Linkwitz Orions beaten by Behringer.... what!!?

That's a familiar looking curve ;) . . . what it doesn't explain is what is happening in our recording/reproduction chain that makes it necessary to roll off speakers like that to make them "sound right".

My limited understanding of psychoacoustics from folks like Bob Katz and JJ:

"The psychoacoustics that JJ and his fellows advocate have to do with real psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up. Other issues regarding the Schroeder frequency of the room. And thirdly the directivity of the loudspeakers as frequency increases."

Read more: Psychoacoustically-Correct room correction - Home Theater Forum and Systems - HomeTheaterShack.com
 
I do not mean to dwell on "expert opinion" as much as the point that objective measures are superior.
I suspect that we all (at least here) give more weight to measured response and distortion than we do to "somebody's opinion". But when I come home from a rehearsal or performance and put on a recording and it sounds "too bright" and "not real" what am I to do with the "measures flat" system?
 
"The psychoacoustics that JJ and his fellows advocate have to do with real psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up.
Yes but . . . in the concert hall the reverberant field is rolled off relative to "direct" sound because of air absorption over the far greater path length. It's actually enough to make several dB difference, and helps us "locate" the band in the hall. Common microphone placements also capture that difference, so it is "cooked in" in the recordings. Our typically "acoustically small" listening rooms do not have path lengths long enough to support that effect (to any appreciable extent).

So what is happening when we roll off both the direct signal and the already rolled off reverberant signal from the recording? Why is that necessary? Is it because the close-to-the-band microphone placement simply picks up more "direct" highs than we would normally hear further back in the concert hall? Is it because our listening rooms often have an upward tilt in reflectivity that has to be compensated for (even more than the common upward tilt in absorbancy)?
 
My limited understanding of psychoacoustics from folks like Bob Katz and JJ:

"The psychoacoustics that JJ and his fellows advocate have to do with real psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up. Other issues regarding the Schroeder frequency of the room. And thirdly the directivity of the loudspeakers as frequency increases."

Read more: Psychoacoustically-Correct room correction - Home Theater Forum and Systems - HomeTheaterShack.com

Thanks for that link...you made my day!
 
Pano, Mitch curves are "Audiolensed", very easy to do ;)
Chris

Chris, not sure if you are kidding... In my case, it was hard to do: The Importance of Timbre in Sound Reproduction Systems - Blogs - Computer Audiophile

Thanks for that link...you made my day!

Cheers! Maybe you will enjoy Bob Katz's agony of defeat and thrill of victory :)

"Let me start by saying that the correction that I have gotten is the best sounding room correction I've EVER heard, analog OR digital, in my 43 years of professional listening! Which means it is now one of the best-sounding stereo systems I've ever heard!"

Audiolense 4.6 and JRiver MC18---Summary of my testing and debugging so far (long, detailed post) - Audiolense User Forum | Google Groups

and

Summary of my testing and debugging so far (part two: correction procedure, measured response) 1/29/13 - Audiolense User Forum | Google Groups
 
"The psychoacoustics that JJ and his fellows advocate have to do with real psychoacoustic issues regarding the lesser importance of reflected sound in the ear's sensitivity as the frequency goes up. Other issues regarding the Schroeder frequency of the room. And thirdly the directivity of the loudspeakers as frequency increases."

I do not believe that our sensitivity to reflections decreases with frequency. In fact my understanding is that it does the opposite. It actually peaks at about 2 kHz, falling slowly above there and below there down to the lower frequencies where, in a small room, we would be completely incapable of detecting a reflection simply because the period of the tone has expanded beyond the delay time of the reflection.

There is a very good natural reason why our ears have a heighten sensitivity between 500 and 5000 Hz - this is the region above which the head is becoming small compared to a wavelength (below 500 Hz the head is small and the ear signals are highly correlated - localization capability is decreasing) and below which the frequencies are too high to yield good detection in the cochlea. (Detection of very high frequency signals is very ambiguous.)
 
It does seem . . . peculiar . . . that the asserted unimportance of "constant directivity" between 100 and 1000 Hz does seem associated with the (in)ability of the product to accomplish it. This especially so since "baffle step" is something that all of us are able easily to hear and "objectively" measure, and it is commonly "corrected" for (really that should be in double or triple quotes since where it exists it cannot actually be "corrected") . . .

Maybe its simply a rational design objective to the directivity control. And I did not say to 1000 Hz, I said below ABOUT 500 Hz. (See post above)

To me "baffle step", which you seem to believe is so audible, is a farce. If one measures the speakers in-situ and designs the crossover based on this in-situ measurement then there is no "baffle step" to "correct for". BS (and that acronym is appropriate) is just a simple way of describing the diffraction from a solid object that is accounted for when the speaker is measured in this object. Of course if you compare one speaker without "correction" to another "with correction" (weather it needs it or not) they will sound different. In what way does that prove the validity of the concept?
 
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To me "baffle step", which you seem to believe is so audible, is a farce. If one measures the speakers in-situ and designs the crossover based on this in-situ measurement then there is no "baffle step" to "correct for". BS (and that acronym is appropriate) is just a simple way of describing the diffraction from a solid object that is accounted for when the speaker is measured in this object. Of course if you compare one speaker without "correction" to another "with correction" (weather it needs it or not) they will sound different. In what way does that prove the validity of the concept?

Earl, I find this comment rather odd. What you are effectively saying is that the crossover needs to be designed for the specific room and speaker placement. That's great if you can do it, but not so great if you can't. When you sell a kit what conditions are the crossovers optimized for? speakers against the wall? speakers out from the wall?

Baffle step is a real phenomenon, a designer needs to decide how they will deal with it. Yes the crossover can be designed to take care of it without a specific "baffle step" circuit added, but this is just semantics. If a speaker has a narrow baffle and is out from the wall and does not have some form of baffle step compensation built into the crossover then it will almost certainly sound unbalanced.

As a designer one needs to decide just how much or how little baffle step compensation needs to be put into the design considering the intended usage of the speakers, obviously the ideal is where it is completely optimized for the room and placement that they will be used in, but for a more general design some decision has to be made, often this decision is to use around 3db.

I myself when I did my mtm's started off with no baffle step compensation, in the end I re-did the crossover with quite a lot of baffle step compensation (though there is no traditional BS circuit).

You will see from the attached graph that the gated response is quite different, the revised crossover sounds much better than the original (original is the blue trace). The tradeoff was the drop in sensitivity. I was originally going to do line level BS compensation which would be a better option if the speakers were moved from room to room, but I decided I liked the result I got with the crossover implementation so have left it as is. These speakers would now no doubt sound unbalanced if I put them up against the wall.

So looking at the two graphs, am I wrong to say that the second (black trace) has baffle step compensation? and that it is a farce to say that the speaker without it would be audibly unbalanced?

Tony.
 

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Somo body please define ACCURATE.

When designing a speaker any number of design objectives can be specified and when the speaker is completed it can be measured against those objectives and a level of accuracy in meeting those objectives can be defined by looking at the deviation from the reference the objectives define. But with a souind field produced in your room, how do you define accuracy? There is no reference. You most likely don't even have any idea of what the recordign engineer though it should sound like, let alone what the actualy perfromance sounded like. In your room it either sound the way you like it or it doesn't. There is no measure of accuracy because there is no reference to compare it to.
True. I like to try a whole variety of music. If generally only a few types of music sound emotionally moving to me, then it is an indicator that something is wrong. I also like to involve people that nit pick. Sometimes I learn valuable experience from these people.

In aircraft design, we like to use what is called a Copper Harper rating chart for each mission and maneuver. I think a similar rating scheme can be used in audio evaluation. There is also what we call an Engineering Pilot whom needs to understand what a pilot is talking about when they feel something is not so good in the way an aircraft handles and relate it with technical the technology terms. Audio development has yet to accomplish that link to integrate into the development life cycle of products.

Theoretically, if you compare the wavelet of the recorded music and a recorded system playback, you can see how the sound is altered. The less it is altered, the more accurate the system is.
 
B&K response - see Figure 5: http://www.bksv.com/doc/17-197.pdf

As an ex recording/mixing engineer for 10 years, this was/still commonly used in control rooms where at least one set of monitors was tuned/eq'd to this response.

For example, mastering engineer, Bob Katz, uses a fine tuned version of the B&K curve (no pun intended), where the response is flat to 1 kHz, and using 1 kHz as the hinge point, plot a straight line to -6 dB at 20 kHz.

I use this on my old school 3-way box with a 15" woof and mid/high horn/compression drivers. The measured 1/6 oct response looks like this at the listening position some 9 1/2 feet away:

stereofr_zps24b06dac.jpg


I find the tonal balance to be just about right for any program material. Assuming that the rooms decay time has been sorted with bass traps, absorbers, and diffusers as required.

In my experiments with AS, I find that the closer together the speakers match in FR, across the audible range at the listening position, the better the AS. This assumes that the speaker setup follows the same guidelines as how it was mixed (i.e. equilateral triangle per attached guidelines).

I also found that response to 20 kHz, albeit -6 db down also made a difference in AS. i.e. it took me 3 different HF compression drivers to find one that actually could do this without being -15 dB down at 20 kHz at the listening position.

Anyway, lots of fun :)

Cheers, Mitch
This is very interesting. I assume your drop is because you may find the sound more fatiguing without the slope. When we do power line conductive noise measurements on equipment, I always see a noise level rise when the ground line is connected (Class I) compared without the ground line (Class 2). This basically tells me the circuit and layout is less optimum. The result is the sound is harsher. Generally, people will sought to adjust the speakers because it seems the easiest thing to do. To get to the source of the problem, one really needs to get into every part of the audio playback system.
 
Though not truly "accurate", the best way that I've found was to make my own binaural, and non binaural recording as a point of comparison.

Binaural on headphones represents *more* accurate, then sample the non-binaural on loudspeakers.

It's still utterly subjective however, but I live in the real world and don't require absolute precision.
I feel that Binaural on headphones as well in terms of spacial presentation. I personally feel that the headphones have more potential in accomplishing the best fidelity as well.
 
Constant directivity (CD) does this. It has flat power and flat response along any axis. No direct radiating tweeter does this, so adding a second one might seem to help, except that it has the same problem as the first one so it really doesn't as John said.

I have always thought that SL claim that CD was desirable - except that his speakers aren't CD.
Equalized full range driver @ 0 and 30 degrees.

An externally hosted image should be here but it was not working when we last tested it.
:p
Red is SPL, green is phase
 
I do not see the widening at LF as an issue as long as it is not narrowing above say 500 Hz as any direct radiating speaker does. CD is the goal, but CD to ever lower frequencies is basically impossible unless it is fairly wide directivity. Below some frequency the need for CD goes away and clearly in the modal region it is meaningless.

Ok, yes but he is referring to power response being important to AS.
That means having just as much acoustic energy at 5kHz as 100Hz.
So if I have a perfect omni......theoretically, (i know they don't exist), it will not need BSC. A box speaker will need BSC so we are taking 6db of acoustic energy away from mid to high frequencies. That's 6db less illuminating the room.
Uneven power response.
 
B&K response - see Figure 5: http://www.bksv.com/doc/17-197.pdf

As an ex recording/mixing engineer for 10 years, this was/still commonly used in control rooms where at least one set of monitors was tuned/eq'd to this response.

For example, mastering engineer, Bob Katz, uses a fine tuned version of the B&K curve (no pun intended), where the response is flat to 1 kHz, and using 1 kHz as the hinge point, plot a straight line to -6 dB at 20 kHz.

I use this on my old school 3-way box with a 15" woof and mid/high horn/compression drivers. The measured 1/6 oct response looks like this at the listening position some 9 1/2 feet away:



I find the tonal balance to be just about right for any program material. Assuming that the rooms decay time has been sorted with bass traps, absorbers, and diffusers as required.

In my experiments with AS, I find that the closer together the speakers match in FR, across the audible range at the listening position, the better the AS. This assumes that the speaker setup follows the same guidelines as how it was mixed (i.e. equilateral triangle per attached guidelines).

I also found that response to 20 kHz, albeit -6 db down also made a difference in AS. i.e. it took me 3 different HF compression drivers to find one that actually could do this without being -15 dB down at 20 kHz at the listening position.

Anyway, lots of fun :)

Cheers, Mitch

I have to agree with all of this since all of my mixing rooms have this curve thanks to Bob Hodas advice, and his calibration skill. My own personal hometheaters and multichannel music rooms also have this same curve.
 
I feel that Binaural on headphones as well in terms of spacial presentation. I personally feel that the headphones have more potential in accomplishing the best fidelity as well.

unfortunately, it would be too easy. We do not hear only with the ears, but with the whole body. Take this basic physics out of the equation, and the real experience illusion will vanish... Can you feel the music? ;)
 
unfortunately, it would be too easy. We do not hear only with the ears, but with the whole body. Take this basic physics out of the equation, and the real experience illusion will vanish... Can you feel the music? ;)
This is an interesting question. A few months ago, I went to experience live performance and a really expensive audio system in the same room. During one live performance, I could feel my chest resonate with very specific notes of the cello. I never experience this with any audio system. Aside from that, using earphone/headphone that are properly designed can provide a more realistic spacial presentation. If the performance is behind, you will hear it behind you. Since the earphones are so close to the ears, it is easier to produce the realistic SPL dynamics. Additionally, there is no interaural effect you get using speakers, you get not room influence. I have recorded from at my ears and listened to playback through earphones, that is what convinced me.
 
And I have a strong bias for good imaging and great dynamics.

I think it's a good thing you acknowledge it is a bias/preference. I put most weight on clarity, timbre and a lack of coloration. In my opinion a higher-, more or less constant directivity speaker performs better on those aspects as well. But like A_twinkle remarked earlier, well designed speakers with lower directivity can sound very good as well. Currently I very much enjoy the Olympus speakers (see sig.) again, but I also very much enjoy listening to a pair of DIY CBT's I've got (based on a cheap and actually quite lousy wideband driver), or even conventional box-speakers.

btw. I just started reading Thinking, Fast and Slow. I'm at chapter 4 right now and I love it so far!