# How do room modes show up on a waterfall plot?



## bjs

My room is 17 feet x 14 feet x 8 feet high. I used this page (http://www.bobgolds.com/Mode/RoomModes.htm) to get the theoretical room modes.

I then put my subwoofer in the right front corner and my mic dead center of the three dimensional room space (ie 8.5 feet from the end, 7 feet from the side and 4 feet high) and generated a waterfall plot. I expected to see strong nulls at the 3 fundamental room axial modes but don't. In fact, other than the room clearly being "modal", I can't correlate much of what I see with the room mode calculator.

This leads to a related question. At this point I'm not trying to figure out the best place for the subwoofer but to understand how to measure and analyze room modes in preparation for treatment. What do you pros do when faced with treating a new room? How do you approach the investigation/analysis in order to determine what treatment is required?


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## bpape

Well, dead center is the worst place to sit and the corner is the worst place for a sub. Try the mic 62% back from the front wall and at ear level with the sub in the corner and that will tell you what you have to deal with.

Bryan


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## bjs

Well yes, but that wasn't quite my question.:dunno:

As mentioned I'm not trying to find the best listening/sub position at the moment...I'm trying to learn how to indentify and measure room modes. Can it be done with waterfall plots? If so, how do the room modes show up on a waterfall plot?

For example, the calculator says that the axial modes should be strong. And the first modes the strongest. And they are listed as 33 and 40 hz for example yet on the waterfall I see little evidence of their existence. I'm assuming strong peaks and strong nulls are room modes. Yet when I compare them with the calculator the peaks/nulls either don't correlate or correlate to what should be a very weak mode. It's not making a lot of sense...hence my questions...!


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## thewire

To identify room modes you can place the subwoofer in your listening postion, and the mic in the corner.

Bryan, is 25Hz a low enough room response that corner placement will cause issues? I read that corner placement is not good for low frequency capable subwoofers and since my sub only does as low as 19Hz (but I only hear 22Hz) I'm not sure it would not matter. Do you have an opinion on this? I have four subwoofers and placing each on my left & right wall they are not equidistance from placing each on my front and back wall (center). Each left and right subwoofer would end up having less lower ouput. Would it be better to place on in each corner? Currently I have one in each front corner, a DIY center channel subwoofer, and one behind each seat in the back row. I could perhaps equalize the left and right to match the front and back but that is the only option I have to adjust besides volume level. Would require allot of eq that way. 

Is it more important to have them interacting with room modes postively, versus flatest possible response by placement?


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## bjs

This thread is about the analytic tools used for measuring and identifying room modal reponses, most specifically the waterfall plot. Can we keep it on topic please.


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## thewire

Yes sorry. Says right here page 331 in my book under chapter "Modal resonances in enclosed spaces" subject "Experimental verification".



> To evaluate thier relative effects, a swept-sine-wave transmission experiment was set up. In effect, this measures the frequency response of the room. Knowing that all room modes terminate in the corners of a room, a loudspeaker was placed in one low tri-corner and a measuring microphone in the diaganal high tri-corner of the room.


Source - Master Handbook of Acoustics fourth edition by F. Alton Everest. If this is vilolation of copyright, mods please delete.


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## bpape

bjs

Sorry. I understand your desire to be able to identify the room modes. In looking at your plot, there is a severe dropoff below about 65Hz. Are you running speakers and subs both below 80Hz? I promise I will address your question but until I can see a valid response plot. Something is awry which may be hiding things. Can you please expand the vertical axis?

Yes - you can use waterfalls to help identify modal issues. However, you will see other things which look very similar which are not axial modes. You may find yourself in a situation where you SHOULD see a mode where other factors are causing a cancellation of that modal peak (or a reinforcement of a null).

Bryan


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## brucek

> understand how to measure and analyze room modes in preparation for treatment.


bjs,

I'm afraid you're looking in the wrong place to achieve your goal. Waterfall plots are useful for frequencies below ~200Hz to analyze the modal response of a room. Room treatment has fairly limited effect at those frequencies, and normally is directed to solve problems above that band (size of the treatments being the limiting factor at low frequencies where room modes dominate).. 
(BTW, it's a good idea to switch your waterfalls to logarithmic mode rather than linear, and use a vertical scale of 45dB-105dB). 
Also note, that theoretically calculating room modes, (as compared to the real world where all these modes mix and cancel, etc), isn't that useful. You need to measure....

Above the modal response area, RT60 is used as the indicator, and really only when the room is large, but this is certainly the frequency range where room treatment is effective (~100-500Hz).
RT-60 displays the decay across the entire band or a selectable section of it (i.e.octave bands). It's basically reverberation time, the time it takes a sound to decay 60dB or 1,000,000th from its initial impact, or sound pressure level. It's very revealing and will show the fruits of the work you're doing. Think of what a church sounds like and you know what high RT60 is.

Generate the RT60 from the impulse response screen (after a full range sweep) and then take a look at the RT-60 tab. The scale is frequency versus time (in portions of a second from 0 to 1 seconds).. Ignore below 200hz. A flat line of about 0.3 seconds would be considered ideal. 

brucek


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## bpape

Realistically, the idea is to place your sub and then play with seating position until you smooth the best you can. After that, we play with sub positioning to try to further refine the response. Next comes the mains but this also has to blend with the sub.

Once you've done all that, then you'll know what's left to deal with and should have a good idea of what is causing the specific problems - which leads to how to treat them.

Bryan


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## Guest

The cumulative spectral decay plot, often known as the waterfall plot, can _indeed_ be used to identify room modes!

Rather than simply display the frequency response, you also have the advantage of seeing the behavior over a slice of time (but, avoiding the math and the exact methodology employed in their generation, please be aware that this is not truly a time domain measurement!).

How you produce them is dependent upon quite a few parameters. And you will soon see that there is a bit more to this than simply taking ‘a measurement’.

Before we get into the process, being cognizant of a few basic characteristics of tuned enclosures will be helpful.

The physical size of a sound wave (wavelength) determines how the wave will interact with its environment.

If the wavelength is large compared to an object that the wave encounters as it propagates, the wave will diffract around the object. If the wavelength is small compared to the size of the object, the object becomes an obstruction that reflects the wave and a shadow is formed behind the object. 

When sound is radiated between two surfaces that can reflect the wave, the short wavelengths (relative to the spacing of the surfaces) produce “flutter” echoes as the sound reflects back and forth between the surfaces. 

As the wavelength becomes larger and the frequency decreases relative to the obstructions encountered, the wave can ‘meet itself’ coming and going. 

If the maxima (peak/node) and minima(null antinode) occur at the same location for each wave, a “standing” wave is formed. This standing wave can be thought of as a stationary sound wave, even though the waves themselves are actually moving. The sound pressure distribution between the two surfaces becomes fixed. One’s movement within the room allows one to move to and from areas of acoustical high pressure (peak/nodes and low pressure (null/antinode) as uneven areas of increased or decreased sound intensity at the affected LF resonance frequencies and their integer multiples.

It is also useful to remember that in a closed tuned pipe, the fundamental high pressure regions, or nodes, will occur at each terminated end. It the tuned pipe is open, a low pressure antinode will be present at the open end of the pipe. The room functions as a tuned pipe with resonances based upon the LF fundamental and its integer multiples referred to as harmonics. 

And easy way to experience this is if you have a listening room that has one rear 'closed' corner while the other rear corner/wall opens into an adjacent room. This is common in many living rooms or dens. 

If you play a bass heavy source, walk to the enclosed corner and you will note overly emphasized muddy bass while at the other rear open corner the bass will be the opposite' - much sparser and lower in intensity.

Any deviation from an ideal rectangular shape is going to cause a variation in the distribution of the various modes. Additionally, irregularly shaped rooms, especially those with alcoves, or opening into adjacent rooms will result in a complex interaction of tuned spaces - referred to as 'coupled spaces'. The response will be a very complex combination of each individual region summed non-linearly with all of the other possible acoustic ‘spaces’ - with each combination and permutation of possible configurations coming into play. 

The mathematical modeling of such a complex space is very complex requiring significant computational capabilities (for a very simple example, witness the run times of RPG's Room Optimizer program which can often last several days!- and note that this program assumes a most basic ideal rectangular space without any variations such as even flush doors or windows!).

As with comb filters, room modes are harmonically related and distribute evenly on a linear scale. In addition to predicting the modal response, Room Optimizer can also determine the optimum placement of woofers and full-range loudspeakers – assuming that you have control over constructing a room that closely complies with the model room assumptions.

Rooms with irregular geometry require finite element methods to calculate their modal response. These calculation-intensive applications are just beginning to fall within the reach of desktop computing capabilities. To my knowledge, the best current tool for running calculations for coupled spaces is in the EASE acoustic modeling software, despite its having a few restrictive issues with small acoustical spaces (note – the distinction between a large acoustical space and a small acoustical space and their associated characteristics as defined by Manfred Schroeder (and others) are well established and important to note!!) - and we are anxiously awaiting an anticipated upgrade to this capability in the next release as the advance in computational power makes the calculations a little more feasible - but STILL daunting at best.... But back to the real world...

And depending upon your overall goals (which are not necessarily as simple as just optimizing a single seat in a dedicated room!), there are a variety of methods to explore this complex environment... And I will hazard to say that variations in the results of this measurement are reflective of the manner in which it is conducted and of the windowing capabilities and limitations of the equipment used to perform it. (As I often see them, I am assuming that programs such as ETF/RPlusD are sufficiently capable to do this – but many are not! So caveat emptor!)

There are a variety of common methods that I will touch on briefly as they provide insights into different aspects of the systems performance.

In order to determine the general characteristics of the room itself, measurements are taken with the loudspeaker and microphone positioned in diagonally opposite corners of the longest dimension with a time window large enough to provide about 1Hz of frequency resolution. Various modes ‘tend’ to “pile up” in the corners of the room and the placement of the speaker in a front lower corner excites the most modes. Placing the microphone in the diagonally opposite upper corner along the largest dimension measures the ‘most’ modes. 

(Note! Contrary to popular opinion, the corner is Not the location of maximum velocity for all of the nodes! But it is the closest practical common loci to capture all 3 modes in a closed corner in a single measurement!)

And while typically a swept frequency signal is used to drive the room, depending upon the capabilities of your particular measurement rig, it is interesting to correlate this with a stimulus consisting of pink noise over time.

And while this can give you a good idea of the room, it does not necessarily provide the specific information you will need to specifically address the issues of specific interest to your application, as you are going to need to deal with the listening position(s) as well. It is also important to measure the modal response using the actual woofer and listener locations, since all modes may not be equally excited under these conditions.

Software programs such as RPG’s “Room Optimizer” can aid in determining the best locations for loudspeakers and listeners (…assuming the restrictions in room design.)

It is with this later aspect of measuring the actual locations that the process becomes a 'bit' more complex. Not only is this an iterative process, but if the room is a complex shape, you are going to be a busier with more measurements.

Further complicating the process is the actual combinations and locations of the sub(s) that you are going to be using. And while this subject is far beyond the scope of what I hoped would be a short post, you can find useful resources by Floyd Toole regarding subwoofer and multiple subwoofer placement to mitigate room modes, who has several papers available at the Harmon website regarding the placement of multiple subs in a rectangular room. (If you have difficulty finding these, PM me.) Additionally, for a single location, general multiple LF/sub array orientation guidelines can be summarized as (be aware that these guidelines extend to larger arrays in large acoustical spaces as well, but are applicable to two units in a small acoustical space as well. Please note that these deal primarily with the summing characteristics of the units, with acknowledgment to room boundaries, rather than the focus on room nulls!: 

1. For most venues, sub arrays should be stacked vertically to narrow vertical coverage and maintain horizontal coverage.
2. If subs are to be split horizontally, they should be as far apart as possible (> 5 wavelengths).
3. If subs MUST be arrayed horizontally, consider the Bessel array.
4. Cardioid subwoofers can be useful for controlling rear radiation.
5. Remember that the polars are frequency dependent, and a good polar at one frequency will produce the opposite effect at twice (or half) that frequency. For this reason, it may be necessary to use multi-way subwoofer systems with each passband optimized for the desired polar. 
6. The room’s response will interact with the subwoofer array response, and can nullify the benefits of the array.
7. A rigid boundary near a subwoofer (floor, wall, ceiling or combination) will produce a reflection that can be treated as an additional low frequency source. The principles of subwoofer spacing apply, and the best coupling takes place within 1/8 wavelength of the surface, producing a mirror image that couples with the sound source as though separated by 1/4 wavelength.
(P. Brown)

So, being cognizant of the above references (sort of like the proverbial “and then a miracle happens” so often employed in your physics class lectures), the next steps are iterative, meaning that we may have to repeat the measurements many times evaluating many combinations of sub locations and listening positions, as the primary problem is that we will be located in an antinode or null. Neither traps nor EQ can correct for a null.

So a fundamental goal is to locate the sub in a position relative to the primary seating positions where they are not located in a null. Likewise, the means of mitigation to attenuate the nodes and damp the resonance are interactive with the room, meaning, as you apply various LF trap topologies, they change the room response and subsequent locations of the nodes. So iterative means are necessary to optimally place and (re)tune high-Q LF traps – of which tuned enclosure and resonant panel Helmholtz resonators are optimal in that they selectively attenuate the fundamental resonate frequencies while preserving the finite mid and high band acoustic energy in the small acoustical space to be addressed as necessary at a later point in the tuning of the room. 

The priority list for achieving good low frequency behavior in small listening rooms, in order of priority, is:

1. Room shape
2. Loudspeaker/listener placement
3. Room treatment
4. Electronic equalization (limited to the mitigation of ‘peaks’/nodes)


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## Guest

*RT60s in a small acoustical space.....hmmmmm.*

RT60s in a small acoustical space.....




As mentioned above, the CSD/waterfall plots are most useful for the identification and mitigation of LF room modes.


But what do we use to examine the behavior of a small acoustical space above the frequencies dominated by modal behavior where the wavelengths at issue become small relative to the obstacles they encounter?

Unfortunately, there persists the notion of using RT60/30 and their variants in small acoustical spaces. This was a rather common notion 35 years ago, but has since been eclipsed by a greater understanding of acoustical behavior in what is defined as a small acoustical space (as distinct from, yup, you guessed it, the large acoustical space!)

It is time we moved on to understand and to embrace the new acoustical model suggested by Schroeder that has been rigorously verified, for all of our mutual benefit.


In a small acoustical space, the use of RT60s (and any similar variant, regardless of weighting) is completely erroneous, and indicates that we are misapplying a measurement that has no correlation in physical fact in a small acoustical space.

Fundamental to this is an understanding of the distinction between a large and a small acoustical space. These terms are not used casually, as they have very specific conditions which define and differentiate the two. And a failure to distinguish between the two will result in some rather unfortunate consequences. 

This distinction is not only the subject of much of Manfred Schroeder’s work, but it is a fundamental concept that underlies all of his work.

And to reduce the distinction to its most basic functional difference, a large acoustical space features a developed statistically random reverberant sound field where reflections at any location are equally probable to radiate from any direction. Conversely, a small acoustical space LACKS a statistically random reverberant sound field where reflections at any location are equally probable to radiate from any direction – and instead, is DOMINATED by focused specular reflections which are definitely identifiable as decreet phenomena as a vector with both direction and intensity and a discreet time of arrival and decay that an be measured and all characteristics identified and isolated via such measurements as the envelope time curve (ETC).

The RT60/30, etc. are ONLY suitable to measuring the decay times of a statistically random reverberant sound field. As such, they have no place in the small acoustical space. If one only has a hammer, all of the world begins to look like a nail, and that is _precisely_ what has happened to the mis-application of the RT60! Instead, the proper tools that accurately identify the real phenomenon in the small acoustical space are to be utilized.

Someone might attempt to use a rock to drive screws, but few would be so bold as to suggest that this methodology is suitable for use in the art of cabinetry. And if you are going to expend the effort and the money to do the job, it is incumbent that we use the proper tools that afford us accurate atomistic insight into the actual phenomena of focused specular reflections, rather than simply using the wrong tool suitable for measuring that which does not exist in a small acoustical space.

Understanding this also leads one to use absorption surgically instead or randomly and with prejudice! As being able to see what is actually occurring has a profound impact upon one’s acoustical world view. And as we have had the tools to do exactly this now for over 30 years, it is time that more in the world of acoustics move up and embrace the current models. After all, even the most stalwart of nay saying classical physicists have begrudgingly admitted that there just might be something to quantum electrodynamics!

Its time to put the idea of using RT60/30/etc. in small acoustic spaces, to bed!

To quote from *Sound System Engineering*, 3rd Ed.[/B] Pp 178-9:

_“Small Room Reverberation Times

To quote the late Ted Schultz (formerly of Bolt, Beranek and Newman)
“In a large room, if one has a large sound source whose power output is known, one can determine the total amount of absorption in the room by measuring the average pressure throughout the room. This total absorption can then be used to calculate the reverberation time from the Sabine formula. This methods fails badly in a small room, however where, a large part of the spectrum of interest lies in a frequency range where the resonant modes of the room do not overlap but may be isolated…In this case the microphone, instead of responding as a random sound field (as required for the validity of the theory on which these methods depend), will delineate a transfer function of the room… It does not provide a valid measurement for the reverberation time in the room.”*

What is often overlooked in the attempted measurement of RT60 in small rooms is that the definition of RT60 has two parts, the first of which is commonly overlooked.

1.)	RT60 is the measurement of the decay time of a well-mixed reverberant sound field
Well beyond Dc (the critical distance).
2.)	RT60 is the time in seconds the reverberant sound field to decay 60 dB after the sound source is shut off.
Since in small rooms, there is no Dc (critical distance), no well mixed sound field, hence no reverberation but merely a series of early reflected energy, the measurements of RT60 become meaningless in such environments.
What becomes most meaningful is the control of early reflections because there is no reverberation to mask them.”


What you have instead is a small acoustic space dominated by room modes (which are not reverberation!) and focused spectral reflections – which by definition are anything but a well-mixed reverberant sound field wherein the arrival of reflected signals from every direction is equally probable.

In the small acoustical space, there is no mixing, nor homogeneous, statistical reverberant sound field! In fact, in a small acoustical space, as the lowest frequency that can effectively develop across the largest dimension can easily increase to ~500Hz – compared to a large acoustical space where such frequencies are often below 30Hz!

In such environments, intelligibility can be degraded by specular reflections that must be isolated and corrected directly, not statistically. 


*Note, It is equally proven that the fundamental form of Sabine’s expression cannot be modified so as to become correct for large absorption. Per Sabine’s Reverberation Time and Ergodic Auditoriums, Wm. Joyce, Journal of the Acoustics Society of America, Vol. 58, No. 3, pp. 643-655,Sept. 1975. 
_


Imagined non-existent RT60s, or some fascinating “similar but scaled down analysis” has no basis in fact or physics! And they are of no valid use here. The simple persistence of acoustical energy is of little use in a small acoustic space! 

An RT60 _is_ useful in a large acoustic space as the statistical reverberant field is sufficient to mask specular reflections. But this is NOT the case in a small room where the specular reflections EASILY dominate over the all but nonexistent statistically reverberant sound field (where they exist only at frequencies of interest to UHF engineers, dogs and bats!).

The simple persistence of the acoustical energy fails to provide ANY insight into the specific nature of specific focused specular reflections which are the real issue of concern! Such a measurement fails to reveal the intensity, arrival time, or any information that aids in the identification and behavior of, let alone provides specific information regarding the effectiveness of the surgical treatment of any focused specular reflection – and by their nature such a measurement would be location specific as you are NOT in a statistical reverberant field! And an average over many locations would be completely nonsensical! In fact, such a concentration leads only to the shotgun application of absorption – precisely that which we are trying hard to avoid!

It is the persistence in the belief that completely invalid variations of a technique designed to evaluate that which does not exist in a small room that has held back a real understanding of acoustic behavior in small acoustic spaces! Regardless of how one squints and tilts their head when evaluating such nonsense in a small acoustic space.

I realize that this steps on the toes of a few here. But its time to move on to the new tools and to use the measurement techniques which we have at our disposal to better and more accurately understand the physical phenomena of acoustics and sound for all of our benefits.


I will be posting this in a separate independent thread as well (with a few additional comments as to _the source_ of this common misconception), as this topic comes up in _many_ threads...


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## bpape

I assume you're addressing me and that's fine. 

I will agree that RT60 and RT30 do not apply to small spaces. One can, however, use the same calculations to determine RELATIVE persistence across the frequency spectrum with the end result only being used to properly address what portions of the spectrum may need more or less absorbtion (broadband or targeted). They do not present accurate target ranges specifically for the reasons you've addressed and should not be used for such.

That's all I'm going to say on the matter as I'll not have another thread turned into an uncivilized debate. 

Bryan


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## bjs

A lot of good comments and interesting points of view. Coolio! 

I verified that my original waterfall plot was correctly taken except I forgot to apply the mic calibration My observation remains: the axial room modes do not show up as nulls which I would have expected given the centered mic position. However I'm not so fussed about this given the comments in this thread...clearly real world rooms behave in a much more complex manner than these simple models suggest.

Anyway, older and wiser, I will now take the advice to "just measure it" and go from there! :doh:

Although I'm not sure I ever got an answer on how room modes show up on a waterfall plot. Presumably peaks and dips with ringing are modal responses. But is that the criteria for identifying them? REW purports to find them...does anyone know what it looks for? Maybe no-one cares...but I would have thought non-modal induced peaks and dips would be handled differently (ie less inclined to EQ them, more inclined to absorb or diffuse them etc). :scratchhead:


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## Guest

The modes 'show up' in a waterfall plot as 'persistent peaks' - meaning that they have a greater amplitude and they persist in time - in other words they appear as a ridge. Also, as I am not sure of your expectations, you are going to see the gross summations of the total modal behavior! Which is what you need to address anyway. 

Remember, just because you have a measurement, that does not mean that you are finished. 
Instead, exactly 'how' the measurement is made, including the capabilities of the tool, have a great impact upon your final results, and such seems to be the case here.

Just a few quick cursory observations…{as I don't have MatLab on this computer and I quite literally don't have the time to do all of the manual trig to determine each effective reflective path length relative to the side wall and ceiling (and you cannot rule out the rear wall and possibly floor) to determine the corresponding time of travel from source to each first incident reflection point to microphone in 3space...}

(Besides, I have become spoiled as TEF and EASERA will display the time/distance variable breakdowns for various absolute and relative reflections if we just run a corresponding ETC of the same setup! ;-) )

Likewise, I have no idea of the window and resolution settings or limitations of your rig...

So....

Below ~50 Hz I question the measurement, which I suspect is a direct function of the windowing/ resolution of the particular measurement system. 

But you nevertheless appear to have resonances centered ‘about’ the 32, 37, 47, 63, 85, 110, 135, 160, 180, 223 Hz points. (Another comment here…using the log scale is great if your display covers a bandwidth of 20kHz, but for simply a 200 Hz window, use the linear scale! However in such a limited window, I would prefer not to compress 100 Hz of response in the display and obscure measuement…But that is a personal choice…)

Additionally, you appear to have severe comb filtering effecting your measurement due to the superposition of the reflection(s) from the ceiling and side walls as indicated in the severe notches at 120-145Hz and 190-210 Hz regions. 

Again, not to belabor the point, but at the same time also acknowledging that you cannot ignore the facts; without the additional abilities of the ‘larger’ rigs such as TEF using time delay spectrometry, you lose significant ambient noise immunity with the lack of tracking filters to effectively adjust the windowing capabilities and to effectively 'remove' the destructive reflections. So again, your measurement rig and technique is going to become interactive with the response you are attempting to measure. (you will see again and again that the larger tool sets provide many capabilities and much additional derivative information – and hence this what also determines their additional cost… And to go one step further to illustrate this, in large halls, as running a million chirps tends to drive one crazy, with the TEF/TDS we are able to put something on that we like to listen too over the system under test while we simultaneously run sweeps; whereas with MLS and other techniques, noise immunity is a severe limitation. This is not to say that you absolutely need them, but when one is trying to make a living doing this and your goal is to get in and get out as efficiently as possible, versus having the luxury to take as long as you like as it is a hobby, the returns translate into real considerations ;-) ) 

For the wavelengths you are measuring, your window is going to have to be large, and this is also necessarily problematic for any measuring device, exacerbating your technique and control of associated variables which will vary (to a degree) with your tool.

Noting that you are not so much measuring the room’s response as you are measuring the locations response, the deviations from the old chart is not at all surprising. They will not be identical to the total summed room response. If you are desirous of comparing the calculator’s response, you want to drive the room and in order to do this I might suggest placing the microphone ‘in’ the pressure zone of the diagonally opposite corner facing away from the speaker.

In the listening position you are going to be measuring variations from that overall room measurement. For instance, if you are in an axial null, the tangential and oblique modes will dominate, etc. Hence why the process becomes iterative with modifications to the listening position and position of the sub in an attempt to minimize the summed destructive interference amidst a complex environment.

Anytime you make a measurement you are going to need to step back and determine if the results are reasonable. Obviously there is a quality/quantity that is unknown, or why would you be measuring it… But never cease to question things that look too good to be true. For example, if you are measuring the response of a 3 inch driver and you see a prodigious ‘bump’ in output at 60 Hz, would you reasonably expect a 3 inch driver to output significant energy at 60 Hz? If you are anywhere but Bose (sorry, couldn’t resist!) you might start looking for a source of 60 cycle hum in your measurement rig.

Another aspect to consider is the test’s repeatability. Can you reliably repeat the experiment and achieve the same results. Can another person, given the same unit under test?

I am not able to tell yo9u exactly what the performance parameters of your tool are. I simply don’t know, so this will necessarily remain a 50,000 foot analysis…

But a few issues to consider in any acoustic measurement environment that often show up to bite you in the posterior if you are not vigilant…

First is the uncertainty principle…When a measurement is taken, there is some limit to the resolution, or how much detail we can resolve. If we make measurements with a resolution of say 1ms, then we are going to be unable to see any events that occur in a shorter time frame than 1ms. Now this doe not necessarily mean that we can’t see them! In fact we may, but they will be imprecise and distorted – a blur, just like if you need glasses and try to read the fine print that is beyond your limits of visual resolution.

In audio, if we make measurements with a resolution less than say 1 kHz, then any details less than 1kHz will be blurred. Likewise, if we repeat the experiment with a resolution of say, 500Hz, then more details between 500 Hz and 1kHz will become clearer, but we will be unable to clearly see anything below 500 Hz.

Because time and frequency are reciprocals, as our acuity in one domain increases, the corresponding accutiy in the other domain decreases. This is a function of the relationship, and not our equipment. Thus, as our acuity in one domain approaches infinity, our acuity in the other approaches zero. 
This relationship is unity, meaning that we are bound by this relationship. As we gain acuity in one domain, we lose it in the complementary domain.

To bring this to bear on the situation we are facing here… If we wish to measure, say the response of a speaker with a resolution of 20 Hz, our resolution is 1/20, or .05 seconds. At STP(standard temperature and pressure), this .05s corresponds to a distance of 56.5 feet. This means that any reflections within that distance will be included in the measurement an will contribute to a false reading relative to our initial goal. If the resolution is changed to say, 500Hz which corresponds to a time of .002s and a distance of 2.26feet, true anechoic measurements can actually be taken. 

Technologies such as TDS (time delay spectrometry) exhibit adjustable tracking filters which track the stimulus and bound the reception of the response signals and filter out reflections outside of the bounded region, effectively enlarging our scope by filtering out destructive interference and enlarging our scope.
But nevertheless, we are all subject to the same fundamental issues – regardless of the technologies that may enhance one technique over another in a particular environment.

And I believe we are seeing aspects of some of them here. For instance, when 2 (or more!) signals arrive at the measurement microphone with nearly identical levels with one slightly delayed in time, we experience a phenomenon known as comb filtering. And depending upon our resolution, this may or may not be readily apparent, but it is nevertheless real! This is exactly the result when we set too large a time window and a reflection is included along with the response in which we are really interested. Such notching is far apart if the signals arrival times are close, and closely spaced in the arrival times are far apart. On a log scale this can be difficult to see. And hence one of the reasons linear scaling is becoming increasingly popular.

This type of interference is common whether the source be two speakers or a real and a virtual source such as is generated by diffractive or reflective sources.

When making frequency measurements, the frequency resolution must be considered. In order to measure a frequency, there must be sufficient time to measure at least one full cycle. Thus, if have, say, a frequency resolution of 500 Hz, the period of one cycle is 1/500 or 2ms. A general rule of thumb is that the data will be reliable from a frequency that is equal to ½ of the resolution frequency. Thus, with a resolution set at 500Hz, we can reasonably expect the data from 250Hz up to be sufficiently reliable. The problem arises when we do not recognize this relationship and we assume the additional data that will be displayed to be reliable – as far too often happens!!!

Noise is another critical issue. (TDS is highly immune to noise, so that unless you are using very high sweep rates in very noisy environments, noise is not a problem.) However for other platforms, and always for ETCs as a wider filter bandwidth is required, In many cases noise is interpreted as part of the system under test. Level adjustments can help. 

{In the other thread it was asked to what TEF, TDS, EASERA, etc. referred. Here it might be useful to mention that one of the advantages to TEF/TDS is the ability to effectively and accurately measure down to dc- 0Hz – in addition to ‘negating’ or extending many of the heretofore mentioned limitations. And thus it is an excellent instrument to also measure the complex impedance of a device…which leads us to another entire realm of Nyquist and Heyser spirals displayed in 3space and an entirely amazing and feature rich environment that has yet to be fully explored!}


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## bjs

mas said:


> The modes 'show up' in a waterfall plot as 'persistent peaks' - meaning that they have a greater amplitude and they persist in time - in other words they appear as a ridge.


Given room modes are standing waves, one would expect they could also manifest as initial depressions that ring on into ridges. After all, the microphone may not necesarily have been placed in a peak of the standing wave.





> ...but for simply a 200 Hz window, use the linear scale!


I also prefer linear in this case, especially for room modes which are linearily related. However REW doesn't display waterfalls in linear scale properly so I have to use log.

I tried the standard scaling used here at HTS but found it poor for waterfalls. A 45dB floor is much too low creating a great deal of artifacts obscuring the key results. Especially in my noisy measuring environment!





> Additionally, you appear to have severe comb filtering effecting your measurement due to the superposition of the reflection(s) from the ceiling and side walls as indicated in the severe notches at 120-145Hz and 190-210 Hz regions.


Quite correct, and front/rear walls too! Attached is the ETC plot for this same position. Unfortunately ETC plots are not perceptually uniform in frequency. Lower frequency reflections are harder to see and easily masked by higher frequency reflections. However since I tested the subwoofer with a wider than normal bandwith the reflections are visible. The flutter echos are perhaps reasonable given the room has nothing on the walls...just hard flat gyproc surfaces.












> Another aspect to consider is the test’s repeatability. Can you reliably repeat the experiment and achieve the same results. Can another person, given the same unit under test?


Well, like Britney Spears...I did it again...! A week later and this time with higher frequency resolution. I probably put the speaker back within a few inches of the original test and the mic also (mic is probably within an 1inch). And I used a different mic this time... a cheap Radioshack SPL meter (it probably cost me $30 some thirty years ago). And I've learned how to properly save and upload my graphs so it looks a little different this time. Having said all that, the results look very similar.








Your comment about stepping back and checking the results is a good one. The ridges at 16hz and again at 24hz are due to heavy industrial air conditioning nearby.

Anyway, it seems I have enough tools ready and they make enough sense to me that I'm actually ready to start measuring and treating my room! I start a new thread for that and see where it goes. People can watch me learn "on the job" as I'm sure I'll do some daft things at times...but that is the great thing about this site...a lot of experience and expertise for the newbies!


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## Guest

Just a comment and a suggestion that I hope will help in the future...

That response is looking much nicer...:bigsmile:

I've not spent much time with equipment that displays the particular format of the ETC that RoomEQ seems to show - of an averaged line connecting the amplitude peaks of the various reflected specular signals as opposed to also displaying the individual constituent reflections.... as discreet events in time and amplitude...

In order to identify (and verify) the actual sources of each peak and null, you can take a small piece of OC703 or equivalent, and standing in a spot behind the mic, preferably in its null so as to avoid contributing to the measurement with your reflection or shadow - verified by repeating a known good measurement - and holding the absorptive material such that it intercepts what is assumed to be the signal pathway. If you have assumed correctly, subsequent measurements with the absorptive material impeding the pathway will show a reduction in the specific ETC response gain corresponding to the particular arrival time of the reflection. 

I suspect this has already been suggested elsewhere in detail, but if not, I offer it simply as a helpful suggestion...

And once you have addressed the room modes, you will want to switch your point of view to the ETC and the time/amplitude perspective rather than the frequency perspective.

Have fun!


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## jerome

Very interesting reading mas! I'm not sure I got the more complex details but please continue explaining us newbies how things work :yay:

Completely OT but I was just wondering: in short, what is your background? Should we be aware of which company your work for?


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