Acoustic Measurements: Understanding Time and Frequency

FrantzM

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Apr 20, 2010
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Hi
Having read this and digested it , including the contributions of micro and Terry, I am reinforced in my belief that the best way to deal with bass in most rooms is multiple sub-woofers. I dd it in my former system and that was the best bass I have heard anywhere .. my new system is about to be finished, more modest that the last one. i expect it to be however even better performing with these new information.
Those are the kind of discussions I love about the WBF. We learn and can repeat the experiences. Repeat is the operative term...

One last question why trying to solve issues very low in frequency with absorption may seem inefficient .. Doesn't their absorption in the highest frequencies help in eliminating in "drowning" out if you will, the amplitude of the ringing at those higher frequencies.. IOW it seems the ideal solution would be EQ in the bass coupled with some absorption?
 

JonFo

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Jun 11, 2010
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No question that reductions in level via EQ affect the intensity and duration of the decay (ringing) of low frequencies in-room.

The question is: is that the best approach?

Ideally, one would like to get the *same* amount of initial energy into the room, and not have it ring.

IIRC, bass energy needs to be measured with a minimum window of 150ms to capture their longer wavelengths. So Frequency response plots measure the sum of the energy over time within the window.
Waterfall shows us an approximation of the decay elements over time to help expose the ‘ridges’ where the room is interacting strongly with the sound source.
Now, if I damp the causes of the resonances in the room via treatments, I can still have a high level of sound source output at those frequencies without the deleterious effect of resonances and attendant ringing ‘smearing’ the sound.

Just using EQ to reduce the amount of energy introduced into the room to minimize the engagement of the resonance also minimizes the impact of that frequency (or note) since we have pulled down its relative level.

Overall, it’s better than no EQ, but far from optimal.

To me, the best approach is to treat the room, use multiple subs to smooth out the energy distribution, and then, as a final touch, use room EQ.

That approach will preserve much more of the original sound impact at the critical resonance frequencies, and works better across a much broader area of the room.

It is this later point that is critical, as EQ generally has its optimal effect only at a specific point, the rest of the locations are a compromise. Room treatments tend to have a beneficial effect over a broad range of seating locations.
 

microstrip

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One last question why trying to solve issues very low in frequency with absorption may seem inefficient .. Doesn't their absorption in the highest frequencies help in eliminating in "drowning" out if you will, the amplitude of the ringing at those higher frequencies.. IOW it seems the ideal solution would be EQ in the bass coupled with some absorption?

FrantzM,

It seems so. At least we see that most of the rooms famous for their sound quality in this forum have bass trapping. OK, I know that this is not an objective statement, the owners and friends could be all just living an illusion. But with the exception of those lucky people using the services of Keith Yates, that includes intensive, extensive and expensive simulation with proprietary measurements, I do not see any scientific approach to bass trapping elsewhere. Perhaps there are a few others I do not know about. Since the measurement used by 99% of people (waterfall at low frequencies with high frequency resolution) is flawed we find we are still in the dark ages of room treatments at bass frequencies.
 

jkeny

Industry Expert, Member Sponsor
Feb 9, 2012
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As a novice in this area forgive my naive questions.
Firstly have these two different approaches to dealing with room modes been compared in listening tests - Eq Vs Absorbers? Any studies or results? I know Amir has attested to the effectiveness of the Eq approach but wonder if he compared it to absorber in a real listening environment?

Secondly, one has to be careful about digital equalisers - the underlying assumption here is that the equalisation is without blemish. This may not be the case in that the equaliser itself can introduce pre and/or post-ringing. An interesting paper (1990) from Michael Gerzon, mathematician & audio expert, goes into more detail about this [url]http://www.audiosignal.co.uk/Resources/Why_do_equalisers_sound_different_USL.pdf[/URL]
 

microstrip

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I have just found that the book Recording Studio Design by Philip Newell, an author I have referred to several times and I first found in an article in HifiCritic, an UK audiophile magazine, includes a chapter about audio measuring in large and small rooms. From what I could read in google books it looks rather interesting - I have just ordered the book. Although written for professionals, not consumers, the basics and science should not depend on target. Also, I hope that as soon as I will start posting about the Schroeder plots WBF members will take me more seriously! ;)
 

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amirm

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A quick note: whether you use multi-sub, eq or acoustic products, you are taking away bass energy. with multiple subs, one's output is cancelling the output of the other at times which obviously reduces efficiency. EQ as I have shown, involves pulling down levels. And acoustic products work by converting sound energy to "heat." So it is not an issue of one not doing it and the others doing it.

The fix by the way, at least partially, is once you have smooth response, bump up the level of bass some. We tend to like a boost there anyway. Alternatively you can pull down the level of your main speakers.

In the Todd Welti there is an efficiency factor for some configurations. For example a two-sub configuration I am using is actually more efficient than a 4-sub configuration.
 

bblue

Well-Known Member
Apr 26, 2011
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I have to say, I believe the whole concept behind this EQ-solves-all discussion to be seriously flawed. It's all fine mathematically, but not acoustically, and doesn't even take into account what the presence and performance of the equalizer brings to the table. Nothing good, usually.

Through my recording and mixing in a variety of studios along with assisting in design of same, I learned from several different mentors a basic theme: that the basic aural characteristic of any room where recording or serious listening is to take place should be as neutral as possible before any electronic assistance is applied. The reasoning, most important during recording, is to not allow the room characteristics to influence negatively the character of the instrument(s) being recorded. But the same can be said about influences a room has on the playback as well.

The waterfall plots (including those with the natural room noise correctly superimposed) show the resonant decay of any given frequency. The length of that decay at each significant frequency influences all sounds produced in that room, regardless of their level. Long decays in a frequency center (the result of a room mode) will extend its audible time, and that by itself gives an entirely different character to how an instrument falling in that frequency range will sound. In the event a resonance is very low level, it may be lost at the bottom of the dynamic range of the room.

Think of it like this. You have a room with a large resonant peak at 70hz, and want to record an upright bass or cello. The room will significantly modify the character of the bass/cello by resonating with it at that resonant frequency. Eq later during mixing will not correct the problem; that note or range will always sound different than the neighboring frequencies. You can lower its level, but the character remains the same. The same thing would be true if you were in the middle of a room null produced by mode cancellations, and were recording an instrument. At certain notes the cancellation itself will have a unique decay which could not be recovered entirely by eq in the problem area.

The same issues apply in a listening room. Unless you acoustically treat the room to eliminate or significantly reduce (at least major) aberrations, the eq will only be a partial solution: level of the frequencies, not the perceived character of them.

The notion that you lose bass when using bass traps also seems faulty. You only lose what shouldn't be there in the first place -- resonances in the room. The actual bass produced by the speakers does not diminish.

Maybe I'm missing something? I don't know, but this all seems wrong to me.

--Bill
 

Raffles

New Member
Aug 12, 2012
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I'm still somewhat bemused by this thread. Is the contention that time domain correction is wrong mathematically and/or in practice, and EQ is better? Or that EQ is an easier or less expensive way of achieving the same practical result as time domain correction?
 

amirm

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Apr 2, 2010
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I have to say, I believe the whole concept behind this EQ-solves-all discussion to be seriously flawed. It's all fine mathematically, but not acoustically, and doesn't even take into account what the presence and performance of the equalizer brings to the table. Nothing good, usually.

Through my recording and mixing in a variety of studios along with assisting in design of same, I learned from several different mentors a basic theme: that the basic aural characteristic of any room where recording or serious listening is to take place should be as neutral as possible before any electronic assistance is applied. The reasoning, most important during recording, is to not allow the room characteristics to influence negatively the character of the instrument(s) being recorded. But the same can be said about influences a room has on the playback as well.

The waterfall plots (including those with the natural room noise correctly superimposed) show the resonant decay of any given frequency. The length of that decay at each significant frequency influences all sounds produced in that room, regardless of their level. Long decays in a frequency center (the result of a room mode) will extend its audible time, and that by itself gives an entirely different character to how an instrument falling in that frequency range will sound. In the event a resonance is very low level, it may be lost at the bottom of the dynamic range of the room.

Think of it like this. You have a room with a large resonant peak at 70hz, and want to record an upright bass or cello. The room will significantly modify the character of the bass/cello by resonating with it at that resonant frequency. Eq later during mixing will not correct the problem; that note or range will always sound different than the neighboring frequencies. You can lower its level, but the character remains the same. The same thing would be true if you were in the middle of a room null produced by mode cancellations, and were recording an instrument. At certain notes the cancellation itself will have a unique decay which could not be recovered entirely by eq in the problem area.

The same issues apply in a listening room. Unless you acoustically treat the room to eliminate or significantly reduce (at least major) aberrations, the eq will only be a partial solution: level of the frequencies, not the perceived character of them.

The notion that you lose bass when using bass traps also seems faulty. You only lose what shouldn't be there in the first place -- resonances in the room. The actual bass produced by the speakers does not diminish.

Maybe I'm missing something? I don't know, but this all seems wrong to me.

--Bill
Well, let's take a look at this graph where it shows the frequency response of the identical speaker but from different seating positions:



Which line is most faithful the recording below transition frequency? The blue, red or green? What if I changed the room? Would the lines change?

I think the obvious answer is that the room that we put the speaker in completely changes the character of the sound that is coming out of the speaker in low frequencies. It can create swings of up to 30 db. That is 6X different level perceptually (and far more as far as measurement is concerned).

Now, the point you are making is that during recording there were also such peaks. That is very correct. Alas, we don't know what that curve looked like. Not from one venue, or the other million that is used to produce music. So in no way can we ever match what was heard during the recording. All we can do is be faithful to the groves or the bits on the source. That faithful reproduction says don't have your room make up its on "EQ" and change the tones. Because when it does, it doesn't sound right. Somehow as humans we are able to tell that. When I was testing the sub and I played guitar music into it, I could easily tell it was "wrong" when the peaks were there. When I took them out, it sounded much more like a guitar string without the overhang and boominess and seemed to come and go. From Dr. Toole:

"Since the true nature of the original sound cannot be known to listeners one cannot say “it sounds as it should.” But listeners routinely volunteer opinions on scales that are variations of like-dislike, which frequently have a component of emotion. Descriptors like pleasantness and preference must therefore be considered as ranking in importance with accuracy and fidelity. This may seem like a dangerous path to take, risking the corruption of all that is revered in the purity of an original live performance. Fortunately, it turns out that when given the opportunity to judge without bias, human listeners are excellent detectors of artifacts and distortions; they are remarkably trustworthy guardians of what is good. Having only a vague concept of what might be correct, listeners recognize what is wrong. An absence of problems becomes a measure of excellence. By the end of this book, we will see that technical excellence turns out to be a high correlate of both perceived accuracy and emotional gratification, and most of us can recognize it when we hear it."

As to whether EQ is a natural or unnatural fix, I can't agree :). There is no magic in an acoustic material that makes it more right than anything else. Acoustic material can also shape the sound just like an EQ. Here is a simulated absorber:



Clearly the absorption rate varies with frequency and there is even some ringing/vestiges of "comb filtering." Who says that is a good way to get rid of the energy in the room? Maybe it is more natural that we don't energize the room to start than to try to absorb it later! :)

To be honest, yes, an EQ can be poorly implemented and cause its own set of problems. Graphic EQs are one such example. But DSP is pretty cheap these days and we can do a very good job.

Hope I didn't miss your point :).
 

amirm

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I'm still somewhat bemused by this thread. Is the contention that time domain correction is wrong mathematically and/or in practice, and EQ is better? Or that EQ is an easier or less expensive way of achieving the same practical result as time domain correction?
What time domain tool you have in mind? And how does it work?
 

microstrip

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Amir,

I was just re-reading this post from the REW creator in the REW forum. As far as I can see a specified window is shifted along the samples - the time is not really divided in slices and the processing is more complex than just considered in the F. Toole book. Do you know how does this affect the resolution limitations of waterfall analysis? Also as far as I could see the examples shown in the book were taken using a Techron TEF-12 manufactured around 1986, I can imagine that modern instruments have better characteristics.

http://www.hometheatershack.com/forums/rew-forum/59776-waterfall-window.html

I have now re-analyzed my data with 100 and even 66ms windows. The differences between the measurements taken with room treatment and without it seem stable and are comparable with what I got using 3.3 Hz frequency resolution (300 ms).
 

amirm

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Apr 2, 2010
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Amir,

I was just re-reading this post from the REW creator in the REW forum. As far as I can see a specified window is shifted along the samples - the time is not really divided in slices and the processing is more complex than just considered in the F. Toole book. Do you know how does this affect the resolution limitations of waterfall analysis? Also as far as I could see the examples shown in the book were taken using a Techron TEF-12 manufactured around 1986, I can imagine that modern instruments have better characteristics.
There is nothing different in that explanation. What he is covering is an advanced topic I mentioned in my first post I was skipping over. Namely taking a continuous waveform and then truncating it in time as to show that slice, will cause distortion because the real signal did not abruptly start and stop. To fix this in signal processing, we use a slope so that we gradually go to no samples. These are also called "window" functions, not to be confused with the window time selection in REW. The use of windows *functions*means that the actual time resolution is worse than what we think they are. The number of samples in reality is more than what the window setting indicates.


http://www.hometheatershack.com/forums/rew-forum/59776-waterfall-window.html

I have now re-analyzed my data with 100 and even 66ms windows. The differences between the measurements taken with room treatment and without it seem stable and are comparable with what I got using 3.3 Hz frequency resolution (300 ms).
You mean there is always a difference? That is to be expected since you did change the room :).
 

Raffles

New Member
Aug 12, 2012
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UK
What time domain tool you have in mind? And how does it work?
No particular tool, but simply the idea that the source signal has been convolved with the room's (and speaker's) impulse response when it reaches the listener, and that this can be reversed by deconvolving the source with the impulse response before playing it over the speaker. The black art would be in finding the inverse impulse response that works over a larger area; we would probably want to limit it to the lower frequencies for this reason. But what appeals to me is that it seems to start from a mathematically 'correct' position, and we simply temper the correction until it gives us what we want.

I'm guessing that this is basically what all the time domain correction products are derived from, but with various clever black arts being practised on the basic inverse impulse response.

Is there something inherently wrong with this approach that simple EQ'ing of the source can overcome?
 

amirm

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No particular tool, but simply the idea that the source signal has been convolved with the room's (and speaker's) impulse response when it reaches the listener, and that this can be reversed by deconvolving the source with the impulse response before playing it over the speaker.
That's how a number or auto-EQ systems work. But the fact they operate in time domain (i.e. deconvolution) doesn't mean they are doing any time correction, or anything different than a parametric EQ. It is just two different ways of accomplishing the same thing.

The black art would be in finding the inverse impulse response that works over a larger area; we would probably want to limit it to the lower frequencies for this reason. But what appeals to me is that it seems to start from a mathematically 'correct' position, and we simply temper the correction until it gives us what we want.
Indeed, that is what separates the boys from men. :) One technique which draws from the psychoacoustics that I talked about is to reduce the resolution as frequencies go up. What complicates things is that you don't know the speaker response so speaker directivity could derail you too. What looks like a dip may not be correctable because it is caused by the off-axis response being poor.

I'm guessing that this is basically what all the time domain correction products are derived from, but with various clever black arts being practised on the basic inverse impulse response.

Is there something inherently wrong with this approach that simple EQ'ing of the source can overcome?
No, as I mentioned you are describing actually how these systems work. It is just that the implementations are different. Audyssey and some free auto eq systems work this way. JBL Synthesis that I use works on programming a large bank of (high quality) IIR filters for each resonance and/or general EQ. While vendors talk about one being better than the other, ultimately the decisions that they make is the important bit, not how one goes about inverting the response.
 

Raffles

New Member
Aug 12, 2012
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Hi Amir

...the fact they operate in time domain (i.e. deconvolution) doesn't mean they are doing any time correction, or anything different than a parametric EQ. It is just two different ways of accomplishing the same thing.

Is that always true? As I understand it, a time domain impulse response can be represented in the frequency domain as a frequency & phase response. As far as I know, a parametric EQ doesn't give you control over phase, whereas a 'time domain' correction (which may be computed in the frequency domain) inherently controls phase as well.

At low frequencies the two methods may end up looking very similar, but the time domain correction has the potential to be more flexible and subtle, it seems to me.

(But I may be wrong on this!)
 

ack

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May 6, 2010
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The same issues apply in a listening room. Unless you acoustically treat the room to eliminate or significantly reduce (at least major) aberrations, the eq will only be a partial solution: level of the frequencies, not the perceived character of them.

I am in complete agreement with Bill and JonFo: fix the problem, don't work around it; eventually there will always be room ringing until treated. The EQ electronics will inevitably impose their own sonic character. Just fix the room and use multiple subs to smooth aberrations out - even the one I use brings my 50Hz hump from +14dB down to +3dB and smooths everywhere else in its operational range out. These are tried and true techniques for decades now.
 

amirm

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I am in complete agreement with Bill and JonFo: fix the problem, don't work around it; eventually there will always be room ringing until treated. The EQ electronics will inevitably impose their own sonic character. Just fix the room and use multiple subs to smooth aberrations out - even the one I use brings my 50Hz hump from +14dB down to +3dB and smooths everywhere else in its operational range out. These are tried and true techniques for decades now.
An EQ applied to the sub, only changes the character of the sound going to the sub, if there is such a thing. I can guarantee you that the 3 db hump is audible. I am pretty sure if I add or take out whatever character there may be in a good eq to that sub you won't even notice it! I am able to turn filters on and off and the only thing I hear is the amplitude of that resonance changing, and ringing going away. Of course my mains are not effected so they do whatever they do and bring majority of the sound I hear.

So to be clear, in this context we are not saying take an EQ and put it across your entire signal, 20 to 20,000 Hz and go make everything flat. There are issues above transition frequency that could make use of EQ poor. One example is speaker directivity where the off-axis frequency response has dips in it. Once we combine that with the direct sound, we still have a dip. If one attempts to fix that dip with EQ, both the direct and reflected sound change and we will be back to square one. While EQ can partially compensate, clearly it cannot in the absolute. It is these situations, and use of lousy "graphic" EQs that has given this technology a bad name. Inverting a resonance added to the sound with an inverse correction up stream does not do harm. Whereas leaving it their for the sake of being natural, does.

It would be a very unusual situation where placement alone gets you there without EQ. The sub outputs can add in phase and cause peaks that you need to take down somehow. It is very hard at those low frequencies to do that with room treatment alone. The room treatment will also change the character of the sound anyway as I showed in the graph of the absorber.
 

microstrip

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(...) You mean there is always a difference? That is to be expected since you did change the room :).

Amir,

I should have been more precise - the differences seem to agree very well with the expected reality.

IMHO, we are moving away from my concern - the utility and reliability of waterfalls. Most of the time I have seen them used as a diagnostic tool for detecting long decays at some frequencies, and as an auxiliary tool for room treatment. Perhaps I have misunderstood you, but I got the feeling you questioned this application, mainly because of the frequency and time limitations of the method of measuring the decays. However, in my naive measurements I found that using a 100 ms window with 10 Hz resolution mixed with the 300 ms and 3.3 Hz I could identify and correct some issues of my room using bass traps, even at the very low frequency of 37 Hz. Asking a direct question - do you consider that I only lucky, and these waterfalls were useless?

Another question - why do you consider that we need very high time resolution? Should we assume that small irregularities along the decay curve at individual frequency zones are more important than the general trend?
 

amirm

Banned
Apr 2, 2010
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Seattle, WA
Amir,

I should have been more precise - the differences seem to agree very well with the expected reality.

IMHO, we are moving away from my concern - the utility and reliability of waterfalls. Most of the time I have seen them used as a diagnostic tool for detecting long decays at some frequencies, and as an auxiliary tool for room treatment. Perhaps I have misunderstood you, but I got the feeling you questioned this application, mainly because of the frequency and time limitations of the method of measuring the decays. However, in my naive measurements I found that using a 100 ms window with 10 Hz resolution mixed with the 300 ms and 3.3 Hz I could identify and correct some issues of my room using bass traps, even at the very low frequency of 37 Hz. Asking a direct question - do you consider that I only lucky, and these waterfalls were useless?
Let me put it this way. I say you are hearing the reduced amplitude of the peak that is visible in the SPL graph at 56 Hz or whatever it was. The fact that its ringing also went down was a consequence of the same thing so while seeing it in the waterfall may be reassuring, it didn't provide more insight there. As to 34 Hz reduction in time domain only, I remain dubious that anything good was done since the amplitude of the resonance is there. And if there is one thing we know is that we absolutely hear such frequency response variations.

Another question - why do you consider that we need very high time resolution? Should we assume that small irregularities along the decay curve at individual frequency zones are more important than the general trend?
Yes. I have not seen your displays for short and long time window. But in my testing there is a huge difference in the tail (see my examples earlier). BTW, in one of the waterfall displays in Dr. Toole's book he is using 40 msec. I don't recall now which one it was but did notice the short time. BTW, to see the difference I am talking about, you need to calibrate the noise floor. If that is set wrong, then it might look like it is not doing what I said.
 

ack

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May 6, 2010
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It would be a very unusual situation where placement alone gets you there without EQ. The sub outputs can add in phase and cause peaks that you need to take down somehow. It is very hard at those low frequencies to do that with room treatment alone. The room treatment will also change the character of the sound anyway as I showed in the graph of the absorber.

Just take a look at my system's first page, where I have the response graph at the listening position. It's even smoother now than what's shown there - no EQ or even room treatments, just clever positioning; room treatments to come...
 

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