History of Multi-Sub & Sound Field Management (SFM) for Small Room Acoustics

F

Floyd Toole

Acoustician and Wine Connoisseur
Goliath,

If you read my latest JAES paper "The Measurement and Calibration of Sound Reproducing Systems" in the July/Aug issue, 2015 you will understand my perspective on "room correction". It is a free download, so AES membership is not a requirement: go to www.aes.org, click on "publications" click on "open access" type "Toole" and download 30 pages.

It is not as straightforward as many would think. Two ears and a brain process sound very differently from an omni mic and an analyzer.
There are two exceptions:
1. one is at low frequencies, where EQ is definitely useful, within the constraints that I mention in my article.
2. is to equalize the anechoic on-axis frequency response of the loudspeaker.
 
G

Goliath

Full Audioholic
Goliath,

If you read my latest JAES paper "The Measurement and Calibration of Sound Reproducing Systems" in the July/Aug issue, 2015 you will understand my perspective on "room correction". It is a free download, so AES membership is not a requirement: go to www.aes.org, click on "publications" click on "open access" type "Toole" and download 30 pages.

It is not as straightforward as many would think. Two ears and a brain process sound very differently from an omni mic and an analyzer.
There are two exceptions:
1. one is at low frequencies, where EQ is definitely useful, within the constraints that I mention in my article.
2. is to equalize the anechoic on-axis frequency response of the loudspeaker.
Hi Floyd,

I should have been more specific. I was specifically referring to low frequencies only regarding the MiniDsp Dirac live.

Haven't read your latest JAES paper but will do. Thanks for letting me know.
 
F

Floyd Toole

Acoustician and Wine Connoisseur
Goliath,

Good! Because doing "room correction" to hit a target curve above about 200-300 Hz introduces a significant risk of making a good loudspeaker less good. Great self control is required not to want a smooth room curve :)
 
A

awdio

Audioholic Intern
I am curious as to what processor you were using in 1988 as part of your 7 channel home theater?
 
F

Floyd Toole

Acoustician and Wine Connoisseur
awdio,

I installed seven channels before it was popular - because I simply knew that it would be better than five, and enormously better than stereo. My first surround processor was a Shure HTS. It had decorrelated L & R surround channels, an excellent idea that was copied by THX when they came along. I drove the rear channels in parallel with the sides in the beginning. I intended to add a delay device between the sides and the rears, but soon the necessary solution appeared and I had a Lexicon CP-1 driving all 7 channels, allowing me to custom design acoustical spatial settings for stereo recordings. I learned a lot. Dr. David Griesinger, the designer, did a great job on that processor and several that followed .
 
nathan_h

nathan_h

Audioholic
Thanks for the link to the AES paper. It raises again the question in my mind about why none of the consumer room correction systems allow a user to adjust the system response below just 200hz.
 
TheWarrior

TheWarrior

Audioholic Ninja
Because the room itself and arrangement of speakers to LP influence the bass modes and standing waves present in every room.

No way for a simple mic to understand what is going on in an unknown room with unknown speakers.
 
nathan_h

nathan_h

Audioholic
Because the room itself and arrangement of speakers to LP influence the bass modes and standing waves present in every room.

No way for a simple mic to understand what is going on in an unknown room with unknown speakers.
If this is in response to me, I guess I asked my question poorly. I will try again.

Most or all of the consumer grade and many professional level room EQ systems try to attack the complete audio spectrum.

But all of the research indicates that the real gains are to be found only below the Schroeder frequency which is the approximate transition between

1. reverberant room behavior above (Toole's research indicates correction filters are not ideal) and
2. discrete room modes below (Toole's research indicates taming modes with EQ is useful).

Given that Toole's findings are not at variance with a lot of other research, once would think that commercial offerings would stop conflating the two.
 
F

Floyd Toole

Acoustician and Wine Connoisseur
nathan_h This is only my guess, but I think they want to give the customer the feeling that they are the ones responsible for everything you hear. You and Warrior are correct in that the right kind of EQ can do useful things at low frequencies, but if that is all the algorithm does it raises the question in the customer's mind- if that is good, wouldn't full bandwidth EQ be better? Maybe if one has truly awful loudspeakers, but they are getting harder to find these days. If you know what the loudspeakers are doing - i.e. have full sphere anechoic data on the loudspeakers - then a target curve at middle and high frequencies makes some sense, but only then. This is done in JBL Synthesis, but they use their own loudspeakers that are fully documented. It is just a check to see if the speakers are driven correctly and to ensure that there is nothing stupid about the room acoustics - the "room EQ" and SFM portions are targeting lower frequencies.

It is interesting that most of the providers of full bandwidth room EQ schemes do not make loudspeakers. If they did, they might have a bit more respect for what really good ones can do without room EQ :) In teaching my courses at CEDIA I find that most of the attendees have tried broadband room EQ, but very few continue to use it.

I will repeat, ad nauseum, my assertion that we don't need full bandwidth room EQ, but we do need user friendly bass and treble controls to restore good spectral balance from recordings that are so variable. Broadband EQ aims to be a fixed tone control, and that is not what is needed. The target curve, whatever it is, is substantially determined by the recording you are listening to, and that needs to be adjustable in real time.
 
nathan_h

nathan_h

Audioholic
Thanks for the additional explanation.

nathan_h This is only my guess, but I think they want to give the customer the feeling that they are the ones responsible for everything you hear. You and Warrior are correct in that the right kind of EQ can do useful things at low frequencies, but if that is all the algorithm does it raises the question in the customer's mind- if that is good, wouldn't full bandwidth EQ be better?
Here is hoping that someone realizes there is a market for making it possible for "smarter" consumers to choose to just EQ lower end.

Right now, many of us do this via having a PEQ on the subwoofers (which are positioned intelligently to produce the most uniform experience across the main seating positions), to tame the most egregious room modes.

But that typically leaves the important 100hz to 200hz -ish range untouched, and that range could benefit from some EQ in most domestic rooms, if I am reading the literature correctly.

(Heck, this limited range for EQ could just be a user-selectable option and not even the default. Just having the option would be a huge win.)

Maybe if one has truly awful loudspeakers, but they are getting harder to find these days. If you know what the loudspeakers are doing - i.e. have full sphere anechoic data on the loudspeakers - then a target curve at middle and high frequencies makes some sense, but only then. This is done in JBL Synthesis, but they use their own loudspeakers that are fully documented. It is just a check to see if the speakers are driven correctly and to ensure that there is nothing stupid about the room acoustics - the "room EQ" and SFM portions are targeting lower frequencies.
Great demonstration of what can be done. Seems like some of that could make its way into consumer level gear.

...In teaching my courses at CEDIA I find that most of the attendees have tried broadband room EQ, but very few continue to use it.
That's very telling.

I will repeat, ad nauseum, my assertion that we don't need full bandwidth room EQ, but we do need user friendly bass and treble controls to restore good spectral balance from recordings that are so variable. Broadband EQ aims to be a fixed tone control, and that is not what is needed. The target curve, whatever it is, is substantially determined by the recording you are listening to, and that needs to be adjustable in real time.
I feel this way more with music than with movies. Music seems to be all over the place, compared with how movies are mastered. That's not to say that movies are as consistent as one might want, but there seems to be far less variation for films.
 
F

Floyd Toole

Acoustician and Wine Connoisseur
nathan_h Yes movies are more consistent, largely because there are standards for setting up the dubbing stages and cinemas. However, if you read my AES paper it is clear (a) that they are not consistently calibrated, sometimes not at all and, (b) the target curve (the X-curve) is not what is needed for home theater - or even cinemas, as it turns out :-(
 
mtrycrafts

mtrycrafts

Seriously, I have no life.
...


Here is hoping that someone realizes there is a market for making it possible for "smarter" consumers to choose to just EQ lower end.

...
Well, someone came up with an extremely flexible one that you can apply to any frequency you need to correct, all in the digital domain.
http://www.sweetwater.com/store/detail/FBQ1000?adpos=1o1&creative=55686342961&device=c&matchtype=&network=g&gclid=CjwKEAjw_oK4BRDym-SDq-aczicSJAC7UVRt-aih3tWfj7pCo84nrvhvQbN4HSgIBzwGkFNVlQYv8hoC59_w_wcB

Very powerful at selection the frequency and Q.
 
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nathan_h

nathan_h

Audioholic
Good point. I guess the miniDSP line of products also has some DIY options that could fit the bill. I have used those for sub EQ before.

I was sort of hoping someone like Audyssey or Dirac or Yamaha would allow the option to only EQ below the Schroeder frequency since that is the place where EQ does more good than harm.

But your point is good. The tools exist for a DIY approach.

Now I forget, does REW allow one to determine the Schroeder frequency? Then one could avoid using any correction filters it recommends that are above that point.

Well, someone came up with an extremely flexible on that you an apply to any frequency you need to correct, all in the digital domain.
http://www.sweetwater.com/store/detail/FBQ1000?adpos=1o1&creative=55686342961&device=c&matchtype=&network=g&gclid=CjwKEAjw_oK4BRDym-SDq-aczicSJAC7UVRt-aih3tWfj7pCo84nrvhvQbN4HSgIBzwGkFNVlQYv8hoC59_w_wcB

Very powerful at selection the frequency and Q.
 
F

Floyd Toole

Acoustician and Wine Connoisseur
nathan_h and others - if you are able to make high-resolution steady-state LF measurements, you will find the resonances. It is very likely - based on my lifetime of experiences - that there are only a few, probably only one or two problem resonances (peaks) and a bunch of dips. You cannot equalize the narrow dips because they are non-minimum-phase destructive-interference phenomena. So you don't want an automated equalizer that might try to fill them - there are some "smart" algorithms that are programed to ignore them, but you have to know that. All you really need is a manually programmable parametric EQ that allows you to zero in on the problem peaks, adjust the frequency, the Q (bandwidth) and dial the peaks down to a moderate level. That's it.

If you want to look at waterfalls, this will be confirmed. But really, if you look at the back curve in a waterfall and see a peak, you will see ringing. No peak, no ringing. Waterfalls do not add information, they confirm what can be inferred from the room curve in a photogenic way. This assumes that the parameters of the waterfall are set right.

As for equalizing above the transition frequency (the Schroeder calculation is designed for large reverberant performance spaces and yields the wrong frequency in small rooms), it is a good thing - up to a point. Above the subwoofer frequency one enters the domain of adjacent boundary effects (Chapter 12 in my book), which includes what some people call the Allison effect. These can be equalized, and often show up as dips, but they are broad, low-Q dips when you do the necessary spatial average over the listening area. So, knowledgeable equalization is useful up to a few hundred Hz if used with restraint. I would advise reducing the Q of the filters at higher frequencies though.

So, equalizing resonances for the benefit of the sweet spot needs one mic location, while identifying the adjacent boundary issues requires multiple mic locations. Only with multi-sub solutions will the EQ reliably work for multiple listeners.
 
nathan_h

nathan_h

Audioholic
nathan_h and others - if you are able to make high-resolution steady-state LF measurements, you will find the resonances. It is very likely - based on my lifetime of experiences - that there are only a few, probably only one or two problem resonances (peaks) and a bunch of dips. You cannot equalize the narrow dips because they are non-minimum-phase destructive-interference phenomena. So you don't want an automated equalizer that might try to fill them - there are some "smart" algorithms that are programed to ignore them, but you have to know that. All you really need is a manually programmable parametric EQ that allows you to zero in on the problem peaks, adjust the frequency, the Q (bandwidth) and dial the peaks down to a moderate level. That's it.
I'm going to assume that you prefer not to talk about specific commercial solutions, which is fine/understandable.

For other folks in this thread, I will comment that I have found the DSPeaker Anti Mode products to be just about he only "automatic" tool that tames key peaks in the subwoofer realm, without doing harm or getting overzealous.

http://www.dspeaker.com/en/products/anti-mode-8033.shtml

But the MiniDSP, if one is comfortable with translating REW measurements into filter parameters, seems to achieve very similar results. Pros: You can visibility into exactly what the EQ is doing with the MiniDSP because YOU are doing it. Cons: If you blindly follow the REW recommendations about filter parameters, you may not get what you want.

I have one of each and depending on what I am trying to achieve, I use one or the other. If it is simply taming a sub's in room response, the DSPeaker Anti Mode is quick and easy. The MiniDSP, on the other hand, lets me do other things, like take a stereo signal and do bass management with it (pulling the low frequencies to a separate output) and EQ a little, a lot, etc, whatever I want.

As for equalizing above the transition frequency (the Schroeder calculation is designed for large reverberant performance spaces and yields the wrong frequency in small rooms), it is a good thing - up to a point. Above the subwoofer frequency one enters the domain of adjacent boundary effects (Chapter 12 in my book), which includes what some people call the Allison effect. These can be equalized, and often show up as dips, but they are broad, low-Q dips when you do the necessary spatial average over the listening area. So, knowledgeable equalization is useful up to a few hundred Hz if used with restraint. I would advise reducing the Q of the filters at higher frequencies though.
When I first read your book, I'll bet I completely did not grok this. I need to go revisit it. Thanks for the reminder/pointer.

So, equalizing resonances for the benefit of the sweet spot needs one mic location, while identifying the adjacent boundary issues requires multiple mic locations. Only with multi-sub solutions will the EQ reliably work for multiple listeners.
I have finally got the 'multi subs' religion. I first heard / read about the idea nearly 10 years ago. I bought a second sub, put it on the front wall with the first and said "Well, that doesn't seem to have made any difference!" and sold that second sub.

Some time after that I read the summary of Welti's research/articles (there was a nice PPT making the rounds that summarized the AES paper, I believe) and understood my error. It wasn't just about dropping a second sub into the room at random but doing so in a deliberate fashion. I got a second sub, again, and placed it in the opposite corner of my rectangular room from the first sub, and that made a ton of difference. A little EQ to tame peaks and one is golden.

--

Thanks for sharing your learnings. I'm off to do my homework.
 
mtrycrafts

mtrycrafts

Seriously, I have no life.
nathan_h and others - if you are able to make high-resolution steady-state LF measurements, you will find the resonances. ...
Would a test disc with 1 Hz increment per track considered hi-rez?
 
nathan_h

nathan_h

Audioholic
More than enough.

I suspect the caveat is: Smoothing beyond a certain point can be misleading, hence the notion of high-resolution.
 
A

andy_c

Audioholic
If you are measuring the steady state room curve (i.e. amplitude but no phase) for each sub separately at a listening location, the summation cannot work predictably. I may be wrong, but I sense that some people, and maybe some algorithms are doing this. Maybe somebody knows . . .
There have been some interesting new developments in this area with the Room EQ Wizard (REW) measurement freeware. Traditionally, it could make measurements that preserve the relative delays between different subs at the measurement position only when using an analog mic and a loopback timing reference. But in the last few years, USB mics have become the measurement mic of choice because of increased simplicity and decreased total cost. A loopback timing reference is not possible with a USB mic. Until recently, when not using a loopback, REW would shift each computed impulse response so that its peak is at t=0. This effectively removed the relative delay information between sub measurements at a given listening position, rendering attempts to form their complex (phasor) summation at that position in other software invalid. However, as of 5.15 beta 3 and later, it added an "acoustic timing reference", a chirp starting at 5 kHz from a chosen full-range speaker, whose impulse response peak is shifted to t=0. This forms a timing reference for the other impulse responses, restoring the relative delay information that would otherwise be lost. This feature restores the ability of third-party software to correctly compute the phasor sum of multiple sub measurements at a given listening position. More information can be found here.
 
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