# Waterfalls



## brucek

I've read some posts lately that I haven't really agreed with.

Point # 1 is regarding the effectiveness of equalization at low frequencies (15Hz-100Hz). The question posed is how can a parametric filter possibly correct room resonances. The assertion being that an EQ filter only lowers the relative SPL level in the room at that frequency, and as a result the ringing may be reduced since it drops into the noise, but it can't really correct the problem. Sorry, I don't agree..............................

Point # 2 deals with the practice of applying REW's smoothing feature to measured low frequency responses before addressing filter creation. 

One of the most useful aspects of REW, and the one that seems to get ignored the most (especially with regard to the above) are the waterfall plots. Actually, I spend more time looking at LF waterfall and LF decay than I do frequency response graphs. They tell you so much more about the response of your room. How could they not, they add another dimension. The frequency response graphs completely ignore time. The effectiveness of smoothing can't be examined without a waterfall plot.

I performed a couple experiments that I think addresses the two points above.

Point # 1. 
We've all looked at waterfall plots. They use the familiar horizontal axis of frequency and the vertical axis of SPL level that the frequency response graphs use, but they add a third dimension of time. 

The waterfall is derived from the impulse response by shifting the impulse response window to the right by a proportion of the time range to generate each succeeding slice. It has to be generated before you can observe it, so you need to click the 'generate waterfall' for the measure you want to see..

The slice slider on the waterfall graph page shows you slices of time from the first response at zero time and then time is increased as you move to slice 30. So if you have the Time Range (ms) set to 300 ms and the resolution Window set to 300ms, each slice is 10msec after time zero. This shows what the microphone hears as time moves on from the initial sweep. If you have a resonance that tends to decay very slowly, you'll see it in the waterfall. This is what we're trying to reduce with the equalizer filters.

Whenever you create a waterfall plot of a measure, you can set the slider to zero and then slowly move it out to 30. This is the decay of the signal in the room. It will tell you quite a bit where the room resonances are.

Anyway, to better demonstrate that the behaviour of the BFD EQ filters have an effect in the time domain as well as the frequency domain (that matches the modal response of a room), I connected the BFD into a loopback cable of my soundcard and used this setup to take frequency response measurements.

Below is the result of a measurement of the BFD (with the filters bypassed) using the waterfall plot. The two dimensional frequency plot would show a simple flat line of course, but the waterfall plot adds the time dimension. Each slice in the waterfall is 10msec. After 130msec, the return is down in the noise.

*Waterfall plot of a BFD with all filters bypassed.*









Now I enter a single filter into the BFD. It is a 40Hz filter with a Gain of +15dB and a Bandwidth (BW) of 10, and I do a response measurement.

Below is the expected frequency response of the BFD with the single filter added.

*Frequency Response plot of a BFD using a single filter of (40Hz, Gain +15dB, BW 10)*









But now I look at the resulting waterfall plot of that single filter below. 

Look familiar? Sure it does. 

It looks like a room mode resonance of any REW measurement at subwoofer frequencies. And it should. It's because the EQ filter, just like the modal resonances of a room, has a time response that acts like a 2nd order biquad. If I apply an EQ filter with the same Q and opposite gain of a room mode, I would completely counteract the effect of the mode. See the time component of the filter (just like a room mode). It rings out, and still isn't in the noise after 300msec. You see, EQ filters don't just affect level. This is why they're so effective at equalizing at modal frequencies below 100Hz. Yes, it is listening position dependant, and only valid at the point where the response was measured, but because of the long wavelengths of low frequencies, the region around that area is fairly large. This is in opposition to higher frequencies where equalization is a bit of a waste of time, since the effective region is so small that eq is impractical.

*Waterfall plot plot of a BFD using a single filter of (40Hz, Gain +15dB, BW 10)*









As a side note, you can see what a completely terrible idea it is to add a gain filter to boost the level of a sub at low frequencies. You do nothing more than emulate a room mode at the gain frequency

So, if I enter a second filter into the BFD of (40Hz, -15dB, 10BW) to counteract the existing filter of (40Hz, +15dB, 10BW), the result is the measured response below. The room mode is completely nullified.

*Waterfall plot plot of a BFD using a two filters of 
(40Hz, Gain +15dB, BW 10) & (40Hz, Gain -15dB, BW 10)*









And so, if we applied a counteracting filter in the BFD that matched a room's modes, the effects of the resonance is completely removed.

continued in the next post.......................


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## brucek

continued from above...............................

Point #2.

In keeping with the information in the post above, let's take the response of a BFD that uses a single filter of (40Hz, +15dB, 10BW) and let REW recommend a counteracting filter to remove it. Let's see if REW works.... 

Fairly obvious results, that REW recommends to add a counteracting filter of (40Hz, -15dB, 10BW). If I enter that filter, just as I previously did above in point #1, the result is a flat response with a flat waterfall. Below is the response graph of the room resonance filter and the REW filter recommended and the corrected response. The waterfall, of course, is the flat smooth response pictured above.

*FREQUENCY RESPONSE OF A SIMULATED ROOM MODE AND REW's RECOMMENDED FIX*








Now, lets add some smoothing to the response measurement of a BFD filter of (40Hz, +15dB, 10BW). This response represents a typical room mode from an actual measure. Instead of letting REW recommend a filter to counteract the mode, I'll add some smoothing to the response first and then see how REW does in recommending filters. 

I'll enter those filters into the BFD and see what the resulting response is. I won't waste too much time on the frequency response graph since it doesn't really tell much of a story - I'll look at the waterfall and see if the filters I have applied after smoothing remove the resonance.

Below is the graph of the (40Hz, +15dB, 10BW) room mode with 1/2 octave smoothing added. The picture also shows the filter that REW recommends to enter into the BFD. No amount of manual intervention gets the corrected response any better.

*FREQUENCY RESPONSE GRAPH OF A SMOOTHED SIMULATED ROOM MODE AND REW's RECOMMENDED FIX*








Below is the picture of the BFD with the room resonance filter combined with the smoothed filter recommendation.


*FREQUENCY RESPONSE GRAPH OF ROOM MODE AND REW's RECOMMENDED FIX AFTER FILTERS ENTERED INTO BFD*








The revealing picture is the waterfall of course. It shows quite conclusively that filters optimized against a smoothed response will have settings that don't accurately match the room's modes. Look at the ringing out of that resonance.......

*WATERFALL OF ROOM MODE THAT WAS SMOOTHED AND FILTER USED THAT REW RECOMMENDED*









brucek


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## Blaser

These are very good remarks Brucek!! Bravo!
So, is it assumable that equalization can be more practical, cheaper, easier for room mode decay (ringing) as well as FR below say 80 Hz than room treatment? I am talking about a single seat of course.


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## brucek

> So, is it assumable that equalization can be more practical, cheaper, easier for room mode decay (ringing) as well as FR below say 80 Hz than room treatment?


Well for sure. Certainly equalizing to remove modal resonances is what REW is all about. The listening area can be measured at mutliple positions and the results averaged if you like or you can choose a single position for the best results. The effective area will depend on the frequency of the mode. Acoustic treatment can improve the response over a wider area, but at those low frequencies <100Hz, the size of the treatment would be ridiculous. For less than 100Hz, use EQ. Above that, to about 500Hz (so I understand), use room treatment.

brucek


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## thxgoon

Very informative post, thanks!

So what in your opinion would be the best way to set up eq under 100hz. Would you not use the smoothing before assigning filters to the BFD?


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## brucek

> Would you not use the smoothing before assigning filters to the BFD?


I would not use smoothing. The author of REW also highly recommends not to use smoothing before assigning filters.

I would take the raw measure and use REW to find peaks, assign filters and optimize. Enter those filters and then then see how the waterfall looks after a remeasure.

brucek


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## toecheese

Thank you Bruce! I disagreed with some recent things as well and you touched on some of them.


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## Blaser

Brucek,

I have once made the assumption that room treatment cannot be as effective as equalization below 80 Hz in another forum and I was strongly attacked by the members as well as by a very well know person (that I do respect a lot) in this industry.
You have encouraged me to make my own experiment in my room and to explain my point of view (based on real world measurements), and discuss it with those who might have some doubt.

Let's begin with a quick description of my room which is approx. 1800 cuft (5 m *3.6 m * 2.7 m). My subs are located on the front wall while I am seating about 0.7 m from the back wall. Before applying any eq. my problem was mainly the axial mode which I have measured to be about 18 db at 36 Hz approx.

Thanks to this wonderful and best forum that I know, I bought the FBQ 2496 which corrected this peak problem.... (but is it only the 36 Hz peak problem???, we'll see).

My analysis will be about the effect of equalization on the axial mode decay (36 Hz ringing) through LF waterfall(which will add the parameter of time as explained by Master Brucek). Take a look at the unequ. and eq. graphs below. For the sake of *keeping things fair*, I will keep the SPL of 36 Hz the same for both equ. and unequ. graphs. 

Continues in following post....


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## Blaser

The reason why I kept the 36 Hz SPL the same for both graphs is because, the value after 300 ms can be too low or too high due to an initial smaller or higher SPL.

Now take a look at the LF waterfall for my unequalized room and equalized and both together. The difference is self-explanatory, although I have raised the amps output by 18 db after equalization (to keep the same value as the unequ. graph at 36 Hz.) the ringing is much less at 36 Hz, decay is faster and of course this will be less "boominess", and "tighter" sound.

Conclusion: Not only the equalized FR is of course of much supperior quality than the unequ. but LF waterfall graph is much better (about 12 to 14 db less SPL at 36 Hz after 300 ms), not to mention that the other frequencies are heard too, and the one note bass has left its place to a more pleasant listening experience.

Why is that? I don't know. Propably you Brucek may explain how this was electronically done.

Continues in next post...


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## Blaser

Some may mention that between 10 and 15 Hz, Waterfall is worst when equalised. I say no!
In the beginning I mentioned that for comparing data, we have to *keep things fair*. At 0 ms and at 10-15 Hz, the unequ. SPL is 52 db while it is 70 db with the equ. curve. The difference is 18 db (those that I have added by turning up the amps during the equ. graph). And something weird (is it) is happening: at 300 ms the 18 db difference is still the same, why? because no filter was used at these frequencies!! Anyway, any low decay at 20 Hz and below is wellcome for me:bigsmile:

To confirm the above, below are the LF decay curves as well.

Some will claim that this is for only one seat. Again I say no. Check the very first post of Brucek about wave length of LF and you will note it does not change so much (due to their nature) unless the distance is great. Moreover, I am using dual subs aligned on the front wall, and this makes the FR quite similar at any lateral place of the couch, so these results will also apply to 4 or 5 seated people seating laterally in the room, and propable another raw for those who have larger rooms than mine (and of course larger room may sound even better).

How much money and room space give up (if possible at all) can we do similar results with bass trap?

Thank you!


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## brucek

Yeah, the equalization certainly seems to have nearly eliminated the resonance at 36Hz. The fact that the signal doesn't persist in the room after the other frequencies have gone, accounts for the better overall bass and a less muddy sound I'm sure.

You may even try taking successive measures while watching the small ringing out signal (shown in yellow) and manually tweak the FBQ a bit. Try a single click higher and lower for the parameters of frequency and bandwidth and see if you can get the center frequency and shape of the resonance any better (although it's quite good now-that's a very nice response).

I also wonder if a bit of sub phase or sub distance setting would help your crossover dip (shown in green)?









brucek


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## Blaser

Regarding the dip at 80 Hz, I am not too worried about it as it is not very wide. Nevertheless I will do some more tweeking later on.


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## brucek

In talking to blaser, he pointed out that I didn't explain too well in my last or original post the reason why an equalizer is helping the decay of the resonance and needed further explanation how this is done electronically?

Well, in my reading about 2nd order biquad filters, poles and zeros, and minimum phase systems, some of the assertions by the author of REW do make some sense in that regard.

I tried to explain this in my original post, but probably fell short since perhaps I don't fully understand it myself.

The modal response of a room acts exactly like a 2nd order filter and matches the BFD generated filters in all aspects. At modal frequencies, a room resonates in gain and Q exactly as if you fed a sub signal through a 2nd order parametric filter. This fact allow us to fashion an identical 2nd order filter with the opposite gain and bandwidth that matches the room mode so it will completely disappear (at the point of measurement). 

This doesn't apply outside the low frequency range where signals are no longer considered minimum phase, where primary reflections (second order) from the walls, ceiling and floor arrive at the listening position anywhere in the room with a phase shift of quite a bit less than a cycle. So, the effective limit here of about 80Hz-100Hz is reasonable for equalization in most rooms...(an 80HZ signal has a wavelength of about 14ft)............


I suppose I could also add here that REW may also not suggest a filter for a peak that you think it should have tried to correct. This may be a result of the peak not having the attributes of a modal resonance (this is what REW is looking for). You'll likely find that if you examine the waterfall at that peak that REW ignored, there would be no ringing out in the time domain. There's nothing stopping you from adding your own manual filter into REW and then into the BFD. It would be wise to examine the waterfall after that though.

There also may be modal resonances very close together, perhaps as a result of equal lengh and widths in a room or other reflections. REW could have trouble finding these, so you may have to do it manually. It's tricky and requires a back and forth measurement and tweaking as you watch the waterfalls.

JohnM may refute any of my ramblings in this thread if they are misleading or plain dumb.... :reading:


brucek


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## JohnM

Looks good to me, Bruce, and a nice job by blaser as well though there is scope for further improvement in the control of the modes around 36-40Hz in blaser's room. It will be interesting to see how the tweaks work out.


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## Blaser

Thanks John. When I have time, I'll try to tweek and see, but I don't expect much better result. Indeed equalization has already removed 12 db after 300 ms which is already very good,and propably not easily possible with any other method.


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## Sonnie

You guys are making my head hurt... really bad... :hissyfit:

I'm gonna have to read this 3-4 times you know. :foottap:


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## SteveCallas

Good stuff - excellent in room response blaser :T


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## Chrisbee

As I have frequently mentioned I boost at maximum at 20Hz to compensate for my high Fs 32Hz non-spec drivers. (AE IB15s) 

BFD= 20Hz + 16dB BW= 120/60.

Here's a waterfall I generated from an older file. Extended to 1000ms and cropped to 80Hz to keep the interesting VLF area in view.










It seems that boosting with the BFD does indeed increase ringing.


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## JohnM

You won't see any ringing from the filter at that bandwidth setting, it is so broad it boosts a very wide range. The peaks in your measurement are most likely room-related. To see the effect of the filter alone connect the BFD in a loopback to your soundcard and measure it that way.


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## Doug Plumb

"You'll likely find that if you examine the waterfall at that peak that REW ignored, there would be no ringing out in the time domain."

This would be impossible. Its impossible to have a peak without the resonance or the ringing. Sharp peaks have sharp slow decay resonances. Dull peaks have quickly decaying responses or resonances.

If you know the frequency response then the impulse can be calculated then a waterfall can be calculated.

Really, the best way to look at response is with the single frequency response. The waterfall is prettier but of little practical use.

(I just had to say this after reading that, not interfering with REW thread)


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## Blaser

Chrisbee,

You need to compare your graph to the same initial 20Hz SPL (without BFD) and then compare after 1000ms. Otherwise you will never know.

Indeed, ringing at 20 Hz is not necessary a bad thing...It could be amazing:devil:


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## thxgoon

Wow, I envy all of you and your room response. I'm still working on mine but here is what I've go so far.... note this is going out to 933ms. At 300 virtually all frequencies were still there...










It's interesting to see on the waterfall that right before and after my lull at 56hz, there is are substantial nodes! I'm not too sure how to deal with this yet.


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## brucek

> This would be impossible. Its impossible to have a peak without the resonance or the ringing.





> Really, the best way to look at response is with the single frequency response. The waterfall is prettier but of little practical use.


Doug, we'll just have to disagree on just about everything in your post. 

I wish I could find some better examples to show you that very similar peaks can have completely different persistence in time, but I could only find the response below (I've seen far better examples). 

I find the waterfall plots very useful in identifying peaks that ring-out for a long time. I use REW to match that peak and reduce its ringing as low as possible. Other peaks that look very similar in the 2 dimensional frequency response plots that don't ring-out as much have nowhere near the impact to the sound.

Here's a quick example of two peaks that are very similar - at least they look that way using a simple frequency response plot. With the single slice 2 dimensional plot of frequency and amplitude, the two peaks look about the same. 

Now examine what happens in a three dimensional plot when I move the slider to 30 slices to add time derived from the impulse response. One just keeps on going. And you hear that too. Using REW to match that room mode eliminates that ringing out in time..

Waterfalls are far more than pretty pictures....... 

ONE SLICE








30 SLICES








brucek


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## Doug Plumb

"Doug, we'll just have to disagree on just about everything in your post."

I'm sorry to hear that. I program waterfall plots and have studied these response functions in detail as an engineering student. We spend about 4 courses per year for two years studying the response function in various courses from math to controls to DSP and analogue filtering. 

NASA or any other company does not ever use waterfall plots to evaluate the response of their systems and the response of their control systems is similar to what we have here. I have never seen a waterfall plot in any serious controls theory, electric circuit theory paper or in any other form. Its never used for a reason - its useless !! They are only in audio, they look pretty but mean little. A well known Phd acoustician has agreed with me on this point and doesn't use them for the same reason. 

They can obfusgate the situation when people mis interpret noise or the mathematical distortions in gating as an actual physical affect, as you have above.

Every single peak in the waterfall plot is due to a peak in the frequency response. If there is a section in the waterfall that displays a decaying portion without a peak in the frequency response then its noise or some other affect.

In some non mimimum phase systems you could possibly see a decaying part that does not show as a resonance in the first slice but the resonance must appear in later slices. This is very unlikely and absolutely unlikely in a modal region room response.


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## Blaser

Well...at least the ears do not lie. There is not need for a very sophisticated program to check for HT audio stuff. Anyway, the waterfalls in REW is a very useful program, and if correctly understood (I do not mean you don't) and utilized will with no doubt help enhance ringing (at low freqs of course). 

Did you take the time to see the improvement clarfied by the waterfalls taken in my room...They definitely confirm the HUGE improvement that was done in my room. And at least, this is a practical measurment...:T


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## JohnM

Doug Plumb said:


> NASA or any other company does not ever use waterfall plots to evaluate the response of their systems and the response of their control systems is similar to what we have here. I have never seen a waterfall plot in any serious controls theory, electric circuit theory paper or in any other form. Its never used for a reason - its useless !! They are only in audio, they look pretty but mean little. A well known Phd acoustician has agreed with me on this point and doesn't use them for the same reason.
> 
> They can obfusgate the situation when people mis interpret noise or the mathematical distortions in gating as an actual physical affect, as you have above.
> 
> Every single peak in the waterfall plot is due to a peak in the frequency response. If there is a section in the waterfall that displays a decaying portion without a peak in the frequency response then its noise or some other affect.
> 
> In some non mimimum phase systems you could possibly see a decaying part that does not show as a resonance in the first slice but the resonance must appear in later slices. This is very unlikely and absolutely unlikely in a modal region room response.


That is overstated Doug. As an example, I'm sure you have seen a great many instances where what appeared to be a single peak in the frequency response is the result of two or more modal resonances, it is only in the waterfall or by gating later parts of the impulse response that this becomes readily apparent. Whilst a waterfall plot can be misinterpreted, so can just about any other data representation - not least the frequency response itself, whose appearance in this context is entirely dependent on the window positions and durations used to generate it. It is very easy to produce frequency responses of widely varying appearance from the same impulse response. Waterfalls and similar/related data visualisations which portray combined frequency and time domain behaviour are very widely used in some fields of analysis, e.g. seismic data in oil/gas/mineral exploration.


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## Doug Plumb

> frequency response is the result of two or more modal resonances, it is only in the waterfall or by gating later parts of the impulse response that this becomes readily apparent.


I think that in many cases this is room noise that is also reinforced at certain frequencies by the dimensions of the room. Phase interactions between "modal noise" and signal may cause the waterfall to do weird things and be misinterpreted. I don't trust low level waterfall data because it can be corrupted by noise.

If you have a resonance and can't correct it with one filter then you need more than one, this becomes apparent only after one filter is used and you cannot ideally dampen the resonance. This occured in my room at about 140 Hz. There are three sharp resonances right around that area, easily identified with curve fitting.

Oil exploration, etc look almost exclusively at non minimum phase behavior under which conditions a waterfall plot may apply. I think that generally we should encourage users of our kind of software to look only at frequency response because the distortions associated with waterfall plots are usually poorly understood.


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## clubfoot

But isn't this why the room acoustics is so important? One should not have to apply more than one filter to correct a resonance at one frequency if the room is doing its job! My belief is if you have to apply multiple filters for a narrow frequency spike or bump, it's the room or a very poorly designed sub driver or enclosure "ringing". Granted it still has to be corrected for the system to sound "right" but not multiple filters.

My engineering degree is not in acoustics, so this is just my opinion.


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## Blaser

You cannot have a reasonable room treatment if say you have a problem at 30 Hz...


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## Doug Plumb

> One should not have to apply more than one filter to correct a resonance at one frequency if the room is doing its job! My belief is if you have to apply multiple filters for a narrow frequency spike or bump, it's the room or a very poorly designed sub driver or enclosure "ringing". Granted it still has to be corrected for the system to sound "right" but not multiple filters.


My degree is in EE not acoustics as well - but even if you design a room you cannot guarentee not getting multiple modes at one frequency or a narrow band because rooms are never perfectly square nor built to precise dimensions. Higher order modes are dense & sensitive to small differences in planned and actual construction.

You don't need multiple filters to correct - you can just put a big notch in for low frequencies if you want.


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## clubfoot

Doug Plumb said:


> My degree is in EE not acoustics as well - but even if you design a room you cannot guarentee not getting multiple modes at one frequency or a narrow band because rooms are never perfectly square nor built to precise dimensions. Higher order modes are dense & sensitive to small differences in planned and actual construction.
> 
> You don't need multiple filters to correct - you can just put a big notch in for low frequencies if you want.


Absolutely,...if the room is not rectangular (non divisible sides) and very well constructed then its influence on the sound becomes unpredictable and more drastic (read treatments) measures have to be applied to "dampen" resonances, or at least tame them to the point where something like a BFD can control them without degradation in sound quality.

I still don't believe that using a big notch filter would be a final answer,...temporarily, yes. But long term you're going to get tired of it and will want to address what is causing that big bump.

So this is where I agree with brucek and the use of waterfall plots because we are dealing with an analog device that behaves like a car's suspension or sub in this case (spring & shock absorber). Remember the big old "American" sedans and how their suspensions behaved,...excellent frequency response most of the time, but throw in a curve or mid corner *bump* and they became unsettled, and that's why I believe in this instance waterfall plots are useful.


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## Guest

Hi,

I would like to understand whether to intervene:
I don't succeed in flattening the 153 Hz blade (DEQ2496)
(2 of my Ht satellites about 6 ft convergent, microphone to the center)



Red waterfall toslink input;
Green waterfall DEQ equalization + FBD - 6 [email protected] 1/60 oct
Yellow waterfall DEQ equalization + FBD - 30 [email protected] 1/60 oct
Orange waterfall DEQ equalization + FBD - 30 [email protected] X 3 1/60 oct.

where am I wrong?

thanks

EDIT amrvf:

excuse me, 
soundcard not calibrated!!! (not loopback but microphone signal!!!)


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## brucek

I would say the peak is far too narrow to even hear.

Filters of 1/60th of an octave are far too narrow to be useful. Small microphone movements at that frequency would change the result.

Looks like the signal may be room noise.

brucek


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## Chrisbee

I'm back. Thanks for the earlier responses.

Here are some fresh waterfall plots of my 4 x 15" IB in a very non-parallel 27' attic with multilevel floors, an open stairway in the floor. Ceilings, floor and 45 degree walls are all boarded.

*Listening position no BFD:*









*Listening position with BFD including +16dB boost @ 20Hz 120/60.*









*Nearfield with BFD.*









Any useful thoughts?


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## clubfoot

All I can say is WOW


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## Ivaols

Brucek wrote earlier in this thread:

"As a side note, you can see what a completely terrible idea it is to add a gain filter to boost the level of a sub at low frequencies. You do nothing more than emulate a room mode at the gain frequency"

In Chrisbee`s second waterfall in the post above he has gained 20 hz with 16 db. I cant see that this has emulated a room mode at the gain frequency. The room mode seems to be the same as the higher frequencys.


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## Chrisbee

You may find this an interesting comparison: Measured at the listening position:










Here I have replaced my original 32Hz drivers with four brand new 16Hz drivers from AE.

BFD settings have been reset from the massive +16dB @ 20Hz boost required before.

I wanted to continue the new slope into the infrasonic to match my normal listening levels. Fletcher-Munson, Equal loudness etc.

A wide trough existed between the 12Hz room mode and ~25hz. I filled this with another quick and dirty boost filter at 20Hz.. 

Trial BFD Filters are:

20Hz + 5dB BW35. 
25+2Hz -6dB BW20. 
40Hz -4dB BW30 
50+5Hz -3dB BW20


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## JohnM

Ivaols said:


> Brucek wrote earlier in this thread:
> 
> "As a side note, you can see what a completely terrible idea it is to add a gain filter to boost the level of a sub at low frequencies. You do nothing more than emulate a room mode at the gain frequency"
> 
> In Chrisbee`s second waterfall in the post above he has gained 20 hz with 16 db. I cant see that this has emulated a room mode at the gain frequency. The room mode seems to be the same as the higher frequencys.


That is because the filter Chris used was very wide, spanning 2 octaves (BW 120/60). Modes have narrow bandwidths, typically 1/10th of an octave or less. Using narrow filters to boost the response creates the same extended ringing problems as room modes.


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## DrWho

In the first post...

Am I the only one that finds it odd that the decay rate of your BFD is over 100ms? I understand that some shape is expected for a bandwidth limited spectrum, but _that_ long?


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## brucek

> Am I the only one that finds it odd that the decay rate of your BFD is over 100ms?


I don't believe that it's the decay rate of the BFD at all.

A loopback cable (with the BFD removed from the circuit) will show exactly the same result.

Isn't the 100ms that you refer to a function of the duration of the impulse response analyzed?

For example, if connect a loopback from line-out to line-in of the soundcard and sweep a measurement to 200Hz, and increase the duration of the gate out to 1000ms (1Hz frequency resolution), I would get the following result.









But if I decrease the gate to 100msec (10Hz frequency resolution), I would get the following result.









brucek


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## DrWho

> Isn't the 100ms that you refer to a function of the duration of the impulse response analyzed?


An impulse, by definition, has a length of 0 (it's instantaneous). This is impossible in a bandwidth limited system, so one would expect to see something a little bit longer...but it shouldn't be longer than say 1ms - certainly not 100ms.

Are you saying that you only did the sweep up to 200Hz? If so, that might explain it. Why aren't you sweeping to the top limit of your soundcard?

Regardless, simply increasing the volume on the BFD is going to make it seem like it takes longer to decay. The fact that your EQ shows the same thing simply verifies that it is achieving an amplitude change...not so much changing the ringing at that frequency.


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## brucek

> An impulse, by definition, has a length of 0 (it's instantaneous).


Yes, but REW doesn't use impulses to measure the response. The impulse response is derived from the system's transfer function, which is determined from the system's response to the sweep. In this case I am interested in an end frequency of 200Hz for the response and waterfall plots. An end frequency in REW is set to the highest frequency desired (in this case 200Hz), and the resulting sweep will span from 0Hz to twice the frequency set (with an overall limit of half the soundcard sample rate) to provide accurate measurement for the selected range (in this case 400Hz).

Sorry, I'm not sure what you're disagreeing with, or even what point you're trying to make? :huh:

brucek


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## DrWho

> Am I the only one that finds it odd that the decay rate of your BFD is over 100ms?
> 
> 
> 
> I don't believe that it's the decay rate of the BFD at all.
Click to expand...

Exactly, so how do *you* reconcile the measurements to correlate with reality?

What I'm trying to point out is that the EQ does not ring or resonate or whatever you want to call it. It is not changing over time. You are misinterpreting an artifact of the waterfall calculation.

For a given point in the room where reflections arrive within a wavelength of the frequency in question, the system _could_ be simplified as minimum phase. EQ only works in minimum phase situations.

However, you have not shown that the standing waves have been removed. In fact, they are most certainly still there. The easiest way to verify that the standing waves are still there is to locate the nulls of the standing waves...which do not move to different positions in the room or change frequency as EQ is applied to the signal.

I suppose you could just not sit in the nulls, but it's not quite so easy. The point I was making above is that you're only minimum phase for a single location in space. If you move the microphone even a few inches (like the width of your head), your EQ no longer works perfectly. Sure, some might argue that it is a small compromise, but the point is that it's not perfect. Proper acoustical treatment gets rid of the standing wave, which in turn improves the situation at far more locations in the room (it's not perfect either, but the compromises are much less).

One more comment which has to deal with psychoacoustics. When you walk into a room, there is a certain sonic characteristic to that room, even with no music playing. When you use EQ on the music to try and not trigger behaviors in the room, you end up sending conflicting signals to the listener who is partially expecting that bass guitar to sound like it would sound in their room. In other words, when you sit there listening to music, you are partially going to be identifying some attributes of the sound as being the room and not the music....if your music is precompensated for the room, and then your ears compensate what they hear from the room, then you end up with double compensation and the music sounds disconnected. Perception is definitely different for everyone, but the point is that it's much easier when the room isn't imparting **** on the sound because then there is never any amount of the listener trying to filter it out - it just sounds way more natural.


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## brucek

> The point I was making above is that you're only minimum phase for a single location in space.


Yes, and I completely agree and have said the same thing in my post #13 that I'll reprint below:

_The modal response of a room acts exactly like a 2nd order filter and matches the BFD generated filters in all aspects. At modal frequencies, a room resonates in gain and Q exactly as if you fed a sub signal through a 2nd order parametric filter. This fact allow us to fashion an identical 2nd order filter with the opposite gain and bandwidth that matches the room mode so it will completely disappear (at the point of measurement). 

This doesn't apply outside the low frequency range where signals are no longer considered minimum phase, where primary reflections (second order) from the walls, ceiling and floor arrive at the listening position anywhere in the room with a phase shift of quite a bit less than a cycle. So, the effective limit here of about 80Hz-100Hz is reasonable for equalization in most rooms...(an 80HZ signal has a wavelength of about 14ft)............
_



> Proper acoustical treatment gets rid of the standing wave, which in turn improves the situation at far more locations in the room (it's not perfect either, but the compromises are much less).


Mmmm OK, I don't know the size of the treatment that might be required though and the WAF of such treatments at the frequencies in question. Personally, I think EQ is an acceptable and effective route to follow. You and I have different opinions on that I guess. We'll leave it at that.

brucek


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## DrWho

How often does it show up where EQ predictions are made that do not result in the measured outcome? If the system were minimum phase, wouldn't the measurements correlate exactly with the predictions? Whenever this happens, one might argue that making the frequency response as flat as possible doesn't lead to the most accurate reproduction. Of course this notion is nothing new - there are articles dating back into the 70's.

There are also articles that discuss windowing limits for low frequency waterfalls too - it's a tradeoff between time and frequency resolution (can't have both at the same time). I'll see if I can't find that article too, but basically the data is meaningless without a proper window and obviously in this case, the wrong window is being used because it's showing decay rates that cannot be true.


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## JohnM

The main factors determining the shape and behaviour of the waterfall plot are the types and durations of the window functions used. The main window function is a right half window whose type is set by the Low Freq Decay window selection in the Analysis settings (default is Tukey 0.25) and whose duration is governed by the window control below the waterfall graph. The key to the observed behaviour of the waterfall is the window function applied to the left edge, which is not user-selectable. It is a left half Hann window whose duration is half the selected window duration (making for a total duration of 1.5 times the value in the control). This value was arrived at empirically by trying various settings to find a value that makes it easy to distinguish modal effects in measured data. I may make this a user-configurable parameter in a future build, but the current settings work well for the intended purpose of this plot.


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## PeteD

brucek said:


> As a side note, you can see what a completely terrible idea it is to add a gain filter to boost the level of a sub at low frequencies. You do nothing more than emulate a room mode at the gain frequency


Brucek, I do not agree with this. I believe boosting versus cutting should have no impact on creating resonance. The resonance is a function of the room. If you need "boost" at a given frequency, the room is absorbing that frequency, which is why you need the boost. Now, maybe there is some effect with really narrow bandwidth (I am an engineer, but not an acoustical or EE so I won't claim to know).

What about boosting within an area of a large cut at a level less than the cut? Surely that would not create a resonance, right?

To prove your point (I would do it, but I am still doing manual reading imported in REW), I think you would need to create a frequency response curve for a real room using two methods: 1) no boosting, and 2) with boosting at selected low frequencies. Then compare the waterfall plots between the two. I think any area of resonances are likely to be the same and induced by the room, not dependent on whether you used boost or cut in a given area.

Regards,
Pete


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## MakeFlat

I read that Floyd E. Toole (of Infinity) stated that leveling a peak will mitigate a room mode because doing so changes the phase as well. I am not expert enough to prove or disprove it but it seems to make sense. The proof, as he said, is in listening.

Now on a slightly different twist: suppose there are two identical subs except that the phase on one sub is changed to oppose the other at a resonance frequency, keeping the phase the same at all other frequencies. Has anyone tried it?


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## Wayne A. Pflughaupt

PeteD said:


> Brucek, I do not agree with this. I believe boosting versus cutting should have no impact on creating resonance. The resonance is a function of the room. If you need "boost" at a given frequency, the room is absorbing that frequency, which is why you need the boost.
> 
> What about boosting within an area of a large cut at a level less than the cut? Surely that would not create a resonance, right?


I believe that John clarified here that you are essentially correct. Boosting a null shouldn’t result in ringing that’s any (or at least much) worse than it would be if the null wasn’t there.



MakeFlat said:


> Now on a slightly different twist: suppose there are two identical subs except that the *phase on one sub is changed to oppose the other at a resonance frequency*, keeping the phase the same at all other frequencies. Has anyone tried it?


Are you talking about *phase* or *polarity*? I believe phase is more-or-less a time alignment issue – not sure you one could make it “oppose.”

Regards,
Wayne


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## nick72

Happy Holidays everyone!

This thread has been very interesting and very timely as I have significant room issues that I would like to correct and I’m considering room treatment. I haven't been able to find much info on understanding waterfalls plots. 

So statements such as the following….

Blaser:


> So, is it assumable that equalization can be more practical, cheaper, easier for room mode decay (ringing) as well as FR below say 80 Hz than room treatment?


BruceK:


> Well for sure.


…really got me interested. My primary purpose on understanding this stuff is to determine if I need room treatment for my room to handle the basss ringing, since I have a BFD. 

As a novice in the room acoustics realm though, this thread has left me (and perhaps others with regard to using waterfalls plots in relation to room ringing), a little confused.

So, if its ok with you guys, I was wondering if I could pose a few more simpler questions in this thread, since I’m sure I’m missing something here.

From BruceK’s original message…



> Point # 1 is regarding the effectiveness of equalization at low frequencies (15Hz-100Hz). The question posed is how can a parametric filter possibly correct room resonances. The assertion being that an EQ filter only lowers the relative SPL level in the room at that frequency, and as a result the ringing may be reduced since it drops into the noise, but it can't really correct the problem. Sorry, I don't agree..............................


I could be wrong, but it appears to me, that Bruce took the output directly from the BFD to apply filters to a flat frequency response. When the gain was increased by 15db at 40 Hz, a whopping 300+ ms of ringing was observed. 

My initial though was: How is this possible from a direct output of the BFD to REW? No room reverberations had a chance to occur. 
Am I confusing two different acoustic effects?
But later in the thread, Dr. Who and BruceK, and others tried to clarify this. The BFD is likely not really introducing that much of a decay. 
But if the BFD isn’t really introducing 130 ms of ringing to the flat signal, then is it really introducing 300+ ms of ringing when the 15db gain is applied at 40 Hz?
If not, where does this leave us in relation to Point #1, if we can’t really trust the waterfalls plot?
Am I just reading this all wrong? 
Am I missing something basic here.:dizzy:

Finally, should I not be using a waterfalls plot in determining how bad the room accoustics are for my living room?

Needless to say, any insight would be greatly appreciated..
Thanks!


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## Wayne A. Pflughaupt

Nick,

brucek’s graphs were presented as a visual aid for his narrative. No, the Behringer does not add ringing to an electrical signal, nor does any other equalizer. It’s the room that does that.

Don’t let all the theory confuse you. It’s easy enough to take your own REW readings and compare equalized to baseline and see what you get.



> Finally, should I not be using a waterfalls plot in determining how bad the room accoustics are for my living room?


REW can certainly show you the effects of equalizing and treatments, if you apply them.

Regards,
Wayne


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## Blaser

Just check my small experiment supported with graphs at the beginning of the thread ...This is not a simulation but true measuements


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## PeteD

Wayne A. Pflughaupt said:


> I believe that John clarified here that you are essentially correct. Boosting a null shouldn’t result in ringing that’s any (or at least much) worse than it would be if the null wasn’t there.


Thanks Wayne. I guess my point was that if cuts help with an area of higher SPL and resonance, shouldn't boost help with the opposite situation? It could be the room or a lack of low frequency capability of the subwoofers, but sometimes a little boost is useful and I do not think it would introduce resonance.

Now, there are of course other reasons to use boost judiciously or not at all, including maximizing the signal to noise ratio through the BFD (although I believe one has to look at the entire system's S/N ratio, also), and the fact that boosting a null eats up your sub amp's headroom (needlessly, if you really have a null that does not respond). However, a little boost can be used to treat a low line level output from the BFD, which was my problem with using just cuts.

Unfortunately, I think there needs to be a case-specific evaluation by each person as they tweak their system. Luckily, there forum is available to guide people along the way.

Pete


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## Blaser

PeteD said:


> However, a little boost can be used to treat a low line level output from the BFD, which was my problem with using just cuts.
> 
> Unfortunately, I think there needs to be a case-specific evaluation by each person as they tweak their system. Luckily, there forum is available to guide people along the way.
> Pete


I agree with you about boosting to get back the signal that was reduced by cuts. This will have its problems but the benefits are greater IMO.


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## MakeFlat

Wayne A. Pflughaupt said:


> Are you talking about *phase* or *polarity*? I believe phase is more-or-less a time alignment issue – not sure you one could make it “oppose.”
> 
> Regards,
> Wayne


Perhaps polarity is the right way. Here's what I'm thinking. Run the second sub through a bandpass filter with center frequency the same as the room resonance. The second sub would have the polarity inverted. Sorry I don't have a second sub to try.


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## DrWho

I should mention that there is absolutely no doubt that the BFD is NOT introducing any kind of ringing when the filter is boosted. What we're seeing is precisely the effects of the window used to generate the waterfall...not what's happening in real life.

I brought it up to point out that changes in amplitude can pretend to show differences in ringing (as so clearly demonstrated for us). Since we know that no ringing is added when a frequency range is boosted, we can use the graphs to normalize what constitutes no change in decay when amplitude is changed.

In other words, any EQ added to your subwoofer is going to show the exact same difference on the waterfall that the EQ made....and it's not because the EQ magically made the room ring less. Likewise, any natural amplitude variations with the subwoofer will also show up as ringing (and dips will show up as decaying faster).


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## JohnM

DrWho said:


> I should mention that there is absolutely no doubt that the BFD is NOT introducing any kind of ringing when the filter is boosted. What we're seeing is precisely the effects of the window used to generate the waterfall...not what's happening in real life.


That's not correct. The windowing effects are the cause of the response across the freq band not dropping sharply to the noise floor when no filters are active, but every EQ filter (gain or cut) rings at its centre frequency, the higher the Q (the narrower the bandwidth) the longer it rings. The ringing is easily observed in the impulse response by setting up the BFD in loopback and making a measurement with a sharp filter, it is more obvious with boost than cut but both ring. It is because of this time domain behaviour that EQ filters are able to counteract the ringing of modes, the decay of the EQ filter's attenuation over time matches the decay of the mode's gain, when the filter is properly set to match the mode's bandwidth and gain.


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## PeteD

JohnM said:


> That's not correct. The windowing effects are the cause of the response across the freq band not dropping sharply to the noise floor when no filters are active, but every EQ filter (gain or cut) rings at its centre frequency, the higher the Q (the narrower the bandwidth) the longer it rings. The ringing is easily observed in the impulse response by setting up the BFD in loopback and making a measurement with a sharp filter, it is more obvious with boost than cut but both ring. It is because of this time domain behaviour that EQ filters are able to counteract the ringing of modes, the decay of the EQ filter's attenuation over time matches the decay of the mode's gain, when the filter is properly set to match the mode's bandwidth and gain.


So, can I assume that would part of the key to using the waterfall to tweak your response...adjusting the bandwidth of the filters to modify the waterfall plot?

What is the ideal decay rate (or realistic desired decay rate where sound quality is not compromised - i.e. we obviously aren't listening in anechoic rooms), or do we just want it to be relatively even across the frequency response?

I guess could also be said that boost in particular should be performed at as wide a bandwidth as possible, with subsequent cuts to included peaks, instead of boosting individual dips - which is more likely to introduce ringing?

This is a very informative thread - it motivated me to pick up a sound card yesterday.

Pete


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## JohnM

PeteD said:


> So, can I assume that would part of the key to using the waterfall to tweak your response...adjusting the bandwidth of the filters to modify the waterfall plot?


Exactly, though centre frequency and gain might also need tweaking for best results.



PeteD said:


> What is the ideal decay rate (or realistic desired decay rate where sound quality is not compromised - i.e. we obviously aren't listening in anechoic rooms), or do we just want it to be relatively even across the frequency response?


Ideally decay would be uniform across the band, for domestic rooms the ideal is thought to be around 250-300ms, though there is more tolerance of extended decay times at low frequencies than high. Look for discussions about RT60, a measure of decay time (more relevant for large venues but often discussed in the context of domestic rooms also).



PeteD said:


> I guess could also be said that boost in particular should be performed at as wide a bandwidth as possible, with subsequent cuts to included peaks, instead of boosting individual dips - which is more likely to introduce ringing?


Right again


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## Wayne A. Pflughaupt

JohnM said:


> Ideally decay would be uniform across the band, for domestic rooms the *ideal is thought to be around 250-300ms,*


But at what amplitude would the signal start? And at what noise floor ? Those things dramatically affect the apparent rate of decay that a waterfall graph shows. Without a set guideline for signal amplitude and noise floor parameters, a 250-300 ms “ideal” is highly approximate and arbitrary. You can make a waterfall look "worse” by raising the SPL level of the measurement. You can make it look "better" by lowering the SPL level. 









*Baseline Measurement








Measurement SPL Level Increased








Measurement SPL Level Decreased*​

In the same manner, waterfalls can also be made to look worse or better by raising lowering the graph’s floor. I really don't like the arbitrariness of this whole thing..

And where do phase changes from filtering fit into the picture? I lifted this from Rane’s Exposing Equalizer Mythology note. The bolded text (emphasis mine) pretty much reads like what you’ve often stated here at HTS, except he’s talking about phase shift, not modal ringing. Comments?


_ Phase shift is not a bad word. That it has become a maligned term is most unfortunate. This belief stands in the way of people really understanding the requirements for room equalization.

The frequency response of most performing rooms looks like a heart attack victim's EKG results. Associated with each change in amplitude is a corresponding change in phase response. Describing them as unbelievably jagged is being conservative. Every time the amplitude changes so does the phase shift. In fact, it can be argued that phase shift is the stuff that causes amplitude changes. Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others.

An equalizer is a tool. A tool that allows you to correct for a room's anomalies. *It must be capable of reproducing the exact opposite response of the one being connected. This requires precise correction at many neighboring points with the associated phase shift to correct for the room's opposing phase shift*. It takes phase shift to fix phase shift. Simple as that.

One way people get into trouble when equalizing rooms is using the wrong type of equalizer. If an equalizer is not capable of adding the correct amount of phase shift, it will make equalizing much more difficult than it has to be. The popularity of the many constant-Q designs has come about because of this phenomenon. *Equalizers that produce broad smooth curves for modest amounts of boost/cut make poor room equalizers*, and good tone modifiers. They lack the ability to make amplitude and phase corrections close together. Lacking the ability to make many independent corrections with minimal interference to neighboring bands restricts their usage primarily to giving a shape to an overall response rather than correcting it. Serious correcting requires sharp constant-Q performance, among many other things.

*Only by adding many precise, narrow phase shift and amplitude corrections do you truly start equalizing a system's blurred phase response.* You do not do it with gentle smooth curves that lack the muscle to tame the peakedness of most rooms. Broad smooth curves do not allow you to correct for the existing phase shift. Its just that simple, you must pre-shape the signal in both amplitude and phase. And that requires narrow filters that preserve their bandwidths at all filter positions._

Regards,
Wayne


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## atledreier

That seems to somewhat contradict your general advice to use broad filters, Wayne. I use your advice with good results, but whatever works.. I'd like to know more about this.


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## JohnM

The amplitude and phase response of EQ filters is inextricably linked. What the passage says is use narrow filters to counter narrow effects, i.e. modal resonances. To counter a mode the corresponding filter needs to match it precisely, which is why smoothing should not be used when viewing responses with the aim of addressing room modes. The waterfall plot is a good indicator of whether a filter's bandwidth and centre frequency are correct to deal with a mode - it is difficult to do this with the initial response alone as the overlapping effects of the various modes can make it hard to separate them, as the waterfall progresses the strongest modes (which most need correcting) stand out as the response around them decays, making it easier to work out the filter settings needed to address them.

All that is not an argument against broader filters for response shaping, that is another tool in improving the overall response and the only one that can be usefully applied above about 200Hz as modal eq filters would only be effective over a very small area at such frequencies. Both should be combined with careful placement of speakers and listener to start with the smoothest response that can be practically achieved in the space and whatever room acoustic treatment can be accommodated, even sympathetic choice of furnishings, drapes etc can be very beneficial.


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## Wayne A. Pflughaupt

Okay, so basically phase shift and modal ringing are the same thing. So in essence what the Rane man is saying here...

_ In fact, it can be argued that phase shift is the stuff that causes amplitude changes. Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others. _

...is that change in modal ringing (phase) from an equalizer is no big deal. I agree. 




atledreier said:


> That seems to somewhat contradict your general advice to use broad filters, Wayne. I use your advice with good results, but whatever works..


I get the impression he’s mainly talking theory more than practical application, because...

_Only by adding *many precise, narrow phase shift and amplitude corrections* do you truly start equalizing a system's blurred phase response. _

...for one, I don’t know of anyone who makes an equalizer like that – i.e., allows for numerous highly-precise filters.

Second, I assume he would know that it would be impossible to achieve this for every location in a room.

Third, from what I understand, room modes as we know them in our little rooms are not a significant problem in the large rooms you typically see in Rane’s world.

Regards,
Wayne


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## Wayne A. Pflughaupt

JohnM said:


> To counter a mode the corresponding filter needs to match it precisely...


Am I the only one who has “repeatability” problems? A measurement I take say, today is a little different than what I got back in the summer, or even last week - different enough for REW to generate slightly different modal filters. Which are the precisely-perfect filters for the room mode?

Regards,
Wayne


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## terry j

Wayne A. Pflughaupt said:


> Am I the only one who has “repeatability” problems? A measurement I take say, today is a little different than what I got back in the summer, or even last week - different enough for REW to generate slightly different modal filters. Which are the precisely-perfect filters for the room mode?
> 
> Regards,
> Wayne


How much natural variation would you get between measurements over a period of time, given that the mic is probably not exactly in the same spot, small changes may or may not have been made in the room, different ambient conditions etc ?

I know the bass is not as twitchy as the mids and highs etc when the mic moves just a bit, but are the minor changes in the bass from different mic positions enough to account for what you are talking about?


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## atledreier

I'd say screw EQ for anything over deep bass, and treat the room!  That's what I do, and I'm very happy. I do one filter at 33Hz and that's it!


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## Blaser

atledreier said:


> I'd say screw EQ for anything over deep bass, and treat the room!  That's what I do, and I'm very happy. I do one filter at 33Hz and that's it!


You are one lucky man! :T


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## JohnM

Wayne A. Pflughaupt said:


> Am I the only one who has “repeatability” problems? A measurement I take say, today is a little different than what I got back in the summer, or even last week - different enough for REW to generate slightly different modal filters. Which are the precisely-perfect filters for the room mode?


Filter settings are likely to need tweaks to get the best match, the waterfall plot is helpful in assessing what tweaks are improving the result. For good results the centre frequency needs to be within 1% of the mode's frequency, the bandwidth settings is less sensitive. Once the right values have been found they won't change unless the room is changed in some way (altering the modes), or the sub is moved, or you measure at a different position.


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## Wayne A. Pflughaupt

JohnM said:


> Filter settings are likely to need tweaks to get the best match, the waterfall plot is helpful in assessing what tweaks are improving the result. For good results the centre frequency needs to be within 1% of the mode's frequency, the bandwidth settings is less sensitive. *Once the right values have been found they won't change unless the room is changed in some way (altering the modes),* or the sub is moved, or you measure at a different position.


Sure, the room isn’t going to change. However, the transducers involved – the speakers and elements in the measurement microphones - are not as stable as the room or the REW platform. Their physical (and consequently electrical) properties are altered with changes in temperature, humidity, etc. 

As such, when you take a second REW reading six months or a year later you'll find it doesn't look quite like your original one. A waterfall graph generated today with last year's filters isn't going to look as good as it did back on the day you fine-tuned the filters for minimal ringing.

As an example, here is a graph that shows three readings I took at two-month intervals. Same location, the only change was the time of year and consequently the room temperature and humidity. Which graph should I generate the precise modal parametric filters for? Should I re-do this excercise and re-equalize every few months???









*Three REW readings, taken at two-month intervals*​

Naturally, even though the graphs read different, there was no perceptible difference in the way my subs sounded. Which is why I can’t help but question the relevance of the whole “modal equalizing” theory.

This next graph shows the effect of temperature on the performance of the measurement mic. It was a cold winter day (for Houston at least), and it was 73˚ inside the house. After taking a measurement, I set the SPL meter on the floor where sunlight was streaming in a window. The thermometer registered 86˚ in the sun. After a half hour or so I took another reading. The red trace is the reading from the warmed-up mic.









*Two REW readings, mic element at 73 vs. 86 degrees*​

So again, which modal filters would be correct? The ones generated for the 73˚ mic, or the 86˚?

Is there any particular reason why these graphs are displayed in a 300 ms window and not some other value?

Regards,
Wayne


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## JohnM

Wayne A. Pflughaupt said:


> Is there any particular reason why these graphs are displayed in a 300 ms window and not some other value?




That's just a convenient starting point that is often suitable for looking at low frequency behaviour, can dial in whatever window width and time span you find appropriate, bearing in mind that shorter windows decrease frequency resolution (the resolution is shown next to the window control).


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## PeteD

I have a few more questions:

1. When I generate LF waterfalls for a given response, the starting SPLs at a given frequency are higher than appear in the LF decay and filter adjust tabs. Is this correct and why is this?

2. About waterfalls in general, do higher SPLs generate slower decay/more ringing with all else being equal? (i.e. Does RT-60 change with decibel level?) It seems like "yes", as there is more energy in the room that must be dissipated. If "yes", what is the ideal SPL to test at to examine decay in order to see if ringing is present? (see question 3).

3. To look at RT-60, wouldn't you have to test at levels greater than 60 dB above the room's background noise (i.e. very loud, since a quiet home is considered 40 dB)?

Lastly, I am interested to hear comments on this quote (link below) from Richard Bird of Rives Audio:
_"Lastly, let’s take a look at our listening rooms. Listening rooms, have short reverberation times relative to our church, but long reverberation times relative to the control room mentioned above. Reverberation times for listening rooms vary depending on listener preferences including listening levels, and whether or not it is designed for multi-channel or 2 channel use. This is also an area where we can scientifically measure the reverberation times, but it’s somewhat of an art to get the reverberation times correct for a particular listener—not unlike the differences in studios—what are the goals and what do we want to achieve. Some people feel that the listening room should not interact with the speakers, and it should perform much like the control room. Having listened in rooms designed like this, I can tell you I completely disagree. It takes away the kinetic energy of the music and having no interaction with the room has a very unnatural sound. In general we would like to achieve something between 0.34 and 0.39 seconds for an RT-60 from about 200 Hz on up. RT-60 measurements in small room acoustics below 200 Hz have little meaning and are flawed by the energy buildup of room modes. To evaluate bass response below 200 Hz, we have to use other methods."

Here is the reverberation time for an ideal 2 channel listening room:[see link for graph]_

http://www.positive-feedback.com/Issue12/rives2.htm

I will definitely be doing some more reading on this stuff...

Thanks,
Pete


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## atledreier

Regarding #2, I think the -60 part means the time it takes for the spl to reach -60dB compared to your initial output, so it shouldn't vary with level. The time it takes to reach the noise floor will vary with level, though.


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## JohnM

PeteD said:


> 1. When I generate LF waterfalls for a given response, the starting SPLs at a given frequency are higher than appear in the LF decay and filter adjust tabs. Is this correct and why is this?


No, that's not correct. The waterfall is often generated using a shorter window than that used for the response on the Filter Adjust graph, which reduces its frequency resolution and hence peaks that are very close to one another might merge into one taller peak, but the overall level of the response should be the same. 



PeteD said:


> 2. About waterfalls in general, do higher SPLs generate slower decay/more ringing with all else being equal? (i.e. Does RT-60 change with decibel level?) It seems like "yes", as there is more energy in the room that must be dissipated. If "yes", what is the ideal SPL to test at to examine decay in order to see if ringing is present? (see question 3).


No



PeteD said:


> 3. To look at RT-60, wouldn't you have to test at levels greater than 60 dB above the room's background noise (i.e. very loud, since a quiet home is considered 40 dB)?


No. RT60 is generated from analysis of the impulse response, the S/N of the impulse response can be much higher than the S/N of the room through the process gain of the method used to generate it. For example, each doubling of sweep length adds approx 3dB to the S/N of the impulse, as does each doubling of the number of averages. To generate an RT60 figure the impulse response S/N needs to be better than 65dB since the decay is based on the decay of the response over the range from -5 to -65dB, if the S/N is lower than that then a reduced range can be used, e.g. RT30 uses -5 to -35dB, RT20 uses -5 to -25. In all cases the figure being estimated is the time it would take for sound to decay by 60dB but it is based on the measured decay over different ranges of the impulse response.



PeteD said:


> Lastly, I am interested to hear comments on this quote (link below) from Richard Bird of Rives Audio:
> _"Lastly, let’s take a look at our listening rooms. Listening rooms, have short reverberation times relative to our church, but long reverberation times relative to the control room mentioned above. Reverberation times for listening rooms vary depending on listener preferences including listening levels, and whether or not it is designed for multi-channel or 2 channel use. This is also an area where we can scientifically measure the reverberation times, but it’s somewhat of an art to get the reverberation times correct for a particular listener—not unlike the differences in studios—what are the goals and what do we want to achieve. Some people feel that the listening room should not interact with the speakers, and it should perform much like the control room. Having listened in rooms designed like this, I can tell you I completely disagree. It takes away the kinetic energy of the music and having no interaction with the room has a very unnatural sound. In general we would like to achieve something between 0.34 and 0.39 seconds for an RT-60 from about 200 Hz on up. RT-60 measurements in small room acoustics below 200 Hz have little meaning and are flawed by the energy buildup of room modes. To evaluate bass response below 200 Hz, we have to use other methods."_


_I'd agree that a very "dry" (fast decay time) room is not good for music, though it can work well for movies. I also agree that for low frequency behaviour RT60 measurements are meaningless._


----------



## PeteD

Thanks JohnM. I see what you are saying. I thought maybe there was a minimum SPL needed to excite the rooms resonances, so variation might be seen with different SPLs. It may be more of a background noise issue (furnace, washer, dryer, etc.), in what few graphs I have generated.

Regarding number 1, please look at the graphs below. These show filtered response from 10 to 200 Hz on the same scale. I generated the LF waterfall from the frequency response. Why do the SPL in waterfall look higher?



















Below what frequency are you less worried about resonance?

Thanks,
Pete


----------



## brucek

> Why do the SPL in waterfall look higher?


Because you have used a different impulse response post window for the two graphs. The frequency response graph is controlled by the IR Windows popup ICON. Click it and look at the right window resolution in msec. Then look at the Waterfalls TAB and see the Windows resolution in msec used there. They will be different. I belive the default for response is 500msec and Waterfall uses 300msec. Change them if you like...

Increase the time and see the Frequency resolution change (gets smaller). You can change the Window on the frequency response popup and hit Apply Windows and you'll see your response graph change.

brucek


----------



## PeteD

Hi Brucek.

You were right about the settings, but the SPL still look different. Maybe I am missing something else?

Here they both are with right window at 300ms (the target line has been manually adjusted, so it may not be the same):



















What setting is recommended for adjusting filters?

Thanks,
Pete


----------



## JohnM

PeteD said:


> Regarding number 1, please look at the graphs below. These show filtered response from 10 to 200 Hz on the same scale. I generated the LF waterfall from the frequency response. Why do the SPL in waterfall look higher?


Because you are not allowing for the perspective of the waterfall plot. The SPL axis markings on the LHS of the graph are for the frontmost slice of the waterfall, if you look at the lines on the back wall of the waterfall you can see the 96 and 88dB lines and the rearmost slice lines up between them as it should. You can also use the slice slider to move the rearmost slice to the front to more easily see the actual levels, or change the perspective settings via the "More waterfall controls..." panel.



PeteD said:


> Below what frequency are you less worried about resonance?


I don't understand that question. Modal resonances are most significant at low frequencies, below about 200Hz. At higher frequencies they are still present but there are so many their individual effects are no longer easily resolved and dealing with them requires room treatment.


----------



## brucek

> Maybe I am missing something else?


As John says, move the 1-30 slider on the Waterfall plot over to 1 and you will have identically response charts.....

Perhaps best you start a new thread in the REW section to discuss your filters etc..... it will get more play there.. 

brucek


----------



## PeteD

JohnM said:


> Because you are not allowing for the perspective of the waterfall plot.


I knew it had to something very simple!


----------



## PeteD

JohnM said:


> I don't understand that question. Modal resonances are most significant at low frequencies, below about 200Hz. At higher frequencies they are still present but there are so many their individual effects are no longer easily resolved and dealing with them requires room treatment.


Sorry for the very poorly worded question. I was looking for you to expound in this statement:

_"I'd agree that a very "dry" (fast decay time) room is not good for music, though it can work well for movies. I also agree that for low frequency behaviour RT60 measurements are meaningless."_

How do we use these waterfall plots, especially at low frequencies if the RT60 become meaningless?

Also, regarding your point on rapid decay being OK for home theater (this is because the surround effect gives ambience I presume). I think one could argue that movie sound track VLF tend to less discrete than music in general and are not as negatively impacted negatively by reverberation at low frequencies. Also, most of the VLF comes from the subwoofer, not from the surrounds, so wouldn't reverb in the low frequencies match better with the surround ambience in the higher frequencies? Total conjecture on my part...

Thanks,
Pete


----------



## PeteD

brucek said:


> Perhaps best you start a new thread in the REW section to discuss your filters etc..... it will get more play there..
> 
> brucek


I am going there soon, I want to get some readings with less background noise, first. I was just playing around yesterday to generate the plots above.

Pete


----------



## brucek

> I want to get some readings with less background noise, first


Use the multiple sweeps (up to 8) and longer sweeps (up to 1M) in the Measure screen to reduce noise considerably.

brucek


----------



## JohnM

PeteD said:


> How do we use these waterfall plots, especially at low frequencies if the RT60 become meaningless?


The waterfall plots don't show RT60, they show the actual decay of each frequency across a band, which is what we need to see at low frequencies. RT60 is a single figure that summarises the decay across the entire band or some section of it (octave bands are fairly common), but a single number doesn't give any information about which specific frequencies are decaying slowly and which are decaying normally.


----------



## DrWho

Hey Pete, are those graphs showing the behavior after EQ?


----------



## PeteD

Hi Mike:

Yes, I quickly put together 5 or 6 filters, including +7 dB of boost at 20Hz, with a bandwidth of 120/60 - to compensate for cuts in the upper bass region, necessitated by a -12dB/octave x-over on my receiver and the fact that I have the same AE-IB -15s (4 in an IB, Fs of 32Hz, stiff spiders) that Chris Bee had before his recent upgrade. Without some boost, my receiver output is too low.

I will post some additional sweeps in a separate thread soon - my furnace is right behind my front wall/IB and it was on during measurements, along with some other stuff.

I am not sure of the relevancy of any readings from the RS meter below 10Hz, but using the curve (according to date stamp data) from the Shack which has factors down to 7Hz, I have output down to 2Hz (96 dB) on the curves shown above, with a dip at 5Hz.

Pete


----------



## spudbudy

*waterfalls*

hi gang just looking for a little advice on my recent setup of my BFD with the filters and a buttkickers with mains all in play.


----------



## Guest

ok, i'm new to this forum and hopefully this won't be a dissasterous post. i'd like to chime in on the idea of correcting a room via eq rather then acoustic treatments.

i know it's been a while but i started reading this post at the beginning.

my thinking goes like this. say you have a room mode at 60hz of +25db. you correct his with eq. now what do your ears hear?

if you are about 7 ft from your speakers you should first hear a direct signal. right? takes about 7ms to get to you since sounds travels about 1 ft/ms. so the first thing you hear is a signal with a major deficit at 60hz due to corrective eq.

after the initial direct signal all the other reflections come in. so for a mode you need the wave to go across the room and bounce back over itself. right? constructive interference (as i was taught). for 60hz we are talking about a room around 10ft across. so for the first standing wave to occur it should take longer then 7ms. maybe 20ms to hit a wall and come back.

ok, so you hear -25db at 60hz for the first 13ms. then the room mode kicks in and things are corrected.

seems like the same thing would happen on the way out as well. in other words the bass would be 13ms late arriving at its proper level and 13ms late decaying back down. think about this on something like a kick that is transient. i can tell you little things like 13ms on a kick are noticable. 

13 ms might not seem like a long time but it sounds like one. just try putting 13ms delay on one channel and listen.

thats why i can't go for the the total eq solution. it's probably fine for minor stuff but not 20-30bd corrections. that just won't sound the same.

there are a few studio construction books (jeff cooper, f. altom everest) as well as bunch of recording pros that support this view.


----------



## Wayne A. Pflughaupt

Welcome to the Forum, Chris!

That's an interesting perspective, and it certainly seems logical. Perhaps our "salvation" is that I've never seen anyone with a 25-30 dB room mode, and we've seen hundreds of in-room measurements on this Forum. Have you been able to discern some delay when using an equalizer to tame a room mode?

Regards,
Wayne


----------



## brucek

Well, EQ is generally used where the room is dominated by the room's modal response. It certainly is only effective at the point of measurement (which would be the main listening postion). If a filter is applied that is the same Q and the opposite gain, then that resonance would be eliminated at that listening position. That's basically it.

brucek


----------



## JohnM

Hi Chris,

The direct sound is affected roughly as you describe, which is why EQ'ing the direct path is rightly frowned upon if not correcting for anomalies in the reproduction of the speaker itself, but you are forgetting the frequency you are considering. At low frequencies we are unable to perceive such short duration effects, our ears detect the total energy integrated over much longer periods than the time it takes sound to travel the length of a domestic room. The transient effects you detect in a kick drum, for example, are in the higher frequency content of the spectrum of that sound, not in the fundamental, and those higher frequencies which determine our perception of the transient are not being altered by the EQ filter. What is being corrected is the total energy which reaches us, which restores our perception of a balanced sound.


----------



## Guest

ok, maybe not a 30db mode. more like a +15 db at 60hz and a -15 db at 120hz. i guess it would be better to say unbalanced. that's what i've read and i can measure in my own room. sad.

as far as the low freq stuff goes. well, i don't know. i've moved a kick 20 ms and noticed a difference in the performance. i don't feel like my ear is doing an averaging at low frequencies. 

if that were true (at low frequencies we can't easily detect transients) i wouldn't be able to here a truncated kick (7ms long) with a high pass killing everything above 100hz. i know i can. i just did it.

confusing mr. bigglesworth. and i just spent 1k on room acoustic treatment. 

i feel like these arguments are coming out of 2 camps. one is theoretical and one is about having critical listening skills. i have a small foot in both i suppose.


----------



## Guest

and i must confess. i have never used eq to tame a mode (they are wild misundertsood beasts that must run free). maybe i should try before my acoustic treatments arrive. then post that.


----------



## JohnM

cporro said:


> if that were true (at low frequencies we can't easily detect transients) i wouldn't be able to here a truncated kick (7ms long) with a high pass killing everything above 100hz. i know i can. i just did it.


I have no idea how you arrive at the conclusion that the energy integrating aspects of hearing at low frequencies should make you deaf.



cporro said:


> i feel like these arguments are coming out of 2 camps. one is theoretical and one is about having critical listening skills. i have a small foot in both i suppose.


Chris, this is not being done as an academic exercise. If I didn't care passionately about good sound I wouldn't have spent the last several years writing analysis software to help people improve the sound they get from their systems and making the software available for free. I don't believe there is any conflict or incompatibility between striving for the best sound quality and striving to understand the acoustic principles and effects that contribute to it, quite the contrary.



cporro said:


> i have never used eq to tame a mode (they are wild misundertsood beasts that must run free). maybe i should try before my acoustic treatments arrive. then post that.


Absolutely, we are great believers in measurement here.


----------



## Guest

not trying to give you a hard time. i'm one of those types that needs to reason things through or else i don't feel i know them. i start out with what i know (or think i know) and run through the scenario. if it doesn't add up i start asking questions like columbo.

btw, much much thanks for this useful program. i am just starting to get my head around it. right now i am loving the waterfalls. seems just about everything i need is in that one. dig how i can see the the issues with the speaker transform into issues with the room


----------



## parsley

Am I right in thinking that a waterfall plot helps to differentiate the response of the speakers (the first or early slices) from that of the room (the later slices)?

Mind you, for low frequencies e.g. 30 Hz where wavelength = 11.3 m and one cycle = 33 ms, is it possible to pick up the direct sound from the speaker and correctly identify its amplitude and frequency before the room response affects things at the measurement point?

Finally, in some of the waterfall plots shown previously (e.g. on page 2 of this thread) there seemed to be examples of the energy at a given frequency _increasing_ a small amount with time. Is this likely to be a measurement/analysis artifact, or background noise, or non-linearity (e.g. something starts shaking and re-radiating energy at different frequencies)?

Thanks for the great software.


----------



## brucek

> Am I right in thinking that a waterfall plot helps to differentiate the response of the speakers (the first or early slices) from that of the room (the later slices)?


No, the room is very much a part of the first slice (which is the same plot as the frequancy response) since the impulse response is gated (windowed). The next 29 slices are the decay of the signal over time in the room at the listening position. If you want the simple response of the speaker without the room influence, it's best to do a near-field away from boundaries (or go outside for the test).

brucek


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*

*EDIT by Wayne Pflughaupt:

The next few posts (#97-#111) are some relevant posts that have been moved here from another thread that got sidetracked. They have been edited as needed to maintain relevancy to this discussion.*



On another note, the love many people seem to have for DSP and room EQ is curious to me. Before I start, I must say that DSP and EQ can be a useful tool to setting up a system, but can be very misleading when used as a "be all end all" for deciding what sounds good.

First off, the goal, at least for me, and I would think the goal would be for most, is to recreate sound as close to what the original artist had intended to be heard. In this case the best scenario in a playback system is a completely transparent, uncoloured sound. Now I realize that not everyone has the means to design a system that is capable of this ideal, but the thought that this can be achieved with a mediocreor poor system and some EQ and DSP is severly flawed. 

Now if your goal is not to recreate the most accurate representation of the source material, but to alter it to suit your aesthetic, such as having huge amounts of bass coming from your car system where you listen to hip hop, none of this concerns you.

Instead of writing out my argument myself I'm copying an explanation that is far better written that something I could compose, and states the issue much more clearly than I could. Note: I'm not referring to DSP used in crossover design, or creative desicions such as reverb. This is only reffering to the use of DSP and EQ to correct deficiencies in poor quality, audio equipment, speakers and rooms.

Here's the article. In particular I'm referring to section 3.
http://sound.westhost.com/articles/dsp.htm#s40


You cannot correct Time with Amplitude.


----------



## tonyvdb

*Re: Is there a real benefit to preamps or two channel amps in HT?*



macrae11 said:


> On another note, the love many people seem to have for DSP and room EQ is curious to me. Before I start, I must say that DSP and EQ can be a useful tool to setting up a system, but can be very misleading when used as a "be all end all" for deciding what sounds good.
> 
> Now I realize that not everyone has the means to design a system that is capable of this ideal, but the thought that this can be achieved with a mediocreor poor system and some EQ and DSP is severly flawed.


You make some good points in your above post my only comment is on this section. You have to remember that an EQ or the so called room correction system is to get the sound coming out of the speakers in a room to sound like the original recording as close as possible. The problem is that room acoustics severally effects the sound, a "perfect room" should need little correction of the sound and a flat response is what you would get "if" the speakers are designed properly without any color. The issue is this is rarely the case. The other problem is that some recording studio's dont follow the rules and tweak there system so it is not flat to begin with and adds color to the sound even before it is mastered.


----------



## Wayne A. Pflughaupt

*Re: Is there a real benefit to preamps or two channel amps in HT?*




macrae11 said:


> You cannot correct Time with Amplitude.


Assuming Rod Elliot was referring to equalization, we get a contrasting (conflicting?) view from Rane’s Exposing Equalizer Mythology by Dennis Bohn (bold emphasis added):

_“Phase shift is not a bad word._ It is the glue at the heart of what we do, holding everything together. That it has become a maligned term is most unfortunate. This belief stands in the way of people really understanding the requirements for room equalization.

Associated with each change in amplitude is a corresponding change in phase response. Describing them as unbelievably jagged is being conservative. Every time the amplitude changes so does the phase shift. In fact, it can be argued that phase shift is the stuff that causes amplitude changes. *Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others.*”​
I think what Mr. Elliot failed to consider is that equalizers also introduce phase changes, which probably accounts for how it’s usually possible to EQ phase-related response problems around a sub’s crossover region.

Overall though, that was a very good article. Thanks for linking it. :T I especially liked the part about mics not "hearing" the way our ears do. That's probably why you don't get much of an audible improvement equalizing subs beyond smoothing out the worst problems. I.e., piling on lots of "minutiae" filters smoothing out every little ripple in response you probably won't be able to hear the difference with them in or out. I sure can't. 

Regards,
Wayne


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*



tonyvdb said:


> You make some good points in your above post my only comment is on this section. You have to remember that an EQ or the so called room correction system is to get the sound coming out of the speakers in a room to sound like the original recording as close as possible. The problem is that room acoustics severally effects the sound, a "perfect room" should need little correction of the sound and a flat response is what you would get "if" the speakers are designed properly without any color. The issue is this is rarely the case. The other problem is that some recording studio's dont follow the rules and tweak there system so it is not flat to begin with and adds color to the sound even before it is mastered.


Yes unfortunately as was in the article I posted, EQ can never fix room deficiencies. A "perfect room" should need no correction, regardless of the speakers used. However there are precious few of these rooms in the entire world. I can only think of one of the top of my head which might even be considered. 

As far as the recording studios, changing the response of their monitoring systems, this is generally frowned upon, and rarely happens. More often is the case that perhaps the studio can't afford proper monitoring or room treatment. Either way it is rather irrelevant as this is the point of creation. If the artist got things how he wanted to sound in that room and it translates well to other systems than the project, and the room that it was created in was a success. Proving again, that in the creation of art, there are no rules.



DS-21 said:


> Except in the bass, as the article you cited notes, which is why pretty much everyone here understands and uses parametric EQ on their subwoofers. But that fact is precisely why the really interesting room correction DSP software (Audyssey, Meridian Room Correction, TacT/Lingdorf, etc.) all work both in the time and frequency domains. Moreover, they do so taking after taking measurements from multiple locations within the putative listening area of the room. So Rod Elliot's excellent article, while entirely correct at the time he wrote is, has fallen a little behind the times.


I'm not familiar with how these softwares could correct in the time domain to correct a room node that cause a(for example) low D to sound twice as long as it sounds in the program material.

I will research it a bit more, but if you have a simple explanation that could enlighten me, I would welcome it.


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*



Wayne A. Pflughaupt said:


> Assuming Rod Elliot was referring to equalization, we get a contrasting (conflicting?) view from Rane’s Exposing Equalizer Mythology by Dennis Bohn (bold emphasis added):
> 
> _“Phase shift is not a bad word._ It is the glue at the heart of what we do, holding everything together. That it has become a maligned term is most unfortunate. This belief stands in the way of people really understanding the requirements for room equalization.
> 
> Associated with each change in amplitude is a corresponding change in phase response. Describing them as unbelievably jagged is being conservative. Every time the amplitude changes so does the phase shift. In fact, it can be argued that phase shift is the stuff that causes amplitude changes. *Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others.*”​
> I think what Mr. Elliot failed to consider is that equalizers also introduce phase changes, which probably accounts for how it’s usually possible to EQ phase-related response problems around a sub’s crossover region.
> 
> Overall though, that was a very good article. Thanks for linking it. :T I especially liked the part about mics not "hearing" the way our ears do. That's probably why you don't get much of an audible improvement equalizing subs beyond smoothing out the worst problems. I.e., piling on lots of "minutiae" filters smoothing out every little ripple in response you probably won't be able to hear the difference with them in or out. I sure can't.
> 
> Regards,
> Wayne


You are correct, but I belive Mr. Elliot was thinking on a slightly different line. Of course when dealing with EQ's amplitude and phase are related(with the exception of a linear phase EQ). In the real world(no EQ) they are not so incontrovertably linked. You can have the identical phase of two signals with two completely, even opposite amplitudes. Also the time which Mr. Elliot and yourself are referring to are on different scales. a 10 degree phase shift at 80 Hz (wavelenghth of 4.2m) takes a little over a 1000th of a second. Room nodes can cause differences in time of an 80Hz note of a second or more. So an EQ(even a very poor "phasey" EQ) can come nowhere close to adjusting that kind of time.

PS. Someone please correct me if my math is wrong, it's been quite some time since I've had to work out any formulas.:huh:


----------



## Wayne A. Pflughaupt

*Re: Is there a real benefit to preamps or two channel amps in HT?*




macrae11 said:


> I'm not familiar with how these softwares could correct in the time domain to correct a room node that cause a(for example) low D to sound twice as long as it sounds in the program material.
> 
> I will research it a bit more, but if you have a simple explanation that could enlighten me, I would welcome it.


From the little I’ve seen, they don’t really have much effect on time domain. I haven’t seen any comprehensive in-room testing, other than Ethan Winer’s EQ vs. Bass Traps extravaganza. The test has some problems, though, mainly that the sheer number of measurements to sift through is mind numbing, and the guy who set up the equalizer did a really, really bad job of it. 

Ethan also did an evaluation of the effectiveness of the Audyssey MultEQ DS-21 mentioned, which claims to reduce extended signal decay (aka ringing). His conclusion was that it was a mixed bag as far as dealing with ringing was concerned.

Since there seems to be a dearth of any comprehensive in-room testing, other than these two (which are the only ones I’m aware of) I’ve done some of my own, although I haven’t explored it fully or posted any results here at the Shack. After studying waterfalls ‘til I’m bleary-eyed, I’ve found pretty much the same thing as Ethan, that it’s a mixed bag using an equalizer as a fix for extended low frequency room decay.

From what I’ve seen, the improvement waterfalls show for modal (or “time-domain”) EQ filtering has only been apparent in the short-duration 300 ms window. When the window is lengthened to 600 ms, any advantage modal filters showed over other equalizing techniques pretty much vanished. 

Here is a sample to show what I’m talking about. The Hz markers represent peaks that REW found, and the “Modal Filters” waterfalls have REW-recommended filters set at those frequencies (except for the red 49 Hz marker). The “Smoothing Filters” graphs show the result of filtering that ignored REW’s recommendations and used four filters that merely smoothed response. Notice that this “faster decay, but only up to a point” effect is especially noticeable for the 26.9 and 23.5 Hz markers. As you can see, once the window is lengthened to 600 ms, any apparent advantage of modal filtering over smoothing virtually disappears. (Hit F11 and you should be able to get all four graphs on your screen at the same time.)















Modal Filters (Left) vs. Smoothing Filters (Right) @ 300 ms















Modal Filters (Left) vs. Smoothing Filters (Right) @ 600 ms​

You can see the same effect in Ethan’s “with and without Audssey” comparison graph. Notice the broad peak between 20-56 Hz, where Audyssey seems to improve the rate of decay in the short term, but ultimately does not reduce or eliminate it in the long term. Also note that even significantly reduced gain of the same peaked area effected by the equalizer did not reduce the ringing.










Regards,
Wayne


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*



Wayne A. Pflughaupt said:


> From the little I’ve seen, they don’t really have much effect on time domain. I haven’t seen any comprehensive in-room testing, other than Ethan Winer’s EQ vs. Bass Traps extravaganza. The test has some problems, though, mainly that the sheer number of measurements to sift through is mind numbing, and the guy who set up the equalizer did a really, really bad job of it.
> 
> Ethan also did an evaluation of the effectiveness of the Audyssey MultEQ DS-21 mentioned, which claims to reduce extended signal decay (aka ringing). His conclusion was that it was a mixed bag as far as dealing with ringing was concerned.
> 
> Since there seems to be a dearth of any comprehensive in-room testing, other than these two (which are the only ones I’m aware of) I’ve done some of my own, although I haven’t explored it fully or posted any results here at the Shack. After studying waterfalls ‘til I’m bleary-eyed, I’ve found pretty much the same thing as Ethan, that it’s a mixed bag using an equalizer as a fix for extended low frequency room decay.
> 
> From what I’ve seen, the improvement waterfalls show for modal (or “time-domain”) EQ filtering has only been apparent in the short-duration 300 ms window. When the window is lengthened to 600 ms, any advantage modal filters showed over other equalizing techniques pretty much vanished.
> 
> 
> Regards,
> Wayne


Thanks for that Wayne. I still can't say that I fully understand the system before I do my own in depth research, but this certainly clears things up.

I also think it somewhat proves my point. The DSP system can lower the initial output of the trouble frequencies, but still does not affect the longer term effects of the room.


----------



## Wayne A. Pflughaupt

*Re: Is there a real benefit to preamps or two channel amps in HT?*




macrae11 said:


> Thanks for that Wayne. I still can't say that I fully understand the system before I do my own in depth research, but this certainly clears things up.
> 
> I also think it somewhat proves my point. The DSP system can lower the initial output of the trouble frequencies, but still does not affect the longer term effects of the room.


I agree. Some of my esteemed colleges here (and on other Forums) are believers in time-domain equalization, but they haven’t posted or otherwise offered any evidence to back it up.



> Again I think it's great that perhaps these DSP systems sound great to you. I might like them as well. All I'm saying is that they're not correcting issues in your room, and they are colouring the original signal in it's attempts to do so.


I think it would be more accurate to say that they’re attempting to correct the room’s _effect_ on the speaker’s response, coupled with the natural response of the speaker itself (i.e. equalizing above ~500 Hz).

Regards,
Wayne


----------



## brucek

*Re: Is there a real benefit to preamps or two channel amps in HT?*

*macrae11 says:*


> I also think it somewhat proves my point. The DSP system can lower the initial output of the trouble frequencies, but still does not affect the longer term effects of the room.


*Wayne A. Pflughaupt says:*


> I agree. Some of my esteemed colleges here (and on other Forums) are believers in time-domain equalization, but they haven’t posted or otherwise offered any evidence to back it up.




Guys,

Just as a parametric filter operates in the frequency domain, it also has a time response that acts like a modal resonance in a room.

It's easily demonstrated (see below) by doing a frequency response sweep of a Behringer parametric equalizer and adding a single filter. 

The time tail is evident, is it not?

I don't think anyone can argue that this filter is also operating in the time domain?

*Waterfall plot plot of a BFD using a single filter of (40Hz, Gain +15dB, BW 10)*









From that thread I noted the following:

_But now I look at the resulting waterfall plot of that single filter below. 

Look familiar? Sure it does. 

It looks like a room mode resonance of any REW measurement at subwoofer frequencies. And it should. It's because the EQ filter, just like the modal resonances of a room, has a time response that acts like a 2nd order biquad. If I apply an EQ filter with the same Q and opposite gain of a room mode, I would completely counteract the effect of the mode. See the time component of the filter (just like a room mode). It rings out, and still isn't in the noise after 300msec. You see, EQ filters don't just affect level. This is why they're so effective at equalizing at modal frequencies below 100Hz. Yes, it is listening position dependant, and only valid at the point where the response was measured, but because of the long wavelengths of low frequencies, the region around that area is fairly large. This is in opposition to higher frequencies where equalization is a bit of a waste of time, since the effective region is so small that eq is impractical.
_

I remember a post (from this thread) by JohnM (author of REW) some time ago that was the trigger that really started me thinking about this subject and after lots of playing around with my equipment and investigating, I'm a believer (one of the few, it would seem).

John said:
_This is a fundamental misunderstanding of minimum phase systems. Room modes (in particular those below a few hundred Hz) are minimum phase phenomena (substantially behaving as 2nd order biquads) and are effectively countered by 2nd order biquad correction filters such as those implemented in the BFD pro and many similar parametric equalisers. It is, however, unnecessary to engage in lengthy academic discourse on the topic since measurements clearly show the effectiveness of the approach._

brucek


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*

Thanks for your input Bruce. This thread has certainly gotten off topic from the OP, and it's mostly my fault I'm afraid. I apologize and I hope no one minds, as I think there is some good discussion going on here.

To get back to you Bruce, I'm assuming by the looks of your waterfall graph that you used a broadband white noise burst to create this waterfall measurement.
Here's my thought:
You're not extending the time that the 40Hz signal is present for. All you are doing is making the starting point louder, so more of the signal is heard before it falls into the noise floor. The decay curve appears to be identical, just a higher output, which shows that you haven't actually changed any time issues.

Now certainly it is easy to eliminate a filter boost by a subsequent identical but reversed filter cut. It might even sound identical to not using a filter at all, if it is a high quality digital low phase EQ. But this isn't how things work in the acoustic world.

If you were to introduce that same cut filter into program material going into a room, you would lower the initial attack of that note in the room, and the decay would then take the same amount of time for the sound level to decrease. But since the level of that frequency started at a lower point, it will fall into the noise floor sooner, having it be not audible at the correct time as if the room wasn't playing a factor. So yes the note will appear to be shorter but only because it started at a lower volume.

One thing that I think people forget is that the first instant of sound you hear coming from the speakers is not affected by the room. It is affected by how far away you are from the speakers, and the room temperature, moisture etc, but it has nothing to do with the walls in your room. Hence if you're EQ'ing for the room, you are also EQ'ing the direct signal. So that first attack of sound you hear is going to have a 15 dB cut at 40Hz.

Now one thing I will concede is that in a non severe instance of this, very few filters doing very moderate attenuation, with a very narrow Q, it is possible that this will be inaudible to the listener, and may improve the perception of ringing in the room. This is however, still a band-aid solution to a surgery problem.



Having said all, this I'll give a personal disclaimer: It's been several years, since I've really done much acoustic design, and I may be a bit out to lunch. However the laws of physics haven't changed, at least not to my knowledge, and I don't think we're at a point where we can throw the acoustic treatment out the window yet. At least until there is a way to EQ the room reflections separately from the direct signal.


----------



## brucek

*Re: Is there a real benefit to preamps or two channel amps in HT?*



> I'm assuming by the looks of your waterfall graph that you used a broadband white noise burst to create this waterfall measurement.


Room EQ Wizard - REW 

brucek


----------



## Guest

*Re: Is there a real benefit to preamps or two channel amps in HT?*

Hi Bruce

Sorry I knew what software you were using, I was just wondering what type of signal was used to create the initial flat waterfall. After reading some more about REW it seems it uses a logarithmically swept sine wave.

Thanks


----------



## Wayne A. Pflughaupt

*Re: Is there a real benefit to preamps or two channel amps in HT?*



brucek said:


> I don't think anyone can argue that this filter is also operating in the time domain?


Sure we can. It’s impossible to make a case from an electronically generated, faux "room" mode that displays _amplitude only_ but has no time domain element from reflections, boundary interference, etc. For instance, is it going to look different if you install some bass traps in the room? :huh: It's merely a computer-generated loop that has no real-world significance. 

Since I unfortunately did not save the REW files I used when I ran this experiment a while back, I decided to do it over and this time use a longer window for the waterfall. This time I let REW do the filtering and two modal filters (i.e. optimized for time domain) for my 42 Hz room mode were crafted, the primary one being centered at 42 Hz with a 10/60 bandwidth and cut 12 dB. 

As before, a 300 ms window showed the same short-term reduction in ringing with modal filtering. But at 600 ms – different story. Here are the results with the 42 Hz peak level matched after equalization (as described in the linked thread). (Hitting F11 should get them both on-screen for easier comparision).
















Baseline (Top) vs. Modal Filters, Level-Matched (Bottom) @ 600 ms​

In fact, even with _no_ level matching, we can see that modal filters made only a minor difference in reducing the 42 Hz mode's long-term ringing:
















Baseline (Top) vs. Modal Filters , No Level Matching (Bottom) @ 600 ms​

If you want to convince us otherwise, show us your in-room waterfalls (long window, please). IMO Ethan’s Audyssey report puts the matter to bed, unless someone can show a fundamental problem with his methods or procedures. As his chart above shows, Audyssey did not reduce ringing. I'm sure my experiments aren't as sophisticated as his, but I'm getting the same results every time.

Regards,
Wayne


----------



## atledreier

*Re: Is there a real benefit to preamps or two channel amps in HT?*

Even though the 'evidence' here suggest this and that, the reality is that good EQ in the time domain, like Audyssey and RoomPerfect (Both of which I've extensively used) makes for better sounding systems, especially in a well treated room. I can't explain it, I won't even try to, but the fact still remain. It sounds better.


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## brucek

> If you want to convince us otherwise, show us your in-room waterfalls (long window, please)


I suspect I could find a suitable example of REW removing a modal resonances, but I don't think that would convince you.

I do take exception to the level adjusting in your example though. If I had a signal that was at +20dB and I reduced it by some means by -20dB to eliminate its effect, you can't raise the result by +20dB and say, "see, it's still there".

REW is fairly strict in what it targets to eliminate. The peaks must be minimum phase modal resonances. Sometimes the peaks can be a combination of two or more closely spaced resonances and the filter is only seeing one of them and so the tail carries on as it decays. Sometimes REW may not exactly identify the center frequency or the exact bandwidth of the response. I think you have to play around to get it exact. The waterfall plot is certainly the display to use for that purpose.

You've obviously proved the point that the ringing out was reduced in your example and so the point is proven that the filters operate not only in the frequency, but also the time domain. I'm sure with a bit of playing around you could completely remove the room mode, if indeed it was one. Again, this is all at the point of measurement only.

My "faux" tests were designed to show that if the filter is perfectly placed, then the result is perfection.

brucek


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## Wayne A. Pflughaupt

*Re: Is there a real benefit to preamps or two channel amps in HT?*




atledreier said:


> Even though the 'evidence' here suggest this and that, the reality is that good EQ in the time domain, like Audyssey and RoomPerfect (Both of which I've extensively used) makes for better sounding systems, especially in a well treated room. I can't explain it, I won't even try to, but the fact still remain. It sounds better.


Sure, I don’t think there are many who will claim that Audyssey (or any equalizer) won’t make an audible difference (hey, I’m a big equalizer buff myself). And certainly there is a time domain element with Audyssey, just like there is with any equalizer – to re-quote Rane’s Dennis Bohn, “Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others.” The debate seems to be over exactly how or why. Which I’m starting to think is kind of silly. It’s like, “Okay, it works. Who cares why?” 

At the end of the day, an equalizer merely tames a room mode by depriving it of energy – i.e. reducing its amplitude.	If it also happens to truncate the near-term signal decay time, that’s certainly a nice side effect. But if you could somehow totally eliminate the extended decay without reducing the amplitude, it would still sound bad (just try applying one of those “electronic room modes” to some headphones, where there is no ringing issue). So in my mind it’s the amplitude aspect that makes the audible difference, not the time domain.

Regards,
Wayne


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## Wayne A. Pflughaupt

brucek said:


> I suspect I could find a suitable example of REW removing a modal resonances, but I don't think that would convince you.


Try me.  After all, I became a skeptic by examining the graphs.  I fully recognize that my own room is all I have to work with, and that things could be different in another room – or even _every_ other room! (Probably should have mentioned that sooner - sorry...  )



> I do take exception to the level adjusting in your example though. If I had a signal that was at +20dB and I reduced it by some means by -20dB to eliminate its effect, you can't raise the result by +20dB and say, "see, it's still there".


It’s valid comparison because we don’t keep our systems at a static level. We turn them up and down. It’s well known that signal decay times are directly related to signal level (see graphed examples in my previous post). 

Level matching simply takes that factor out of the equation for examining the signal decay effects of modal filters. Otherwise a cutting filter has an unfair advantage merely by the effect of reducing amplitude: “See, it reduced the ringing!” Umm, not necessarily. Reducing the signal gain _looks like_ an improvement in the decay time to the untrained eye...







​
...but that is not the same as a reduction in the actual _rate of decay_ (related to RT-60).







​
Regards,
Wayne


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## Guest

Ok whether room EQ works or not, whether it reduces ringing or not, whether you like the sound of it or not, right now I don't care. Can anyone answer this question.

How can an EQ cut that is significant enough to reduce a modal resonance in a room, not affect the direct signal?


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## brucek

> How can an EQ cut that is significant enough to reduce a modal resonance in a room, not affect the direct signal?


Well, it does affect the direct signal. The EQ adjusts the direct signal to obtain the desired result at the point of measurement (which can be a result of direct and reflected signal). The EQ results are only effective at the point where the response was measured and a small area around that point depending on the frequency.

brucek


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## JohnM

To elaborate on Bruce's comment a little, unless you live in an anechoic chamber or sit a foot from your one speaker, you never get to hear just the direct signal from the speaker. You cannot escape the contribution the room makes to what arrives at your ears.


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## Guest

Absolutely you never just hear the direct signal. However you do hear the direct signal first by itself before any room reflections get to the listening position. So if you make a 15dB cut at 40Hz, don't yo think that would be audible in the direct signal? And if you do that the room node is still there correct? So you're not "fixing" the room node you're just sending less signal in that frequency range to excite the room node. 

Am I wrong in any of my logic? I'm not trying to be facetious, just trying to get a better understanding of your logic in this scenario. 

Thanks


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## Blaser

*Re: Is there a real benefit to preamps or two channel amps in HT?*

Wayne, 

I usually agree with Brucek on many points and specially the waterfall thread where I have made practical measurments in my room to demonstrate and confirm Brucek's thread opening theoretical analysis.

It was close but missed....He came alone :rofl2:


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## Blaser

Wayne A. Pflughaupt said:


> Level matching simply takes that factor out of the equation for examining the signal decay effects of modal filters. Otherwise a cutting filter has an unfair advantage merely by the effect of reducing amplitude: “See, it reduced the ringing!” Um no, it didn’t...


Wayne,

Look at my measurements in the first page again. It did.


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## JohnM

macrae11 said:


> Absolutely you never just hear the direct signal. However you do hear the direct signal first by itself before any room reflections get to the listening position.


Actually, you don't  It's tempting to think that, and to think of sound as zero dimensional particles spraying out of the loudspeaker, but that does not reflect reality. 

The 40Hz signal you refer to has to start somewhere. For convenience, lets say it starts at zero and begins its sinusiodal shape from there. It takes 6.25ms to reach the first peak of the first cycle, by the time it gets to the end of just the first half cycle of its very first period 12.5ms has elapsed, any surface with a path difference of less than 12 feet or so is already part of what is arriving at your ear. When you couple that with the effective gate time of the ear at such frequencies, 50ms or so, you find that it is not possible for us to separate the direct sound from the contributions of the reflecting surfaces at such frequencies.

Similarly an EQ filter at 40Hz is having no effect on the signal until there is some 40Hz present for it to attenuate. When 40Hz content starts to appear the filter response has its own evolution in time which, if the filter is properly matched to the 40Hz room mode it is countering, mirrors the evolving boost the mode is contributing. The net effect is you perceive 40Hz as it would have sounded free of the effects of the mode. It is very easy to form misconceptions about the actions of filters when viewing only their steady-state frequency responses and to forget that this is just a convenience to help us understand their effect, the filters operate in the time domain on time domain signals and have their own onset and decay times that need to be considered.

What must not be forgotten is the positional dependence of these corrections, however, which broadly speaking scales with the wavelengths being considered. For modes along the principal axes the benefit is broader across the direction of the axis and shorter along it, for example a correction for the first length mode would be valid across the whole width of the room at the point for which it was measured and generated, but would have a high sensitivity to movement along the length of the room.


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## Guest

I'm not exactly sure what difference the wavelength of the source has on the question at hand. By the time a 40Hz tone reaches it's first half cycle, it would have already reached the listener, unless they're farther than 14 feet away. 

Also this is the first time I've ever heard of an "ear gate time". I'm interested in this, as it may explain why I'm not understanding you. I did a quick google search for it but nothing came up. Do you have a link of some studies, or more information about this phenomenon?


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## JohnM

The wavelength of the sound makes a huge difference. Talking about a 40Hz tone "reaching the listener" is to use language that treats the tone as if it were a tennis ball that has been fired from the speaker to the listener at the speed of sound. Think about what is actually arriving at the listener, the beginning of a pressure variation that will take another 6.25ms to reach its first maximum and 12.5ms more to reach its next minimum, during which time the reflections from many surfaces of the room will also be arriving and adding their pressure contributions to create the composite signal that our hearing system turns into a perception of sound. We are not even able to perceive the pitch of the tone until several cycles of it have arrived at our ear, which for the frequencies under discussion takes a long time during which the room's contributions have been added. The fraction of a cycle that constitutes the direct signal before the rooms contributions begin to be added is too little for us to perceive the sound independently of the room effects.

"Gate Time" is my application of impulse response parlance to hearing, for more detail look for "temporal integration" but be aware that the mechanisms of hearing are very complex and studies tend to focus on the vocal range, there are few studies of low frequency performance. Here is a paper that employs a model of hearing for onset detection that mentions the 50ms integration time: http://www.music-ir.org/evaluation/mirex-results/articles/onset/ricard.pdf and here is one of the very few studies of hearing at low frequencies: http://projekter.aau.dk/projekter/retrieve/9897666?format=application/pdf and finally here is a paper that goes into a lot more detail about temporal integration, but not in the low frequency range: http://web.ics.purdue.edu/~mheinz/Formby et al JSHLR-2002.pdf


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## Wayne A. Pflughaupt

blaser said:


> Wayne,
> 
> Look at my measurements in the first page again. It did.


Yes, I’ve seen it.  But it’s a short-duration 300 ms window. I’ve also seen the same improvement in a 300 ms window, as I noted here.

By any chance, did you save your REW readings from that session, that you could run through a longer 5-600 ms window? I’d sure be interested in seeing them.

Regards,
Wayne


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## terry j

wow, thank you very much for your explanation John! Funnily enough on another forum recently we kinda spoke of 'how long it takes for the brain' to recognise a frequency.

Simply for sake of illustration and ease of mental calculation, let's say it's twenty cycles. That means of course that a 20 K tone is recognised in 1/1000 sec, but it would take a full second (thousand times longer) to hear a 20 hz tone.

At the time I remarked that I thought that was interesting and I had thought of that before, but also said that I couldn't really see any relevance to that train of thought, till now that is when you gave your explanation.

(It is entirely possible that my mental construct of the time taken is completely wrong, but nonetheless because of Johns answer the real world implications of this difference in recognition times really makes sense to me now)


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## Blaser

Hi Wayne,

I do so many measurements I am not sure if I can still find it. Nevertheless, I can do them again with a 600 ms window, no problem. I'll do it on Saturday.


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## Guest

Hi John

I wanted to take some time to read through and attempt to fully understand the papers that you presented however there are still some things that I'm not convinced of from your arguments. In your first article that you posted I don't see it directly referring to integration time. It does mention 50 ms, as that is the rough time where humans start to get out of the Haas effect. (Actually the Haas effect is around 30-40 ms) In attempting to model human hearing, they eliminated peaks that were closer than 50 ms, because they are stating that they aren't heard and there for aren't relevant. I do find it interesting that they don't keep the first detected peak, even though that could potentially be the peak that humans would “hear”.

Also they state that sounds with slow attacks are not represented well by this algorithm they created. Since almost all sounds that fall in the frequencies we're discussing have a slow attack time, I think this article is more or less irrelevant in this discussion. 

The second and third articles seem to have less to do with our discussion albeit they had some interesting reading. They are quite long however, and I can't state that I explicitly understood every single paragraph that was written. If there is a specific section to support your case, then perhaps you could point it out to me.

In the third article it seems to me to state that temporal integration doesn't have anything to do with a delay of sensation at various frequencies but rather a variation of sensation based on the duration of tones.

According my understanding of your hypothesis of “gate time” there would be a noticeable delay of lower frequencies from higher frequencies. I'm just guess here, but if you're saying that there's a 50ms “gate time” at 40Hz, than it might be logical to assume that there is a 100ms gate time at 20Hz, and a 25ms gate time at 80Hz?

Using this assumption, I consider the proposition quite preposterous. From listening experience I can with great certainty say, there is no noticeable delay of lower frequencies as my understanding of gate time seems to imply. 

If I am changing your words to mean something different than what you intended, please point out my flaws in reasoning.


In regards to the first part of your post, you claim that humans need to hear an entire cycle, or at least a half cycle of a waveform in order to identify it. To my understanding I don't believe this is true. I tried to find some research to this effect, but haven't found what I was looking for as of yet.

I do propose a real world situation though. If humans need to hear an entire cycle of a waveform would they be able to hear low frequencies in headphones. Obviously the size of waveforms are far to large to reside in the space between a headphone and an eardrum. Now I know your response to this, is that the waves would go through their pressure variances, from compression to rarefaction in that space without the full waveform being present at any one time. This still presents the problem, that the waves of lower frequencies would be heard later in reference to higher frequencies, resulting in a glissando effect with everything we heard. Hearing high frequencies and then having the low frequencies be heard later in time. In order to hear 20Hz there would have to be a delay of approximately 50ms causing a discreet delay. (I think this might be a more accurate measurement to use for “gate time” since a 20Hz wave has a wavelength of approximately 17 metres, which would take approximately 50ms to propogate.)

I don't believe there is such a delay although I am going to conduct my own listening tests to determine if I am correct or not. 

As it is, I'm going to be on location all of next week, so I probably won't be able to respond to any of your responses, although I will try to read them, and then respond next weeknd.

Thanks for this interesting and entertaining debate. I am certainly learning some new things, and stretching my brain in ways that it hasn't been stretched since I was in school.

Cheers

Andrew


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## terry j

macrae11 said:


> Hi John
> 
> In regards to the first part of your post, you claim that humans need to hear an entire cycle, or at least a half cycle of a waveform in order to identify it. To my understanding I don't believe this is true. I tried to find some research to this effect, but haven't found what I was looking for as of yet.
> 
> Andrew


Hi Andrew, in no way am I John!! so I too am interested in his response.

I would however be very surprised that we could, by definition, recognise the frequency of a signal before one complete cycle. 

(mental construct here) If a signal starts at the traditional '0' point, and starts rising then it will rise to a certain level, (talking sine waves for ease here) start to fall towards the bottom of the wave (in picture terms). OK, perhaps this might be the earliest point that we can say we have sufficient information for us to predict what the remainder of the signal will be, which is of course an inverse of the preceding. In other words, perhaps it is possible after all to only need half a cycle to recognise the signal.

Maybe you are right? Huhh, talked myself around to admitting the possiblity you are correct in the space of a half baked post!

Will wait for John!


----------



## JohnM

Hello Andrew,



macrae11 said:


> According my understanding of your hypothesis of “gate time” there would be a noticeable delay of lower frequencies from higher frequencies. I'm just guess here, but if you're saying that there's a 50ms “gate time” at 40Hz, than it might be logical to assume that there is a 100ms gate time at 20Hz, and a 25ms gate time at 80Hz?


Your understanding of that is incorrect, and attempting to extrapolate that to a simple frequency dependence magnifies the misunderstanding. Perception of sound is not an instantaneous process, it is affected by what is presented to the ear over a period. A demonstration of that is Temporal Masking, in which a loud sound can prevent us perceiving sounds that arrive up to 100ms after it (not surprising) but also sounds that arrived up to 20ms _before_ it. That does not mean that we simply delay everything to see what might come next, but that our perception of what arrives is affected by all the arrivals within a time window. There is no perceived delay.



macrae11 said:


> In regards to the first part of your post, you claim that humans need to hear an entire cycle, or at least a half cycle of a waveform in order to identify it. To my understanding I don't believe this is true. I tried to find some research to this effect, but haven't found what I was looking for as of yet.


Think about the spectrum of a short segment of a tone. How much of the tone do you think would be needed before the fundamental can be seen on a plot of the tone's spectrum? The inner ear acts somewhat like a spectrum analyser, with different parts of the spiral portion sensitive to different frequencies. Until enough of a tone has arrived at the ear there is no content at the fundamental to be detected, just some spectral content related to the evolving envelope of the sound. As if that was not problem enough, people seem divided between those who even perceive fundamentals at all and those who rely almost entirely on overtones (google "missing fundamentals") but in any case sufficient signal needs to be received to establish the spectral content for the auditory system's pattern matching to work on.

Meanwhile to return to the original question  as to whether applying a filter to the direct sound at a low frequency means what we hear sounds correct or not, let's consider a very simple case of a 40Hz tone arriving at the ear accompanied by a single reflection from a wall a few feet away. For convenience let's have a path difference that corresponds to a quarter wave, 6.25ms, and allow the reflection to be as large as the direct signal. The signals arriving at the ear are then the direct sound from the speaker and a quarter wavelength delayed version. The sum of these two, using basic trig identities for sums of sines, is a sine wave at the original frequency with a 1/8th wavelength phase shift and an amplitude of sqrt(2) times the original, so as far as the listener is concerned the tone has been made louder. To get the tone to the level it would have had without the effect of the reflection we need to reduce the level of the original sound by 1/sqrt(2). In doing that the 1/8th wavelength phase shift remains, but the level is corrected and the listener is none the wiser. 

In the more general case of an enclosed space there are only specific frequencies, the modal resonances, at which the multiple reflections from the room's surfaces generate a stable standing wave. The effect at those frequencies is to alter the perceived level of those tones according to the amplitude of the standing wave at a given location in the room. Altering the level of the original sound correspondingly gets us back to the level a tone would have had without the room's influence, with the proviso that nothing can be done for locations where the amplitude of the standing wave is zero and more generally it is inadvisable to boost the signal in locations where the standing wave amplitude is lower than the original signal.


----------



## Guest

JohnM said:


> Hello Andrew,
> 
> Your understanding of that is incorrect, and attempting to extrapolate that to a simple frequency dependence magnifies the misunderstanding. Perception of sound is not an instantaneous process, it is affected by what is presented to the ear over a period. A demonstration of that is Temporal Masking, in which a loud sound can prevent us perceiving sounds that arrive up to 100ms after it (not surprising) but also sounds that arrived up to 20ms _before_ it. That does not mean that we simply delay everything to see what might come next, but that our perception of what arrives is affected by all the arrivals within a time window. There is no perceived delay.


John, I understand that sound perception is not instantaneous. But in your initial post speaking of gate time, implied that temporal masking was frequency dependent. eg


JohnM said:


> When you couple that with the effective gate time of the ear at such frequencies, 50ms or so,


 This statement you made, seems to me like you are saying that there is different times for the brain to process signal depending on the frequency. This would cause a delay _relative_ to higher frequencies.



JohnM said:


> Meanwhile to return to the original question  as to whether applying a filter to the direct sound at a low frequency means what we hear sounds correct or not, let's consider a very simple case of a 40Hz tone arriving at the ear accompanied by a single reflection from a wall a few feet away. For convenience let's have a path difference that corresponds to a quarter wave, 6.25ms, and allow the reflection to be as large as the direct signal. The signals arriving at the ear are then the direct sound from the speaker and a quarter wavelength delayed version. The sum of these two, using basic trig identities for sums of sines, is a sine wave at the original frequency with a 1/8th wavelength phase shift and an amplitude of sqrt(2) times the original, so as far as the listener is concerned the tone has been made louder. To get the tone to the level it would have had without the effect of the reflection we need to reduce the level of the original sound by 1/sqrt(2). In doing that the 1/8th wavelength phase shift remains, but the level is corrected and the listener is none the wiser.


Well the first reflection wouldn't be the same amplitude as the direct signal.(or at least if it is the listener has more acoustic issues than a little EQ could ever fix!) The issue here again I don't think is just about amplitude, it's about time.


JohnM said:


> In the more general case of an enclosed space there are only specific frequencies, the modal resonances, at which the multiple reflections from the room's surfaces generate a stable standing wave. The effect at those frequencies is to alter the perceived level of those tones according to the amplitude of the standing wave at a given location in the room. Altering the level of the original sound correspondingly gets us back to the level a tone would have had without the room's influence, with the proviso that nothing can be done for locations where the amplitude of the standing wave is zero and more generally it is inadvisable to boost the signal in locations where the standing wave amplitude is lower than the original signal.


Here's the real issue(at least the one I've been talking about) modal resonances causing standing waves. Not only do these standing waves cause amplitude differences(which could be fixed with an EQ) but they also cause time differences. Notes ring out differently than they would without the influence of the room, lasting longer than they are supposed to. I don't think EQ can fix this. EQ will only lower the starting point of the frequency fundamental, which will cause it to dip into the noise floor sooner. Thus giving a perceived "correct" duration to the note, but not actually fixing the problem.


----------



## JohnM

There are substantial frequency dependencies in the way we process sound. That does not mean the differences in how we process content at different frequencies would cause us to perceive a delay between low and high frequencies, it is simply part of how hearing works. We have no other way of perceiving, and frankly for the lowest frequencies how could there be one? How could any instrument determine the frequency of a sound before it has sufficient signal on which to base that determination? At low frequencies a wider time period is used to build our perception. 

Altering the level of the reflection in the simple example only changes the amount of the signal boost.

Finally then, back where we started. Yes, modes affect the time domain. They have rates of attack and decay that depend on their bandwidth. They are accurately modelled as 2nd order systems. So are IIR filters. It is fundamentally wrong to say "EQ will only lower the starting point". The filter has a time domain response just as the mode does, both in the build-up of its attenuation as content at its centre frequency begins and the decay of that attenuation. In a properly configured biquad filter the filter's zeroes cancel the poles of the room mode, leaving the net effect of the poles of the filter itself, which decay faster than those of the mode. The mode's slow decay has been replaced by the faster decay of the filter. These topics are discussed in great depth in some papers, for example Meridian's "The Loudspeaker–Room Interface – Controlling Excitation of Room Modes" from AES 23rd International Conference, Copenhagen and "Modal Equalization by Temporal Shaping of Room Response" by Matti Karjalainen at the same conference.


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## Wayne A. Pflughaupt

> In a properly configured biquad filter the filter's zeroes cancel the poles of the room mode, leaving the net effect of the poles of the filter itself, which decay faster than those of the mode. *The mode's slow decay has been replaced by the faster decay of the filter.*


That certainly goes along way towards explaining the faster near-term decay times I've seen with the waterfalls. I expect that at some point the mode's "residual" decay overtakes and swamps the filter's effect, which is why things don't look so good when comparing long-duration windows? (see my graphs in Post #102) (Of course, one could argue by that time the signal is so low it doesn't matter...)

Regards,
Wayne


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## Blaser

blaser said:


> Hi Wayne,
> 
> I do so many measurements I am not sure if I can still find it. Nevertheless, I can do them again with a 600 ms window, no problem. I'll do it on Saturday.


Fortunately I didn't tell which Saturday:innocent:
Been very busy. Will do it next Tuesday (I hope to have a replacement day off). Wayne has made me curious to find out what happens after 600 ms :nerd:


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## brucek

*Wayne P says in part:*


> .......From the little I’ve seen, they don’t really have much effect on time domain. I haven’t seen any comprehensive in-room testing...............
> 
> .......From what I’ve seen, the improvement waterfalls show for modal (or “time-domain”) EQ filtering has only been apparent in the short-duration 300 ms window. When the window is lengthened to 600 ms, any advantage modal filters showed over other equalizing techniques pretty much vanished..............
> 
> .......Some of my esteemed colleges here (and on other Forums) are believers in time-domain equalization, but they haven’t posted or otherwise offered any evidence to back it up.............
> 
> .......If you want to convince us otherwise, show us your in-room waterfalls (long window, please)...........


Well, I am a believer in the theory that EQ'ing operates in the time domain and removes modal resonance at the listening position, and so I finally found a few minutes on this rainy day to try and satisfy Waynes thirst for an example in real life. It was a bit rushed, but I think I succeeded. 

I decided my main system would work fine since it has a few big modal resonances that REW takes care of quite nicely.

Since I have my new laptop with USB soundcard integrated with that main system I didn't mind testing it out.

I felt that the best way to show results would be to simply pick a single resonance out of my raw response and let REW recommend its filter(s) and I would enter them and then tweak a bit while watching only the waterfall plot to reduce the ringing out to zero. I would only work on a single resonance to avoid any confusion.

As I have observed in the past, the ringing was reduced *in the time domain to basically zero*.

I used a 550ms time window instead of Waynes request of 600ms, since the ringing was simply gone by 550ms in the raw response. If I look at longer time out to 1000ms, that doesn't change.

I used our standard graph limits of 45dB-105dB. Once a signal is below 45dB, it's gone.......




*RAW FREQUENCY RESPONSE FROM MY LISTENING POSITION*
I've circled the resonant peak I chose to work on. REW identifies it as a peak and recommended two filters. One at 55.7Hz and another around 80Hz that I entered but didn't modify. The only filter I modified was the 55.7Hz
Once I entered the REW recommendation, I played only by watching the waterfall. I moved the center frequency to 55.2Hz and the bandwidth by one notch and the gain by a couple db (if I remember correctly) until I was satisfied. I played until the ringing out was gone.










*RAW WATERFALL RESPONSE*









*RAW WATERFALL RESPONSE (with 45Hz-85Hz horizontal axis)*
An expanded scale view.











*RAW (green) AND FILTERED (red) FREQUENCY RESPONSE OF ONE PEAK*









*FILTERED WATERFALL RESPONSE*










*FILTERED WATERFALL RESPONSE (with 45Hz-85Hz horizontal axis)*
Expanded scale view.










*OVERLAY COMPARISON OF RAW PEAK AND FILTERED WATERFALL (with expanded axis)*
A nifty view where you can you see that the green signal that used to ring out to 550ms is now gone.









I don't know what more I can do to prove it. The peak and the ringing is simply gone without too much effort.
I know this is only valid at my listening position, but it is valid.

I could now go and work on all the other peaks etc and hopefully reduce most of the time domain problems....

You do have to be careful when you start looking down below 45dB level, that you may see some new ringing signal in certain circumstances. This can be caused by low level noise in the room from furnaces and fans and refrigerators etc.

brucek


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## Blaser

Nice job Brucek, this is quite in line with what I found in my experiment as well. 
Nevertheless your experiment would be more illustrative if the SPL at 55.7 Hz is the same at 0 ms for both equalized and unequalized measurements. Pls take 2 measurements (equalized and unequalized) and overlay waterfalls with and without eq. so that they start at 55.7 Hz *at the same SPL*. This will give us a direct idea of the effect of equalization on the room mode regardless of SPL.

I couldn't do my measurements today, I hope I will do them soon.


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## brucek

> Pls take 2 measurements (equalized and unequalized) and overlay waterfalls with and without eq so that they start at 55.7 Hz at the same SPL.


hehehe, no I won't be doing any more work on proving this theory. I've beat it to death.

I've proved it first with the theoretical lab experiment and Wayne said it had to be real world to convince him.

So, I used my real world HT with a real modal room peak and added a filter to bring it down to the real level that I would use and the ringing is 100% gone.

Enough is enough............ 

brucek


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## Blaser

:rofl: I can guess what state of mind you're in after that very long debate... LOL

I'll do some more measurements and post them later on to definitely close this issue as far as the 600 ms is concerned.


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## penngray

> I don't know what more I can do to prove it. The peak and the ringing is simply gone without too much effort.
> I know this is only valid at my listening position, but it is valid.
> 
> I could now go and work on all the other peaks etc and hopefully reduce most of the time domain problems....
> 
> You do have to be careful when you start looking down below 45dB level, that you may see some new ringing signal in certain circumstances. This can be caused by low level noise in the room from furnaces and fans and refrigerators etc.


This thread is a great!!

But I have a dumb question....you say that filtering removed your problem but doesnt it only remove it at at SPL level, if you increase the SPL wouldnt the time domain problems re-appear?


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## brucek

The filter removed a modal peak down to the level of the rest of the non peak area. Makes no difference after that if you turn up the wholesale level. It all moves up........

brucek


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## penngray

> It all moves up........


I know it all moves up but isnt there there still something in the 400 ms range down below 48 dBs that will move up too, creating something audible?

Im just wondering how this correlates back to room treatments and if filtering replaces treatments at all?


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## Blaser

Brucek,

I knew this would be brought out... This is why I asked about level matching yesterday


----------



## thewire

penngray said:


> I know it all moves up but isnt there there still something in the 400 ms range down below 48 dBs that will move up too, creating something audible?
> 
> Im just wondering how this correlates back to room treatments and if filtering replaces treatments at all?


No because the human hearing is based on what it chooses to focus on. For example if you are in a room of people all talking at different levels, you can focus your attention on what the person in front of you is saying. If our hearing to into account every frequency from every direction and at every audible range we would be distracted as our hearing perception would go into hearing overload. If the ranges and frequencies are closer together, they become more difficult to discern from one another and become more like noise, less like what we define as human sound. 

As for replacement room treatments with a BFD that is entirely up to the person with the room. Like putting a fake fish tank in your home with little floating plastic fish. It is entirely an opinion on which works best.


----------



## thewire

Here is how gnuware.com says it which is better than I could. I only know this stuff mostly from reading medical books and it a little more difficult to translate than bellow. 

Chapter 2. Audio Fundamentals -



> 2.3.2. Temporal masking
> 
> In addition to auditory masking, which is dependent on the relationship between frequencies and their relative volumes, there is a second masking that comes into play, based on time rather than on frequency. The idea behind temporal masking is that humans also have trouble hearing distinct sounds that are close to one another in time. For example, if a loud sound and a quiet sound are played simultaneously, you would not be able to hear the quiet sound. If, however, there is sufficient delay between the two sounds, you will hear the second, quieter sound. The key to the success of temporal masking is in determining or quantifying the length of time between the two tones at which the second tone becomes audible, i.e., significant enough to keep it in the bitstream rather than throwing it away. This distance, or threshold, turns out to be around five milliseconds when working with pure tones, though it varies up and down in accordance with different audio passages.
> 
> This process also works in reverse; you may not hear a quiet tone if it comes directly before a louder one, so premasking and postmasking both occur and are accounted for in the algorithm.


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## Wayne A. Pflughaupt

brucek said:


> Pls take 2 measurements (equalized and unequalized) and overlay waterfalls with and without eq so that they start at 55.7 Hz at the same SPL.
> 
> 
> 
> hehehe, no I won't be doing any more work on proving this theory. I've beat it to death.
> 
> I've proved it first with the theoretical lab experiment and Wayne said it had to be real world to convince him.
> 
> So, I used my real world HT with a real modal room peak and added a filter to bring it down to the real level that I would use and the ringing is 100% gone.
> 
> Enough is enough............
Click to expand...




brucek said:


> The filter removed a modal peak down to the level of the rest of the non peak area. Makes no difference after that if you turn up the wholesale level. It all moves up........


Well, without before-and-after gain matching, there is no way to confirm whether or not you really accomplished a reduction in _ringing_. All you’ve accomplished is an exercise in gain reduction. You reduced the peak by at least 14 dB. _*Of course*_ the waterfall is going to look better! I would think it would be obvious that a signal with a lower SPL level is going to “fall off the map” – i.e., get down to the noise floor – faster than a signal with a higher level, and as such the former will look "better" in a waterfall graph. It’s RT60 basics.

Take another look at my graphs from Post #60. Doesn't the second one look great? All I did was reduce the measurement by 10 dB and bingo, I have a fabulous-looking waterfall!










*Baseline Measurement









Measurement SPL Level Decreased*​



penngray said:


> I know it all moves up but isnt there there still something in the 400 ms range down below 48 dBs that will move up too, creating something audible?


I don't know about "creating something audible," but what's happening below the graph's lower limit certainly needs to be looked at. Stay tuned.


In order to show that a decrease in ringing has been accomplished, a waterfall graph needs to indicate that an improved _rate of decay_ has been initiated. If not, all you've accomplished is merely a reduction in gain at the targeted frequency. Compared to a baseline graph, an improved rate of decay would be seen as increased spacing between the slices, indicating that the signal level is attenuating faster. This is what improved ringing (faster decay) looks like:



















Note the significantly faster rate of decay above 140 Hz with the lower graph, which added bass traps to a room: At about 15 slices, the signal has dropped as much as 50 dB at some frequencies, in less than 200 ms, compared to the baseline which shows decay times at twice that rate or more. You simply can't get this kind of "action" with an equalizer. (Note, the reduction in decay time came with no electronic attenuation of the signal, as you would get with equalizer filters. Any decrease in signal peaks you see are merely the effect of absorption from the traps.)

That said, I'm basically satisfied with the explanation John gave as to how an equalizer can make at least _some_ improvement in ringing with a room mode (emphasis added):


JohnM said:


> In a properly configured biquad filter the filter's zeroes cancel the poles of the room mode, leaving the net effect of the poles of the filter itself, which decay faster than those of the mode. *The mode's slow decay has been replaced by the faster decay of the filter.*


So once again, an improvement in ringing will show a faster rate of decay, not just a reduction in gain. To put it in hopefully simpler terms for our non-technical readers: A room mode has a slow rate of decay, and canceling it with a suitable (i.e. matching) EQ filter that has little or no decay factor results in a faster rate of decay for the mode - i.e., reduced ringing. It makes sense, and it's readily apparent in the long-term 600 ms window comparing a base vs. equalized graph I presented in Post #109 (hit F11 to get both fully on the screen):
















Baseline (Top) vs. Filtered, Level-Matched (Bottom) @ 600 ms​

Even though there is no improvement in _long-term_ ringing, you can see by the increased gap between the first nine slices in the upper half of the yellow box that there is at least a _short-duation_ improvement in the rate of decay. Moving the window from 600 to a 200 ms highlights the short-term improvement even more (hit F11 to get both on-screen):
















Baseline (Top) vs. Filtered, Level-Matched (Bottom) @ 200 ms​

In the equalized (Filtered) graph, we can see from the wider spacing of the slices at and around the filter center (~42 Hz, indicated by the marker) that the signal is indeed decaying faster, compared to the baseline graph. The base graph's signal is down only ~18 dB at 200 ms, while the filtered graph's signal is down ~32 dB at 200 ms. While that's not nearly as dramatic as the ~50 dB that can be obtained with a slew of bass traps (not that you can readily get any that work down at 42 Hz), it is at least a notable difference.

Two-hundred ms seems to be about the point where time-domain improvement of the filter is "spent" - at least in my room. The rate of decay beyond that point levels off substantially. (That point appears to be a bit sooner than 200 ms in Blaser's red graph in Post #9, and the rate of decay seems to level off with the signal down only ~18-20 dB). Just a guess, but it seems to me the reason is that residual ringing from the slow-decaying room mode eventually swamps the filter's effect. Either that, or the filter wasn't quite matched to the resonance, or it may depend on how deep the required equalizer cut needed to be (by necessity, determined by the severity of the mode). Nevertheless, it also seems that by the time the signal level is that far down (about 32 dB, remember), it could be argued that the desired effect has been accomplished and any residual ringing doesn't matter much, from an audible perspective. (I'm surprised no one has challenged me on that point!) But I'll leave that for the acoustics experts to debate.

The only thing I'm left pondering at this point: I have my doubts that every little up and down rip in measured response meets the definition of a true room mode, as "having a time response that acts like a 2nd order biquad that can be counter-acted with an EQ filter with the same Q and opposite gain."

For instance, check this graph comparing three measurements I took at two-month intervals (top and bottom traces shifted for the sake of clarity):









Three Baseline Readings, Each @ Two Months Apart​

Note that my 42 Hz room mode stays pretty constant in level and shape, while everything above and below that point changes noticeably: Peaks and dips appear and disappear, they change in severity and shape, their frequency centers move back and forth, etc. That's probably why aside from the 42 Hz mode, I've seen virtually no change in ringing anywhere above or below that point, using any configuration of filters, as shown in graphs I presented previously in Post #103.

I've also taken readings at various locations in my room, and five of six readings showed the 42 Hz peak, while areas above and below the peak varied considerably:








Baseline @ Five Locations​
So apparently a mode can be a fairly static phenomenon in the room, while the rest of the response curve is not (as anyone who's taken measurements around their room can verify).

So, it seems to me like a good approach would be to use REW to identify the true room modes - i.e. the one(s) that show an improved rate of decay with the proper filter - and merely smooth the rest of the response curve, as outlined in my Minimal EQ article, since those filters won't be addressing true "minimal phase modal resonances" anyway.


The reason that level matching after equalizing matters is that you won't get the full waterfall picture if you don't. Keep in mind that everyone's graph has varying peak and average levels, but we're all using _the same 45-dB floor_. Without level-matching, you can't tell for sure if or how well you've improved ringing, or if you've merely pushed any residual "tail" that might be present down below the graph's floor - especially if you happen to be starting with a relatively low level to begin with (the peaking frequency in brucek's graphs, for instance, is 4-5 dB lower than mine) and/or are needing substantial cuts. 

Ethan Winer tells me that waterfall measurements need to be done at fairly high SPL levels, as nulls can be 30 dB deep. Before equalizing, the 75 dB reference we establish with REW's Check Levels routine is determined primarily by your sub's _peaking frequency_. In like manner, when you calibrated your sub to the rest of your system before you equalized, any peak in response was the determining factor for the level you ended up setting it at. Peaks can require substantial cuts up to 10-15 dB or more from the equalizer. 

After equalizing you typically have to re-adjust your sub's level upwards to get it properly blended with the rest of the system, do you not? Since that's the level you're going to be operating your sub at anyway, it makes no sense to check your waterfall at the substantially-reduced level you initially ended up with after equalization. If you don't level-match the peak using the method I've described in previous posts - using a sine wave tone at the frequency center to get SPL back to the level it was before - you should at least re-run REW's Check Levels routine before taking your "after-EQ" sweep, if you're interested in comparing "before and after" waterfall readings.

As an example, you can see what I'm talking about in these graphs, which I presented earlier in Post #109. The lower graph has had the 42 Hz peak filtered /attenutated a full 10 dB, and you can see that the ringing "tail" is reduced to barely-visible, compared to the level-matched graph (F-11 to get both on screen):


















Level-Matched (Top) vs. No Level Matching (Bottom) @ 600 ms​

As you can easily visualize in the lower graph, if you push the level down another 5 dB, either with equalization or the receiver's volume control, the "tail" will fall below the graph's floor, giving the illusion that ringing has been totally eliminated. Had you known the "tail" was present, as level matching would have shown (upper graph), you may have been able to tweak the filter parameters and _truly_ eliminated it. 

Thus in many situations, like if you start with a lower-than-usual level and your room mode needs a severe cut, you won't know for sure if you have improved or eliminated the ringing without reclaiming the level lost to the equalizer (again using brucek's graphs as an example, after about -15 dB of equalization, the signal level [red circled area] is a mere 25 dB above the graph's floor). 

Which brings us to another issue that seems to have escaped everyone's notice: The 45 dB lower limit we use for our graphs _is better suited for basic frequency response curves_ than analyzing waterfalls. For waterfalls, 45 dB is an unrealistically high floor. Most quiet residential rooms are going have a noise floor 10-15 dB lower, or even more if you have a soundproofed room or live in a rural location. After all, how can you adequately evaluate any potential improvement in ringing and signal decay, via equalization or other means, if the graph's lower limit is set artificially high? Wouldn't you prefer to know how your room is behaving all the way down to its true noise floor?

Here again is the non-level-matched graph posted above that showed a minimal residual "tail," along with the same graph with the floor extended down to a more reasonable 35 dB. Note the 42 Hz marker.

















45 dB Floor (Top) vs. 35 dB Floor (Bottom) @ 600 ms​

If you have a really quiet room, or one that's soundproofed, you'll probably be better served with a 30 or even 25 dB graph floor. 

Regards,
Wayne


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## Guest

*How to measure impulse response outside REW*

I dont have a means to bring the PC to the living room to make any measurements via REW.


I see that REW has an option to import the impulse measurements. So I was wondering if there is a way to measure the impulse response using SPL just like I can measure the FR using test tones and SPL


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## brucek

> So I was wondering if there is a way to measure the impulse response using SPL


No.

brucek


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## fastuc

Hi,

sorry for my bad english but i'm an new italian user. :jump:

i have a question, because for me there is an very important error in this misuration waterfalls....but the error is make by the software: to do the waterfall the software produce a sine sweep wave, then the loudspeakers produce only one frequency every moment...

BUT

never the loudspeakers produce only one frequency, they produce many frenquencyes at same time!!!

then: do you know if with REW is possible to obtain a waterfall diagram with a istant pink noise signal source?

thanks and sorry for my bad english

_riccardo_


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## JohnM

Riccardo,

The sweep is used to determine the impulse response of the system, the waterfall is then produced by analysing that impulse response. Other signals can be used to determine the impulse response, including MLS and Periodic Noise sequences, but logarithmic sweeps have advantages in rejecting noise and distortion that makes them very well suited to the task. You can find more info here: Transfer Function Measurement


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## thxgoon

Wayne A. Pflughaupt said:


> So once again, an improvement in ringing will show a faster rate of decay, not just a reduction in gain. To put it in hopefully simpler terms for our non-technical readers: A room mode has a slow rate of decay, and canceling it with a suitable (i.e. matching) EQ filter that has little or no decay factor results in a faster rate of decay for the mode - i.e., reduced ringing. It makes sense, and it's readily apparent in the long-term 600 ms window comparing a base vs. equalized graph I presented in Post #109 (hit F11 to get both fully on the screen):
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> Baseline (Top) vs. Filtered, Level-Matched (Bottom) @ 600 ms[/center]
> 
> 
> Even though there is no improvement in _long-term_ ringing, you can see by the increased gap between the first nine slices in the upper half of the yellow box that there is at least a _short-duation_ improvement in the rate of decay. Moving the window from 600 to a 200 ms highlights the short-term improvement even more (hit F11 to get both on-screen):


I've been following this thread for some time and have had some trouble trying to understand how active eq can affect a modal response. My beliefs are more in line with Wayne's in that a decrease in ringing would show up as a steeper slope on the waterfall chart. As a counterpoint to this you provide the measurements above. 

The corrected image does indeed look like it has a steeper rate of decay, however I noticed that the slope to the left and right of the mode are the same in both plots. To me, it only looks as though the amplitude of the mode has been reduced to the point where the ringing now exists underneath the natural decay of the other frequencies around it, and it can still be seen to ring out at about the 50db mark, the same as the uncorrected graph. I would argue that decay time has not been changed, only amplitude. 

What I have a hard time believing (regardless of how an eq filter's characteristics relate to a room mode) is that once a signal leaves the loudspeaker and enters the room, it is at the mercy of the room's acoustics. How can an eq filter affect this without some kind of active feedback cancellation? This doesn't make sense to me. Am I missing something?


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## brucek

> What I have a hard time believing (regardless of how an eq filter's characteristics relate to a room mode) is that once a signal leaves the loudspeaker and enters the room, it is at the mercy of the room's acoustics. How can an eq filter affect this


If you believe that the rooms modal response at a particular frequency alters the signals amplitude and time exactly the same as a filter alters the signals amplitude and time, then you know that the two completely cancel each other. If you don't believe this, then they will not. I happen to be a believer.... and I'm not alone.... 

brucek


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## JohnM

thxgoon said:


> What I have a hard time believing (regardless of how an eq filter's characteristics relate to a room mode) is that once a signal leaves the loudspeaker and enters the room, it is at the mercy of the room's acoustics. How can an eq filter affect this without some kind of active feedback cancellation? This doesn't make sense to me. Am I missing something?


The EQ filter alters the signal before it leaves the speaker. If the filter is properly configured the alteration is the opposite of the effect (time and frequency) the modal resonance has on the signal. The feedback happened when the response was measured and the required filter settings were determined. It is important to recognise that the effect is valid for the point at which the measurement was taken, other places in the room that do not measure the same will not react in the same way.


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## Wayne A. Pflughaupt

thxgoon said:


> The corrected image does indeed look like it has a steeper rate of decay, however I noticed that the slope to the left and right of the mode are the same in both plots.


 Correct. There was no filter applied to affect the area outside the yellow box. It will exhibit no change.



> To me, it only looks as though the amplitude of the mode has been reduced to the point where the ringing now exists underneath the natural decay of the other frequencies around it, and it can still be seen to ring out at about the 50db mark, the same as the uncorrected graph. I would argue that decay time has not been changed, only amplitude.


Certainly, there is an amplitude factor; as explained in Rane's Exposing Equalizer Mythology article, “Amplitude, phase and time are all inextricably mixed by the physics of sound. One does not exist without the others.”

If you check the graphs I posted just below those two you referenced in your post, you can see that the rate of decay was indeed improved out to 2-300ms; with the longer 600 ms, window you can see the rate of decay did not improve for the full duration of that window. This may well be because my filter wasn’t a perfect match for the mode, something John has repeatedly emphasized needs to occur for maximum decay reduction. It looks like from the graphs John presented here that you can improve the rate of the decay all the way out to 500 ms or so (note especially the 25 and 60 Hz modes), if you spend enough time tweaking the bandwidth, frequency center and gain parameters. (Not sure I can hang with 12 notch filters between 20-65 Hz, though!  )



> What I have a hard time believing (regardless of how an eq filter's characteristics relate to a room mode) is that once a signal leaves the loudspeaker and enters the room, it is at the mercy of the room's acoustics. How can an eq filter affect this without some kind of active feedback cancellation? This doesn't make sense to me. Am I missing something?


There’s nothing to it really. As John explained in this post, a mode is slow-decaying. For example, remember how I noted (in the post you referred to) that you could see that a reduction in decay time had been achieved with the equalizer because there was more space between the slices? Take a look at this graph of the 42 Hz mode (with no EQ applied) from that post, and you can see just the opposite effect:










Note that mode’s slices are _closer together_ than the areas to each side of it. That shows the mode is indeed decaying at a slower rate than the other areas. The EQ filter matched for that mode has little or no decay element, because it’s electronic, whereas the mode is acoustic. So when you apply the filter, you’ve robbed the mode of energy and essentially defeated it. The signal now decays the same rate as everything else, as you can see in this w/ EQ graph that was posted just below the one above.










Anyway, hope some of this makes sense...

Regards,
Wayne


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## thewire

What is the ideal temperature range for taking waterfall measurements? I'm guessing that bellow 66 degrees is too low. We don't heat the side of the house with the HT much, but I can warm it up with the wall heater prior to measuring if that will help any.


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## Wayne A. Pflughaupt

The correct temperature would be the same one used when you're watching a movie. Doesn't make much sense to measure under any other conditions, does it? 

Which is yet another reason I don't lose sleep over waterfalls: Response readings (and the waterfalls generated from them) will change with temperature and humidity changes.

Regards,
Wayne


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## thewire

I will keep the temp at around 73 degrees then. The humidity isn't even registering on the meter in my HT right now. If I keep the wall heater on scorch mode as it is now, it doesn't seem to be making the clacking noises, so I think I will be in good shape. I had expected the room to get much warmer with the wall heater on overnight.


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## Blaser

Do you think temperature and humidity variation (within a slight range acceptable to human body) might have a noticeable impact on waterfalls??


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## thewire

Blaser said:


> Do you think temperature and humidity variation (within a slight range acceptable to human body) might have a noticeable impact on waterfalls??


I did not notice any change after heating the room to room temperature. The foundation walls (every side) are a bit cold right now, and they used to make the rooms drop ceiling creak pretty bad during the winter months. I am now trying with some humidity assuming that a Vicks Machine is up to the task. What scale should I use? This is 780ms and an 85dB target. I do not hear 20Hz unless I put my ear up to the subwoofers, or I am in the next room where things rattle. Occassionaly my seat feels like someone is kicking it. This happened recently when I was watching a movie with some others in the room, and I looked behind me and they said "nope that was the movie" and there legs were extended horizontal over to the other end of the couch. I'm not sure if it to do with the 20Hz but I don't hear it, and I am unsure exactly what Berkline has done in the couches for this LFE to occur. When I record a sample of 20Hz from a movie at reference level with my RS meter, it seems almost impossible no matter what scale I use. There is what sounds like extreme clipping down low that sounding a bit like somewhat running a board across a picket fence. It's something I don't hear while watching movies. Maybe I should stick to waterfalls at the 75dB target..


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## JohnM

Blaser said:


> Do you think temperature and humidity variation (within a slight range acceptable to human body) might have a noticeable impact on waterfalls??


In a word, no. The effect will be on the speed of sound, which will shift the resonant frequencies correspondingly. There is almost no effect from humidity, between 0% and 100% humidity the speed of sound in air changes by less than 0.4%. Temperature has a larger effect, with almost a 6% change between 0C (32F) and 30C (86F), so the frequencies would shift by the same amount, but the rates of decay would not be affected.


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## thewire

The room was heated up very slightly with all the steam going into the room. That is the only thing I could think of that might have changed my measurements. I did not see any changes to the waterfalls that looked apparent. Thanks.

Red = 4 measurements 8:24am 73.9 degrees - humidity not registered 
Blue = 5 measurements 11:15am 75.7 degrees - humidity not registered


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## Blaser

That's what I thought as well John.
Steven, I believe the difference could come from the measurment itself and anyway, that is too low to consider.


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## Wayne A. Pflughaupt

JohnM said:


> Blaser said:
> 
> 
> 
> Do you think temperature and humidity variation (within a slight range acceptable to human body) might have a noticeable impact on waterfalls??
> 
> 
> 
> In a word, no. The effect will be on the speed of sound, which will shift the resonant frequencies correspondingly. There is almost no effect from humidity, between 0% and 100% humidity the speed of sound in air changes by less than 0.4%. Temperature has a larger effect, with almost a 6% change between 0C (32F) and 30C (86F), so the frequencies would shift by the same amount, but the rates of decay would not be affected.
Click to expand...

The issue isn't the effect of temperature and humidity on the speed of sound. The issue is their affect on the transducers – the speakers and measuring mics. Their diaphragms, cones, windings, etc. contract and expand with temperature changes, which will alter their performance. This is why any professional sound company worth their salt will require the room to be fully air conditioned as it will be used before calibrating a system. 

Here again is the graph on the effects of temperature changes that I posted a few pages back. It was a cold winter day (for Houston at least), and it was 73˚ inside the house. After taking a measurement, I set the SPL meter on the floor where sunlight was streaming in a window. The thermometer registered 86˚ in the sun. After a half hour or so I took another reading. The red trace is the reading from the warmed-up mic.









*Two REW readings, mic element at 73 vs. 86 degrees*​

Sure, there will be no _actual_ change in the rate of signal decay, but any changes in the _measured response_ (for whatever the reason) that register will result in a different waterfall. In the illustration above, with response being 1 dB or higher across the board for the 86˚ reading, the waterfall would have looked worse.

Regards,
Wayne


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## Blaser

Wayne,

BTW, Will that make a discernable enough difference??


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## Wayne A. Pflughaupt

It’s highly doubtful, but then I can’t speak for everyone on the planet. Fortunately, our ears don’t hear the same way mics “do” – they’re much more forgiving.

Regards,
Wayne


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## viccmw

The last post was quite awhile ago.... but this was such an interesting read 

I guess that EQ is indeed possible to address room modes to some degree in the low frequency - otherwise devices like Antimode 8033 would never have found acceptance . 

Another thing IMHO is that the mic should be of high precision as using an EQ filter requires the system to identify the correct room mode frequency - am I correct? Would an RS SPL meter cut it (if we want to address room mode through REW)? 

Another thing that was mentioned often times is that this measured waterfall response is relevant only to the location of the mic, so was wondering exactly how small spot would such EQ benefit - main LP only? Audience to the immediate left and right of the main LP do not enjoy it?


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## JohnM

viccmw said:


> Another thing IMHO is that the mic should be of high precision as using an EQ filter requires the system to identify the correct room mode frequency - am I correct? Would an RS SPL meter cut it (if we want to address room mode through REW)?


The quality of the mic would not affect the determination of frequency, only amplitude. An RS SPL meter would do just fine.



> Another thing that was mentioned often times is that this measured waterfall response is relevant only to the location of the mic, so was wondering exactly how small spot would such EQ benefit - main LP only? Audience to the immediate left and right of the main LP do not enjoy it?


Depends entirely on the room, to find out just measure in the positions of interest.


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