To stir up a hornet's nest...

:D

Can a more HT oriented sub like an ED A2 - 300 be EQd to be more "musical" at the cost of output? You can certainly flatten the FR in a given room with an EQ, and perhaps also correct for distortion, but is that all there is to it? Can the elusive "group delay/transients" be changed? Or is that primarily the result of box/driver combination?

The tricky part is to first define the meaning of "musical". You're not going to be able to change group delay and I'm not sure how you would correct for distortion. You can keep distortion minimized by knowing the limits of the sub and operating under those limits.

In my opinion, if you can define "musical", you can then get close to achieving it through flattening the frequency response.

You can EQ it to sound, in your room, in your opinion, better with music and call that "EQ'd to be more musical". Maybe get 20 people to do that and see what they all did to see if there is some trend.

yes, this is why the svs ultra has a sealed mode

To stir up a hornet's nest...

:D

Can a more HT oriented sub like an ED A2 - 300 be EQd to be more "musical" at the cost of output? You can certainly flatten the FR in a given room with an EQ, and perhaps also correct for distortion, but is that all there is to it? Can the elusive "group delay/transients" be changed? Or is that primarily the result of box/driver combination?

EQ'ing a sub is one of the best things you can do. A properly balanced FR can do wonders to how a sub sounds for music. Even the best subs can sound like garbage with a bloated lower end or null.

I personally think all subs should be EQ'd (or at least any sub that's not in a perfect room, if there is such a thing). One thing that you will gain from a properly EQ'd sub is a better blend with your main speakers. A good blend with you main speakers goes a long way to sounding "more musical".

If you're missing my point: EQ your sub regardless of it's transient or group delay performance. You WILL be pleased with the results. Then you will know if your sub is musical or not and you can improve the sub from there.

I highly recommend a BFD and Room EQ Wizard. ;)

I highly recommend a BFD and Room EQ Wizard.
Yes, ... and bass traps first though.

Once you get a capable sub that can reproduce accurately at sufficient SPL, ....it's all about the room.


Tim

yes, this is why the svs ultra has a sealed mode

So, group delay and transient response are changed by using sealed mode with the PB-13 Ultra?

As usual, there is not a shred of evidence from SVS to back up your supposition. But, maybe you figured out how to make all those Def Techs sound musical. :rolleyes:

yes, this is why the svs ultra has a sealed mode

Does the OP have a eD sub or a SVS?

If the OPs sub is ported he can stuff the port with a sock to make it more "musical", which sometimes is also referred as "tighter". Oh, no there is that subjective word "tigher"

Room treatments are always great but it has to be done right...One or two bass traps may not help at all. Just warning the OP before he has one and thinks it didnt help.

No sub yet, just considering options.

Thanks! :)

EQ'ing a sub is one of the best things you can do. A properly balanced FR can do wonders to how a sub sounds for music. Even the best subs can sound like garbage with a bloated lower end or null.

I personally think all subs should be EQ'd (or at least any sub that's not in a perfect room, if there is such a thing). One thing that you will gain from a properly EQ'd sub is a better blend with your main speakers. A good blend with you main speakers goes a long way to sounding "more musical".

If you're missing my point: EQ your sub regardless of it's transient or group delay performance. You WILL be pleased with the results. Then you will know if your sub is musical or not and you can improve the sub from there.

I highly recommend a BFD and Room EQ Wizard. ;)
You seem to think that EQ is the ideal solution for most all subwoofers. Actually, EQ should only be used to "tweak" the response by a few db *after* everything else has been addressed. First, the sub should be placed properly. You can use Room EQ Wizard to help find this spot. Then the listening position should be optimized with respect to the sub. Again REQ can help. Next, the room itself should be addressed with bass traps.

Once you have the frequency response tuned with sub location, listening position and room treatments, the final response can be tweaked with a few dB of EQ. You shouldn't add more than ~3 dB of boost and you shouldn't cut more than ~6 dB off the peaks.

Boost can add strain to the sub amp and driver. 3 dB of boost means the amp needs to double it's output at that frequency. It also means the speaker must travel further to try to produce more output. Boosting a true null is ineffective anyway. All it does is add more energy to the cancellation.

Cut will reduce the level of the initial wave, which is the wave you *really* want to hear. The initial wave is where the slam and punch are, so you don't want to reduce the level of the initial wave too much. Cutting the intitial wave by 6 db is removing about 2/3 of the energy at that frequency from the initial wave. Sure most of that is added back by the room reflections, but they're delayed and at a lower level than the initial wave.

Also any boosts or cuts apply to the entire room, not just the listening position you are EQ'ing for. Therefore large boosts or cuts will negatively impact other listening positions.

EQ is a tool that can improve the final response. But it should not be the first tool out of the toolbox, and it should be used sparingly.

Craig

You seem to think that EQ is the ideal solution for most all subwoofers. Actually, EQ should only be used to "tweak" the response by a few db *after* everything else has been addressed. First, the sub should be placed properly. You can use Room EQ Wizard to help find this spot. Then the listening position should be optimized with respect to the sub. Again REQ can help. Next, the room itself should be addressed with bass traps.

Once you have the frequency response tuned with sub location, listening position and room treatments, the final response can be tweaked with a few dB of EQ. You shouldn't add more than ~3 dB of boost and you shouldn't cut more than ~6 dB off the peaks.

Boost can add strain to the sub amp and driver. 3 dB of boost means the amp needs to double it's output at that frequency. It also means the speaker must travel further to try to produce more output. Boosting a true null is ineffective anyway. All it does is add more energy to the cancellation.

Cut will reduce the level of the initial wave, which is the wave you *really* want to hear. The initial wave is where the slam and punch are, so you don't want to reduce the level of the initial wave too much. Cutting the intitial wave by 6 db is removing about 2/3 of the energy at that frequency from the initial wave. Sure most of that is added back by the room reflections, but they're delayed and at a lower level than the initial wave.

Also any boosts or cuts apply to the entire room, not just the listening position you are EQ'ing for. Therefore large boosts or cuts will negatively impact other listening positions.

EQ is a tool that can improve the final response. But it should not be the first tool out of the toolbox, and it should be used sparingly.

Craig

Craig,

That is a lot of great information. Thank you.

I only suggested the tools (BFD and REW), I didn't go into any detail on how to use them. It would be up to the original poster to figure that out (hometheatershack BTW ;)).

Not everyone is going to be able to use room treatments in their living spaces. Bass traps and such are most often VERY intrusive to the room when it's not a dedicated music/HT room.

Am I a fan of EQ'ing a sub? YES I AM! :D

IME, the best thing you can do to make your sub better for music.

Get it out of the corner location.

Look at this: http://verkkokauppa.planeetta.net/ep...cts/8033B-0001 and there is a thread going at: http://www.avsforum.com/avs-vb/showthread.php?t=1028464. I ordered one and it will be here on Monday, 6/9.
Bill

Look at this: http://verkkokauppa.planeetta.net/ep...cts/8033B-0001 and there is a thread going at: http://www.avsforum.com/avs-vb/showthread.php?t=1028464. I ordered one and it will be here on Monday, 6/9.
Bill

Top link not working but looking forward to your impressions. :)

Yes, ... and bass traps first though.

Once you get a capable sub that can reproduce accurately at sufficient SPL, ....it's all about the room.


Tim

I second the BFQ2496 with its dual 20 bands of parametric EQ and if you wanted 40 bands you can always loop the BFQ2496 thou I use it for sub bass extension and the sub bass for LFE.1 so I’m using the two channels of the EQ and Aiming for critical flattest smooth frequency response over the sub range.

A frequency sweep should be taken from each LCR and S and then lastly the sub bass because the blending may make for muddy sound, well not really only an excessive peak where it shouldn’t be, could be anywhere in the sub bass range?

To stir up a hornet's nest...

:D

Can a more HT oriented sub like an ED A2 - 300 be EQd to be more "musical" at the cost of output? You can certainly flatten the FR in a given room with an EQ, and perhaps also correct for distortion, but is that all there is to it? Can the elusive "group delay/transients" be changed? Or is that primarily the result of box/driver combination?

no you cant unless you can teach it to sing, or maybe play the piano.... then it would be musical!

can you make a 4th order system sound like a second order system? with filters, of course... tune low and roll that bad boy off, 2nd order seems appropriate, lol... i couldn't tell the difference in tests.

seriously don't use the "m" word, you'll confuse everyone! And the only transient response that matters is the transient time constant it takes me to come back here and slap you guys around for using the "m" word!

:)

Cut will reduce the level of the initial wave, which is the wave you *really* want to hear. The initial wave is where the slam and punch are, so you don't want to reduce the level of the initial wave too much. Cutting the intitial wave by 6 db is removing about 2/3 of the energy at that frequency from the initial wave. Sure most of that is added back by the room reflections, but they're delayed and at a lower level than the initial wave.


I'm not sure I follow you. Are you saying that if I EQ the frequency response flat at my listening position that it really isn't flat because the first wave + reflections hitting me are not equal to what the first wave would feel/sound like if the first wave was at the same level and there were no reflections?

I'm just trying to understand what it is you're saying because I don't understand the logic.

Up until just now - I was under the impression if I EQd the subwoofer flat at a certain listening position than what I was hearing would be a flat response. If I moved my head six inches that might change but now you're saying that if I have made cuts to the response than I am not actually getting a flat response because ???? and the slam and punch would be diminished ??? Considering the size of bass waves and the speed of sound this doesn't make a lot of sense to me but I admit I have a lot to learn so I am anxiously awaiting your response.

if I EQ the frequency response flat at my listening position that it really isn't flat because the first wave + reflections hitting me are not equal

It depends on how the EQ works. If the EQ is just modifies the amplitude response, the attack-phase of a dampened frequency may start slower than before. However, it is possible to model the filter accurate enough to gradually drop the output of the dampened frequency so attack is not affected: the filter will 'wait' for the wave to fill the room and only then starts to dampen the frequency. That way you get the desired attack, but get rid of the room's influence.

The same applies to decay. The EQ can even generate anti-phase sound to make the frequency decay faster. The more accurately the EQ system can model the room response, the better (less affected by the room) the attack and decay times are. A couple of Hz is not enough, you need half a Herz.

Btw, if you want to pedantic, you should re-calibrate the system whenever the room temperature has changed more than a couple of degrees, because the speed of sound and thus the frequency of the standing waves change.

It depends on how the EQ works. If the EQ is just modifies the amplitude response, the attack-phase of a dampened frequency may start slower than before.
I agree except I would change "slower than before", to "at a lower level than before".

However, it is possible to model the filter accurate enough to gradually drop the output of the dampened frequency so attack is not affected: the filter will 'wait' for the wave to fill the room and only then starts to dampen the frequency. That way you get the desired attack, but get rid of the room's influence.

The same applies to decay. The EQ can even generate anti-phase sound to make the frequency decay faster. The more accurately the EQ system can model the room response, the better (less affected by the room) the attack and decay times are. A couple of Hz is not enough, you need half a Herz.
Agreed. This is the difference between a standard graphic or parametric EQ (which uses IIR filters), and systems like Audyssey, Anthem ARC, Trinnov, etc., (which use FIR filters) to correct in the time domain as well as the frequency domain.
____________________________________________________________

I'm not sure I follow you. Are you saying that if I EQ the frequency response flat at my listening position that it really isn't flat because the first wave + reflections hitting me are not equal to what the first wave would feel/sound like if the first wave was at the same level and there were no reflections?
If you use a standard parametric EQ, (SMS-1, BFD, etc.), which only corrects in the frequency domain, you'll end up with "flat frequency response" but a decrease in level of the initial sound wave. Think about how "peaks" occur in a room. The initial wave arrives at the listening position followed by reflected waves. When the reflected waves arrive in-phase, they combine to add "constructive reinforcement" at that frequency. To reduce the constructive reinforcement, the only thing a parametric EQ can do is to reduce the level of the initial wave. The reduced level of the initial wave decreases the amount of gain added by the reflections. You get "flat frequency response", but with a reduced initial wave.

Up until just now - I was under the impression if I EQd the subwoofer flat at a certain listening position than what I was hearing would be a flat response. If I moved my head six inches that might change but now you're saying that if I have made cuts to the response than I am not actually getting a flat response because ????
You'll get a flat "frequency" response...

and the slam and punch would be diminished ???
... but at the expense of power in the initial wave. FFR is generally better than peaked response. However, using excessive cuts to flatten the response could leave you with smooth bass with reduced impact, punch and slam. This is why I suggested cuts of no mare than 6 dB.

It's also why it is suggested to use bass traps to flatten the response in your room. If you reduce the level of the reflections with sound absorption, you reduce the "constructive reinforcement" without affecting the level of the initial wave. Once the room is improved, then EQ can be used to "tweak" the final response with gentle boosts, (no more than 3 dB), and gentle cuts, (no more than 6 dB).

I recently switched from an SMS-1 to an Audyssey based EQ system. I also installed bass traps in the front corners and the ceiling. The Audyssey/Bass Traps system is smoother AND it has more impact than the SMS-1 EQ'd system. However, I must also disclose that I added a second subwoofer, (a 2nd JL Audio F112) to the system at the same time, so this is not a direct "apples to apples" comparison.

Craig

There are many good points there, Craig. Folks frequently ignore what's going on in the time domain because it's a harder concept to grasp than an easily plottable frequency response. In addition to using EQ excessively to correct room responses, things like near-field placement and/or subs located far from the mains will cause all kind of time smearing which cannot all be corrected no matter what you do.

If you use a standard parametric EQ, (SMS-1, BFD, etc.), which only corrects in the frequency domain, you'll end up with "flat frequency response" but a decrease in level of the initial sound wave. Think about how "peaks" occur in a room. The initial wave arrives at the listening position followed by reflected waves. When the reflected waves arrive in-phase, they combine to add "constructive reinforcement" at that frequency. To reduce the constructive reinforcement, the only thing a parametric EQ can do is to reduce the level of the initial wave. The reduced level of the initial wave decreases the amount of gain added by the reflections. You get "flat frequency response", but with a reduced initial wave.


While conceptually this may seem probable, the reality is that we have to remember how we hear we need to look at how different wavelengths interact with the room/space we place them in. Before we get too far, let me not confuse the fact that it is most certainly still preferred to find mechanical means of damping room modes with various acoustic devices, but this is often not practical, and even with this there will be some variation in response with both frequency and location.

The above assertation goest against most of the research published by Harman and many others. At higher frequencies above say 300-500Hz (depending on the room), this is true, but below 100-300Hz the modal energy and the direct radiated power of the subwoofer start to merge in the way we hear. That is not to say a single location frequency response is the only thing that matters, but the reality is that a conventional parametric filter affects both the magnitude and time domain response. In the vast majority of spaces, this effect is complimentary to the response observed in the room at the location of measurement, and standard PEQ will help both the magnitude response AND the time domain response.

It may also help to remember that we tend to be much more sensitive to the decay of sound vs. the rise-time. The more complicated problems all related to methods of employing EQ to improve the sound over multiple listening locations where the response differs with the best subjective results. The "initial wave" talk really only applies to the high frequencies where we will often talk about the direct sound field and differentiate between early and later arriving reflections and how they have different impacts on the sound. In this sense, subjective effects will differ as the reflections range from about 5-30ms after the direct sound arrival.

There are many good points there, Craig. Folks frequently ignore what's going on in the time domain because it's a harder concept to grasp than an easily plottable frequency response. In addition to using EQ excessively to correct room responses, things like near-field placement and/or subs located far from the mains will cause all kind of time smearing which cannot all be corrected no matter what you do.

While ALL effects can't be perfectly corrected, different distances and group delays of different speakers and subwoofers can be largely corrected for or the differences minimized over some area with commonly available delays and filters.

The following was posted a while back by Bossobass, (who unfortunately doesn't post much here anymore). It's a little convoluted because it's a quote of a quote of a reply to a question. The original quote is from Dick Pierce. However, the points made go to the heart of the matter I was discussing above:

As far as my thoughts on using PEQ to 'fix' an in-room response, I'll re-post an excerpt from posts on the subject by Dick Pierce which was forwarded to me by Brucer over at AV123 a long while back and is in line with my thoughts on the subject as well as the tutoring I received about PEQ way back as a bass player in various studio situations...


quote:Tom Ascher <u15310@uicvm.cc.uic.edu> wrote:
> It appears that the primary use of equalizers is to get rid
> of room resonances that result from systems being in spaces
> considerably smaller than auditoriums or theaters where music
> is generally heard.

That might well be the most popular use, but it's the wrong use.
It's a waste of an equalizer and it is garaunteed NOT to do the
job.

> But, the broader issue is that any room can be considered an
> extension of the speaker enclosure and will introduce
> resonances at various frequencies throughout the audible range
> and that it really is necessary to have some way of "voicing"
> a system to a given room.

No, any room CANNOT be considered an extension of the speaker
enclosure. This is because of the inherent delays due to room
resonances put these effects well outside of the realm
associated with enclosure issues.

> Of course, speaker selection and placement, mounting, use of
> room treatment materials are all part of adjusting a sound
> system so it sounds good in a given room. But, there remain
> some problems that really cannot be dealt with short of some
> form of equalization.

Let's look at the falacy of this overall approach. Let's assume,
for the purpose of discussion, that the axial frequency response
and the power response of the speaker is perfect: it is dead
flat from 20 to 20 kHz anechoically. Let's further assume that
we want to maintain that response at the listener's ears. (We
can generalize the assumptions to say that whatever the response
of the speaker is, flat or otherwise, we want to preserve it
by the time it reaches the listeners ears.)

Now, let's play the loudspeaker in the room. Sit 3 meters away
from it. The first thing to reach your ears is the direct sound
from the speaker, completely unaltered by the room. Whatever the
response of the speaker is anechoically, THIS is what reaches
your ears first.

Now, let's say your room has a nasty resonance at some
frequency. In order for that resonance to affect what you hear,
the energy from the loudspeaker has to travel to where that
resonance is formed, it has to excite that resonance and that
resonance has the effect of dramatically extending the
reverberation time at that frequency, and the delayed,
reverberated result now has to travel the remaining distance to
your ear in order for you to perceive it.

And it arrives at your ear, AFTER the unaffected direct sound
made it and stimulated your ears.

Now, put an equalizer in and try to correct that resonance.
Let's say that the resonance cause a 15 dB peak in the response
at 300 Hz (not likely, but let's pretend). So you take your
equalizer and dial in a -15 dB hole in the electrical signal
going to the speaker at 315 Hz...
Now, let's play the loudspeaker in the room. Sit 3 meters away
from it. The first thing to reach your ears is the direct sound
from the speaker, with its new 15 dB hole in the response,
COMPLETELY UNALTERED BY THE ROOM. Whatever the response of the
speaker is anechoically, NOW modified by a 15 dB hole in the
response, THIS is what reaches your ears first.

And at the same time, the energy from the speaker travels about
the room, and STILL excites the room resonance (but at a lower
level) and STILL gets delayed and reverberated (sorry, but the
reverberation time at the resonance is not changed by the amount
of energy you put in it).

So, instead of a perfect speaker affected by a room, you now
have a speaker with a big whopper 15 dB hole in the response
that's somewhat less affected by the room.

A room's acoustical problems are fixed by dealing with the
causes of the room's acoustical problems, not by screwing up the
signal fed into it.

The problem is that MOST of the instrumentation available to
most people for "measuring" room response are completely
inadequate to the task. 1/3 octave real-time analyzers, pink
noise generators and sound level meters, warble tones and all of
it simply lose ALL the important time information that describes
WHY the room behaves the way it does. It hides much f the
information altogether.

You have a room with a nasty side-wall reflection? Propose how
an equalizer eliminates that reflection and it's effects? Bad
floor-ceiling slap echo? How and why would an equalizer fix it?
(Hint: it won't). Reverb time of the room about 1/4 second
EXCEPT at 200 Hz, where it's 1 second? How does something which
operates in the electrical frequency domain correct for energy
storage and delay in the time domain, especially when that agent
is no longer causally connected to the stimulus?

Yes, you're absolutely right:

"It appears that the primary use of equalizers is to get rid
of room resonances that result from systems being in spaces
considerably smaller than auditoriums or theaters where music
is generally heard."

But they are ineffective at doing so.

--
| Dick Pierce

Craig

While ALL effects can't be perfectly corrected, different distances and group delays of different speakers and subwoofers can be largely corrected for or the differences minimized over some area with commonly available delays and filters.

While it's true that the timing of the primary wave can be largely corrected, and even that only around the sweet spot, it is not possible to correct the delays of all of the reflected waves which are originating from all kind of different places in the case of subs in many locations away from the main speakers. As always, there are trade-offs which may have to sometimes be made. Also, the lower the crossover, the smaller is the spectrum of sound affected.

The following was posted a while back by Bossobass, (who unfortunately doesn't post much here anymore). It's a little convoluted because it's a quote of a quote of a reply to a question. The original quote is from Dick Pierce. However, the points made go to the heart of the matter I was discussing above:

Craig

Hi Craig,

The dialog from Dick Pierce is correct, but only above 300-500Hz in most listening rooms, and certainly is not correct below 100-300Hz.

The explanation clearly notes why 1/3rd octave equalizers and full range EQ using simple RTA measurements doesn't work, and fell out of popular use. For the past 20 years we have had measurement methods to examine both time and frequency domain in specific slices through various gating and windowing techniques. In the past 5-10 years, this has become much more readily available and less effective with computer based measurement systems. Exactly as I explained above, the concept Dick is talking about breaks down at lower frequencies. Frequency and wavelength matters. When we look at any source of reflection or impediment to sound propagation, we have to consider its acoustic dimension as it relates to frequency. A 2' x 2' x 6" solid block directly in the path of sound travel will be very effective in stopping or reflecting 10kHz, but will be entirely ineffective at 100Hz.

My TEF measurement system allows me to look at what are called ETC graphs, which show energy vs. time in dB and will clearly show reflections at later bumps in response after the direct or first arrival. Only when distances and arrival times become significant with respect to frequency can you see secondary bumps in the response. At high frequencies this can occur with relatively short distances, where at lower frequencies the distances where this occurs increases significantly, and for most HT systems I've tested, you can never identify a late arrival, but rather it melds into a long decay.

All of the research agrees with this, and one important fault in the above arguement is that PEQ does have both frequency and time domain effects. As is correctly noted, this is behavior will be of no benefit to a 2kHz reflection from a side wall, but all published theory and follow up research/testing shows the corresponding time domain behavior of the correction filter to be of benefit to the time domain in the modal range.

Hi Craig,

The dialog from Dick Pierce is correct, but only above 300-500Hz in most listening rooms, and certainly is not correct below 100-300Hz.

The explanation clearly notes why 1/3rd octave equalizers and full range EQ using simple RTA measurements doesn't work, and fell out of popular use. For the past 20 years we have had measurement methods to examine both time and frequency domain in specific slices through various gating and windowing techniques. In the past 5-10 years, this has become much more readily available and less effective with computer based measurement systems. Exactly as I explained above, the concept Dick is talking about breaks down at lower frequencies. Frequency and wavelength matters. When we look at any source of reflection or impediment to sound propagation, we have to consider its acoustic dimension as it relates to frequency. A 2' x 2' x 6" solid block directly in the path of sound travel will be very effective in stopping or reflecting 10kHz, but will be entirely ineffective at 100Hz.

My TEF measurement system allows me to look at what are called ETC graphs, which show energy vs. time in dB and will clearly show reflections at later bumps in response after the direct or first arrival. Only when distances and arrival times become significant with respect to frequency can you see secondary bumps in the response. At high frequencies this can occur with relatively short distances, where at lower frequencies the distances where this occurs increases significantly, and for most HT systems I've tested, you can never identify a late arrival, but rather it melds into a long decay.

All of the research agrees with this, and one important fault in the above arguement is that PEQ does have both frequency and time domain effects. As is correctly noted, this is behavior will be of no benefit to a 2kHz reflection from a side wall, but all published theory and follow up research/testing shows the corresponding time domain behavior of the correction filter to be of benefit to the time domain in the modal range.
Thanks Mark. I appreciate your input. Are you saying that, for example, a 15 dB cut at 60 Hz to reduce a modal ring will be more beneficial than detrimental?

Craig

Thanks Mark. I appreciate your input. Are you saying that, for example, a 15 dB cut at 60 Hz to reduce a modal ring will be more beneficial than detrimental? CraigDepends, in part, on the time signature of the filter.

Depends, in part, on the time signature of the filter.
OK, let me be a little more specific. If one has a narrow peak at 60 Hz of 15 dB (at the listening position), would it be more beneficial than detrimental to use a standard parametric EQ, (i.e., SMS-1, BFD, Rane, etc.), with a center frequency of 60 Hz, a narrow Q, and 15 dB of cut?

Craig

I can't exactly join in the lingo of this conversation (over my head) but it seems most logical that the larger (and lowest) frequencies would not be affected much by bouncing off a wall.

I can't exactly join in the lingo of this conversation (over my head) but it seems most logical that the larger (and lowest) frequencies would not be affected much by bouncing off a wall.
Ahh, but they are. That is how room resonances are created, by sound bouncing off the walls and adding constructive or destructive interference, AKA "peaks" and "nulls".

Craig

OK, let me be a little more specific. If one has a narrow peak at 60 Hz of 15 dB (at the listening position), would it be more beneficial than detrimental to use a standard parametric EQ, (i.e., SMS-1, BFD, Rane, etc.), with a center frequency of 60 Hz, a narrow Q, and 15 dB of cut?
CraigHere's why I was not specific: That may work although one might have to play a bit with the Q and cut.

The reason is that, to be predictive, we need much more information. We would like to know more about the decay properties of this peak and what the overall response is in that frequency region. Also, it would be good to know the time-domain profile of the filter.

Ahh, but they are. That is how room resonances are created, by sound bouncing off the walls and adding constructive or destructive interference, AKA "peaks" and "nulls".

Craig

Sorry I meant in terms of their impact to the listener. I understand how the interaction can cause problems.

Here's why I was not specific: That may work although one might have to play a bit with the Q and cut.
When you say one would have to play with the Q and the cut, I understand the Q part, but how much cut do you think is acceptable? Even with a 15 dB peak, 15 dB of cut would decimate the initial sound wave generated by the sub at that frequency. If one were playing at reference level of 115 dB LFE, the initial soundwave at 60 Hz would now be only 100 dB.

The reason is that, to be predictive, we need much more information. We would like to know more about the decay properties of this peak and what the overall response is in that frequency region. Also, it would be good to know the time-domain profile of the filter.
We said it was modal ringing, so assume a long decay time. Assume the overall response in the rest of the frequency range is relatively flat, (I know, unrealistic, but for the sake of assumption), and assume an IIR filter, like those used in most parametric EQ's.

Craig

When you say one would have to play with the Q and the cut, I understand the Q part, but how much cut do you think is acceptable? Even with a 15 dB peak, 15 dB of cut would decimate the initial sound wave generated by the sub at that frequency. If one were playing at reference level of 115 dB LFE, the initial soundwave at 60 Hz would now be only 100 dB. There's the rub. Are you correcting for the initial magnitude or the effects of the decay/mode? How are you measuring? If you are using pinknoise, you are measuring both effects. If you are using a pulse analysis, you can separate them and see what needs attention. Meridian's position is that decay is more pernicious (especially in the bass) and that magnitude errors will be somewhat ameliorated by decay/mode correction.

As for your concern about absolute levels, I think (not certain) that this is not the full analysis because the room modes are still in the room. So an electronic cut of a peak in the response is indicative of excessive room gain at that frequency. Cut the electronic output by that amount and the room fills it in. In other words, one should be looking at the electronic+acoustical system/room output.


We said it was modal ringing, so assume a long decay time. Assume the overall response in the rest of the frequency range is relatively flat, (I know, unrealistic, but for the sake of assumption), and assume an IIR filter, like those used in most parametric EQ's.I suggest you take a look at Meridian's white paper on this (Lexicon does similar but, afaik, doesn't offer a scientific analysis). It explains why MRC tries to match the filter characteristics, particularly the time parameters, to the correctable modes.

When you say one would have to play with the Q and the cut, I understand the Q part, but how much cut do you think is acceptable? Even with a 15 dB peak, 15 dB of cut would decimate the initial sound wave generated by the sub at that frequency. If one were playing at reference level of 115 dB LFE, the initial soundwave at 60 Hz would now be only 100 dB.That's using a mixture of steady state conditions (15dB of cut, a tone at 115dB SPL) and a transient condition (the initial wave). The filter does no cutting until there is content at 60Hz for it to work on. As that content begins to appear, so the filter begins to work, but the sound is also propagating across the room and the contributions of the rooms surface reflections start to be added in. Bearing in mind our inability to determine the direction of a 60Hz tone our ability to discriminate the initial arrival must be in doubt, particularly when the "arrival" is such a slow event, given that the 60Hz pressure wave has a cycle time of 16.6ms. When does such a (relatively speaking) slow phenomenon "arrive" at a point in the room? Where is the perceived start of the 60Hz wave? In the first quarter of the very first 60Hz cycle as the pressure builds to its first peak it is already spread over a 4 foot distance, I don't think we are able to perceive an arrival for that as we would a millisecond or two of a tone at a few kHz. By the time our ears and brain figure out there's some 60Hz about the room is making a significant contribution, which the filter is balancing out.