room treatments started ?'s

[seti]

My room was live as the grand conyon but after putting up some curtains it is much better. I have 11ft ceilings so I put 9ft curtains along the wall behind the speakers and one more set on a side wall both curtains cover windows. The main wall is mostly covered. I also have a 9x12 rug that is going down on my wood floors which should also help.

These were the first changes I could easily make but now I would like to take a more precise aproach to the room. I have a mac laptop is there any software that I could use which utilizes the internal mic? I need something relatively easy to understand.

Thanks

[mas]

The most inexpensive time domain software is ETF/RPlusD ($150-$300). http://www.acoustisoft.com/index.html

You will have to run it under Windows.

Room EQ Wizard will not address room anomalies as EQ is only effective in treating minimum phase direct sound - thus eliminating its use for room correction as the room introduces non-minimum phase reflected signals that are summed with the direct signal (a process called superposition) and which cannot be effectively EQ'd.

No, you will not be able to use the integrated microphone. Its response characteristics are not suitable for calibrated measurements.

[damonrpayne]

Seti, several of us are embarking on the room treatment journey right now. I am placing an order for some components to work with RPlusD. It won't be super cheap but probably cheaper than other upgrades you might have done in the past.

[seti]

There never is an easy way to do this hmmm. I wonder if I could find someone locally to come over and run a quick test on the room. I could lay my hands on a windows laptop but do they offer demo software as I would only use it once or twice?

Damonrpayne, I saw Front 242 long ago live in portland and also caught KMFDM the same month. I can't remeber the name of the club sure was cool though.

[damonrpayne]

RPlusD is availble in demo form. I have not yet used it enough to determine what the limitations of "demo" are.

You'll need some additional hardware besides the software. As soon as my stuff comes in I will have an article about this up on KlipschCorner.com. I may be willing to lend out the gear at a later date, I spent a total of $190 on an outboard sound card for my laptop, mixer, mic, mic cable.

[PrestonTom]

There never is an easy way to do this hmmm. I wonder if I could find someone locally to come over and run a quick test on the room. I could lay my hands on a windows laptop but do they offer demo software as I would only use it once or twice?

Unfortunately, it is probably not that simple. You will probably need a sound card that is full duplex, the ones built into notebook computers are typically only half duplex.

It is best to double check on this,

-Tom

[rcarlton]

Snip

Room EQ Wizard will not address room anomalies as EQ is only effective in treating minimum phase direct sound - thus eliminating its use for room correction as the room introduces non-minimum phase reflected signals that are summed with the direct signal (a process called superposition) and which cannot be effectively EQ'd.

Snip

I don't understand this statement. How can the graphs of the room response not be reliable?

[damonrpayne]

It would be pretty sweet if mas wanted to write a KlipschCorner article on why parametric room EQ will never be as good as room treatment.

[]

[WMcD]

I believe he is saying the following.

There can be two types of, let's say, drop outs or low level (sometimes peaks) at a given frequeny in a room.

1) One is where there is a direct sound path from the speaker to your ear (microphone). You can boost or lower the signal at the speaker to equalize. There are not two paths to your ear. Therefore, there is not two signals adding or subracting according to their realtive phase. So I guess he is saying this is a minimal or non phase issue. ;If there are not two signals coming into the microphone or ear, there is no phase issue.

2) Remember when you have two signals at the same frequency overlaping, or super imposed, they "add" according to their relative phase and magnitude at the ear or microphone. If they're in phase (zero degrees different) and equal magnitude they add to a 3 dB hump.

But if they are 180 degrees different in phase and equal magnitude, they add to zero. This is a bit remarkable. The peaks are relatively small when signals of the same frequency add. The valleys are down to zero when they are in opposite phase. Note that we are always talking about signals of the same frequency, here.

One of the two signals is direct from the speaker. The other is reflection off a wall. The relative phase depends on how far the two have traveled and their wavelength. The path length difference.

It is almost impossible to convince anyone of this until they do an experiment themselves. Play a single tone (say 200 Hz) through a single speaker at a comfortable level. Then walk around the room. You will find some spots where it seems the speaker stopped working. A few feet away it sounds strong. And if you change the frequency, the nulls will move to another location.

You can sit at your listening position an listen to a swept tone. You'll find the nulls as some frequencies are exactly where you're sitting.

[Note that this is related to what PWK and others were saying about horn length differences. There are, potentially frequency response errors in the crossover zone. But we don't hear small delays of 2 mS or so, otherwise.]

You might say, thats okay about ragged response at my sweet spot. I'll just use an equalizer to boost those frequecies. But this doesn't work. It doesn't work because you're boosting both the direct sound, and the reflected sound (which actually comes from the speaker in the first place). The null, or node, is the same whether you use a milliwatt or a kilowatt to start. You just can't equalize to solve this problem because you can't un-boost the reflection alone . . . unless you use room treatment on the reflection alone. This is the answer to the question of why room treatment is better than equalizing.

mas is also pointing out something about measurements made at a single point. The graphs can show us the result. It can't tell us exactly whether it arises from a reflection out of phase, or it the speaker itself. Maybe looking at various waterfall graphs we can infer the problem and then solve it with absorber or defractors on the walls. This is a frequency response issue.

It is of course much more complicated in CAUSATION than what I describe because we have multiple walls and multiple reflections. And different delay times.

It is also more complicated in SOLUTION even if we solve the frequency response issue. Note these phase issues ARE related to reflections and time. But frequency response is not enough.

This is because of the Hass (?) effect. We want to be rid of early reflections so the first 10 milliseconds or so of sound from the speaker is not contaminated with reflections at all. OTOH, we do want later (in time) diffuse reflections.

Note the conflict in all the above. It may be why the subject tends to be confusing. It is only with a consideration of all the interactions that we can build a good room. mas is going to give us some technical publications on the subjects.

Best,

Gil

I edited a little bit.

[gaspr]

Thanks for the very informative posts guys. I think that I am really starting to understand the basics of this.

Garth

[mas]

First, let me thank Gil for his great summary!

So...what can you EQ?

We have inadvertently stumbled onto a large and important topic, but I will try to toss a few ideas out...but this topic does deserve a more thorough treatment and it will help to have a greater understanding of a few underlying topics... And hopefully we can address this issue in greater depth soon after we deal with a few other issues that we will be tossing out beginning this weekend... ;-) (Mysterious, aren't I? ;-)

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First, may I suggest the link below. Superposition is a rather abstract concept to grasp...until the light goes on, and then it will seem rather obvious. And I can just imagine you saying : Sure, thats easy for him to say! ;-)

It is often presented graphically with linear waveforms, which while useful, is somewhat hard to translate into the world of sound. For sound, it is often easier to refer to the format normally encountered when using a wave tank spherical waveforms reminiscent of dropping a pebble into a puddle.

For a nice example of the interference patterns that this odel presents and which I think you will find very useful to understanding the effects on polar patterns, may I suggest this site:

http://id.mind.net/~zona/mstm/physics/waves/interference/twoSource/TwoSourceInterference1.html

May I suggest playing with the two sources down near the bottom edge of the 'tank', and switching between the small, medium and large setting to make the interferfence patterns more prominent. Note that the regions of 'dark bands' are nulls - regions where the polar response is absent for the particular frequency viewed. In other words, that frequency's acoustic energy has been cancelled and you would not hear it in those regions.

Now, jumping back into the EQ question this is actually a pretty large topic, but I will try to summarize it here without going into too much depth or explanation. Also, note that this also makes reference to large room acouistics which do not apply to small room acoustical environments such as a home theater or home litening room - for small rooms(unlike large acousitical spaces) please understand that reverberent fields do NOT exist! More on this to come!

Allow me to cite an article from the winter 1989 Syn-Aud-Con newsletter:

What can an Equalizer Equalize?

The question What can an equalizer equalize? needs to be asked. Some claim to equalize the room. Is this possible? We think not.

When an electronic or passive equalizer is installed in between a mixer and a power amplifier we need to know that all it can equalize is the electrical signal being sent to the loudspeaker.

What comes out of the loudspeaker?

What comes out of the loudspeaker is called direct sound level, Ld. Early reflections from the floor, walls, and ceiling are called the early reflected level, Lre, and late-in-time, homogenous mixing sound is called the reverberant sound level, Lr.{Please note, Lr is present only in a Large Acoustical Space not your home theater or listening room.}

When an electronic equalizer is employed it not only alters the Ld at the listener position, but also the sound power level, Lw, of the loudspeaker. {Note: If interested, Sam Berkow has a number of articles referring to the need to reduce a subwoofers crossover point with increases in gain. PM me. ) This in turn affects Lre and Lr but has no effect on Ln {Lamb - Ambient noise floor you may see it expressed symbolically as Ln or Lamb}. The question over the years has been how much can I alter Ld without throwing the baby out with the wash? Experience has shown that the answer is not much. Certainly not enough to drop a specular Lre having a full frequency response.

The Audience Effect

The so-called audience effect came about by people looking at the sound fields with 1/3-octave real time analyzers.

What an RTA sees is the total sound field level, Lt, which is the combined Ld, Lre, and Lr plus any ambient noise present, Ln (Lamb). { Unlike SPL meters and RTAs, time domain analyzers such as TEF are unique in that they display the component waveforms (reflections) and provide detailed information about the individual reflections instead of the simple summed result.}

{You might want to read those last two sentences again! ;-) }

Adjusting a sound system to a uniform Lt may or may not result in a sound you would want to listen to. In many cases the floor reflection can cause an operator to misadjust the direct sound level, Ld. Then when the audience arrives and covers the floor the misadjusted Ld is more clearly perceived and we say the audience affected the system.

Really? Think with me for a minute.

What can the audience do to affect Ld from a sound system? The answer, of course, is absolutely nothing. Therefore, it is clear that the audience can only alter Lre, Lr and Ln. Now, ask yourself the question how can an equalizer adjust Lre, Lr and Ln {independently of Ld}? The answer is that it cannot.

I would hesitate to mention such obvious facts except for the remarkable number of articles appearing to claim the contrary. The only way an equalizer can cause a change in Lre, Lr and Ln is to do damage to Ld in the process.

What Can Affect Lre

Loudspeaker directivity factor, Q, is the classic way to handle unwanted Lre. Using a loudspeakers directivity to stay off of surfaces producing unwanted reflective energy is also one of the most cost effective solutions. The second approach is to use either absorption or diffusion.

Electronic Directivity Control

The increasing use of precision digital delays (i.e., 10 usec per step in contrast to normal digital delays of 1msec per step) to correct mis-synchronized loudspeaker arrays, where the mis-synchronization has resulted in directional lobing of the loudspeaker, demonstrates the importance and validity of directional control. {Additionally, limiting overlap of adjacent sound fields also minimizes the superpositional effects such as comb filtering upon sound fields as well.}

Good Engineering Practices

Today, thanks to advanced analysis in the hands of competent users, good engineering practice has become:

1. Adjust Ld by measuring it alone with a TEF analyzer.

2. Optimize the reduction of Lre, Lr, and Ln levels by means of controlled directivity and measured synchronization of arrays.

3. Fundamentally control Lre, Lr and Ln through traditional use of absorption, diffusion, and noise abatement techniques.

What does all of this mean?

The point I hope I have made is that electronic equalization in the frequency domain cannot correct phenomenon in the time domain outside of the minimum phase period (i.e., a few hundred microseconds). An EQ cannot correct for the summation of two or more out of phase signals. And thus they cannot solve acoustical room problems as so often claimed. Claims to do so should be rightfully regarded as mistaken.

An additional factor to consider:

Also, if I can mention one more issue without going into too much depth here or causing too much additional confusion (OK, I guess I am too late!;-) , comb filtering is the result of multiple sources reproducing the same pass band signal. As these sources are physically separated, the arrival times of the multiple signals are not coincident this difference in time is referred to as a difference in phase. In audio parlance, phase is a fancy name for time.

Thus, what may have been a perfectly flat DC to gamma ray frequency response with one transducer, now exhibits a notched response beginning at a frequency fundamental dependent upon the driver spacing that is the source of the phase (time) difference in the arriving direct signals. This notching is repeated at multiples of this fundamental frequency. And as this notch is actually a null created by the phase being 180 degrees out of phase, the signal is effectively cancelled at those points. And even with the best of equipment, it is difficult to increase the gain of a signal that has been cancelled and is no longer available to be processed!

Let me also mention one more corollary to comb filtering. As comb filtering is but one perspective of the sound in the frequency domain, this same phenomena can also be viewed from the physical realm in the form of the polar response. The polar response corresponds to a lobing of the dispersion of the acoustic energy into a physical space. Where the lobes are present, you hear the sound, where they are absent in the notches/nulls, that energy is absent. What may have been a device featuring a perfectly hemispherical (cardioid) polar pattern corresponding to the perfectly flat frequency response, becomes a frequency dependent pattern of lobes, with the number of lobes being fewer and the Q (the focus think: spotlight = high Q, and broad floodlight = low Q) lower/broader for low frequencies, but increasing in number of lobes and in Q as the frequency increases. {If you have a chance, the next time you go to a concert, if you get there early and you have access to the entire floor, when they are playing a fairly consistent musical selection (especially one with a solo vocalist or a piano), slowly walk from left to right across the floor, from one side of the auditorium to the other and listen. You may be surprised to discover that it seems as if someone is adjusting the EQ the tone controls. As you walk from spot to spot, it will seem like various frequency bands of the music are either exaggerated or missing entirely. What you are experiencing is the entry and exit into and out of the lobes and nulls. EQ cannot fix this. Minimizing the overlap of multiple sources fixes this. But this is what many have tried to fix for years with an EQ. And what they have really done by adjusting the EQ, was to introduce a small amount of phase shift to the signal {*see below}, thus changing the phase relationship between the direct sound Ld and the reflected signals Lre, thus ever so slightly moving the polar lobing and the nulls simply rearranging the furniture (the problem), but NOT fixing the fundamental cause of the problem.

I hope this has helped a bit! It can be a bit confusing at first, but I think you will find it clear after a bit of pondering and experimentation. And after you become aware of it, you will discover just how widespread an issue it is, and how destructive it can be. This (hopefully) will result in your returning to ask OK, so what can I do about it? And that is the subject for another discussion!

*All 2nd-order bandpass or band-reject filters (active or passive) shift phase the same amount. (The bandwidth of this phase shift differs for various 2nd-order responses, but the phase shift is the same.). And when used to create boost/cut

responses, do so with the same phase shift. Different phase responses do exist, but they are a function of boost/cut levels and individual filter bandwidths. That is, there will be less phase shift for 3 dB of boost/cut than 12 dB; and a 1-octave

filter set will have a wider phase response than a 1/3-octave unit (but the number of degrees of phase shift will be the same). Figs. 1 and 2 demonstrate this. In Fig. 1, the phase responses for different levels of boost appear (cut responses are identical but mirror image). This verifies Pennington's rule-of-thumb regarding 10 degrees of phase shift per 3 dB of amplitude change.{T. Pennington, Constant-Q, Studio Sound, vol. 27, pp.82-85 (Oct. 1985}.

Fig. 2 shows the bandwidth variation for this phase shift for wider and narrower bandpass responses.

{Source: Operator Adjustable Equalizers: An Overview; Rane, Inc.}

ConcatEQPhaseShift.CombFilter.3DPolars.pdf

[seti]

Thanks for the responses guys some of this I understand the basics of and the rest is deeper than I want to go or completely understand. I love music and I want it to sound as good as possible in my room without breaking the bank or looking to rediculous. Rather than have to fully understand all this I would rather pay someone to come over and do it for me LOL. I thought perhaps there was an easy way to do this with my laptop with software that would listen to my room and say ok these are your issues deal with it.

I listened to a Raymond Scott compilation cd set last night and it was too damm cool.