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The Birth of Sound Field Management

By Dr. Floyd Toole

But simple rectangular rooms are not everywhere. We needed more flexibility, and besides, my new listening room was not a simple rectangle, and again I was not enthusiastic about filling it with bass traps. This was another case of (personal) necessity mothering an invention. My talented employees Todd Welti and Allan Devantier went to work, and came up with something we call Sound Field Management (SFM). (Welti and Devantier, J. Audio Eng. Soc., vol. 54, 2006). This is much more complicated, involving transfer-function measurements from each subwoofer location to each specified seating location. An optimization algorithm then chews away at the data and comes up with specifications for the signal processing necessary for the signals being delivered to each of the (usually) four subs. Amplitude, delay and one parametric filter will be specified. The goal is to minimize the bass variations among the specified seats so that equalization, if it is necessary, will be equally effective for all listeners. It works superbly. My room is shown in Figure 13.18 and another in Figure 13.19. There are no bass booms. Kick drums are "tight", and it works for all five seats, some of which are in “unconventional” locations – it is a media/family room, not a dedicated theater, but with a 10 ft screen and seven identical loudspeakers it is mighty good. Below is a summary illustration emphasizing the fact that although less power is delivered by the subwoofers, an average of about 10 dB higher sound level is achieved in the room. Some of the subs are loafing, and could be reduced in size and power. Remember, in any normal situation turning up the volume by 10 dB uses 10-times more power. The subs can be flexibly located, although there are preferred locations, and they need not be identical. This is truly room mode control, removing destructive cancellations and liberating more good bass for more listeners. There is no global equalization in the curves shown.

 Room and SFM

Figure 13.18 Multi-Seat Measurements with SFM Optimization

Again, were the room modes eliminated? Yes, but only when the audio system is running, and only for the sounds radiated by it. Now, though, it has been done for several seats, and in practice one finds that the bass is comparably good in the intervening spaces - put on a loop of kick drum and bass guitar and walk around the room. The bass response is tight and very consistent, and the normal sense of a "room" has disappeared. It is not inexpensive, and it is not universally available (pity), but it is a very effective and visually unobtrusive solution.

Editorial Note about Sound Field Management (SFM)
SFM is NOT simple EQ or room correction. I apologize if my article lead you to that conclusion. SFM, or Sound Field Management, is indeed a reference to a very specific Harman proprietary system to reduce seat-to-seat variations. Part of it is the usage of multiple subs. Todd Welti and Dr. Sean Olive's team developed an algorithm that further reduces seat-to-seat variation, which is part of their PC-based “ARCOS” characterization, EQ and processing solution. During the characterization phase, each subwoofer is measured independently at each seat. Then an optimization program determines the signal processing for each one independently only for the purpose of minimizing seat-to-seat variations when they all are running simultaneously. Only after that is completed does the system calculate the coefficients for global EQ, which is really only possible once one has reduced the seat-to-seat variations in response. Harman actually has a patent on it and their tech papers and patent prove that it does fix seat to seat consistency. Again it actually measures delay, phase, level, slopes, etc and computes best possible settings for each sub to get them to integrate optimally across all seats. Once you do that, then you can apply Global EQ to flatten bumps.

multi-sub allows for a more efficient, less unsightly option for low frequency control in small rooms.

Would bass traps have delivered a similarly good result? Of course they would have have yielded a greatly improved bass response. It would just have been a very different looking room. If the physical volume and visual aesthetics of bass traps are not a problem, they work. It is straightforward physics with decades of experience behind it. If you are building a studio or performance space, they are the only solution. But for sound reproducing rooms, now we have some alternatives. Low-frequency absorption is always a good thing, and if you can find ways to incorporate it, do so. It will make whatever else you do work even better.

Conclusion

choicesWhatever solution(s) you choose, it is likely that some bass equalization will improve things. Just don't automatically extend the equalization above a few hundred Hz, because that is where the loudspeaker itself takes over and if you have good loudspeakers you might risk making them worse.

Make your choices, it is a free world.

Summing up, there are several means at our disposal to improve bass in small rooms. Depending on your taste, budget and physical circumstances, any one or combination of these options may help. Low-frequency absorption is always beneficial, and enough of it may be all that is necessary. However, not every situation is compatible with the visual and physical compromises. In those cases, technology may be able to provide satisfaction.

I am pleased to have been a contributor to these optional solutions – motivated by pure, selfish, necessity :)

 

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Recent Forum Posts:

andy_c posts on April 08, 2016 15:23
Floyd Toole, post: 1124354, member: 73854
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.
nathan_h posts on April 04, 2016 17:42
More than enough.

I suspect the caveat is: Smoothing beyond a certain point can be misleading, hence the notion of high-resolution.
mtrycrafts posts on April 04, 2016 17:28
Floyd Toole, post: 1126466, member: 73854
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 posts on April 04, 2016 12:11
Floyd Toole, post: 1126466, member: 73854
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.
Floyd Toole posts on April 04, 2016 11:20
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.
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