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Feature Request: Frequency Dependent Windowing

52K views 115 replies 14 participants last post by  lkwrnflknf4lk  
#1 ·
Dear John:

Many thanks for your continued advancements in performance in REW! If I understand correctly reading the latest beta notes, you have conquered the OSX measurement issues with Java that plagued us before and led me to use ASIO on PC?

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I'd like to make my "annual" request to see if REW can please implement frequency dependent windowing. Attached is a slide from Dr. Uli Brueggemann showing his example.

15 cycles (at 48 kHz I believe) is Uli's favorite setting. It provides sufficient psychoacoustic width for low frequencies and sufficiently "anechoic" width for the ear at high frequencies.

I'm usually using a 500 ms right hand window in REW which is probably psychoacoustically accurate for low frequency measurements. But the high frequency display is too smooth. So I tend to ignore or discount the high frequency measurements as they are not "resolved" enough with such a large window. It would be nice to see it all in one graph.
 

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#3 ·
I took a good look at the variable smoothing. My standard of excellence for sonic performance and psychoacoustic accuracy continues to be Acourate and Acourate Convolver. You might ask "so why are you continuing to use Room EQ Wizard?" The answers are simply: To cross check, and because REW offers a more effective distortion measurement, wonderful waterfall displays and is far easier to use than Acourate!

Also, I really don't know enough about the differences between smoothing and windowing to say with assurance that variable smoothing is a compromise, but it seems that way to me because the ear responds to the direct sound of the loudspeaker above the room's nominal Schroeder frequency and at 20 kHz Jim Johnston says to use a very short window, near 1 ms if possible. If you start with a 500 ms window but you smooth the result is that the same or effectively the same as using a 1 ms window? I don't think so, but I don't have the math to say for sure.


So I tried the new variable smoothing and my conclusions are (based on listening and visual comparison), that variable smoothing is not smoothed enough below 1 kHz, and oversmoothed above 1 kHz. I'm using a 500 ms right hand Hann window in REW. Attached are three images, all of the post-corrected front left speaker, corrected by Acourate.

Attached:

The first is an image of Acourate's psychoacoustic amplitude response display, using 15 samples at 1 kHz for the variable window. I exported a 2448 impulse from REW, loaded it into Acourate. First I then had to do a cut n' window because the impulse was far too long for Acourate to display a frequency response. I windowed it very very wide, so none of the actual information was cut off.

The second is REW's display of the same information, variable window.

The third is with 1/6 octave smoothing. 1/6 octave appears to be closer to Acourate's psychoacoustic measurement. Further investigation is needed, because this is of an impulse-corrected loudspeaker running through a convolver, so it has been considerably "smoothed" before REW could even measure it.
 

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#4 ·
If I understand correctly reading the latest beta notes, you have conquered the OSX measurement issues with Java that plagued us before and led me to use ASIO on PC?
That's right, OS X is now well behaved thanks to Oracle's Java runtime, which is bundled inside the REW download so Java does not need to be installed.

I'm usually using a 500 ms right hand window in REW which is probably psychoacoustically accurate for low frequency measurements. But the high frequency display is too smooth.
That's a little odd, since wide windows usually give ragged HF responses due to the comb filtering effects of reflections that fall within the window.

Variable windowing on its own isn't enough to give a psycho-acoustic response, other factors need to be taken into account such as removing or reducing narrow dips, which remain present in variable windowing (per images below - complex smoothing is mathematically equivalent to variable windowing). Variable windowing and variable smoothing are not equivalent, but smoothing provides better EQ targets from the tests I have done. In REW the variable smoothing uses no smoothing below 100 Hz, then varies from 1/48 octave at 100 Hz to 1/3 octave at and above 10 kHz. At 1 kHz the smoothing is 1/6 octave.

Image


Image
 
#5 ·
That's right, OS X is now well behaved thanks to Oracle's Java runtime, which is bundled inside the REW download so Java does not need to be installed.
That's fantastic, John. A great solution to a nagging problem. I will have to try sampling soon on OSX with REW!

That's a little odd, since wide windows usually give ragged HF responses due to the comb filtering effects of reflections that fall within the window.
Please take a look at the much-too-smooth response above 1 kHz in my image posted just a little above. Notice that even 1/6 octave shows more detail above 1 kHz than variable smoothing in its current incarnation.

Variable windowing on its own isn't enough to give a psycho-acoustic response, other factors need to be taken into account such as removing or reducing narrow dips, which remain present in variable windowing (per images below - complex smoothing is mathematically equivalent to variable windowing).
I'll take your word for it that complex smoothing is mathematically equivalent to variable windowing. So, is REW going to be able to do "complex smoothing"? The inaudible narrow dips are somehow taken care of by Uli Brueggemann. I don't know if it is his choice of window below 100 Hz or some other trick.... Regardless, it appears that with your current variable smoothing, set as you describe:

Variable windowing and variable smoothing are not equivalent, but smoothing provides better EQ targets from the tests I have done. In REW the variable smoothing uses no smoothing below 100 Hz, then varies from 1/48 octave at 100 Hz to 1/3 octave at and above 10 kHz. At 1 kHz the smoothing is 1/6 octave.
....is much too "aggressive" (compared to the perception) below 100 Hz and so needs some smoothing and much too smooth above 1 kHz, does not show as much detail as the ear perceives. I think that the irregularities of the direct response of a tweeter without room reflection issues are audible and of concern.

Regardless of whether you agree with Uli Brueggemann's particular "psychoacoustic" settings, from my point of view it would be nice to be able to set the parameters of REW's smoothing display so that it would be closer in display to Acourate's choices, if that is desired. Do you think that some parameters of complex smoothing could be set by the user and saved as a preset if desired? I do grant that at some point it becomes a completely subjective discussion. Nevertheless, I do have ears, and Acourate is the first system I have EVER encountered (in over 40 years of working with corrected and non-corrected audio systems) which corrects without requiring any further intervention on my part! That's a pretty remarkable endorsement.
 
#6 ·
Dear John: As much as I would like to love variable smoothing, it is not the same as a variable window. I conferred with Jim Johnston and basically he said that with the right parameters the results can be somewhat comparable, but the devil is in the details. I am not getting even close to Acourate's measurements at the high end. Basically I would like to marry a long FFT window length circa 200 to 500 ms below say, 1 kHz, with a short FFT window length circa 10 ms above 1 kHz.

Is there any way we can splice together two (or more) frequency responses? Or can you come up with a couple default pairs of windows? All the psychoacoustic papers by JJ and others make it clear that the ear responds to the earliest signals and not the room (near anechoic) at high frequencies, but the ear integrates the room at low frequencies. This means that a variable window will be far more psychoacoustically accurate. I hope you will be able to accomplish this sooner or later. Thanks!
 
#8 ·
Bobkatz,

Hi, my name is Nick. I very often read this and other forums but rarely do I chime in on a topic. Most of the other people on this forum are more experienced with the general types of questions I read, and I learn a lot by just reading all the Q & A. However, your question of frequency dependent windowing caught my interest as this is something I have a daily working experience with. Have you ever heard of SysTune? It's software from AFMG, a Berlin based software company that makes all kinds of awesome software that's used in the loudspeaker design and installation industry. SysTune is primarily geared toward the pro sound industry, but, I use it. And I work as an installer of custom car audio systems. Frequency dependent windowing, in SysTune this is called the TFC, which stands for time frequency constant. It works like this, when you set the window there are 3 markers that represent the right hand window. One for 8 kHz, one for 1 kHz, and one for 125Hz. Now those are just markers to kinda give you an idea of the "spread" of the TFC. In actuality it is a constantly variable time window without discrete steps. Whatever the window time you set for 8 kHz, it is twice as long for half that freq. For example, say you set your 8 kHz marker for 2ms after the peak of the IR, then your window time for for 4 kHz will be 4ms, 2 kHz-8ms, 1 kHz-16ms, and so on. This way, you can window out reflections but still keep full range response in one graph. Well, window out higher freq. reflections anyway. So as you can imagine, in a car, the reflected energy can be REALLY close to the direct arrival. So with conventional windowing, in a car, if you windowed out the reflections, you'd end up with barely useable data, it would be a time consuming process to properly perform a phase alignment of mid to tweeter in a fully active system for example. And then have to change your settings to see the next driver's interactions as you worked down the freq. scale. But with TFC, it works brilliantly. Another one that is similar to SysTune is Smaart v7. It is a program that uses what's called Multi Time Window, or MTW. however, with Smaart, the multi time window is not user adjustable, so while I do use that software at work for certain things, SysTune is my main program. Room EQ Wizard plays an integral role in my work by allowing me to create compensation files for different mics and inputs, and also for its trace arithmetic, which I use to find the cabin gain of different vehicles. Anyway, thought just in case you hadn't heard of SysTune (or Smaart) that you might find them interesting, as the very mechanism they use to perform their measurements is variable time windowing!
 
#10 · (Edited)
Bobkatz,

Hi, my name is Nick. I very often read this and other forums but rarely do I chime in on a topic.windowing!

Hi, Nick. For your car measurements, windowing out the phasey issues with near walls are very important. But in the car case it's as much for practical reasons as for psychoacoustic reasons since the walls are so close they produce obvious anomalies when the window is too wide. But for psychoacoustic reasons you need a long window to assess how the ear responds to the bass.

I have been using combinations of Acourate, Room EQ Wizard, Spectrafoo and FuzzMeasure for a long time. The beauty and detail of the graphics, the ergonomics as well as the features are all important to me so I can't live without any of them. Of those four, only Acourate has a variable (psychoacoustic) window and is the most accurate of all. And that's what I use when I need to be as precise as possible. But ergonomics are not Acourate's strong point. As you can see I'm on a campaign to lobby the rest of these fine applications to implement a variable window :).

Bob
 
#13 ·
Thanks very much, John. Would a sample response of a real world loudspeaker be ok?

Or, what would a Dirac pulse tell you? ... but that would just look ruler flat. Not sure what you would learn I suppose unless we created some special test signals? Give me some hints as to how we would determine that other than to just do it.

If the job of a sliding window seems difficult to you, what about the idea of two or more frequency responses spliced together, one made with one window and one made with another. I know that John Atkinson frequently splices together two frequency responses, one nearfield near the ports of a loudspeaker and one farfield... so that's an approach.

Best wishes,


Bob
 
#14 ·
A speaker measurement would be preferable. There is no particular difficulty in the basic implementation, I did it back when deciding what kind of variable option to offer (as discussed in this post), but some thought would be needed around how to make the option available and how to allow the settings to be adjusted.
 
#15 ·
Dear John: I'm so glad you are willing to consider this variable feature as it is different from smoothing or averaging.

I can show you a frequency response measurement from Acourate and I could send you the impulse response of those speakers that produced that measured frequency response. Would that help?
 
#16 ·
In terms of how to specify the variable window, one method is to define the number of cycles (Hertz) that it's wide. For example, 15 cycles at 1 kHz translates to 15 ms. And so on.

I am not 100% convinced that approach gives a long enough window for psychoacoustic estimates at low frequencies when it gives a short enough window at high frequencies. However, to my ears, in my room, I have never had to manually touch or correct Acourate's correction filters so they must be doing something right, at least at the low end. The amount of treatment in the room and the room decay time surely must enter in here. My room is well treated so there isn't much to perceive after 150 milliseconds at 100 Hz.

Maybe I can ask JJ and he'll help. But he's constrained by some old NDAs so he may not have an answer. Won't hurt to ask.
 
#18 · (Edited)
OK, here we go. It appears that the "default" psychoacoustic smoothing in Acourate works this way: The window width is a linear interpolation from a low frequency of 16 Hz up to Nyquist. Not sure why Nyquist, but if the Sample rate (most commonly used) is 48 kHz that would be 24 kHz, which isn't too bad as a standard.

He takes this window measure on 1/24th octave intervals.

The apparent "default" window width is specified as 15/15, which means 15 cycles at the low frequency and 15 cycles at the high frequency. These values can be changed, but I've found they produce a result that correlates well with the ear's interpretation of the response. 15 cycles would be 15 ms. at 1 kHz. To the best of what I can determine this is actually the right hand window width, and he uses a Blackman, but some experimentation is in order. Anyway, changing the left hand window size, as long as it is reasonable, does not seem to affect the displayed frequency response.

To compare REW and Acourate I took a screenshot of Acourate's psychoacoustic response of my front left speaker taken at 9 feet distance. I exported the impulse response as a 24 bit/48 khz wav (attached). I then used my usual 500 ms. Tukey with 1/6 octave smoothing to display it in REW. I then saved this as a PNG. I then brought both of these into photoshop, carefully matched the scales, extracted the Acourate curve with the magic wand tool, and overlaid them matching amplitude at 1 kHz. That's the image I've attached.

I'll give you my thoughts about these differences in my next post in this thread.
 

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#116 ·
OK, here we go. It appears that the "default" psychoacoustic smoothing in Acourate works this way: The window width is a linear interpolation from a low frequency of 16 Hz up to Nyquist. Not sure why Nyquist, but if the Sample rate (most commonly used) is 48 kHz that would be 24 kHz, which isn't too bad as a standard.

He takes this window measure on 1/24th octave intervals.

The apparent "default" window width is specified as 15/15, which means 15 cycles at the low frequency and 15 cycles at the high frequency. These values can be changed, but I've found they produce a result that correlates well with the ear's interpretation of the response. 15 cycles would be 15 ms. at 1 kHz. To the best of what I can determine this is actually the right hand window width, and he uses a Blackman, but some experimentation is in order. Anyway, changing the left hand window size, as long as it is reasonable, does not seem to affect the displayed frequency response.

To compare REW and Acourate I took a screenshot of Acourate's psychoacoustic response of my front left speaker taken at 9 feet distance. I exported the impulse response as a 24 bit/48 khz wav (attached). I then used my usual 500 ms. Tukey with 1/6 octave smoothing to display it in REW. I then saved this as a PNG. I then brought both of these into photoshop, carefully matched the scales, extracted the Acourate curve with the magic wand tool, and overlaid them matching amplitude at 1 kHz. That's the image I've attached.

I'll give you my thoughts about these differences in my next post in this thread.
and the same for this one

174862
 
#19 ·
Reactions to the frequency response differences: Well, now that I have the scales perfectly matched in Photoshop, the differences are relatively small, but I think the devil is in the details. I think the most important difference is at the high frequencies, because the red curve is near anechoic and better reflects the perceived response of the loudspeaker. Certainly the slope of the Hf curve is dramatically different. Demonstrating that "one window width does not fit all purposes". At the low frequencies, 500 ms. Tukey + 1/6 octave smoothing is remarkably close to Acourate. At mid frequencies there are differences and depending on how many EQ bands you want to generate, this may or may not be significant.

I can't swear whether the small differences between Acourate at the low frequencies are meaningful or even perceptible, but at least they are visible! In some cases we see 2 dB difference, which could be audible to a keen ear, but don't ask me to take a blind test on this! So let me summarize by saying that probably 1/6 octave smoothing with 500 ms Tukey does a very good job at low frequencies, but not at mid or highhigh. Only by having the ability to correct a system one way or another and observe the results can we reach a valid conclusion.
 
#20 ·
I think the shape of the HF in Acourate from 1 kHz up is probably right on. I say this because the Revel Gems measured in Acourate very closely follows a linear slope from 1 kHz to 20 kHz, which is I think the intent of the loudspeaker. While in the REW result it looks like a complex curve. Acourate's measurement shows that the Revel Gems also slope very linearly and naturally from 1k to 20k.

In Acourate Convolver I can easily implement my preference, a simple linear sloping target which is just about a 1/2 dB lower than the native response of the Revels. Acourate Convolver performs that and subtly smooths out the HF response variations and after correction, the loudspeakers sound very smooth, subjectively flat.

Attached is the Left front response, psychoacoustic measure, before and after correction, 1 dB/vertical div.
 

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#22 ·
No. I use a Josephson C550 which David Josephson confirmed was within spec before shipping.
 
#23 ·
Difficult to explain the high frequency difference then, since using a narrower window (e.g. 10 ms Tukey) lowers the HF response further (there are strong reflections at 10ms and 27 ms that get windowed out).
 
#24 ·
John, is that based on the impulse response that I sent? In that case I don't have a good explanation either. But typically long window time produces a lowered high frequency response since it incorporates more of the room.

Try a 1 ms. Tukey???? Regardless, we have to solve the mystery why Acourate is showing a higher amplitude high frequency response with the same impulse. It couldn't be related to the fact that they do everything floating point and so the 24-bit export I made is a reduction? Seems VERY doubtful as the cause.
 
#25 ·
Perhaps the difference will only be revealed with a continuously variable window? To be more exact, a window that changes every 1/24th octave interval.
 
#26 ·
I just tried a 1 ms Tukey and it produces a big dip from 1k to 3k with this impulse so it's hard to say but it's a lot flatter from 5k on up than the 500 ms, so I think a variable window is the explanation for the difference.

Actuall I see there are two Tukey windows, and with Tukey 0.25 set to 1 ms. it's got that flat character from 5 k on up. But with Tukey 10 ms set to 1 ms it's got the rolloff that I suspect looks more like the Acourate version, but I'd have to line them up to see. It takes a bit of work to bring things into Photoshop for an overlay.

I don't understand the difference between the two Tukey choices....
 
#27 ·
Tukey 0.25 applies the roll-off over that last 25% of the width, Tukey 10 ms applies it over the last 10 ms (so not meaningful with widths < 10 ms).

I don't think windowing can account for the difference, here are the responses from that IR with 500 ms Tukey 0.25, 10 ms Tukey 0.25 and 1 ms Blackman. Narrower windows reduce the HF between 2k and 20k for this response (note that need to be careful about the window start time when applying narrower windows).



I did note that the IR itself appears to have been windowed before export, but with fairly broad windows. It also has large peaks right at the start of the IR.

 
#28 ·
I'm puzzled, too. I don't know what to say or suggest, John. Taking the 2 kHz response as a reference, Acourate is about 4 dB down at 10k but all your choices are 4 to 7 dB down at 10k and in fact as you indicate, the wider window is brighter (with this impulse) than the narrower window.

Any further ideas? Is there any merit to my suggestion that "you may have to implement the variable window in order to prove the concept." Is it possible that a window that varies from 15 ms. at 1k to 0.75 ms. at 20k will have a brighter apparent trace?

Mind you, I've long ago become resigned to the fact that specifying a recommended rolloff requires also specifying all the characteristics of the system you used to measure that rolloff as they are quite individual. However, my premise as you know (and JJ makes it clear in one of his papers) is that smoothing is not the same as a narrow window, because with a wider window you are smoothing more time information than you want to take into account. That's the principle at least.

But apparently not in practice. Then how did Acourate obtain that apparently more correct shape to the curve when measuring these loudspeakers?
 
#29 ·
John: Would you suggest that I take some new parallel measurements with Acourate and REW with the same microphone, same interface, same gain, etc. and then we can compare the look of the raw impulse rather than do an export into REW?
 
#31 ·
I can do all that! By next weekend :). Stand by for the fireworks!
 
#32 ·
I hesitate to break in on this conversation, addressing a topic I find very interesting and a possible change to REW which would be a benefit to us all, if implemented correctly.

I will cut to the chase, give you the short version and then fill in the details with the longer version. I question an assumption made early on, upon which the discussion has hinged and which clearly makes a big difference in the success of the windowing and/or smoothing functions in REW. Specifically that at high frequencies our perception of frequency response is a result of the direct and not the reflected waves, therefore accurate frequency response measurement will window out all of the reflected information. Or, as Bob put it, we hear the speaker not the room. For the purpose of this discussion about accurate measurement of high frequencies by REW, I fear this is not an accurate assumption.

There I have said it. Before you discount me me as a lunatic, please allow me to explain.

Please know that I enter this discussion is a friend, with no intentions of being disruptive, only wishing for the same outcome as everyone else, the most accurate way to have our beloved REW measure frequency response as we perceive it.

The reputations and accomplishments of both Bob and John are not lost on me, nor are those of the authorities in the areas of psycho acoustics upon whose research our progress in this discussion relies. It is with the highest respect that I wish to suggest that an over-simplification has been made in the above assumption.

Of course we are referring to precedence, the Haas effect, well known to us all. I will not restate what is obvious to those involved in this discussion, but will gently - nudge, nudge - remind that it is not an absolute but a tendency, and that the fused direct+delayed sound, as perceived by the ear, can be affected by the delayed sound in terms of spaciousness and timber. As you put it, Bob the devil is in the details, and the finest, most devilish details of audio are what our lives are all about.

My situation is as follows: My speakers are dipoles, electrostatics, and they are set up so that the direct wave to the listening position is quite a ways off-axis of the electrostatic panel. The path of the reflected wave, coming from the wall in front of me, or rather coming from a hard reflective panel carefully positioned on that wall, is virtually directly on-axis of the rear of the panel. Even allowing for the MartinLogan design with its curved panel and broad off-axis radiation pattern, the front wave has significantly attenuated high frequency response relative to the reflected wave.

I will gladly provide measurements, but will start with a simple demonstration. As we speak, my left side electrostatic panel has several layered towels draped over the back, severely attenuating high frequency information reflecting from the wall. The right speaker has no such “attenuator” in place. When I play a mono track through the left speaker only, the LP is receiving primarily the front wave from the left speaker at an off-axis angle of roughly 25 degrees. When I switch that mono track to the right speaker, I hear the direct + reflected waves on at side, and there is a significant brightening of the high frequencies, mainly in the range above 3 kHz.

Bob, the priorities of my listening room and experiments in soundstage construction are a different environment altogether from your mastering room. And my interests and listening preferences have undoubtedly led me to listen for certain kind of detail that you may very well have learned to tune out of your experience by way of room treatment and speaker selection, and by simply paying attention to the sound in a different way. The point here is that there is enough adaptability in our hearing that the perception of devilish sonic details may cover a range of possibilities rather than simply fit the hearing model you have assumed for the matter at hand, that we hear the speaker and not the room.

How can this be? Among the devilish details that need to be considered, I believe, are differences in listening experience and ear training, and differences in listening priorities and preferences, all of which fall into the category of perceptual adaptability. A good look at the details of the research often referred to will reveal that the conclusions reached were not absolutes but covered ranges of possibility with certain probabilities - ah, beloved statistics! - and that a given study, in order to have hopes of a meaningful outcome, would require that limiting assumptions be made at the get-go. That is simply how scientists make progress.

My listening room may be far from typical, but I cannot be the only audio nut playing around with reflections to create a soundstage and hoping to take measurements that accurately show what I'm hearing. Of course there is little doubt that in terms of who has the bigger voice here, and perhaps the greater need for accuracy, Bob wins out, and if such a choice had to be made I would go along with it with a minimum of grumbling. That is meant in the most good-natured way possible.

They say that when you bring up a problem, it is good manners to offer a possible solution. It sounds like what might be called for is a variable setting for that varying windowing function, just the kind of complexity you were hoping for, right John? Unfortunately I have no way at this point to suggest what range of what variable might be useful, or what a possible “alternative setting” might be to what Bob is suggesting - with lots of data to back it up his requests. So if you want to say, “come back when you have some data,” that is fair.

What I really hope is that there is a little room in this discussion for recognizing some range of perceptual possibilities that needs to be accounted for.

Forgive me if I have muddied the waters.

I will take some measurements in the coming days to better illustrate what I am hearing.
 
#33 · (Edited)
Dear AudioCRaver: What is the distance of the Martin Logans from the front wall? How is the front wall treated? As far as I can see, your use of reflections to help create a soundstage has little to do the likelihood that your perception of the direct sound from the loudspeakers is the primary high frequency response.

While we're at it, I'd like to reference a U.S. Patent on my own invention that uses a specific set of delays to enhance ambience, spatiality and soundstage. Somewhere around or under 30 ms is the magic time for that, but the devil is in the details. I just wanted to mention that because since you mentioned Haas I thought I would mention that Haas and I are very good friends :).

I doubt that your ear/brain is wired any differently from other humans, so it's highly largely your perception of the frequency response of your loudspeakers is within less than 10 ms. and the rest of the effects you hear that enhance the soundstage are independent of your judgment of the high frequency response.

Anyway, never fear. The variable window should be user adjustable to account for a range of tastes or other circumstances.

Also, John and I (and anyone else who would like to chime in) have to delve further into the mysteries of Acourate to figure out why its high frequency determination is so much brighter than REW's, when it appears NOT to be window width. Mysteries, mysteries, mysteries.
 
#39 ·
Bob,

Thanks for your kind words and your patience. Please forgive me if I misinterpreted any of your statements, I really try to be accurate about such things. And be assured that my intention is only to contribute to the discussion, not pull it off track in any way.

I realize this is not a discussion about ambiance or soundstage enhancement, that is simply the example with which I am best able to illustrate my point. Thank you for humoring me, I know both you and John are extremely busy guys.

Dear AudioCRaver: What is the distance of the Martin Logans from the front wall? How is the front wall treated?
From electrostatic panel to front wall is 56 inches, a little under 5 feet. The reflected energy reaches the LP after a delay of 8.2 ms. The front wall is finished drywall with slightly textured paint. The speakers are 63 inches apart center-to-center. The reflective panels are flat, unpainted boards, each 12 inches wide, spaced 80 inches apart center-to-center and the entire area of the wall between the panels is treated with absorptive material.

As far as I can see, your use of reflections to help create a soundstage has little to do the likelihood that your perception of the direct sound from the loudspeakers is the primary high frequency response.

While we're at it, I'd like to reference a U.S. Patent on my own invention that uses a specific set of delays to enhance ambience, spatiality and soundstage. Somewhere around or under 30 ms is the magic time for that, but the devil is in the details. I just wanted to mention that because since you mentioned Haas I thought I would mention that Haas and I are very good friends :).
Very cool, I will look it up!

I doubt that your ear/brain is wired any differently from other humans, so it's highly largely your perception of the frequency response of your loudspeakers is within less than 10 ms. and the rest of the effects you hear that enhance the soundstage are independent of your judgment of the high frequency response.
I was referring only to the ability to "learn to listen," or focus, differently, based on interest, need, exposure, etc. The non-sighted person will "hear" a room differently from you or I out of the intense desire to not be tripped or run into a wall or fall out of a window. Your decades of experience in the studio and mastering room have no doubt taught you to hear a lot in a room and speakers that others would overlook. While a few psychoacoustical studies focus on musicians and/or experienced "critical listeners," most focus on the general population. As we learn to listen to finer and finer levels of detail, for whatever reason, I believe it possible, even probable, that some of the psychoacoustical limits defined by those important studies get stretched, sometimes quite a bit.

Anyway, never fear. The variable window should be user adjustable to account for a range of tastes or other circumstances.
Thank you, that answers my concern!

Also, John and I (and anyone else who would like to chime in) have to delve further into the mysteries of Acourate to figure out why its high frequency determination is so much brighter than REW's, when it appears NOT to be window width. Mysteries, mysteries, mysteries.
Indeed, ain't life fun!

Oh by the way, I did not exactly say that "we hear the speaker and not the room". What I said is that authorities such as Jim Johnston have researched that the ear's perception of a loudspeaker's response approaches anechoic at high frequencies, that at high frequencies the brain ignores even the earliest reflections. So a true high frequency measurement should begin to ignore reflections as the frequencies rise.
I do beg your pardon if I put words in your mouth. The phrase is used from time to time referring to all the frequencies in a room above the Schroeder frequency, and while that might not be too inaccurate for general discussion, when referring to the general listener, this is a discussion about picky details as perceived by highly experienced, sensitive listeners, and would be an oversimplification under the circumstances.

Repeating part of your above quote...

...a true high frequency measurement should begin to ignore reflections as the frequencies rise.
...(my added emphasis) I would agree is a more accurate way to state the phenomenon. And just HOW MUCH to ignore it is the perplexing mystery under investigation.

This certainly does not exclude such phenomena as soundstage or ambience enhancement by use of early reflections or bidirectional loudspeakers. In those cases the ear certainly does hear the room. Our perception of the frequency response of a loudspeaker system at lowest frequencies on up through the midrange includes much of the room response as well as the direct response. Hope this helps.
Thank you again for the detailed explanation. I am satisfied that the phenomenon is being treated as a tendency, the sensitivity of which is not fully understood.

I am close to your age, and my audio-centric roles have been numerous, some for pleasure and some professional, including a lot of hours staring at RTA plots and playing with parametric EQ bands just to see what they sound like and how small a change I can detect, and have gotten pretty good at sorting through the details of what the differences might be between sound A and sound B - not unique wiring, as you say, but ability to focus and discriminate as driven by an obsessive curiosity that borders on the ridiculous at times. Well aware that much of the soundstage enhancement effects take place in mid frequencies, the following measurements from my system might help illustrate my point.

All smoothing is 1/12th octave. All windowing referred to is Tukey 0.25 right window.

1. Here is the measurement at the LP of the signal from the front of the electrostatic panel only. The rear signal is being absorbed by thick towels draped over the back of the panel. Note that the plots for 5 ms window and 15 ms window virtually overlay each other above 1 kHz, showing that there is no contribution above 1 kHz from the absorbed rear, reflected signal, which would fall completely within the 15 ms gated window. It says "there is no reflected signal, it is being completely absorbed."


2. With the towels removed and the rear, reflected signal freely combining with the front signal after the 8.2 ms delay, with 5 ms gating. It is like saying "we would not hear the high frequencies above 3 kHz, so let's gate them out of the measurement." If that is true, according to these plots, removing the towels should result in no discernible change to the response above 3 kHz.


3. But I DO hear SOME brightening of the response IN THAT RANGE. The trained ear can focus on that area of frequency response detail, as I am sure you do every day. By opening the window for the combined plot (green), we can see there is quite a bit more information the MIGHT be heard by the TRAINED ear. How much, how high? I don't know (shoulder shrug).


4. As a reference, here is an impulse plot of the direct and reflected sounds.


My ears are not special, and neither are yours. Well, ok, you are Bob Katz, so maybe yours are a little bit special. Laugh with me here. Like A.J., I like what you are doing, just trying to help make sure that one key assumption is being handled properly, and I am quite satisfied now that it is.

I have said enough. You have addressed my concern, and I thank you. If I see a way to contribute to the project, I will try to climb in without swamping the canoe. Otherwise I will follow along quietly, knowing the work to be in excellent hands. Thanks again for listening.
 
#35 ·
Oh by the way, I did not exactly say that "we hear the speaker and not the room". What I said is that authorities such as Jim Johnston have researched that the ear's perception of a loudspeaker's response approaches anechoic at high frequencies, that at high frequencies the brain ignores even the earliest reflections. So a true high frequency measurement should begin to ignore reflections as the frequencies rise.

This certainly does not exclude such phenomena as soundstage or ambience enhancement by use of early reflections or bidirectional loudspeakers. In those cases the ear certainly does hear the room. Our perception of the frequency response of a loudspeaker system at lowest frequencies on up through the midrange includes much of the room response as well as the direct response. Hope this helps.
 
#36 ·
Wayne, without dragging this thing off the deep end, you are not measuring spatial qualities of the soundfield with a pressure microphone as used with REW etc.
You are measuring essentially the sum of pressures at that sample point. What I believe Bob wants is a better psychoacoustic representation of that pressure, as it would be perceived tonally/timbrally.
When you post you LP pressure measurement, it should tilt downwards quite a bit from bass to treble. How do you perceive that HF?
The counter example would be to equalize the amplitude so that it is flat at the pressure mic (at the LP). Try it and tell us how the HF sounds.:)
Needless to say I like what Bob is asking for.

cheers,
 
#37 ·
Bob, can you site some of the research you refer to by Jim Johnston, for those of us, who, like myself, aren't yet as far along in our research / understanding of these phenomenon. As I have mentioned in an earlier post, in my line of work I believe that there is a SEVERE lack of understanding of acoustics and psychoacoustics, and I for one am working hard to change this. But in order for me to do my part, I myself must have as deep an understanding as a possible. Also, I would like to say thank you to everyone who has been involved in this thread, I have never been so intrigued by a forum, this is great!
 
#42 ·
Glad to help, if I can. JJ and his colleagues have published a number of papers. In some areas (particularly FDW) unfortunately, the papers only hint at actual numbers to use. In private conversations with JJ I got a little bit more info and have published as much as I know here. However, you can get a good background on this subject and more by reading the following AES preprints, which you can purchase from the AES website. http://www.aes.org/e-lib/

The papers are: Preprint #7263, 8314, and 8379.