What is the Optimal Measurement Strategy for Practical Living Conditions?

caesar

Well-Known Member
May 30, 2010
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What is the most effective measurement process from a practical standpoint, for normal living conditions vs. a studio?

- Do you get all of your treatments out of the room and measure a baseline?
- Do you measure one speaker at a time? From the listening position? From other positions? From how many positions?
- Do you add your treatments one by one, and re-measure?
- What in the measurement data determines if you need to absorb or diffuse? Specific peaks or nulls?
- How do you know from the data if you should tweak the position of the speaker or move the treatment?
- What data do you look at to determine if it's the floor, ceiling, or sidewalls that are causing problems?
- How do you know from the data how many diffusors to add?
- What don't you know from your measurement data?

Thanks!
 

DonH50

Member Sponsor & WBF Technical Expert
Jun 22, 2010
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What is the most effective measurement process from a practical standpoint, for normal living conditions vs. a studio?


All just my quick and rambling random thoughts, not to be considered an expert opinion.

- Do you get all of your treatments out of the room and measure a baseline?
Hopefully you measure before adding a treatment and that is your baseline. In my case I did that, then a computer crash lost the data. :( I have no plans to pull treatments down, however. At this point I want to optimize the room I have, could remove treatment if needed, of course.

- Do you measure one speaker at a time? From the listening position? From other positions? From how many positions?
I measure the L/R pair at the main listening position, then the L/R pair at other listening positions. I may adjust things to get a better average across the seats but in my case I have enough treatment that things do not vary much.

I tend not to worry much about surrounds more than a basic spot check. Their position is usually fairly fixed by the room layout and I am much less sensitive to their sonic attributes for movies. I will tweak the center to get it aligned with the L/R mains.

- Do you add your treatments one by one, and re-measure?
Well, this is a function of how much time you have and your knowledge of what needs doing. I would typically add a couple of corner bass traps and perhaps treat the first reflection points then measure again. From there it depends on what the measurements show and what I want in the sound. I would not generally add single panels (absorption or diffusion), usually go in pairs in corners and at the sides and ceiling. Unless you mean specific points by "one by one", as in treat the side reflection points, then ceiling, etc. I tend not to do that for lack of time (too busy, too impatient, take your pick ;) ).

- What in the measurement data determines if you need to absorb or diffuse? Specific peaks or nulls?
Cost is often the criteria as diffusors tend to cost much more than absorption panels... However, in general higher-frequency issues can be dealt with by either type so it depends upon my budget and desired sound: diffusion for a more diffuse "all-encompassing" sound field; or, absorbers for a tighter image with sound field essentially determined by the recording. Diffusors must be a significant fraction of a wavelength so tend to get too large to help in the bass frequencies in most rooms, so absorbers are used. Of course, you need a lot to deal with a deep null...

Reverberation time may also play into my decision; I prefer not too long as it muddies the sound so in a very live room where I can't (or can't afford to) place diffusors everywhere I will lean toward absorption.

- How do you know from the data if you should tweak the position of the speaker or move the treatment?
Well, you can calculate wavelengths and see if moving things around will help. For bass room modes about all you can do is move the listening position (or place a sub near-field to help compensate a null). Drawing ray diagrams should determine where treatments should be placed so there should not be a lot of guessing... Again, speaker placement is usually dictated by the room and so minor speaker placement tweaks can be useful but chances are there's a fairly narrow range of adjustment.

- What data do you look at to determine if it's the floor, ceiling, or sidewalls that are causing problems?
Look at the frequencies and compare wavelengths to distances (speaker and reflection points to listening position). For room modes, room dimensions determine where peaks and nulls will be (physically and at what frequencies).

- How do you know from the data how many diffusors to add?
Based on initial measurements you may decide which of the first reflection points to trea. From there it is usually a matter of taste to establish the reverberant field desired. I don't have a specific rule of thumb for the number. Iwill typically treat first reflection points (side walls and ceiling) and the wall behind the listening position unless it is a goodly distance away. I will also treat the wall behind dipole speakers, unless the room is large enough to get them well out from the wall behind to minimize filter/cancellation effects.

- What don't you know from your measurement data?
Most measurement systems work with frequency response, leaving time domain a guess. REW and others do perform decay/reverb measurements so you can help determine how much diffusion/absorption to add at what frequencies but for most audiophiles determining exactly how much reverberation desired is probably not obvious. The final imaging and sound field (spatial field) is often hard to determine from FR and waterfall plots, though I am sure others on WBF are much better at reading and applying waterfall measurements than I (30+ years ago I did not have tools that generated waterfall plots so, while I have used them for RF and more recently acoustics analysis, they are still new to me).

I use a system that performs impulse and step (time) response as well as frequency response and waterfall plots. That said, except to time-align the speakers at the crossover point, I don't do much with it. Largely because my current room is fairly small and has enough issues that I just added absorbers everywhere.

Gotta' run, I am sure my much more learned colleagues will add a lot to this discussion! - Don
 

amirm

Banned
Apr 2, 2010
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Seattle, WA
- What in the measurement data determines if you need to absorb or diffuse? Specific peaks or nulls?
First, really good thread and questions :). I will answer questions as I have time, starting with this one.

One of the fundamental concepts in acoustics is the notion of transitional frequencies. For most residential rooms this is between 200 and 300 Hz. The sound behavior is different below and above this limit. Below the number of reflections in the room are small. This means that they add and subtract from each other in an orderly manner. Orderly is a bad thing in this context because it means it is easy for the direct sound to combine or subtract from its reflections. This will result in large frequency response variations which as you say are commonly called peaks and nulls. The solutions for this "modal region" are electronic equalization, use and optimization of one or more subwoofers, listening room and speaker positioning (and room dimensioning) and different types of low frequency absorbers.

Low frequency absorbers come in two forms: velocity types and pressure. Typical velocity type is the common and cheap fiberglass absorbers. Since they work best when the velocity of the sound is maximum, they are not effective when placed on the wall where the velocity is zero (nothing can move there). So to make them work at all, they need to be quite thick or placed away from the walls as to capture peak velocity of the wave frequencies of interest.

The other type of absorber, is the tuned variety that works at maximum pressure which is nice as that is our wall locations. The drawback is that these work in narrow bands of frequency so you need to get the right device unlike the velocity type that works at broad range of frequencies. Further, these are quite expensive.

Notice that I have not talked about diffusers. Diffusers can be used for low frequencies but it would have to be extremely deep. We are talking about 2-4 feet deep! So as a general rule, they are not used to deal with modal region.

Above, transition frequencies, the number of reflections from a source becomes very large. So large that they mix so well that we no longer see the massive peaks and valleys we see in modal region. Almost all frequency response variations are a function of the speaker here. In this region, sound works a bit like light or billiard balls. It will be directional and will bounce at surfaces and reflect back with the same angle. The point of absorbing "first reflections" comes from this notion. It is here also where you have a choice of using diffusers instead of absorbers.

This area of acoustics can get extremely complicated because the wavelength of sound gets smaller than the distance between our ears. Once that happens, all kinds of new factors come into play. For example the first reflection off the right wall will create a completely different signal in the right ear vs left. Your face creates an acoustic shadow that blocks high frequencies from getting to your left ear. This immediately invalidates single microphone measurements since it cannot possibly capture what the brain is "hearing" with two different signals. See this article I wrote for a deep dive in this area: http://www.madronadigital.com/Library/RoomReflections.html

Many cookbook recommendations of what to do above transition area comes before the age of modern acoustics as reflected in my article. Common recommendations like absorbing the first reflections from right and left turns out to not be what we prefer. Absorbing floor reflections however, is a good thing as is the front wall as those don't have positive contributions.

The above also invalidates the common technique of using ETC or Impulse Response measurements that identify the reflection points and decay. These are all old-school techniques that can cause one to become slave to measurements that don't correlate with what we hear/prefer.
 

amirm

Banned
Apr 2, 2010
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Seattle, WA
What data do you look at to determine if it's the floor, ceiling, or sidewalls that are causing problems?
I mostly answered this in my last reply. Old school acoustics uses Impulse response where reflections show up as higher amplitude spikes. Using the timing of them in the Impulse response and how far folks think they should be attenuated, absorbers are added until the graph is happy :).

But as I mentioned, modern school of acoustics uses psychoacoustics to solve this problem and beautifully so since it does not require any measurements! Floor reflections are bad. But what makes them bad is > 500 Hz. So about 1 inch of absorption in the form of carpet+felt/jute padding and optionally a spacer underneath does the job. Otherwise, proper absorption would call for 4 inches which is not practical in a floor covering.

Front wall has no benefit so it can be absorbed or diffused. Side walls reflections are always always beneficial. If the speaker has really poor off-axis response, it may pay to absorb those reflections although research shows that we like even bad reflections over no reflections!

Once you have done these, you need to pay attention to however you have gone in treating your room. If you keep putting acoustic products in the room you wind up creating a "dead space." A dead simply measurement, pun intended :), is the reverberation time in the room. This should be between 0.3 to .5 seconds. This is a balance between pleasing effect of reflections vs speech intelligibility. Rooms below 0.3 will sound uncomfortably dry.

So measure the room. If you are above 0.6 you have room to add more absorbers for both modal region and for above.
 

microstrip

VIP/Donor
May 30, 2010
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Amir,

IMHO, your great article deserves more that just a small link inside a post. The net audio culture is filled with generous but authoritarian recommendations of the studio pro's, audiophiles should should read this alternative and modern view and take some time to meditate about it.

I suggest that a post with the article link becomes a sticky in the Room Acoustics thread.
http://www.madronadigital.com/Librar...flections.html
 

jkeny

Industry Expert, Member Sponsor
Feb 9, 2012
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Great article, Amir, thank you - brings an intelligent analysis to the room acoustics.

I would also like to point out some recent psychoacoustic research highlighting the role of the temporal coherence in audio scene stability & segregation
"Segregation of complex acoustic scenes based on temporal coherence" http://elife.elifesciences.org/content/2/e00699

"In contrast to the complex acoustic environments we encounter everyday, most studies of auditory segregation have used relatively simple signals. Here, we synthesized a new stimulus to examine the detection of coherent patterns (‘figures’) from overlapping ‘background’ signals. In a series of experiments, we demonstrate that human listeners are remarkably sensitive to the emergence of such figures and can tolerate a variety of spectral and temporal perturbations. This robust behavior is consistent with the existence of automatic auditory segregation mechanisms that are highly sensitive to correlations across frequency and time."

"Although we do it effortlessly, the separation of such mixtures of sounds into perceptually distinct sound sources is a highly complex task. In spite of being a topic of intense investigation for several decades, the neural bases of auditory object formation and segregation still remain to be fully explained"

"The most commonly used signal for probing auditory perceptual organization is a sequence of two pure tones alternating in time that, under certain conditions, can ‘stream’ or segregate into two sources"

All of which is relevant to how coherent & realistic our audio systems sound to use (intelligibility) & how easily we can follow musical strands within the audio presentation.


Also another recent AES paper titled "Sensitivity of Human Hearing to Changes in Phase Spectrum" http://www.aes.org/e-lib/browse.cfm?elib=17068
"Conventional wisdom incorrectly assumes that changes to the phase spectrum of an audio signal are not perceptually relevant. The results of formal listening tests with synthetic harmonic complex signals showed that human beings are not “phase deaf.” The perceived difference resulting from randomization of the phase spectrum can be larger than those from randomizing the magnitude spectrum. Although the mechanism for phase perception is somewhat local in frequency, there are some influences on the perception of neighboring frequencies. The phase of a component at a certain frequency affects the perception of frequencies about one octave above and below. Signals for which the phase between the harmonics is aligned can be described as having a strong low pitch and “buzzy” quality, whereas random-phase signals are perceived to be colored, thinner, and absent of the buzzy quality."

The point being, as your article also highlighted, that a better understanding & modelling of our hearing is crucial to a sensible analysis of how to design & get the best from our audio systems. It's also heartening to see the test signals being used for this research becoming more sophisticated & realistic (hence the new results, probably?)
 

Nyal Mellor

Industry Expert
Jul 14, 2010
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SF Bay Area, CA, USA
Indeed a great article Amir, I read that one in WSR when it was published a while back.

Also people might want to read this one, which is a bit more focused on uses and abuses of the energy time curve (ETC) - http://www.hifizine.com/2011/12/listening-room-reflections-and-the-energy-time-curve/

The frequency response graphs are very interesting in general because modern measurement techniques mean that we add the real (magnitude) + imaginary (phase) components together through a period of time (IR window length). However I am really not convinced the human hearing system works in this way (I'm not sure the ear/brain factors in the phase component, which is the part that causes all the messy comb filtering on the frequency response when the IR is integrated from start to end of IR window), especially above the lowest frequencies.
 

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