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Dr. Toole's - A Rational Approach To Calibrations

A good speaker is one that was tuned to be flat at 1m under anechoic conditions. It has smooth directivity, meaning the off-axis response is spectrally correct. This in turn means that reflections are also spectrally correct.

If we take such a speaker and put it in a room and measure from 1m away, we observe a flat response above transition. We then measure 2m away, and we observe that the frequency response starts to fall. Measure 3m further away, and the response falls even further, and so on. This is where the Harman curve comes from - it is the natural response of a speaker which is anechoically flat in the nearfield, listened to in the farfield. He backs this up by citing preference studies (including some of his own) showing that listeners strongly prefer these speakers.
Hi, don't mean to contradict a good overall post.

I think very few manufacturers would tune tune a speaker flat at a 1m distance, other than maybe for small bookshelves.
In the anechoic chamber, or whatever their available quasi-anechoic test setup is, the speaker should be in the acoustic far-field which is typically at least 3X the largest speaker dimension. (The acoustic far-field is where phase summations have settled into stability, and the speaker obeys inverse square law SPL falloff with distance.. Doesn't have anything to do with room's near-field and far-field definitions.)

The measurement taken at that far-field distance is then simply SPL adjusted to a 1m distance. So 1m is much more about normalizing response sensitivity specifications, than where the measurement was taken.

In a room, the degradation in the response curve as measurement distance is increased from 1m, is predominantly from increased reflections. Can't really see how the Harman curve has anything to do with that.
 
Hi, don't mean to contradict a good overall post.

I think very few manufacturers would tune tune a speaker flat at a 1m distance, other than maybe for small bookshelves.
In the anechoic chamber, or whatever their available quasi-anechoic test setup is, the speaker should be in the acoustic far-field which is typically at least 3X the largest speaker dimension. (The acoustic far-field is where phase summations have settled into stability, and the speaker obeys inverse square law SPL falloff with distance.. Doesn't have anything to do with room's near-field and far-field definitions.)

The measurement taken at that far-field distance is then simply SPL adjusted to a 1m distance. So 1m is much more about normalizing response sensitivity specifications, than where the measurement was taken.

In a room, the degradation in the response curve as measurement distance is increased from 1m, is predominantly from increased reflections. Can't really see how the Harman curve has anything to do with that.
Depends on the speaker.
 
The original LS50 benefits from EQ at around 700-800 Hz and 2300 Hz. It has excellent dispersion, but a few design shortcomings resulting in bumps in the above mentioned ranges.
 
non-minimum phase response should not be corrected
It can be corrected using a mixed-phase approach using FIR filters, seen in autoEQ systems like Dirac, Trinnov, Neumann's MA-1 based system and (I'm not entirely sure) Genelec's GLM. Manually this can be done in Acourate, Audiolense and (again not sure) BruteFIR and rePhase.

Personally I only use REW to measure, any old PEQ to adjust and have used Neumann's MA-1, which is definitely good, but time-consuming and cumbersome. Once Dirac ART is available for computers, I'll switch to that for my HTPC. But that's getting sidetracked. I only mentioned all this stuff for context.
You wrote about identifying frequency, but you seem to be describing localization (like how would one identify higher frequencies through level?). Pitch perception and processing are different. I put together a list of links on psychoacoustics with some selective quotes and summary here: https://www.audiosciencereview.com/...acoustics-self-education-links-sharing.45583/
Through critical bands, which is a function of energy vs. bandwidth. I couldn't think of another single-word term that was descriptive enough. The entire time I was writing that post I was trying to balance technical vs. colloquial language. I think I did ok but not great.
I'm sorry but, I don't see what this has to do with taking the loudspeaker's anechoic data on/off axis and comparing is to the REW readings taken from the room; Equalize up to 300-400 Hz and if necessary treat the HF irregularities with acoustic products to not degrade the speaker. I have read most of your reply in Dr. Tool's book and know it would apply to calibrating loudspeakers; It's just that Dr. Toole's chapter 13 is trying to save the performance of a "good loudspeaker." All of your post can/should be taken into account, just don't break the speaker.
So you wrote this:
If I'm correct, Dr. Toole is saying that very good loudspeakers should not be equalized above transition 300-400 Hz and doing so could degrade a very good loudspeaker.
You absolutely can EQ. What you can't do is "correct" using standard in-room measured data.
 
It is interesting to see different views on the subject. I would think in my simple mind that it is as easy as you will try to do it many different ways, measure it and then find what measures reasonably good and sounds best to you in your room with your own gear. My hypothesis is that we all have gear and rooms that are really incredibly different, and that our goals/focus could be (probably are) as well. Finding a common denominator between all of that is a really difficult task and it seems that it might be causing a lot of misunderstanding.
 
The way I've always understood it, speakers should only be EQ'ed based on in-room measurements below the room transition frequency. Above it, it should be based on anechoic. The reason being the human brain can separate the direct from the reflected sound more the higher up we go, whereas the in-room measurement microphone shows all the direct and reflected sound combined.
 
As Curvature noted in his post, REW, for example, allows you to set the gating window. This means that you can measure in room and get an anechoic-equivalent response above a frequency, and with a resolution, determined by the gating and the distance to the nearest reflective surfaces. You can go ahead and correct that anechoic-equivalent response if an anechoic or Klippel measurement of your speakers is not available. Practically, this means that you can EQ for room correction below 300 Hz or so, and above 1Khz or so based on pseudo-anechoic. Because of poor, frequency dependent resolution caused by the gating window, only do low-Q adjustment until 3-4KHz. If you find a high Q issue above that, measure in several mic positions so see if it goes away or shifts; if it does, leave it alone.

My understanding is that Toole is saying not to EQ based on steady-state response above room-controlled frequencies, and, well, yes.
 
And what if you have 13 speakers and 5 subs?

Hi, I don't begin to understand what relevance that might have to either how a speaker should be calibrated, or even a speaker within a room ?????
 
Hi, I don't begin to understand what relevance that might have to either how a speaker should be calibrated, or even a speaker within a room ?????
Well then, all bets are off. And try to think of 100 different rooms in combination with 1000 different speakers.
 
I have read and thought about many of the subjects Dr. Toole presents in his book, but unfortunately this chapter is about how to avoid degrading very good speakers by using automatic room correction. Dr, Toole does cover the subject of personal preference based on individual hearing extensively else where in his book. However, he does compare a microphone to ears and a brain.

Dr. Toole does say "Consequently, the processors perform equalization corrections including non-minimum-phase acoustical interference irregularities, in order to hit the specified target curve." Could be semantics.

I do agree with Dr. Toole that individuals should tune their loudspeakers to their liking. The reason I post this information is to help those, interested, in not degrading their loudspeakers
No, you are missing the point. I'll explain a little. This is not a semantic issue.

When you "measure" a room using something like REW you are making specific choices about:
  • Bandwidth
  • Level
  • Window
  • FFT length
  • Averaging
  • Other complex calculations
These affect the FR and other graphs. Based on what shows up in the graph, you can then make equalization adjustments using some specific kind of filter. Most manual PEQ uses IIR filters. There are consequences to all this stuff.

Automatic room measurement and adjustment removes your ability to control some or all of these things.

I bring this up because, when you see line on an FR graph, unless you have some sense of what is being calculated, you will not know what physical events are being represented and how.

Next, hearing works, very roughly, in two ways: time sensitivity and level sensitivity, each being different per frequency. These separate into three (but really four) groups:
  1. Below about 1.5kHz, frequencies are identified through the ear/brain phase locking to the physical waveform.
  2. Above around 1.5kHz, this identification takes place through level.
  3. Around 1.5kHz, both of the above mechanisms are active. Ability to resolve and localize sound is reduced.
  4. Below around 100Hz, tactile perception is involved in bass perception. This involves skin pressure receptors as well as proprioception and other sensory pathways I do not understand very well.
For our purposes, only 1 (phase locking) and 2 (level) are important. We do not instantly become aware of particular frequency being played. Phase locking takes time, tending to be longer the lower in frequency the sound, and faster the higher (before it stops working once the frequency gets too high). This means for lower frequencies, what we hear is an average of sound energy over a short period of time. Once level sensitivity takes over at higher frequencies, the ear is mostly responding to the SPL of direct sound (so instantaneous SPL, with a small amount of time required for recovery before responding to the next wave). All of this is contained in what is called the precedence effect, the law of the first wavefront, or the Haas effect.

So, when looking at an FR chart, below the transition frequency, standard IIR filter EQ correction to a target curve will be effective because a standard steady state response will reflect what we hear (although the physical location of room modes complicates this). Around the transition frequency, IIR filter EQ will become ineffective because the peaks and dips shown in a steady state graph are nonminimum phase. Above the transition frequency, the steady state graph no longer reflects what we hear, and correction using IIR filter EQ to a specific target is wrong. Above, the transition frequency, different windows must be used to understand the FR of direct sound vs. the FR of reflected sound.

I apologize for not explaining everything, but like I said, it's all in the book.
I Think I can see where you had taken issue with my post on equalization; I apologize for any misunderstanding on my part; but my position isn't to never use equalization above 400 Hz and I thought I cleared it up in my reply to your first reply “how to avoid degrading very good speakers by using automatic room correction.” It's all in the attachments written by Dr, Toole.

The second time (equalization) was when I said nothing above 400 Hz; I was talking about how I was going to take my first try at calibrating my Focal Aria 936 without using "Automatic Room Calibration" of course if it didn't work out I would have to use other means.

Thank you for your patience, I hope this puts us on the same page, :)

"So you wrote this".... Yes

 
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Speakers that are highly directional by design to minimize room interaction don’t have the same issues with off axis response in appropriate rooms

I can't see how that has anything to do with the point i raised, that speakers are not anechoically measured and tuned to a one 1 meter distance...(again unless very small).

If anything, for a highly directional speaker, it is even more important to measure in the acoustic far-field.
because a highly directional speaker inevitably requires a wave guide of some sorts, which means it's a multi-way needing further driver sections.....
which moves the onset of the acoustic far field father out in distance, simply due to c2c driver section distances.
 
I Think I can see where you had taken issue with my post on equalization; I apologize for any misunderstanding on my part; but my position isn't to never use equalization above 400 Hz and I thought I cleared it up in my reply to you first “how to avoid degrading very good speakers by using automatic room correction.” It's all in the attachments written by Dr, Toole.

I think you have raised very valid points, or at least points in line with how I read Toole's book.
I do not see him as a proponent of automatic room correction at all.
 
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