are misplaced.

And I suspect that my assumption that many still do not understand the fundamental limitations of employing EQ in a non-minimum phase acoustical environment is a valid one. Otherwise we would not be ssing such good questions as the the one that leads this thread. With luck, it just may lead to a better understanding of the problems associated with such an assumption. 

And I would suggest a better use of your time would be to discover why this is the case, rather than simply complaining that I have not reproduced the source material here for which I would be glad to do were it not for EULA of the site!

You will just have to pardon me if I fail to respond in a manner that targets and accommodating _every _question or misunderstanding that _everyone_ of varying levels of understanding might have! LOL! 

But for those who have made an effort and who are still confused, my offer of trying to provide a more targeted explanation remains open if you simply PM me. Of course, you can always complain instead about what you assume my assumptions to be..._if_ you think that adds to your understanding of acoustics...:bigsmile:



Stylized ETC 'regional' diagram (note: the Haas kicker _generally_ corresponds to the direct signal Ld):




Example of an ETC showing before and after snapshots:


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## atledreier (Mar 2, 2007)

I'm sure I'm not alone here, I sure like to have someone 'in the industry' on this board. 

Could you please, in layman's terms if possible try to explain what a minimum phase system/region is? 

And what is a polar anomaly, and what does one sound like? I'm sure it's not another environmental issue? 

I have read a few books on acoustics, but many are not written for a 'selftaught' layman like myself.

I have used EQ myself, in a subwoofer application, with great effect. I currently use Audyssey, which is not 'EQ' in a traditional sense. I would love a deeper understanding of the subject, to better identify problem areas in my own room, so I can address them 'physically', not electronically.


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## Guest (Jul 28, 2008)

Superposition is a fundamental characteristic of wave behavior. Derived from ‘to superimpose’, superposition is the way two or more waveforms sum based u[on their phase offset. If the waveforms are in phase they simply add and the amplitude is greater by 6dB. If they are 180 degrees out of phase, they destructively sum to zero amplitude, effectively canceling each other – a behavior that is more commonly known as active noise cancellation. When the phase is other than unity or 180 degrees, the sum is a complex combination that combines the characteristics of both addition and subtraction.

If two or more signals originate from two or more sources, be they real or virtual (such as a diffractive or a reflective source), superposition is what determines their combined response.

While most may be used to seeing such behavior manifest in waveforms on a device like an oscilloscope or in waves in a string, in 3 space, sound waves exhibit this behavior in two ways of great concern to us.
These two characteristics are comb filtering and polar anomalies. Depending upon the time offset (which can also be seen from the perspective a separation by distance – with the distance corresponding to a time offset), there is a fundamental cancellation that is repeated at integer harmonic intervals in the frequency domain. The frequency response looks like a comb with a repeated series of notches, and hence its name.

The polar response refers to the spatial dissipation of the energy, both in a vertical and horizontal aspect in 3 space. Again, with the in phase summing increasing the gain by 6 dB and the 180 degree out of phase intersections resulting in complete cancellation of acoustic energy – with an infinite number of in between states – especially as we deal with non-sinusoidal waveforms.

Below is posted a picture of the effect of 2 speakers modeled in EASE, and you see both the comb filtering in the frequency domain as well as the spatial polar response – how the summed waveform propagates in 3 space. I have also included a few diagrams from a wave tank which display a 2space dissipation of the polar response of an in phase real and virtual source (wall and speaker) illustrating space loading, real and virtual out of phase sources that combine destructively, and two speakers which also sum destructively.

What does this mean? It means that depending upon your location relative to the two (or more) sources, that certain frequency multiples will be notched – creating the equivalent of a highly equalized signal with certain frequencies missing entirely, others accentuated unnaturally, and still others in between emphasized and deemphasized based upon the characteristics determined by the wave itself and the time/distance (phase) offset.

If you have ever been to a concert with multiple arrays or spaced stacks, the effect of this can be dramatic. Next time, you might want to walk slowly from one side of the hall to the other side – from stage right to stage left about half way back from the stage. You will note various spots where certain frequencies seem to literally disappear, and after moving a few feet more, other frequencies being exaggerated and still other frequencies disappearing. In other words, as you move about, it is as if someone is playing a radical game with an EQ. What is happening is that you are physically moving in and out of polar lobbing nulls and sums, and also experiencing the frequency anomalies that ‘go with’ the spatial polar distribution. And depending upon the physical and time offsets, the distribution will vary in both the horizontal and vertical planes. But as we usually aren’t going up and down stairs while listening, we are usually only concerned with the horizontal plane, But in larger venues where you have balconies, etc., the distribution in the vertical plane can also come into play.

Oh, and to make this even more complex, this superposition occurs in a number of orders of scale – of a variety of magnitudes where we have two or more sources reproducing the same signals. Examples are multi-driver speakers – ESPECIALLY if they employ non-signal aligned passive crossovers where adjacent non-coincident drivers – or a driver and a virtual propagation source such as a cabinet diffraction - reproduce an overlapping passband, and then to the next order of magnitude where we have two speaker assemblies reproducing the same signals (especially as in a concert where multiple speakers are summed for additional gain), to the combination of a speaker assembly and their summation with virtual sources such as their reflected signals off room surfaces. (Note, the origins of the D'Appolito MTM configuration was designed to compensate for exactly such a vertical polar tilt utilizing a 3rd order LR filter.)

Thus the situation is reminiscent of a Russian Matryoshka, or nested doll, set. We have a situation where this phenomenon manifests itself in a variety of orders of magnitude.

So we have big trouble in River City! What do we do? Good question!

First, the superposition of signals that are out of phase are known as “non-minimum phase”. While mentioning this, we are not going to dwell much in the technical definition which involves the H(s) in what is commonly referred to as the ‘S plane’. In this model, In a minimum phase system, each pole in the transfer function must lie in the negative half of the complex S plane. While this is quickly getting beyond the scope of this discussion, it is also jumping smack dab in the middle of one of the most exciting developments that has arisen from the development of the TEF – the ability to not only measure, but to display in 3 space, the complete complex impedance of a system - and the system assumes the form of a rotating phasor existing in the real(X) and imaginary(Y) planes rotating about the time or frequency (Z) axes. The imaginary(Y) axis corresponds to kinetic energy and the real(X) axis corresponds to potential energy. And the rotation of the phasor provides a signature of the complex impedance (reactance) at any time or frequency. These displays are known as the Heyser spirals and Nyquist display and are profound in their implications. From these diagrams, the projections of the total response upon each plane - the polar, imaginary and real planes, provide all of the perspectives available for the various topological mappings of the behavior of the total acoustic (and electrical) phenomenon. (I have included a few VERY simplified examples in the diagrams that are easily generated for a component or a system with the TEF) Note that this response captures the entire complex impedance of the system response.

Additionally, from the phase response, we are able to ascertain regions of minimum and non-minimum phase behavior. But this aspect is a bit beyond the scope of what we are addressing at this juncture…

OK, I realize that I have probably lost a few of you, but I hope that a few will realize that they have just seen a diagram of what imaginary number represent! And they are anything but imaginary! 

Rather they are a view of the phenomena that we normally view in the real domain whose projections onto their respective planes represent the impulse(real) and doublet(imaginary) responses each related by a shift of 90 degrees by the Hilbert transform to provide the complimentary point of view, whereas the view with respect to the frequency domain affords the projection onto the respective planes related as well by the Hilbert transform in the form of coincident(real) and quadrature(imaginary) responses. 
The significance is that one discovers that amplitude and phase are simply two different viewpoints of the same event. And as should become apparent, the Domain Map provides a relational diagram of the various perspectives e now have available to us of an acoustical event.

No longer are we limited to the flatland of the frequency response! And we also quickly discover that the time domain exhibits a fundamental causal relationship relative to the frequency domain, meaning that events in the time domain are determinate over much of what is manifest in the frequency response perspective.



And this is true as well with minimum and non-minimum phase environments. We quickly discover that manipulation of the frequency domain is well-behaved only in a minimum phase relationship. 

In the room, we ONLY have control over the direct signal emanating from the speaker (which is, as you will recall, also subject to the superposition of the various drivers, especially if they utilize passive crossovers which lack substantial ability to align the acoustic origins – the ability to make align the two sources, at least in one plane, in the time domain. A feature often referred to incorrectly as ‘time alignment’. But note, we can only align signals within the time domain! We cannot align time!

So, ideally, only the direct signal can be equalized (cognizant that non-coincident drivers and passive crossovers as well as diffractive factors impart their own degree of superposition and comb and polar anomalies. But we are going to selectively ignore this aspect for now! We have bigger fish to fry here!!

In our example, we can only equalize the direct signal (Ld). But this is not the only signal that we experience in the space of the room! We not only have multiple real sources of a signal, but we have the reflected focused specular reflections in the form or first order early arriving reflections which arrive within what is commonly referred to as the Haas interval ( a subset of the Henry Precedence Effect) – a period of time where our ear/brain lacks the acuity to effectively resolve each arriving reflection into separate signals, with the practical impact being that the signal is smeared and the intelligibility suffers. A classic example of thi that everyone has experienced is in a gymnasium or airport, where the distributed PA system exhibits plenty of gain, but which the resultant sound is utterly incomprehensible. This is precisely due to the lack of controlled distribution with many sources arriving within the Haas time interval subverting our ears ability to resolve each signal separately.

We experience the superposition (summation) of the direct and reflected signals arriving at slightly different times in a variety of ways. This affects us in the small acoustical space in the form of reduced intelligibility, localization and image shift, as well as comb filtering and polar lobing anomalies.
And equalization cannot correct for superposed signals! It may adjust the frequency content of the direct signal, and by virtue of the general use of (R)LC filters, vary the phase a bit, but the reflected signal is still offset in time from the direct signal, and the resulting superposition results in comb filtering and polar lobing anomalies simply shifted a bit relative to the changed direct signal. We cannot alleviate the destructive effect. We can only ‘move the problem around’.

Let me digress and explain the fundamental problem with the RTA, essentially a fancy multi-band sound pressure level meter. They do have a few uses, but they are extremely limited with regards to what we want to accomplish. The problem is that we need to be able to resolve the component reflections into the pieces and parts that comprise the superposed/summed signals. We need to be able to deal with each component piece in order to adjust and maximize intelligibility. And the RTA and SPL meter are unable to do so. Thus we simply get a picture of the entire mixed stew. 

So, we have a mess. 

We need tools that allow use to atomistically identify and focus on each component part of the stew – each reflection in terms of amplitude, arrival time and corresponding distance of travel and locations of their points of incidence as they reflect about the space.


That is where the TEF, EASRA, SMAART and other time domain ‘aware’ tools prove invaluable and make possible what the RTA simply cannot do.. And while we can look at this via many perspectives, the most useful view is the Envelope Time Curve (ETC) which shows the direct signal as well as each specular reflection in terms of their amplitude and time of arrival – and from there, depending upon the tools you use, or the additional busy work you must perform based upon the power of the analytical platform employed, we are able to isolate the component reflections and all of their characteristics. 

(Ignoring the use of the cumulative spectral decay response – commonly referred to as the waterfall plot, and the issue of standing waves were the size of the wavelengths are large compared to the room dimensions, we will move ahead to the behavior of acoustical energy where the wavelengths are small relative to the room dimensions – where sound waves behave more like light waves. )

Based upon this, we proceed to define an initial signal delay gap, where we essentially experience the direct signal anechoically, observing the limitations of the Haas interval and allowing us to maximize our ability to localize the signal as well as the intelligibility of the signal. We establish this initial signal delay gap by judiciously and surgically applying absorption.

Thereafter, we want to establish a well-behaved diffuse semi-reverberant sound field from the reflections arriving outside of the of the Hass interval, which imparts a sense of space as opposed to an overly damped dead zone. Here we must identify. This is where we address the focused specular reflections and reduce their amplitude and reduce their focused characteristics by increasing their randomized diffusion in both space and time with diffusers. The energy in the small acoustical space is finite, and we do not want to overly absorb this acoustical energy as its decay in the overwhelming majority of cases is not sufficient to cause concern. In fact, it is an all too frequently encountered problem that the room exhibits a far to absorbent nature! A significant problem specifically acknowledged in the publications of such notables as D’Antonio, Burger, Davis, Janssen, Cary a list far too long to post here. Instead we want to ‘redistribute’ it in a well behaved manner by manipulating it to create a more diffuse lower energy distribution over time.

Well, I guess this enough for now. This is akin to a 25 word overly reductionist simplification of the many profound implications afforded by the tools and perspectives afforded to us. And I dare it should be readily apparent that the flatland perspective limited to the traditional frequency domain is a very limited point of view! In the immortal words of Dorothy “I've a feeling we're not in Kansas anymore!”


Comb filtering and superposition resulting from 2 spaced sources

Wave tank examples of polar lobing anomalies for a real and virtual (reflected) source in phase; out of phase, and two spaced sources (with non-reflective damped lateral and far end surfaces)



Domain map



Heyser spirals with respect to time and frequency


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## JohnM (Apr 11, 2006)

Thanks for taking the time to assemble that Mark, very well covered.



mas said:


> First, the superposition of signals that are out of phase are known as “non-minimum phase”.


That is perhaps over-stated. Superposition only really becomes non-minimum phase when the delay exceeds the period at the frequency being considered, so whilst any system with a time delayed reflection will become non-minimum phase at a high enough frequency it remains minimum phase below that, which is particularly important in small rooms with correspondingly shorter delays.



mas said:


> While mentioning this, we are not going to dwell much in the technical definition which involves the H(s) in what is commonly referred to as the ‘S plane’. In this model, In a minimum phase system, each pole in the transfer function must lie in the negative half of the complex S plane.


That is not correct, a system with right half plane poles would be unstable which, feedback aside , is not what we are generally dealing with. Non-minimum phase systems have right half plane zeroes (or zeroes outside the unit circle if we deal in the z plane, which is probably more approriate for the sampled data systems we generally deal with). Reflecting those zeroes back into the left half plane/inside the unit circle gives us the equivalent minimum phase system with the same magnitude response.



mas said:


> That is where the TEF, EASRA, SMAART and other time domain ‘aware’ tools prove invaluable and make possible what the RTA simply cannot do.


The focus on RTAs is probably a bit unfair, particularly on the support forum for an acoustic measurement tool which is far from an RTA . Most acoustic measurement software is directed at determining the impulse response and using that as the source of subsequent analysis views, that has been the case for a long time now, paticularly since MLS-based systems appeared and right through the transition to the now more commonly encountered logarithmic sweep based systems we find around us today - even TrueRTA offers a log sweep based frequency response measurement!



mas said:


> And equalization cannot correct for superposed signals! It may adjust the frequency content of the direct signal, and by virtue of the general use of (R)LC filters, vary the phase a bit, but the reflected signal is still offset in time from the direct signal, and the resulting superposition results in comb filtering and polar lobing anomalies simply shifted a bit relative to the changed direct signal. We cannot alleviate the destructive effect. We can only ‘move the problem around’.


That is simply untrue. Even if you decide to reserve the term "equaliser" solely for devices applying IIR-based filters to the signal passing through them equalisation is highly effective in countering the room's modal resonances (which are about as superposed as signals get) in the region where the response is largely minimum phase, below a couple of hundred Hz in domestic-sized rooms. Expanding the term "equaliser" to encompass the many FIR-based devices on offer nowadays that apply (partial) inversion of the impulse response (*not* the frequency response!) brings us into the realm of devices that can and do fully correct for the effects of delayed reflections, albeit with the important proviso that the inversion is only valid within a region of the measurement position and that region becomes ever smaller as frequency increases. I'm not personally a fan of such products, but that is no reason to deny their existence. The scope within which they can improve the system response is well explored in various papers.


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## Bob_99 (May 8, 2006)

After reading through this thread, I feel the Shack owes me an Associate Degree in Acoustics as the very minimum. Seriously though, great information and reflects my thoughts that room treatment was always the primary way to approach the acoustics. Thank you Mark for taking the time to post all this information.

Bob


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## Guest (Jul 28, 2008)

Thanks for taking the time to assemble that Mark, very well covered.

Quote:
mas wrote: View Post
First, the superposition of signals that are out of phase are known as “non-minimum phase”.
That is perhaps over-stated. Superposition only really becomes non-minimum phase when the delay exceeds the period at the frequency being considered, so whilst any system with a time delayed reflection will become non-minimum phase at a high enough frequency it remains minimum phase below that, which is particularly important in small rooms with correspondingly shorter delays.

Quote:
mas wrote: View Post
While mentioning this, we are not going to dwell much in the technical definition which involves the H(s) in what is commonly referred to as the ‘S plane’. In this model, In a minimum phase system, each pole in the transfer function must lie in the negative half of the complex S plane.

That is not correct, a system with right half plane poles would be unstable which, feedback aside , is not what we are generally dealing with. Non-minimum phase systems have right half plane zeroes (or zeroes outside the unit circle if we deal in the z plane, which is probably more approriate for the sampled data systems we generally deal with). Reflecting those zeroes back into the left half plane/inside the unit circle gives us the equivalent minimum phase system with the same magnitude response.

Sorry, but it is correct. To quote: "A physically realizable system is one whose impulse response decays with time. In order for this to be true each pole in the transfer function must have a negative real part. Geometrically, this means thst all poles must be located in the left half of the S plane. Given that this is the case, are there similar locations of the zeros of transfer functions? The answer to this is in the negative. Zeros, whether real, imaginary or complex can be located anywhere without any influence on stable system realizability. However, zero locations do play a role with regards to a system's phase response." 

An analysis of the transfer functions is provided on page 380 of Sound System Engineering, and equalization and filter types are explored relative to this throughout Chapter 14, Signal Processing. 

It further goes into great detail regarding the impossibility of equalizing a system where the transfer function has a zero on the right (positive) side of the S plane, which it can have if it is non-minimum phase, it is impossible to cancel the zero by a superimposed pole. Equalizing minimum phase systems results in the simultaneous correction of phase and amplitude. 

"Equalization should only be applied to minimum phase systems and even when it can be employed, it must be employed with care so as not to force any subsystem beyond its range of of linear operation.

Going back almost 19 years, I have several notebooks of notes and papers as well as discussion notes from multiple workshops and seminars with Gene Patronis (PHd, physics professor emeritus, Ga Tech), Don Keele, Don Davis, Bruce Howse and correspondence with Dr. Ahnert regarding exactly this issue. When this group agrees, I will defer to their combined intelligence.



Quote:
mas wrote: View Post
That is where the TEF, EASRA, SMAART and other time domain ‘aware’ tools prove invaluable and make possible what the RTA simply cannot do.

The focus on RTAs is probably a bit unfair, particularly on the support forum for an acoustic measurement tool which is far from an RTA . Most acoustic measurement software is directed at determining the impulse response and using that as the source of subsequent analysis views, that has been the case for a long time now, particularly since MLS-based systems appeared and right through the transition to the now more commonly encountered logarithmic sweep based systems we find around us today - even TrueRTA offers a log sweep based frequency response measurement!

The mention of RTAs and SPL meters is in regard to tools many commonly use to 'tune' their systems. There are a plethora of additional tools available, The fact is that far too many still use such tools to focus on the frequency response rather than the behavior in the time domain. 

And this thread began with a member focused on the notion of the goal of attempting to equalize a system until it was "ruler flat"! Obviously the limitations of addressing fundamental issues in the frequency is not well understood here. As such, a focus on such limitations is in order.

And many of the systems mention specifically use MLS, including TEF (in addition to TDS), EASERA and SMAART! And we could extend that list much further to include additional systems such as Praxis, CLIO, SysID and others! Quite frankly, having the larger systems, I have not bothered to mess with many of the small systems that have become available except in a passing manner. While they may work just fine, they lack some of the flexibility and additional features that render the more powerful tools so valuable and which comprise a complete multifunction analytical platform. This is not a slight toward the small tools, but it is a tacit acknowledgment that they are more limited in their range of features and tools. 

And TDS remains the prime environment of choice on TEF and EASERA, if for no other reason than its dramatic advantages with regards to noise immunity that NO other platform affords. After all, if one is shooting a large auditorium and addressing incident surfaces requiring the access to points greater than 150 feet in the air, especially when there is a very real potential for making far too many repeated trips to isolate and effect solutions, it sure is nice to be able to quickly and reliably resolve the 3space incident points from the measurement location on the ground by use of software and a laser pointer thus limiting trips up scaffolding and scissor jacks to only those absolutely necessary to affect a solution; as well as to be able to listen to music over the same system while generating literally hundreds or even thousands of otherwise mind numbing sweeps over the course of a day! Besides, in the real world where one must contend with construction deadlines and the noise that would render any other test format, including MLS, moot and unable to proceed, how often does one encounter a hall with no noise, concurrent construction or maintenance in progress?? Time is money, and efficiency, as well as the ability to interface with systems within the common usage constraints of the real world - not to mention the ability to preserve one's sanity while being subjected to a barrage of mind numbing sweeps all day long, can certainly reduce the stress of the task

Quote:
mas wrote: View Post
And equalization cannot correct for superposed signals! It may adjust the frequency content of the direct signal, and by virtue of the general use of (R)LC filters, vary the phase a bit, but the reflected signal is still offset in time from the direct signal, and the resulting superposition results in comb filtering and polar lobing anomalies simply shifted a bit relative to the changed direct signal. We cannot alleviate the destructive effect. We can only ‘move the problem around’.

That is simply untrue. Even if you decide to reserve the term "equaliser" solely for devices applying IIR-based filters to the signal passing through them equalisation is highly effective in countering the room's modal resonances (which are about as superposed as signals get) in the region where the response is largely minimum phase, below a couple of hundred Hz in domestic-sized rooms. Expanding the term "equaliser" to encompass the many FIR-based devices on offer nowadays that apply (partial) inversion of the impulse response (not the frequency response!) brings us into the realm of devices that can and do fully correct for the effects of delayed reflections, albeit with the important proviso that the inversion is only valid within a region of the measurement position and that region becomes ever smaller as frequency increases. I'm not personally a fan of such products, but that is no reason to deny their existence. The scope within which they can improve the system response is well explored in various papers.

Untrue? Really!?

If one had bothered to read the complete post, a specific exclusion of room modes and wavelengths large compared to room dimensions was explicitly stated. 

And again, the equalization of systems with IIR or FIR filters are _limited to minimum phase regions_. And such minimum phase regions, however limited, as well as the precise parametric EQ settings, are quickly identifiable using the TEF - _as was duly noted.
_
Nor was DSP referenced here, as to my knowledge no one is employing it with their consumer parametric or third octave EQ unit! As I attempted to respond to the query of the forum member who asked a question regarding the use of a parametric equalizer for such uses.

Rather the primary purpose was to present a simplified methodology for using the ETC to address the specular reflections that dominate the room for wavelengths that are small relative to the room dimensions...AS WAS SPECIFICALLY REQUESTED IN THE HT ACOUSTICS THREAD AND BY A PREVIOUS POSTER. In fact, despite the fact that_ none_ of what I presented in this post was aimed at addressing room modes, much of the objections focus on that which was specifically excluded from the presentation.


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## atledreier (Mar 2, 2007)

I actually understood some of that! 

What I don't understand is this;

If a program like REW show us the frequency response of the system before the room can have an impact (before the reflections mess it up), how can we se an impact from acoustical treatments?

I see some change with and without bass traps and 1st ref absorbtion, but much more in the waterfall plots, as expected.

If I understand it correctly the frequency response is the 'direct' sound, and the waterfall is frequency response as time passes. Is that a fair approximation?


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## JohnM (Apr 11, 2006)

Mark,

Your statement was


> In a minimum phase system, each pole in the transfer function must lie in the negative half of the complex S plane.


Pole location determines system stability ("physical realisability" if you prefer, though this is not strictly accurate as it is exceedingly simple to physically realise an unstable system complete with its right half plane poles, just introduce positive feedback -> hence the comment regarding feedback in my previous post). Both minimum phase and non-minimum phase (stable) systems have their poles in the left half plane, what distinguishes the two is the locations of their zeroes. Systems _only_ become non-minimum phase if they have zeroes in the right half plane. I'm not trying to turn this into an argument, or to criticise you, merely correcting a statement that could mislead others reading the thread. 



> And this thread began with a member focused on the notion of the goal of attempting to equalize a system until it was "ruler flat"! Obviously the limitations of addressing fundamental issues in the frequency is not well understood here. As such, a focus on such limitations is in order.


That is rather unfair on the original poster. He did title the thread as being a question about theory, spoke about the impossibility of making the response flat and specifically asked what the effect of altering the frequency response might be on the impulse response and the time domain.



> And TDS remains the prime environment of choice on TEF and EASERA, if for no other reason than its dramatic advantages with regards to noise immunity that NO other platform affords.


That's not actually correct. TDS certainly has very good noise immunity, and for equivalent sweep length to sequence length can be as much as 10dB better than using MLS, but logarithmic sweeps also have excellent noise rejection, total rejection of harmonic distortion and do not have the extreme sensitivity to clock jitter and replay/record clock matching that demands dedicated hardware to make reliable TDS measurements. None of that is intended as a criticism of the packages you listed, whose analysis capabilities are indeed superb, but merely an objective comment on the measuring techniques. TDS, MLS and an array of other measurement methods are reviewed at length in this paper: Transfer Function Measurement Using Sweeps

Regarding equalisation and what you refer to as "DSP", we are perhaps divided by our common language and using the same terms to mean different things. Equalisation, as I use the term, is the application of filters to a system - IIR, FIR or both. Most products nowadays implement that filtering in the digital domain on a DSP device, even products performing basic parametric EQ, and "DSP" in perhaps the sense you are applying the term is becoming commonplace thanks to the efforts of companies like Audyssey and Lyngdorf (formerly Tact).

You seem to have taken my comments somewhat personally, and I apologise if I have caused you offense, I assure you none was intended. My aim is only to ensure our forum members get the best information we can provide and minimise the risk of misinterpretation. It is rare for someone of your knowledge and experience to take the time to contribute and it is very much welcomed, I hope you will continue to inform.

Regards,

John


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## JohnM (Apr 11, 2006)

atledreier said:


> If a program like REW show us the frequency response of the system before the room can have an impact (before the reflections mess it up), how can we se an impact from acoustical treatments?
> 
> I see some change with and without bass traps and 1st ref absorbtion, but much more in the waterfall plots, as expected.
> 
> If I understand it correctly the frequency response is the 'direct' sound, and the waterfall is frequency response as time passes. Is that a fair approximation?


The frequency response that REW shows, using the default window settings, includes the contributions of the room - of necessity, since at low frequencies in particular that is our interest. How much of the room's contribution is included in the frequency response is controlled by the setting of the Impulse Response window, as the window is narrowed so more of the reflections are excluded - the trade-off is that the narrower the window, the lower the frequency resolution.

A very useful plot when looking at the effects of absorption is the Energy-Time curve (ETC). This is derived from the impulse response, peaks in the ETC after the initial peak are caused by reflections from the room's surfaces, the time delay betwen the main peak and any particular later peak correspond to the additional distance the sound has travelled in bouncing off whatever surface caused it. You can measure that on the graph, use Ctrl+right click and drag the mouse to see the delta shown as both time and equivalent distance. Looking at the result your absorber panels have on the ETC shows you how effective they are and whether they have been placed on the correct surfaces.


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## atledreier (Mar 2, 2007)

Yes, John, I'm aware of the ETC graph and it's usefulness.  And I understand about the window now. I realized I never quite 'got' that before. Thanks!


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## Wayne A. Pflughaupt (Apr 13, 2006)

mas said:


> Then others who expressed confusion over the limitations I had quickly mentioned stated that others were quite happy with using EQ. As I was concerned with limitations regarding its use for the general use in altering room response other than in the limited situations originally stated (LF modal peaks and minimum phase regions) I am not sure what their point was (as the link did not work for me).


Sorry 'bout that. The link is fixed now, but here it is:

http://www.hometheatershack.com/forums/rew-forum/6691-my-first-rew-please-chime.html

"The point," admittedly subtle, was that the Member in the thread linked was very pleased with the equalization he enacted, despite the declarations of any number of "nay-sayers" that such a thing can't be accomplished. I've been happy with mine, too.



> I then provided several links which address some of the legitimate uses and the very limitations to the use of EQ to which I asserted and was then told that acoustic references from pro-audio sources, because they are considered from 'pro' sources, are somehow invalid in 'amateur' acoustics - a distinction which is curious to me as within the limits of similar atmospheric conditions, large acoustical space and small acoustical space acoustics are the same - be it for pro _or_ amateurs!! A very curious distinction!


I said or inferred no such thing. You might want to  read my post again, perhaps a little more slowly.

Regards,
Wayne


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## thewire (Jun 28, 2007)

Spatial temporal averaging using a RTA is more effecient than using a microphone mulitiplexer. It's also easier.


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## Hipper (Jun 5, 2008)

> How many possess a sufficient understanding of modern acoustical models such that they can delineate the limitations of equalization?


Er. 

I stuck a digital eqaulizer in my system, played around with it and now have a much improved listening experience.

That's all I really know.



It's not all I'd like to know and I have read a bit on acoustics. I think in fact that I have, accidentally maybe, addressed some of the other, non-frequency, problems mentioned by using room treatment and listening near field.

It's nice to know all the above technicalities but is it really nescessary. I think not and shouldn't be condemned for it!


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## terry j (Jul 31, 2006)

seems we here 'fall between two stools'...and can get criticized whatever we do.

We at least are attempting to take steps to do something to correct the most egregious errors in our systems that ALL who have systems are subject to, and about which most dyed in the wool audiophiles have either no inkling of or sneer that 'mere equalizers' can have any positive benefit.

We are at least taking some responsibilty for our systems and not relying on a new 'gold plated silver interconnect with special connectors' to 'fix our sound'. (usual hifi response..buy a new component)

Yet, to some, we are also not going far enough and displaying woeful ignorance...looks like we can't win. And the proof of our woeful ignorance??? in depth obscure mathematical and latest research in the field, which to us mere 'joe blows' looks like it requires one to be fully and actively employed in the profession to be even aware of, let alone able to comprehend and use the data.

Not all of us live, work and breathe stereo (tho maybe we would like to!!), and simply do the best we can in the time available to us.

Seems we can't win.

Do I have anything positive to contribute?? maybe, atm I think so but I guess I'll soon find out heh heh.

I use a deqx in my system, which interestingly enough seems to address at least some of the issues outlined earlier, ie steep linear phase slopes between the drivers and the actual correction of the drivers and their phase relationships themselves, as well as time alignment. So, at least some sort of serious attempt has been made to start with the best speaker platform we can.

After all that, traditionally as a last step we would do peq of the room to provide a 'flat response'. This seems to be the main point of criticism in this thread, and yes (at least to my limited understanding) I can see the point. Nonetheless, even tho it is not 'optimal' and incomplete, I can assure you I would rather listen to it corrected than not.

Recently however a change was made in the methodology of the room correction. Instead of measuring the bass driver (either near field or as anechoically as possible) and then 'correcting the driver' prior to the room correction, instead the bass driver was measured from the listening position for the correction, and so had the room response contained within the bass drivers response. By then correcting the combined room and bass driver (this step is where the dsp also corrects for phase and group delay) an attempt is made to also correct the room induced phase anomolies as well as the bass driver. No doubt technically I have a lot of that wrong (dumb monkey understanding of the complexities, so no correspondence will be entered in to as dumb monkeys can't 'correspond' heh heh) but as I understand it there is now much more 'reading and correction' of time isuues as well as amplitude issues.

$64 question?? A surprisingly big, big improvement. From an FR perspective the result 'is the same' (ie flat or whatever you want), gee big difference in the sound and coherence of the result.

And it's quite easy to see the extra work required to get the result, when measuring and correcting the driver alone it may take say, 'x' seconds for the computation. Doing it the above way, dunno, twenty 'x' seconds or whatever. chugging, chugging, chugging away doing the computation. That must be a result of all the complicated etxra time/phase info contained in the signal.

Dunno if what I've described even comes close to meeting some of the objections outlined here, but I can assure you that without going to uni and immersing myself in the minutiae (undoubtedly important tho it is) I've reached about the limit of what this little black duck is capable of...and I feel I've done more than the overwhelming 'vast majority' of the audiophiles in audiophile land have done.

So some of the comments (or the way they were expressed) do come across as a tad insulting or condescending, esp given what seems to be the degree of expertise needed behind them. I mean I could just as easily get on my high horse and throw terms and concepts from my background and point to the latest research in my field and hence criticise you for not understanding them. Why should you understand or be aware of them?

Same here, not all of us have the luxury of being up to date.


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