Thoughts on "eq'ing" a room

[mas]

When will we move past the notion of equalizing room anomalies?

Probably when no improvements are noticeable...

But you'd have to have your head in the sand to believe that no amount of EQ can compensate for room acoustics. Correct the problems? No, but every sound engineer in the world is going to run different settings on their FOH graphic EQ for every different venue. You can't just set the EQ for the system and expect it to work everywhere.

It's a realm where you move from the ideal world into the real world.

GOOD sound engineers aren't trying to use EQ for correcting room anomalies! They use it for what it does well and that is to adjust the direct sound. But GOOD engineers know this.

Good sound engineers usually don't get to change the venue they're in and must work within the means of the equipment to arrive at as good as sound as possible. But just to stress the point - I purposely never said anything about "correcting" for room anomalies. I specically said "compensate" because you're working within a huge system of compromises. The only reason I made the analogy is to point that there can be subjective improvements. And just for the sake of argument, I'll remind ourselves that at home we're typically only aiming for good sound in a single listening position anyway.

In various venues that change each night, you can't fix the venue. The most you can do is to orient your sound system for optimal coverage minimizing overlap and reflections to the degree you have control (which is often minimal). So you use the tools which are appropriate and you are philosophical about the variables over which you have no control and are simply glad you get to be somewhere else tomorrow! You pull out your SMAART and EQ the minimum phase realms of the response and leave it. You don't worry about room induced anomalies. You develop a thick skin and a wry sense of humor.

And then you thank your lucky stars that you are not sentenced to the Hell known as 'Monitor World' where the soprano wants to sound like Tony Bennet and holds you accountable for the shortcoming!

[DrWho]

Think of a gong or a bell. Rather than a distinct thunk, we experience an extended ringing which is the resonance of the signal that persists in time.

...

Equalization ONLY effects the gain of the initial/direct signal, it does NOT effect the rate at which the signal decays!

Strike a bell softer and it takes less time for the resonance (decaying at the same rate) to drop below the noise floor.

[DrWho]

Btw, I've heard it mentioned that when you sit at the peak of a standing wave, that it becomes a minimum phase situation...

[DrWho]

Equalizing

the room

By Bob McCarthy

http://www.prosoundweb.com/studyhall/lastudyhall/eq.php

But on the issue of equalizing the room a division arose.

All parties agreed that speaker/speaker interaction was somewhat

equalizable. The critical disagreement was over the extent the speaker/room

interaction could be compensated by equalization.

The TDS camp advocated that speaker/room interaction was not at

all equalizable and therefore, the measurement system should screen

out the speaker/room interaction, leaving only the equalizable portion

of the speaker system on the analyzer screen. Then the inverse of

the response is applied via the equalizer and that was as far as

one should go.

The TDS system was designed to screen out the frequency response

effects of reflections from its measurements via a sine frequency

sweep and delayed tracking filter mechanism, thereby displaying

a simulated anechoic response. The measurements are able to clearly

show the speaker/speaker interaction of a cluster and provide useful

data for optimization.

Such an approach can be effective in the mid and upper frequency

ranges where the frequency resolution can remain high even with

fast sweeps but it is less effective at low frequencies. Low frequencies

have such long periods that it is impossible to get high-resolution

data without taking long time records, thereby allowing the room

into the measurement.

For example, the arena scoreboard reflection is 150 ms later than

the direct signal. At 10 kHz, the peaks and dips from this reflection

are spaced 1/1500 of an octave apart. At 30 Hz, they will be only

1/3 octave apart. Thus the scoreboard is in the distant field relative

to the tweeters, and applying equalization to counter its effects

will be totally impractical.

An architectural solution such as a curtain would be effective.

But for the subwoofers, the scoreboard is a near-field boundary

and will yield to filters much more practically than the 50 tons

of absorptive material required to suppress it acoustically.

Many years ago, the FFT camp boldly stated that the echoes in the

room could be suppressed through equalization. Unfortunately, these

statements were made in absolute terms without qualifying parameters,

leaving the impression that the FFT advocates thought it was desirable

or practical to remove all of the effects of reverberation in a

space through equalization.

Practical Implications

It is indeed possible and practical to suppress some of the effects

of speaker/room interaction. If this was not possible, it would

be standard practice to equalize your rig in the shop, put a steel

case around the EQ rack and hit the road. The practical side of

this is that we must be realistic about what is attainable and what

are the best means of getting there.

[mas]

A fascinating selective editing job.

Note a paragraph that was conveniently omitted:

"While it can be proven from a theoretical standpoint that the frequency
response effects of a single echo can be fully compensated for,
that does not mean it is practical or desirable. The suppression
can only be accomplished if the relative level of the echo does
not equal or exceed that of the direct and that no other special
circumstances arise that cause excess delay. (Excess delay causes
a non-minimum phase aberration and is outside the scope
of this article.)"

Oh, and this statement is absolutely incorrect:

"Such an approach can be effective in the mid and upper frequency
ranges where the frequency resolution can remain high even with
fast sweeps but it is less effective at low frequencies. Low frequencies
have such long periods that it is impossible to get high-resolution
data without taking long time records, thereby allowing the room
into the measurement."

These measurements are done all the time! He focuses on sweep rates and completely neglects the tracking filter width adjustments! He speaks like someone who has only read about TDS. And by the way, TEF also includes FFT.

Now, if you care to reference other articles by such folks as Sam Berkow, Pat Brown or David Janssen or many others you might come a bit closer.

As has been mentioned, minimum phase regions CAN be equalized!!! And this is precisely what the TEF PEQ software module provides - and identification of the minimum phase regions and the settings to accomplish precise equalization using a parametric equalizer.

And you may discover why SMAART (A time domain tool developed in large measure by Sam Berkow and marketed by EAW)is the standard for live sound
use.The simple reason is that SMAART has as a fundamental display the
comparison of the resolved signals (and reflections), a comparison of
their phase and displays the minimum phase regions that can be EQ'd.
And then you put the EQ away. It is oriented toward live sound.

BTW, since we are using quotes to make a point, try this one:

http://www.prosoundweb.com/live/labbest/pc/pc.php

Reply
posted by Tom Danley on June 14, 2002

I would say that generally
all filter shapes have an exactly corresponding phase shift (related
to the shape) with an exception being some DSP and all pass filters.

If
one has a loudspeaker who's response rolls off or has a bump, these
undesired "filters" also impose a particular phase shift,
correcting that with an EQ also corrects the phase as most electroacoustic
"things" are "minimum phase" (meaning there
is a specific phase change for a change in amplitude).

If
one wanted to EQ a cancellation notch caused by two different acoustic
paths (one delayed RE the other), one finds that this "filter"
does not have the normal phase change with amplitude and if the
two paths have equal amplitude, the notch is infinitely deep as
well. Using an EQ to put a big peak there does fill in the sides
some but while having little effect on the response, does add a
large amount of phase shift at the EQ point. Since the phase shift
is not corrected by the phase error of the problem filter, it is
left in the signal.

Bottom line is that what you hear so
far as phase usually depends a lot more on what kind of problem
you are trying to fix rather than on what type of circuit was used
to produce the filter. Non minimum phase things like multiple acoustic
paths and so on cannot be fixed with EQ.

One does have to
wonder that since loudspeakers generally do not preserve phase,
how much does that obscure the electronic phase effects.**

Cheers,
Tom
Danley

**Note: This should answer a few who seek to rely on DSP!

[mas]

Think of a gong or a bell. Rather than a distinct thunk, we experience an extended ringing which is the resonance of the signal that persists in time.

...

Equalization ONLY effects the gain of the initial/direct signal, it does NOT effect the rate at which the signal decays!

Strike a bell softer and it takes less time for the resonance (decaying at the same rate) to drop below the noise floor.

...And the rate at which the signal decays does not vary.

If you turn off the system, the intensity of the mode decreases further. And it too is cheaper and more effective than an EQ. And the rate at which the signal decays, reinforced by the room dimensions, still does not vary.

[]

[mikebse2a3]

Step back and remember what a standing wave/room mode is! It is a resonance that is predominate at a particular frequency(ies) determined by the dimensions of the room where the wavelengths are large relative to the room. While we talk of a peak at a particular frequency, the peak which is predominately created by the summing of multiple resonances, the real problem is the persistence of the signal in time.

Think of a gong or a bell. Rather than a distinct thunk, we experience an extended ringing which is the resonance of the signal that persists in time.

Now, the problems with room modes is not so much the gain peak, it is the muddiness, the boominess, created by the persistence in time. So while we may refer to the mode at a particular frequency, the REAL problem is the resonance's persistence in the time domain. And as this resonance is created by the dimensions of the room reinforcing the frequency, it is in effect a tuned pipe, much like an organ pipe.

IMHO based on my experience is that PEQ adjustments when used properly can help within limits in this situation at least for a primary listening position for the "muddiness and boominess created by the persistenance in time". I will say Room Treatment is the proper and ideal way to deal with these issues but when that isn't an option one or two well placed PEQ Filters can have real audible benefits.

In most rooms based on my experience I would only recommend this be attempted below 200Hz and with the aid of some measuring equipment to reach the best adjustment possible. EQing can cause more problems than it helps if not used with some knowledge/experience and some measurement ability.

There are definitly limits on the ability for success with the EQ method and Room Treatment is always the ideal but when compromises have to be made IMHO PEQ adjustments in the lower frequencies can have real audible benefits at a dedicated listening position(by the very nature of the recording methods most often used in two/three channel audio the ideal listening position requires you be in a very limited area)!

So it may be a bit confusing to realize that while we refer to the problem of a room mode being a particular frequency, it is not so much the gain of the frequency, it is the persistence in time at a particular frequency that is the problem.

Equalization ONLY effects the gain of the initial/direct signal, it does NOT effect the rate at which the signal decays!

I believe reducing the gain will result in the initial sound decaying sooner into the noise floor of the room.

Imagine striking the bell hard: The result of the SPL of the Acoustical Output and the Time it takes to decay into the noise floor of the room at the listening position can be measured.

Now Imagine striking the bell softer: The resulting SPL of the Acoustical Output and the Time it takes to decay into the noise floor of the room at the listening position will be less.

So if the room conditions result in certain frequencies(our bell for instance) being able to be developed by the loudspeaker toward the speaker/room combination's maximum efficiency then bringing down the efficiency at those frequencies(with a goal of a more balanced speaker/room efficiency at frequencies in the "modal region" of the room) with a PEQ Filter can bring about a perception of restored SPL balance and the response will in effect decay into the room's noise floor sooner. Since the SPL level is better balanced in relation to the rest of the frequency spectrum and the decay rate into the noise floor is in closer balance to the rest of the frequency spectrum the resulting perception is the sound becomes less boomy and muddy sounding.

mike tn

Edit: Sorry!

I see some things I mentioned have been discussed since I orginally started my post and gotten distracted before finishing and posting it.

[mas]

Step back and remember what a standing wave/room mode is! It is a resonance that is predominate at a particular frequency(ies) determined by the dimensions of the room where the wavelengths are large relative to the room. While we talk of a peak at a particular frequency, the peak which is predominately created by the summing of multiple resonances, the real problem is the persistence of the signal in time.

Think of a gong or a bell. Rather than a distinct thunk, we experience an extended ringing which is the resonance of the signal that persists in time.

Now, the problems with room modes is not so much the gain peak, it is the muddiness, the boominess, created by the persistence in time. So while we may refer to the mode at a particular frequency, the REAL problem is the resonance's persistence in the time domain. And as this resonance is created by the dimensions of the room reinforcing the frequency, it is in effect a tuned pipe, much like an organ pipe.

IMHO based on my experience is that PEQ adjustments when used properly can help within limits in this situation at least for a primary listening position for the "muddiness and boominess created by the persistenance in time". I will say Room Treatment is the proper and ideal way to deal with these issues but when that isn't an option one or two well placed PEQ Filters can have real audible benefits.

In most rooms based on my experience I would only recommend this be attempted below 200Hz and with the aid of some measuring equipment to reach the best adjustment possible. EQing can cause more problems than it helps if not used with some knowledge/experience and some measurement ability.

There are definitly limits on the ability for success with the EQ method and Room Treatment is always the ideal but when compromises have to be made IMHO PEQ adjustments in the lower frequencies can have real audible benefits at a dedicated listening position(by the very nature of the recording methods most often used in two/three channel audio the ideal listening position requires you be in a very limited area)!

So it may be a bit confusing to realize that while we refer to the problem of a room mode being a particular frequency, it is not so much the gain of the frequency, it is the persistence in time at a particular frequency that is the problem.

Equalization ONLY effects the gain of the initial/direct signal, it does NOT effect the rate at which the signal decays!

I believe reducing the gain will result in the initial sound decaying sooner into the noise floor of the room.

Imagine striking the bell hard: The result of the SPL of the Acoustical Output and the Time it takes to decay into the noise floor of the room at the listening position can be measured.

Now Imagine striking the bell softer: The resulting SPL of the Acoustical Output and the Time it takes to decay into the noise floor of the room at the listening position will be less.

So if the room conditions result in certain frequencies(our bell for instance) being able to be developed by the loudspeaker toward the speaker/room combination's maximum efficiency then bringing down the efficiency at those frequencies(with a goal of a more balanced speaker/room efficiency at frequencies in the "modal region" of the room) with a PEQ Filter can bring about a perception of restored SPL balance and the response will in effect decay into the room's noise floor sooner. Since the SPL level is better balanced in relation to the rest of the frequency spectrum and the decay rate into the noise floor is in closer balance to the rest of the frequency spectrum the resulting perception is the sound becomes less boomy and muddy sounding.

mike tn

Just a few notes to qualify a few issues raising a few concerns...:

The PEQ equalization tool is ONLY for minimimum phase regions, where EQ does work. PEQ is an additional TEF software program (module) that compares the various resolved signals (direct and reflected)measured by the TEF and compares gain and phase in order to determine minimum phase regions that are equalizable and supplies precise settings for a parametric EQ. But this software requires the TEF and the a availability of the source time domain measurements.

And a bell or gong is minimum phase. It is not the result of multiple signals varying in time. It is a single coherent source.

But please be aware that the bell or gong is a mechanical resonance.

Its use as an analogy was mentioned ONLY to illustrate the concept of

resonance - the persistence of a signal in time and not to equate the

behavior of the room and its acoustical energy to that particular

mechanical system. It is not applicable to a room where room anomalies are a result of multiple signals varying in phase.

EQ is not effective for non-minimum phase environments. TEF. SMAART and EASERA are able to identify and distinguish minimum phase and non-minimum phase regions, allowing for the surgical use of EQ/ But even so, this will not resolve room anomalies!

For LF below ~300Hz, room modes and standing waves should optimally be treated with traps. Any minimum phase regions, once identified through analysis, can then be equalized.

Again, EQ is limited in its use to minimum phase regions and is not an appropriate solution to non-minimum phase issues. There are tools that identify the EQ'able regions. But one must consider these regions as the exception, not the rule!

[ZAKO]

Yahn !!! Any body ever tell you guys your a freaking bore....You can niggle a subject to death. I think I,ll turn on my BOSE and enjoy the music..

[Coytee]

I think I,ll turn on my BOSE and enjoy the music..

Please be sure your two speakers are in proper phase with each other, lest they end up sounding more like some Mach Ones instead of Bose cubes!

[]

[mikebse2a3]

Yahn !!! Any body ever tell you guys your a freaking bore....You can niggle a subject to death. I think I,ll turn on my BOSE and enjoy the music..

Any body ever tell you if your board you don't have to read or participate in a thread![:^)]

mike tn

[ZAKO]

mike .... Thats not my point.. You keep scraping over old ground for new diamonds of information and getting just the same old rocks. I know how the Intelligencia of "can you top this" goes, but grinding over the same old information makes you about as boreing as reading wikipedia. Its the same old diet over and over with no new perspective. Its like building a new cathredral to hear the same old achoustics.

[mikebse2a3]

mike .... Thats not my point.. You keep scraping over old ground for new diamonds of information and getting just the same old rocks. I know how the Intelligencia of "can you top this" goes, but grinding over the same old information makes you about as boreing as reading wikipedia. Its the same old diet over and over with no new perspective. Its like buildind a new cathredral to hear the same old achoustics.

I understand your point and feel that way too sometimes Maron because for you and many who have alot of experience and knowledge and have been on the forum for many years this is going to be boring.

The other point I want to make is there are also many who are new to the forum and are looking for knowledge and understanding. It's important to spread good and complete information and to correct misinformation and one of the best ways to do this on the forum is to have input from many people and especially from those with experience and background that are willing to share their knowledge.

So basically a part of the forum seems to me will always have subjects that have been talked to death but for people seeking knowledge and are new to a subject, audio and the forum this is one way they will learn.

I hope to always display tolerance and patience for those who are truely seeking knowledge and understanding and I hope those whom I seek the same from will give that to me!

mike tn[]

[ZAKO]

Sigh !! OOOHHH I guess your right.....I just keep searching for newer enlightnment.

[garyrc]

zuzu:

"..... my Radio Shack SPL meter says that ear damage starts to come into play at 80 dB and worsens as you go higher. Most of my listening is 75 dB and a bit lower."

I seriously doubt that 80 dB is damaging IN MOST MUSIC. All SPL figures (except the extremely high, e.g., 140 dB) which are purportedly damaging must specify duration or they are "free floating" and relatively meaningless. Government standards, at least in the chart reproduced below, don't even list 80 dB, and will permit 90 dB for 8 hours, 100 dB for 2 hours, 115 dB for 15 minutes, etc.

Orchestral music, in which the peaks don't last very long, can be played at a higher SPL than music that tends to be "all loud," such as some Rock. A full orchestra often reaches 100 dB for a few moments at a time, with very brief peaks at 110, 115, 120 dB. Paul Klipsch used to claim that one needs 115 dB "at your ears" to achieve what he characterized as the "blood stirring" levels of a symphony orchestra, and I agree, but that 115 dB is only momentary, even if repeated. If one has a very old fashioned SPL meter with a swinging needle, the real peaks may be as much as 13 dB higher than the needle swing indicates (even on "fast"), because of needle ballistics. But those peaks are by their very nature, brief (if they were closer to continuous, the needle would do a better job of registering them).

So it is usually relatively continuous high SPL that causes hearing loss. Much confusion has been caused by the recent attention to impulsive sounds in the workplace ... I think they are concerned with sounds that are not merely impulsive, but of extraordinarily high SPL -- like heavy metal machinery clanking and banging, gunshots, and the like. Employers must provide protection if the impulsive sounds are 140 dB or greater. Real gunshots are so high in SPL that the movie industry usually fakes them because real shots, close up, are so loud that they drive microphone diaphragms back against the plate and produce tame "pops."

The glorious finale of a Beethoven symphony could, very worst case, be considered continuous by government standards because its maxima repeat at intervals of 1 second or less (but really, only rarely) -- even at that level, it could sail along at 115 dB for 15 minutes, when in reality this ff (occasionally fff?) passage probably fluctuates between 90 and 110 dB in the front rows, with a few ultra brief leading edges at 120 dB.

If the variations in noise level involve maxima at intervals of 1 second or less, it is to be considered continuous.

         TABLE G-16 - PERMISSIBLE NOISE EXPOSURES (1)

______________________________________________________________

|

Duration per day, hours | Sound level dBA slow response

____________________________|_________________________________

|

8...........................| 90

6...........................| 92

4...........................| 95

3...........................| 97

2...........................| 100

1 1/2 ......................| 102

1...........................| 105

1/2 ........................| 110

1/4 or less................| 115

____________________________|________________________________

Footnote(1) When the daily noise exposure is composed of two or

more periods of noise exposure of different levels, their combined

effect should be considered, rather than the individual effect of

each. If the sum of the following fractions: C(1)/T(1) + C(2)/T(2)

C(n)/T(n) exceeds unity, then, the mixed exposure should be

considered to exceed the limit value. Cn indicates the total time of

exposure at a specified noise level, and Tn indicates the total time

of exposure permitted at that level. Exposure to impulsive or impact

noise should not exceed 140 dB peak sound pressure level.