# Help with ETC interpretion



## fotto (Jan 17, 2010)

Hi,
I am using ETC to help me locate and treat some reflections in my room. Hoping someone can help me understand how to interpret the measurement so I can determine how many feet the reflecting surface is from the SPL meter, in order to determine treatment location.

Following is a pic of my front left speaker's ETC response. I used the "use loopback as timing method" for this measurement, with both "sub sample timing adjustment" and "decimate IR" unchecked as I was told this gives you the actual TOF of the reflection. Problem is, I just don't know how to read this plot. 

My hardware consists of UCA202 and RS SPL meter. I have the meter going to Rin on the 202, with Rout feeding my AVR. I added an RCA cable for loopback from Lout to Lin on the 202. Is that correct? Couldn't think of any other way the loopback could/should be connected.

Is T1 the loopback signal? If so, that doesn't make sense as the initial peak (T2) indicates arrival of the direct signal at 25ms, which corresponds to around 28ft. That can't be correct as I'm measuring around 12'-13' from the speaker.










Thanks in advance for any clarification offered.
Floyd


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## EarlK (Jan 1, 2010)

SAC is your man to guide you through this .

> ( having said that ) Can we assume that you checked off the box called ; "Use Loopback as Timing Reference" ( under the "Analysis" tab, within the "Preferences" window ?

<> :sn:


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## fotto (Jan 17, 2010)

EarlK said:


> SAC is your man to guide you through this .
> 
> > ( having said that ) Can we assume that you checked off the box called ; "Use Loopback as Timing Reference" ( under the "Analysis" tab, within the "Preferences" window ?
> 
> <> :sn:


Yes, that box was checked.


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## SAC (Dec 3, 2009)

EarlK said:


> Can we assume that you checked off the box called ; "Use Loopback as Timing Reference" ( under the "Analysis" tab, within the "Preferences" window ?





fotto said:


> Yes, that box was checked.


It also appears that you have checked 'Set T=0 at IR peak' which defeats the hardware loopback compensation. Uncheck that and only check 'Use Loopback as Timing Reference".

One might look at the response and suspect something might be amiss in that its 'rare' that a test signal is generated and received by the measurement mic such that the total time of travel occurs in zero time.

Just looking at the measurement, several things may be happening (especially if Set T=0 to IR Peak is not checked.

T1 may be an anomaly of the measurement rig - as the first, second (and a cluster of several more) significant reflection(s) is(are) not typically of higher gain than the direct signal!

If this is the case, then the actual T1 is likely your T2.

And in this scenario T2 is actually the direct arrival. 
But, again as a sanity check, is your mic (& listening position) really placed ~28 feet from the source speaker ? (~25ms x 1.13ft/ms = 28.25 ft = 28 ft)

If T2 is the actual direct arrival signal, then you will likely want to re-window your display to focus on the actual display, and adjust the x-axis time window to the time corresponding to approximately 2x the room length. While this is technically: (length/1.13 ft/ms), the easy way to guestimate this is to set the x-axis time to slightly longer than twice the length. Example: If the room is 20 feet long, the time window can start at ~40ms. Thus one will set the x-axis time window in the upper right hand corner "Limits" window to approximately: Left: .001s -> Right: .040s


_
But first we have have a few things to determine before going further. _

Also, please post the .mdat file so that we can massage the actual measurement results.


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## fotto (Jan 17, 2010)

SAC, attached is the new .mdat file. I just made two new runs and verified that the loopback settings were correct in REW.
My speakers (both L and R) are no more than 12'6" from the SPL meter, both sitting just inside the right and left edges of my AT screen. Following are a couple of pics for reference. Interestingly enough, both sweeps exhibit a rather large spike in ETC approx 2.4ms after the original signal (if indeed that is original signal). I have been under the impression that spike was due to reflections off of my wood columns, which are close to 50% the distance from speaker to listening position. 

Today, I took a bunch of measurements with what I can only interpret as the "blocking method" using a roll of linacoustic (about 50% leftover). Interestingly enough, I couldn't find any location while placing that roll (48" high) in any lateral positions to attenuate the largest spike. I then took a folded up piece of R-19 and measured while holding it to sides, front etc. right around the SPL meter. I was able to get a nice attenuation of that spike only when holding it in front/higher elevation of the meter. I am coming to the conclusion that this common spike between L and R speakers may very well be coming from the ceiling (which is untreated).

Anyway, I was hoping to be able to understand how I can interpret the ETC and use the string method to identify points that this particular reflection (and subsequently others) incident boundary surface is. Just not sure about how to interpret the TOF based on loopback and translate that appropriately.



















Thanks,
Floyd


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## jtalden (Mar 12, 2009)

T1 is indeed the loopback timing reference.

If the test signal is routed through a digital AVR however, or there are any other digital processors in the signal path to the speakers, there is additional processing delay introduced. Most AVRs have a "direct" mode (analog through) that allows the signal to remain in the analog mode. Use that mode to eliminate the extra delay. 

The multichannel input/output path often stays in the analog realm also I think, so that may be a good option.

If you don't have an "analog through" setting on the AVR you could connect the loopback connection from the Left output of the AVR back to the UCA 202 left input and that will add the same extra delay to the loopback signal also. The numbers should then work out *IF* both the right an left speaker distances in the AVR are the same.

[I would expect that most of the extra delay is because of speaker distance settings in the AVR that adds delay to match the arrival times of the various speakers. These times are calculated in a relative manner, not on an absolute basis so if you set all speakers to the same distance there will probably be no distance delays added to any of the channels. The digital conversion delay will still exist however if you don't have the analog through mode .]


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## fotto (Jan 17, 2010)

John, I have an Onkyo TX-NR708 AVR, with FR/FL pre-outs feeding a Rotel 2 channel amp then into my FR/FL speakers. I only have Audyssey enabled (no dynamic EQ or volume) and have the AVR in 2.1 stereo listening mode when taking measurements with the sub physically turned off. The UCA202 connects into the AVR via front panel AUX input, with connection to L or R input depending on speaker being measured. Indeed, I agree there is signal processing delay being introduced which is indicated in the info panel as "System Delay": 29ms.

I guess I thought this would be a bit more intuitive than what you're proposing, in connecting the loopback path out of the AVR? If the system delay is measurable as it appears to be, seems to me it's just a matter of interpreting the resulting ETC timing correctly.

So if T1 is the test signal pulse, then the measured delay through my signal chain is approx 29ms. Direct speaker signal is arriving at T2 to the mic, and reflections are arriving at T2 plus whatever the diff is between T2 and T3.

If that's correct, let's continue to analyze based on the following. Let's assume I have 12 feet from mic to speaker. If my T2-T3 delta is 2.54ms, or approx 2'6", then total delay from that reflection would be 14'6" (12' for direct + 2'6" for the additional distance traveled by the reflected signal) correct?
So, I should be looking for a reflecting surface around 7'6" from the microphone (half the travel time of that signal).

AND if all that's correct, I'm not sure what value the actual TOF is when measuring with loop back (for the analysis I am doing). The measured delay (T2-T3) on sweeps I took NOT using loop back are the same 2.54ms as the loop back method. The only diff I can see is that for non loop back, direct/peak energy spike is referenced at "0" time on X-axis, while in loop back "0" time corresponds to the test signal, while shifting everything else (initial direct energy and reflections) out on the axis.

If one knows their speaker distance to mic, measure the difference from initial signal to reflected signals and do the math, what does it matter if you use loopback or non loop back? 

Maybe I'm just being dense, but I don't get it.


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## EarlK (Jan 1, 2010)

Here's a pic I created using your mdat file .

As you can see , I think this ETC spike is created by a reflective surface very near your speakers .

> I think what I would do is cut a piece of string 14.6" long , firmly attach one end to the speaker & the other end firmly to the measuring position . 
> Then pull out the slackness of the string towards a point on the wall or roof ( until all the slack gets taken up & the angles look good ) . This is the area I would acoustically treat .

> It'll be interesting to hear what SAC says is the best approach/interpretation . In the meantime I need to go back to my copy of  *"Sound System Engineering" * to get some pointers from the grand-dad of ETC .

<> :sn:


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## jtalden (Mar 12, 2009)

Floyd,
I was only addressing why the loopback to impulse arrival time (T1 to T2) did not correspond to the actual distance measured between the speaker and the LP. If you eliminate the AVR delay in one of the ways suggested, it should then agree with your measurement.

I am no expert on the overall general question, but would expect that EarlK's suggestion would work just fine. That is, I don't see why one needs to measure direct distance using the time from T1 to T2 when you can just measure it easily. That method probably has more utility in large venue where the distance is not as easily determined. I am just guessing about that though. 

In summary, I also would expect the 14' 6" string that EarlK suggested to work well enough to locate the reflection point.


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## fotto (Jan 17, 2010)

Thank you both for the replies. John, I most likely misinterpreted your post, sorry.

So, will cut me some string and see if I can track it down


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## SAC (Dec 3, 2009)

fotto said:


> Yes, that box was checked.





fotto said:


> John, I have an Onkyo TX-NR708 AVR, with FR/FL pre-outs feeding a Rotel 2 channel amp then into my FR/FL speakers. I only have Audyssey enabled (no dynamic EQ or volume) and have the AVR in 2.1 stereo listening mode when taking measurements with the sub physically turned off. The UCA202 connects into the AVR via front panel AUX input, with connection to L or R input depending on speaker being measured. Indeed, I agree there is signal processing delay being introduced which is indicated in the info panel as "System Delay": 29ms.
> 
> I guess I thought this would be a bit more intuitive than what you're proposing, in connecting the loopback path out of the AVR? If the system delay is measurable as it appears to be, seems to me it's just a matter of interpreting the resulting ETC timing correctly.
> 
> ...


(All comments are made assuming the examined portions of the measurement to be accurate...)

A system delay of 29ms??????

What? I am totally confused by what is happening here. (...actually, I'm not, but the processing renders large portions of the information questionable at best. I am just confused as to why it is being included!)

Why are we incorporating ANY system processing?

The inclusion of such totally defeats the entire purpose of using loopback if we are simply going to introduce some additional source of wacked system introduced delay. And I would love to have someone explain to me why the front speakers are delayed by 29 ms anyway?

_Turn Audyssey OFF._
All we want at this point is to see the direct unprocessed signals!

If you want to introduce delay in order to align the arrival times of the various drivers, you do this AFTER we evaluate the ETC of the drivers and precisely establish the required delay!

In fact, the most precise way to determine the amount of delay needed for each loudspeaker is through the use of each loudspeaker's ETC.

Thus what we have at this point is a totally invalid time of flight measurement.

All we can do is look a the various energy arrivals independently of time in relation to one another.

And no, the time delta between energy arrivals does NOT correspond to the distance between the various reflection points, as one is direct and the others are not. It merely tells us the difference in signal path lengths. 

I am a bit confused also by the reference to using a role of stuff to block reflections... You use a relatively small piece of absorption (imagine a throw pillow) that is iteratively placed anticipating the vector path of the indirect signals at different points slightly in front of the mic and moved incrementally in a circular motion around the mic in a plane perpendicular to the front wall. You position the absorption at one stationary point per sweep. This is repeated incrementally until you identify the various energy paths and identify the directions of each energy arrival. As you identify each 'direction', you then repeat the process for each, 'walking' the reflection back to the incident surface, further refining the vector path and ultimately identifying the boundary incident point. This process may be repeated in order to identify and determine the vector paths of all signal arrivals from all orientations including the later arriving signals from all orientations relative to the mic.

With the information we have at this point, there is no way to use the string method, as we lack accurate time of flight information.

Looking briefly at the actual ETC, there are quite few significant reflections, including a very curious direct signal!

First, the direct signal:
Be they errors due to diffraction as a result of how the speakers are mounted (or manufactured), or whether the drivers are misaligned, or some other issue yet to be determined cause, you lack a distinct direct signal arrival featuring significant alignment and/or diffraction errors as evidenced by the tight grouping of multiple signal arrivals (at least 4!). ...Talk about issues with group delay! This needs to be addressed.



Secondly, looking at the overall response, you have LOTS of early arriving high gain (less than 10 dB down relative to the direct) energy arrivals within the first 10ms after the direct arrival(s). And far more if the threshold is set to -15 or to -20 dB of the direct arrival within the first ~10 ms.



These need to be resolved into their sources.

But in order to use the string method one must establish accurate acoustical time of flight arrival times for this loudspeaker, and all the rest in the system in order to accurately set system delay times, you need to remove the system processing! Any system processing is done AFTER we have accurately ascertained what is happening prior to system processing. In other words, we need to know what is going on before the system or user starts playing around with the signals.


(If after all that needs to be examined and addressed is completed, and you then want to look at the effect of processing, then you can look at the data as it is currently displayed. But it makes no sense to attempt to determine what the baseline behavior is with processing concurrently attempting to manipulate it !)


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## fotto (Jan 17, 2010)

LOL...I'm such a dolt sometimes. I didn't even think to disable Audyssey as I was previously doing some measurements to determine response after it was run, and quite frankly it's been a while. 

For subsequent measurements, would you include my Rotel amp in the chain, or drive my fronts directly from the AVR?

Interestingly enough, I did end up doing just what you suggest regarding the "blocking" method. I stapled a doubled up piece of R-19 about 12" square to a stick that I could position around what I thought could be the path of the reflection. I was suspect that possibly the reflection was coming off the wood bullnose trim on my stage, so the linacoutic roll was placed there just to rule it out (which it did as there was no effect).

I also ended up doing the string trick based on the previous measurements. After measuring my actual speaker distance to mic at 12' and adding is the delay from T2-T3, total length was around 14'8". Guess where that touched something....right on the ceiling between speaker and mic. I marked it and then held the insulation with the stick while I remeasured. Bingo! That single piece of insulation brought that reflection down to around -10dB! Pic below for measurement with time marked for ref:











Now, as to that nasty direct signal, I am using American Acoustic tower speakers circa 1988 or there abouts Probably safe to say their construction and technology used at the time is a major contributor there. I've spent so much on the theater build, I decided to use these until I could afford an upgrade.

Guess what's on Floyd's list to Santa this year???? Matter of fact, I am planning on getting new bookshelf's in prior to the time I have off in Dec. I have Axiom QS8's for surround and VP150 for center currently. Would take any suggestions on what to consider for front mains if anyone has an opinion on that.

So, will probably put further efforts on hold until I get my upgrade. I've been a little suspicious of the current speakers since day one.


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## aackthpt (Jan 24, 2011)

Fantastic Floyd, thanks for this thread. I hadn't seen ETC used in this sort of super-zoom mode to look at the driver time alignment. New job, new tools! w00t!


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## fotto (Jan 17, 2010)

aackthpt said:


> Fantastic Floyd, thanks for this thread. I hadn't seen ETC used in this sort of super-zoom mode to look at the driver time alignment. New job, new tools! w00t!


Hey...long time no talk SAC's the one who massaged the ETC into that super zoom view so can't take any credit on that.


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## aackthpt (Jan 24, 2011)

fotto said:


> Hey...long time no talk SAC's the one who massaged the ETC into that super zoom view so can't take any credit on that.


Yeah, I know, but it's your thread and speaker, would never have seen it otherwise!


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## SAC (Dec 3, 2009)

fotto said:


> .......
> 
> AND if all that's correct, I'm not sure what value the actual TOF is when measuring with loop back (for the analysis I am doing). The measured delay (T2-T3) on sweeps I took NOT using loop back are the same 2.54ms as the loop back method. The only diff I can see is that for non loop back, direct/peak energy spike is referenced at "0" time on X-axis, while in loop back "0" time corresponds to the test signal, while shifting everything else (initial direct energy and reflections) out on the axis.
> 
> ...


Why are you worried about what the loopback value is? What color are the curtains in your kitchen? (There are ways to determine this, but its not germane to the issue at hand.)

If this is enabled and you have no additional processing enabled in the AVR, the time that the signal takes to travel from source to mic should be the Time of Flight. Period.
That's all you need. From this you can determine the time of flight converted to distance by multiplying the TOF by 1.13foot/ms. The time differentials between direct and indirect signals are of no practical value for this application. ...Maybe when you start setting delay lines...

What is so complex about this?

But let's take this two steps further. You imagine that you "know" the acoustic center of the speaker. Really? This is NOT always true! Is your speaker a signal aligned full range single driver speaker? Is it a signal aligned coax driver such as in a Bag End coaxial? Probably not. Taken top another practical extreme often encountered in pro-sound, what if the source is a multi-element array?

The acoustic origin of the speaker - the point at which the sound 'appears' to emanate from the speaker is NOT the tweeter (as so many want to assume) or necessarily the baffle. Confuse this a bit more by including a horn, where the acoustical origin is at some amorphous spot somewhere inside the throat, determinable only by measurement...And these are 'well-behaved' examples! We haven't even mentioned astigmatism in CD horns where the acoustic center moves with frequency, or the fact that many drivers literally offset at different frequency ranges and move forward to a different position only to move back to the normal excursion range in other frequency ranges (as Don Keele demonstrated at the Atlanta '91 Loudspeaker Design Seminar with the woofer removed from his reference B&Ws!). 

If only we could simply 'define' the acoustic center of a speaker simply by looking at it!

The point is, if you already know the distance (by intuition), why bother? Because you are going to use the MEASUREMENTS to determine the actual vector paths of travel and points of incidence. And these times, and hence derived distances of travel, are referenced to the MEASURED values - the actual acoustic center - not to your intuited acoustic center. You wanted accurate...right?

And the use of the string is a basic conceptual device. It is not the recommended procedure to use as a rule. It provides a concrete demonstration and visual reinforcement of the concept!

Once this is accomplished, I would recommend that you use the blocking technique...unless you simply enjoy monopolizing other folks time and taking several days to resolve the individual reflections paths and points of incidence. In this way the process, once understood based on a good understanding of the concept, can be relatively quickly performed.

And I will go still further...if you are going to do this with any regularity, I find it difficult to imagine not using the Polar ETC and a laser pointer and letting a PolarETC program calculate the 3space coordinates and simply dialing these coordinates into a transit with the mic replaced by a laser pointer and letting the laser pointer point to the incident spot on the wall as fast as you can move the cursor to the reflection and adjust the transit coordinates!

This isn't intended to be a character building exercise. Its a necessary bit of busy work used to identify points of incidence in order to then facilitate treatment options in accordance to one's desired target acoustical response model. In other words, the first few times someone does this its exciting and fascinating. Thereafter its work...and some would rather focus on moving the goal forward and not simply standing in admiration of the hammer they use to complete various tasks - if you catch the analogy.

Thus, to regress a bit, the focus is NOT on the string. Its not on guessing the acoustical center of the source. (And what if the source you were measuring was flown 15 feet over your head? Are you into repeatedly climbing up there on a ladder or scissors jack with your string? And then there are the other points of incidence...ceilings come to mind...what fun.....) the focus is improperly configuring the test platform so that these issues become trivial and one less issue about which to worry - allowing you to focus on the larger issues at hand.

So, I have taken you a bit further than where your current focus lies, but i hope it puts this process into a bit better perspective. We use the loopback correction as we are relying on the _measurement _to provide accurate values. Not because we want to add a bunch of approximate estimations into what is otherwise an extremely accurate process. And such errors, while perhaps not critical in a very short time frame and corresponding distance, become increasingly critical in longer times and distances, where the identification of incident points could easily be off by a few feet - significant enough to effect the treatment and response. And also significant enough to require what could be a pretty extensive period of re-measurement, re-placement of treatment and re-verification. 

For some this time is money. For all its an avoidable pain in the backside.

The point here is not to see how many goofy variables we can introduce and explain (at least not from my perspective...and if it becomes this, there are quite a few neater 'anomalies' to explore...). The purpose is to properly configure the platform, make reliable measurements that avoid the amazing serendipitous journey of exploration, and to cut to the chase and determine the necessary information, namely the effective point of boundary incidence and to move on toe determining what treatment is most effective in obtaining the overall response that is desired.

So try not to get lost in the weeds and try to remain focused on the larger goal. Once that is understood there will be plenty of time to explore and to better appreciate the various associated minutia and variables that one can 'construct' that can be explored.


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## fotto (Jan 17, 2010)

SAC said:


> Why are you worried about what the loopback value is?
> 
> If this is enabled and you have no additional processing enabled in the AVR, the time that the signal takes to travel from source to mic should be the Time of Flight. Period.
> That's all you need. From this you can determine the time of flight converted to distance by multiplying the TOF by 1.13foot/ms. The time differentials between direct and indirect signals are of no practical value for this application. ...Maybe when you start setting delay lines...
> ...


I'm worried about the loopback value/system delay because I have close to 30ms showing in my measurement, which I have not resolved yet. Assuming I will get it resolved, I can proceed with more meaningful efforts based on accurate data. 

So, if you are recommending blocking method as a more productive/useful tool vs. string method, why are we even discussing actual TOF? You simply block the energy to it's reflective source right? What's distance have to do with it when one can't accurately define acoustic center of the speaker (as one end of the length)?

I believe I understand your argument on why it's important to know actual TOF, but you're suggesting resolving methods which don't even need that info, unless I'm missing how you use TOF/total distance of a reflection in CONJUNCTION with blocking.


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## aackthpt (Jan 24, 2011)

fotto said:


> Guess what's on Floyd's list to Santa this year???? Matter of fact, I am planning on getting new bookshelf's in prior to the time I have off in Dec. I have Axiom QS8's for surround and VP150 for center currently. Would take any suggestions on what to consider for front mains if anyone has an opinion on that.


I'm sure you could get plenty of suggestions if you ask in the proper area of the forum. I'd guess that many responses will be motivated by brand loyalty though. One of the thoughts I've had is that if I ever wind up looking for a new array again, I may well look at self-powered studio monitors, ideally using a processor with balanced outputs. I've honestly no idea where one might go to get good quality speaker reviews as I've never seen one that included blind listening against the reference in any fashion. If I don't go with the studio monitor idea, I'd probably DIY something, which I might actually do anyway. Even if I don't manage to educate myself on all of the ins and outs of speakers, there are plenty of published DIY designs, or close enough to DIY (e.g. the stuff from Fitzmaurice).


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## SAC (Dec 3, 2009)

fotto said:


> I'm worried about the loopback value/system delay because I have close to 30ms showing in my measurement, which I have not resolved yet. Assuming I will get it resolved, I can proceed with more meaningful efforts based on accurate data.
> 
> So, if you are recommending blocking method as a more productive/useful tool vs. string method, why are we even discussing actual TOF? You simply block the energy to it's reflective source right? What's distance have to do with it when one can't accurately define acoustic center of the speaker (as one end of the length)?
> 
> I believe I understand your argument on why it's important to know actual TOF, but you're suggesting resolving methods which don't even need that info, unless I'm missing how you use TOF/total distance of a reflection in CONJUNCTION with blocking.


The inordinate delay is due to a mistake in having processing enabled in your AVR! Disable it!

I have repeatedly discussed several methods, beginning with the simplest method that illustrates and reinforces the concept while explaining critical concepts that are fundamental to the response and what are properly displayed by the response that corresponds to reality, to a more abstract concept (blocking), to finally one that is still more expeditious and more abstract (PolarETCs or PET).


And "why are talking about it?....."
For those who don't even know what an ETC response is or to what the various display variables correlate, and who repeatedly ask questions wanting further explanation, and for whom starting off saying that the ETC illustrates the arrival time of various energy wavelets, I sure am not going to begin by saying "look folks, the ETC displays the actual time of flight with great accuracy from the source's acoustical origin to the measurement mic" and then say - "oh, just ignore the time, it corresponds to nothing "real'" in an isolated instance just because I can show you a 'trick' that works for a limited application of the tool in a limited context.

And I am not the one who, when we suggest just focusing on a particular case for the time being, goes off on tangents wanting to delve into other related issues, and then when we attempt to present a more comprehensive introduction, then complains that we have not presented the Cliff's Notes version of the tool.

The ETC can be used for a great many purposes, and all are based upon its ability to display energy arrival times with great precision. Simply because I can specify one instance where we can get 'around' that for a LIMITED application, I am NOT about to reduce the full capability of the response to what in effect is a 'slang' interpretation that happens to work in a limited instance. And as few here have any idea as to what I refer when I mention the Analytic from which all of this is derived and but a subset, there is still more to be understood that extends far beyond this rather elementary feature. What we are doing here is but a very small portion of a much larger world in acoustics.

Just like in math, you start with algebra, proceed to calculus and differential math and, AFTER one has a solid grounding in the various applications and what they represent, THEN they introduce you to Laplace transforms!

And with all due respect, 'we' are still hung up on, and can't seem to get beyond, the inappropriate inclusion of AVR processing as well as misinterpretations of what the time differential between various direct and indirect signal times represent. I dare say the circumstances warrant a few baby steps in order to more fully understand what is actually going on before we simply leapfrog understanding that is necessary in the name of expediency and pretend to be able to intuit acoustic origins, and other nuances of the ETC response- especially as the tool can be used to evaluate very small time differentials as well as large differentials. And in a large number of the cases it is _critical_ that one be able to ascertain the precise time differential to the maximum precision and accuracy. The additional methods that I have mentioned are not applicable. 

And the fact that I can provide an exceptional 'shorthand' method that is handy when performing ONE particular task is not a substitute for understanding what is represented by the ETC response and how it is properly used in the measuring process.

So, I can describe several methodologies (a few of which are 'shorthand' methods that utilize the response), and we can proceed in order so that the concepts become clear, and so that one may also progress to use one of the alternative methods that may prove to be easier. But such techniques are of use AFTER one has a good grasp of the fundamentals, not instead of them. And it would also be nice is we focused on the issue at hand and did not persist in chasing off topic ghosts regardless of how pressing a topic they might seem to be. _After_ one obtains a working understanding of the tool, one can then chase all the ghosts they desire as well as more interesting real behaviors at their leisure.


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## aackthpt (Jan 24, 2011)

SAC said:


> The ETC can be used for a great many purposes, and all are based upon its ability to display energy arrival times with great precision. Simply because I can specify one instance where we can get 'around' that for a LIMITED application, I am NOT about to reduce the full capability of the response to what in effect is a 'slang' interpretation that happens to work in a limited instance. And as few here have any idea as to what I refer when I mention the Analytic from which all of this is derived and but a subset, there is still more to be understood that extends far beyond this rather elementary feature. What we are doing here is but a very small portion of a much larger world in acoustics.


Please start a thread (on any of the fora) to discuss the analytic, indeed possibly even the rest of the "much larger world in acoustics" in a new thread. I think I, and a fair number of others, would perhaps like to read further. Alternately or in addition, I think everyone would appreciate further details on use of ETC (in the appropriate threads) to achieve various acoustical results. I think the process of capturing ETC has been pretty well discussed and worked out by now, so let's take a few more baby steps.


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## fotto (Jan 17, 2010)

Little late updating this.....I did get rid of the long system delay by putting my AVR in "Direct" mode instead of "Stereo" and proceeded to measure and track down some reflections. 

As an experiment, I decided to track down the largest reflection in my ETC measurement of my front right speaker, which was measured at 2.22ms or 2.5 feet after the initial direct speaker signal. The direct speaker signal is normalized at T= 0 on the x axis, with the spike measured at the 2.22 ms later as shown following (note, this was a previous effort NOT using loopback with T=0 at impulse peak):










Next I needed to do some simple calculations in order to use the "string method" to find out specific areas to check where the actual reflection surface is. First I measured the distance from my Radio Shack SPL meter to the actual acoustic center of the speaker being measured. That distance was 12' (as best as I could measure). The reflection being measured takes an additional 2.5' to arrive, which means that the total distance of the reflected signal is 12' + 2.5' or 14' 6".

The concept of using the string method is that you take a length of string which is equal to the distance traveled of the signal (in this case 14'6"), tie off one end to your measurement mic, and the other end to the acoustic center of your speaker. If you have helpers available, they can hold either end. In my case I didn't so I had to gin up tie offs. You then take up (I just slid a finger along the length) the string slack between the mic and speaker extending it in all axis's and note what boundary surfaces it touches.
In my case, it touched the ceiling about 1/2 way between the mic and speaker. Following are a few pic for clarification.

Mic at measurement position with one end of string tied off:










Other end of string tied off at speaker:










Location on ceiling where string touched:










Visual of overall string path with my "treatment stick" holding it up:










I next took another measurement while holding that insulation batt to the ceiling. Following is that measurement with time reference noted. Notice that the spike has been attenuated. Original measurement shown directly following again for ease of comparison.



















I have the same spike occurring on the left speaker at similar time, so appears that a ceiling panel is in my future.


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## EarlK (Jan 1, 2010)

Nice write-up ! Thanks ! :bigsmile:

:T


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