# Feature Request: Frequency Dependent Windowing



## bobkatz

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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## 3ll3d00d

A different form of variable smoothing is in the latest beta - http://www.hometheatershack.com/for...-beta-release-asio-support-13.html#post957161


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## bobkatz

3ll3d00d said:


> A different form of variable smoothing is in the latest beta - http://www.hometheatershack.com/for...-beta-release-asio-support-13.html#post957161


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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## JohnM

bobkatz said:


> 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.


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## bobkatz

JohnM said:


> 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.


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## bobkatz

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!


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## ajinfla

bobkatz said:


> 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!


Hi Bob,
You've been talking to the right people.
Did JJ suggest 1k, or more around 700Hz? TIA

cheers


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## Niick

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!


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## bobkatz

ajinfla said:


> Hi Bob,
> You've been talking to the right people.
> Did JJ suggest 1k, or more around 700Hz? TIA
> 
> cheers


All I remember from talking with JJ is a window width of up to 500 ms below about 200 Hz and very short (1 or 2 ms) at 20 kHz. Yes, the devil is in the details. Windowing is NOT the same as smoothing and in JJ's AES papers he makes that clear. If you are not looking at the earliest sound first it's not the same as smoothing the whole kit and kaboodle!

JJ also told me that with the right windowing that a subjectively flat response will correspond with a measured flat response. Perhaps someday he will enlighten us how to do that. However, I've never experienced that and not being the super mathematician that JJ is I just have learned to accept that I have to use a particular high frequency rolloff with a particular measurement system. As long as the measurement system has variable windowing I can see enough detail in the high frequency response and be confident it is looking at the near-anechoic response of the loudspeaker in the HF region. 

The rest of the devil is in the details .


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## bobkatz

Gennelle51187 said:


> 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


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## Niick

Very nice. +1 for frequency dependent time windowing?


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## JohnM

Hi Bob, if you can send me an example impulse response and a screenshot of how you prefer it to be presented I'll look at some windowing options.


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## bobkatz

JohnM said:


> Hi Bob, if you can send me an example impulse response and a screenshot of how you prefer it to be presented I'll look at some windowing options.


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


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## JohnM

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.


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## bobkatz

JohnM said:


> 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.


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?


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## bobkatz

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.


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## JohnM

bobkatz said:


> 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?


Yes, that would be great, along with a note on whatever settings you used to produce the response.


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## bobkatz

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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## bobkatz

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.


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## bobkatz

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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## JohnM

Was any cal file being used for the mic with Acourate, Bob?


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## bobkatz

JohnM said:


> Was any cal file being used for the mic with Acourate, Bob?


No. I use a Josephson C550 which David Josephson confirmed was within spec before shipping.


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## JohnM

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).


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## bobkatz

JohnM said:


> 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).


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.


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## bobkatz

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.


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## bobkatz

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....


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## JohnM

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.


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## bobkatz

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?


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## bobkatz

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?


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## JohnM

Sounds like a good idea, may also want to compare Acourate's psychoacoustic frequency response with its unprocessed (but 1/6 octave smoothed?) frequency response.


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## bobkatz

JohnM said:


> Sounds like a good idea, may also want to compare Acourate's psychoacoustic frequency response with its unprocessed (but 1/6 octave smoothed?) frequency response.


I can do all that! By next weekend . Stand by for the fireworks!


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## AudiocRaver

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.


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## bobkatz

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.


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## Niick

Now THIS is good stuff!! Best discussion ever??


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## bobkatz

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.


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## ajinfla

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,


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## Niick

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!


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## 3ll3d00d

I have compared fdw in Holm and acourate before and found them to agree when the same window sizes are chosen. Holm shows the window length at each decade so this is easy to equate to acourate of you use the same value for high/low and left/right in acourate.

The Holm guide has a section of how it calculates the windowed response at the end - http://www.holmacoustics.com/downloads/HOLMImpulse/HOLMImpulseUserGuide.pdf

I only did this for one measurement though, I can try another in the next few days with independent measurements to verify.


----------



## AudiocRaver

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.



bobkatz said:


> 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!



bobkatz said:


> 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.


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## 3ll3d00d

Some measurements of my L speaker (which is a fairly small sealed speaker, MK MP150). Mic is at the usual MLP about 3m away, mic pre is an RME FireFace 800, mic is a CSL calibrated EMM6, same levels used for both sweeps & same length sweeps used (approximately... 512k in REW is 10.9s and it was a 10s long sweep in Acourate) & same freq range (10-24000Hz). 

Two wavs attached, one is the acourate sweep itself (in 24 bit PCM form for import into REW) and the other is the same after a 15 cycle FDW applied.

Comparison of REW variable smoothing and acourate 15 cycle FDW below


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## JohnM

That's interesting Matt, thanks - can you also post an image of how the Acourate psychoacoustic response looks in Acourate's display? Or is it the same as the appearance of the imported FDW IR in REW?

Those phase cancellation dips in the FDW trace between 250 and 700 Hz are what put me off including FDW in REW.


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## bobkatz

Gennelle51187 said:


> 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!


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.


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## bobkatz

3ll3d00d said:


> I have compared fdw in Holm and acourate before and found them to agree when the same window sizes are chosen. Holm shows the window length at each decade so this is easy to equate to acourate of you use the same value for high/low and left/right in acourate.
> 
> The Holm guide has a section of how it calculates the windowed response at the end - http://www.holmacoustics.com/downloads/HOLMImpulse/HOLMImpulseUserGuide.pdf
> 
> I only did this for one measurement though, I can try another in the next few days with independent measurements to verify.


Thanks for that Holm reference. Can you please download the impulse file that I used in Acourate and imported into REW and see where you think things are going wrong? That is, if they are going wrong....

Also I will try this weekend to do what John asked me to do towards the end of this thread. It's quite involved because I want to take a whole new set of measurements with Acourate, and it's a day's job or more to follow all the details needed.


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## bobkatz

Dear Audio Craver:

Your thorough answer is very helpful. I'm sure I have more to answer but my first answer regarding removing and adding the towels on the rear of the loudspeaker... Even though they are on the back of the loudspeaker, the towels will contribute to the liveness and deadness of the room at upper mid to high frequencies, so you will perceive a difference. It may be so subtle it won't visually contribute to REW's graph of octave band analysis of RT60 unless you look very carefully.... but RT60 and EDT are the areas which the towels on the back of the Maggies will affect and you will perceive a high frequency difference in the ambience of the room and the liveliness of the room. 

Now this may seem contradictory to my statement that the ear perceives the loudspeaker's high frequencies largely anechoically, but what we are observing with the towels is the effect on the high frequency character of the room. This does affect our perception of the liveliness of the sound character, the part that occurs after the attack of the snare drum, for example. And Schroeder and other authorities make it clear we want to try to even out the reverb time (or EDT) of the room with respect to frequency or the room can be perceived as too dead or too live... even if we think the loudspeakers themselves are still "flat". 

------

Next, regarding the distance of the loudspeakers from the front wall. You've basically said that an attenuated delay of about 8 ms (for 8 feet), below about 1 kHz will be added to the direct sound from the loudspeaker. This will (hopefully) color the ambience in a nice way, and by means of the Madsen effect (see my patent) it will bring out the ambience in the original recording. Bravo... you've set up a bit of an ambience extraction device in your room. This is of course a well-known characteristic of dipoles. Personally I don't like to have that kind of thing built into all my reproduction... it's conceptually like an effect button that's engaged all the time. No offense intended, I've always enjoyed Maggies for what they do and it's very enjoyable, just not my cup of tea. I'll bet you I can find a number of pieces of music where this effect of the rear wall bounce would produce an inferior sound to it not being there, that is, if we could somehow eliminate the reflection in a controlled single-variable experiment ;-)

Hope this message helps and doesn't muddy the waters.


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## bobkatz

JohnM said:


> That's interesting Matt, thanks - can you also post an image of how the Acourate psychoacoustic response looks in Acourate's display? Or is it the same as the appearance of the imported FDW IR in REW?
> 
> Those phase cancellation dips in the FDW trace between 250 and 700 Hz are what put me off including FDW in REW.


You bet those dips would put me off, too. But I've never seen anything that looks like that in Acourate unless it is a comb filter due to a known early reflection. And my room doesn't have that kind of problem, as viewed in my earlier post in this thread. 

Not sure yet what to make of all these discoveries. Is Acourate using some kind of secret sauce?


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## bobkatz

3ll3d00d said:


> Some measurements of my L speaker (which is a fairly small sealed speaker, MK MP150). Mic is at the usual MLP about 3m away, mic pre is an RME FireFace 800, mic is a CSL calibrated EMM6, same levels used for both sweeps & same length sweeps used (approximately... 512k in REW is 10.9s and it was a 10s long sweep in Acourate) & same freq range (10-24000Hz).
> 
> Two wavs attached, one is the acourate sweep itself (in 24 bit PCM form for import into REW) and the other is the same after a 15 cycle FDW applied.
> 
> Comparison of REW variable smoothing and acourate 15 cycle FDW below
> 
> View attachment 94442


I just skimmed the Holm user guide. This guy looks like another FFT genius on the order of John, JJ and Uli. Way over my head. Maybe John can make some sense out of the section in the Holm guide on smoothing using the variable window. Is that the key to not having those comb filter artifacts?


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## 3ll3d00d

bobkatz said:


> You bet those dips would put me off, too. But I've never seen anything that looks like that in Acourate unless it is a comb filter due to a known early reflection. And my room doesn't have that kind of problem, as viewed in my earlier post in this thread.
> 
> Not sure yet what to make of all these discoveries. Is Acourate using some kind of secret sauce?


Remember that macro1 runs both TD-Functions/Psychoacoustics and TD-Functions/Frequency dependent Window so one ingredient in acourate's secret sauce is that psychoacoustic smoothing function. Running either one alone can produce some slightly odd looking responses, indeed running them in the wrong order produces an odd response too. 

The sample I posted is using just FDW hence the spikiness.

I'm not at my home computer atm btw so will post the other pics/responses later. My room is untreated except for soft furnishings btw so it won't be as well behaved as yours, in acourate IACC terms I get ~77% after correction & ~70-71% before.


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## bobkatz

3ll3d00d said:


> Remember that macro1 runs both TD-Functions/Psychoacoustics and TD-Functions/Frequency dependent Window so one ingredient in acourate's secret sauce is that psychoacoustic smoothing function. Running either one alone can produce some slightly odd looking responses, indeed running them in the wrong order produces an odd response too.
> 
> The sample I posted is using just FDW hence the spikiness.
> 
> I'm not at my home computer atm btw so will post the other pics/responses later. My room is untreated except for soft furnishings btw so it won't be as well behaved as yours, in acourate IACC terms I get ~77% after correction & ~70-71% before.


This is a very important and enlightening observation. Not sure what Uli means by "psychoacoustics" in macro 1 and he is necessarily close-lipped about it. I thought it was just an improved implementation of FDW. I spoke to him about the FDW function in the time functions menu a couple of years ago and he recommended using the macro instead as it was more sophisticated. Then for this thread first I tried doing the FDW function in the time window and I did keep getting weird dips in the lower midrange! I thought it was just me not being able to plug in the exact necessary parameters. When I ran the macro I got the much smoother plot you see earlier in this thread. 

After we peruse this thought a little bit I'll ask JJ saying that "every time I try an FDW I get these dips and anomalies in the lower midrange, what's with that?" And maybe I'll ask him about 1/6 octave smoothing with a long window versus FDW and maybe he'll have some gems of wisdom. JJ is under an NDA about all his psychoaoustic discoveries he made while employed by previous employers so he can only say so much.


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## 3ll3d00d

here's a comparison of the options

green is FDW alone
blue is FDW the psychoacoustic
brown is psychoacoustic alone
teal (somewhat hidden by brown) is psychoacoustic then FDW

the obvious impact of the psychoacoustic smoothing is to cut off the dips 









@JohnM is that the view you were after? leaving aside minor differences in formatting (e.g. aliasing, thickness of lines), I think REW and acourate render the FR the same way. The IR does look a bit different in acourate though but obviously it's the same data.


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## JohnM

Yes, just the ticket Matt, thanks.


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## AudiocRaver

Bob,

Your comments have been informative and have given much food for thought. Thanks again.


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## JohnM

I've generated a beta build that has a psychoacoustic smoothing option that uses a frequency-dependent window and dip limiting, there is a Windows version and a Mac version. I don't recommend using the psychoacoustic smoothing for the EQ target match, variable smoothing will produce better results.


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## ajinfla

Thanks John


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## bobkatz

3ll3d00d said:


> here's a comparison of the options
> 
> green is FDW alone
> blue is FDW the psychoacoustic
> brown is psychoacoustic alone
> teal (somewhat hidden by brown) is psychoacoustic then FDW
> 
> the obvious impact of the psychoacoustic smoothing is to cut off the dips
> 
> View attachment 94450
> 
> 
> @JohnM is that the view you were after? leaving aside minor differences in formatting (e.g. aliasing, thickness of lines), I think REW and acourate render the FR the same way. The IR does look a bit different in acourate though but obviously it's the same data.



Dear Senior Shackster (sorry I do not have your name): Thanks very much for that research!

I was going to attempt that same demo this weekend, but I spent the whole weekend working on my surround alignment and did not finish it yet. So I'm very gratified you made this display for us!!!!

I know how to get some of your traces but not all and I'm a bit puzzled by your nomenclature: 

1) "green is FDW alone". Did you get that trace via the Time domain menu option? 

2) "blue is FDW the psychoacoustic". How did you get that display? What do you mean by that?

3) "brown is psychoacoustic alone". I believe that's the file named like: "Pulse48Lpsy.dbl" from macro 1. It's the first half of macro 1. 

4) "teal is psychoacoustic then FDW". I believe that's really the file named like: "Pulse48Lmp.dbl" which you are not showing on your display (it would be the red trace if you checked the box), and this is the file which, to the best of my knowledge, Acourate uses to determine the official smoothed psychoacoustic response of the loudspeaker for DRC in the next macros. Please clarify your steps, thanks.

5) Lastly we need to compare the "mp" trace (which I believe is the one Acourate uses) versus basic 1/6 or 1/12 octave smoothing in Acourate and 1/6 or 1/12 octave smoothing in REW.


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## bobkatz

JohnM said:


> I've generated a beta build that has a psychoacoustic smoothing option that uses a frequency-dependent window and dip limiting, there is a Windows version and a Mac version. I don't recommend using the psychoacoustic smoothing for the EQ target match, variable smoothing will produce better results.


Dear John: You're a quick programmer! Probably a bit too quick as we haven't fully examined what Acourate is doing. There is no "dip limiting" in Acourate.... he solves this some other way.


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## AudiocRaver

Nice work, John.

*Bob:* In post 49, I believe 3ll3d00d meant “...blue is FDW THEN psychoacoustic...”

*Bob:* That is the plot I have been most interested in. Without dip limiting, how does Acourate manage deep dips? Perhaps we do not understand that yet.

*John:* The psychoacoustic in 3ll3d00d’s post 49 acts very differently from the psychoacoustic in REW 5.12beta3, In post 49, It follows the peaks and somehow rejects the dips,. in your 5.12beta3, it does not follow the peaks and does not rejected dips either. Is this the way you intended it to work?


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## 3ll3d00d

bobkatz said:


> Dear Senior Shackster (sorry I do not have your name): Thanks very much for that research!


Hi Bob, I'm Matt (Khan) from the acourate forum :wave:



bobkatz said:


> 1) "green is FDW alone". Did you get that trace via the Time domain menu option?


load Pulse48L to slot 1
run TD-Functions/Frequency Dependent Window with 15/15 15/15 as the parameter values, set output to slot 2



bobkatz said:


> 2) "blue is FDW the psychoacoustic". How did you get that display? What do you mean by that?


load Pulse48L to slot 1
complete the previous step
set slot 2 to active
run TD-Functions/Psychoacoustics (radio button set to psychoacoustic)
set output to slot 3



bobkatz said:


> 3) "brown is psychoacoustic alone". I believe that's the file named like: "Pulse48Lpsy.dbl" from macro 1. It's the first half of macro 1.


load Pulse48L to slot 1
run TD-Functions/Psychoacoustics (radio button set to psychoacoustic)
set output to slot 2

I've never looked at Pulse48Lpsy.dbl directly but I imagine it is the same thing yes



bobkatz said:


> 4) "teal is psychoacoustic then FDW". I believe that's really the file named like: "Pulse48Lmp.dbl" which you are not showing on your display (it would be the red trace if you checked the box), and this is the file which, to the best of my knowledge, Acourate uses to determine the official smoothed psychoacoustic response of the loudspeaker for DRC in the next macros. Please clarify your steps, thanks.


yes it is the same as Pulse48Lmp.dbl, you can compare by 

load Pulse48L to slot 1
run TD-Functions/Psychoacoustics (radio button set to psychoacoustic)
set output to slot 2
set slot 2 active
run TD-Functions/Frequency Dependent Window with 15/15 15/15 as the parameter values, set output to slot 3
load Pulse48Lmp.dbl to slot 4 
compare slot 3 and 4, they look the same to me



bobkatz said:


> There is no "dip limiting" in Acourate.... he solves this some other way.


I don't know how it is implemented but one side effect certainly seems to be that dips are cut off


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## JohnM

AudiocRaver said:


> The psychoacoustic in 3ll3d00d’s post 49 acts very differently from the psychoacoustic in REW 5.12beta3, In post 49, It follows the peaks and somehow rejects the dips,. in your 5.12beta3, it does not follow the peaks and does not rejected dips either. Is this the way you intended it to work?


Make sure the analysis option to allow 96 PPO log spacing is not selected, that beta only processes the psychoacoustic smoothing properly on linearly spaced measurement data (if the option was selected, deselect it then reapply the IR windows to regenerate the data). You should then find it looks very similar to the Acourate plot.



> There is no "dip limiting" in Acourate...


Just my shorthand for the psychoacoustics option.


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## AudiocRaver

Thank you, much better.


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## bobkatz

bobkatz said:


> I can do all that! By next weekend . Stand by for the fireworks!


Well, I didn't, and this thread moved on without me... let me follow it again, now. Aha... I see that I'm going to have to do some more careful comparison with the new beta. Thanks, Matt. I hope I can remember your name or I may revert to "Senior Shackster".


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## Niick

So, I've got a question regarding psychoacoustics in general. I'm wondering, when I read that the human hearing mechanism approaches a more anechoic state of perception as frequency increases, does this have to do with the other often referred to phenomenon that low frequencies are (to us) "omnidirectional", and that high frequencies are directional? Or are these two separate, unrelated phenomenon? Also, is there a chart or graph or something that shows the approximate time/frequency relationship of the psychoacoustic theory. 

I guess I just want to further understand this phenomenon. I'm wondering if this phenomenon is at all applicable in an average car interior, or is it too small, in other words, are the reflected sounds too close in arrival time to the direct sounds to be eligible for psychoacoustic discussion? 

Anybody?


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## subterFUSE

Gennelle51187 said:


> 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!



I have been considering the purchase of Smaart 7Di recently, but them ran across SysTune. Now I am torn as to which program might be better, because they are very expensive. I don't mind purchasing one of them, but buying both would be too cost prohibitive for me.

Smaart had the initial edge for me due to the Mac OS availability, which is my preferred platform.

But from your description, it sounds like SysTune might have an edge for the in-car environment due to the freq. dependent windowing?


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## Niick

Oh boy, that is a SERIOSLY tough question to answer, they are both EXTREMELY awesome software IMO. For me, as I've used bot Smaart v5, v6, v7, and SysTune (and now SysTune Pro), SysTune is a more useful program because I can use it for things that Smaart can't do. Like for instance, one of the things that I used to use my oscilloscope for, which is determining the relative acoustic polarity between drive units in factory systems, or any system where you can't actually see the speaker. With SysTune you can choose "time signal" as one of the display options, and it's quicker than routing my interface's output into the input of my oscilloscope. Also, SysTune has delay analysis and virtual EQ, and offline investigation of stored data, the advantages of which are too numerous to go in depth on right now, but suffice it to say that they allow me to SERIOUSLY cut down on installation and tuning time of higher end, fully active systems. Smaart has none of these things. Although, as long as you know how to use the software, either one will ultimately provide the exact same results. It's just that for me, since I work with sound systems every day, yes, SysTune has the upper hand. If I were going to use recommend either one for personal, infrequent use, I'd say Smaart for sure. SysTune has a STEEP learning curve. Smaart is much more streamlined, intuitive, user interface wins hands-down. Really depends how much you're into audio test and measurement and acoustic analysis, with SysTune, I've learned exponentially more about IRs and FFT in the first few months of ownership than in all the time combined with Smaart, REW, AudioTools for iOS, etc. With either one, it's truly an eye opening experience to be able to CONTINUALLY test a loudspeaker system and make changes and in real time see how those changes affect the resulting measurements/sound. With programs like REW, although free, and totally awesome, you have to take a sweep, make a change, take a sweep, make a change...............truly two very different experiences. Oh yeah, and a lot of what I mentioned as features of SysTune is actually SysTune Pro, which is more expensive than Di, it's just about the same (+\- & $50) as Smaart v7 full version. Also, a lot of guys are using SysTune on MAC, not natively of course, but in bootcamp I think.


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## subterFUSE

Sounds like SysTune is much better because of the real-time measuring tools.

I'm going to grab the demo and try it out.

How many mics are you using?


Sent from my iPhone using Tapatalk


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## 3ll3d00d

JohnM said:


> I've generated a beta build that has a psychoacoustic smoothing option that uses a frequency-dependent window and dip limiting, there is a Windows version and a Mac version. I don't recommend using the psychoacoustic smoothing for the EQ target match, variable smoothing will produce better results.


what's the difference between ERB (equivalent rectangular bandwidth I suppose?) and Psy smoothing? ERB is, visually, noticeably nicer than Psy as the latter has odd humps. It also seems a better option than Var as Var is too smooth at high frequencies. Psy looks a lot like one of the acourate functions tbh albeit one of the ones that, on its own, is not useful and seems to be designed to be used with another function. 

I would say ERB looks particularly good at dealing with HF noise, for example it really removes the cruft from a a gated speaker measurement without actually changing it's real shape at all. It's not as effective for subwoofer use though as it seems you lose a lot of (lower) frequency resolution, it seems more effective >5-600Hz to me.


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## Niick

subterFUSE said:


> Sounds like SysTune is much better because of the real-time measuring tools. I'm going to grab the demo and try it out. How many mics are you using? Sent from my iPhone using Tapatalk


 So, I use a combination of a 3 mic array, and a single mic. All of the mics are Dayton EMM6, one of which was calibrated by Cross Spectrum Labs. I also have a UMM6 from CSL that I use with Smaart Tools on my iPad. But as far as SysTune goes, the 3 mic array allows me to quickly gather 3,6,9,12,15 etc. different measurement points, this way I can quickly/easily determine what is mic placement related anomalies, and what is valid data that needs to be paid attention to. Also, with SysTune you can select all inputs, or any combination of inputs for real time spatial averaging. But for time/phase alignment thru crossover, I select the mic at the middle of the array usually. I used to use a 4 mic array, but it's size and unwieldy shape made it difficult to position, and I see no real disadvantage to 3 vs. 4, but the 3 mic array has definite advantages over the 4 because it's smaller, lighter, easier to position and move. You wouldn't think one mic would make much difference but it does. Especially the added width and weight of the mounting hardware. Keep in mind, that the exact same results could ultimately be realized with a single mic, many of the choices I make in my measurement setup are directly influenced by the fact that I use it to tune systems at work, and so, since I'm being paid to do so, efficiency is key. I don't want to waste customers time or money. So I'm always trying to find ways to make the process as efficient as possible, without sacrificing accuracy or the quality of the end result. 

Edit: Smaart is also fully real time.


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## subterFUSE

Oh. I took you post to say Smaart had to be swept, analyzed, reswept? While SysTune could be viewed and adjusted in real time?

I like the multiple mic array idea because special averaging in a pain in the ___. I hate doing it in REW because of the time it takes to sweep, move, sweep, repeat. Especially this time of year when I'm working in a hot garage.

What's the mic array look like?
Where do you buy one?

I have that same Dayton mic from Cross Specrum Labs. Might get a few more.

I have a miniDSP USB mic, too. But I can't use it for IR with REW because it's USB.


Sent from my iPhone using Tapatalk


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## JohnM

3ll3d00d said:


> what's the difference between ERB (equivalent rectangular bandwidth I suppose?) and Psy smoothing? ERB is, visually, noticeably nicer than Psy as the latter has odd humps. It also seems a better option than Var as Var is too smooth at high frequencies. Psy looks a lot like one of the acourate functions tbh albeit one of the ones that, on its own, is not useful and seems to be designed to be used with another function.
> 
> I would say ERB looks particularly good at dealing with HF noise, for example it really removes the cruft from a a gated speaker measurement without actually changing it's real shape at all. It's not as effective for subwoofer use though as it seems you lose a lot of (lower) frequency resolution, it seems more effective >5-600Hz to me.


ERB follows the profile of the ear's Equivalent Rectangular Bandwidth, which is pretty much 1/6th octave above 1 kHz and levels out to 25 Hz or so at low frequencies. I'm not sure it has much practical application though. The variable smoothing option is deliberately quite smoothed at HF (1/3rd octave above 10 kHz) to provide a good input for the automatic EQ. The psychoacoustic smoothing in the beta version is to emulate the response Bob provided, a 15 cycle frequency-dependent window combined with what looks like a form of dip limiting consisting of taking the greater of the response or the 1/3rd octave smoothed response. That won't make it into a release version, I have other ideas about what psychoacoustic smoothing should consist of and frequency-dependent windowing will be provided as part of the IR window options.


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## Niick

subterFUSE said:


> Oh. I took you post to say Smaart had to be swept, analyzed, reswept? While SysTune could be viewed and adjusted in real time? I like the multiple mic array idea because special averaging in a pain in the ___. I hate doing it in REW because of the time it takes to sweep, move, sweep, repeat. Especially this time of year when I'm working in a hot garage. What's the mic array look like? Where do you buy one? I have that same Dayton mic from Cross Specrum Labs. Might get a few more. I have a miniDSP USB mic, too. But I can't use it for IR with REW because it's USB. Sent from my iPhone using Tapatalk


Oh, ok gotcha. Yeah, I guess I should have been more clear, Smaart is just like SysTune in that respect. Totally real time.


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## Niick

JohnM said:


> . That won't make it into a release version, I have other ideas about what psychoacoustic smoothing should consist of and frequency-dependent windowing will be provided as part of the IR window options.


Right on John! That is absolutely awesome. I'm a big fan of frequency dependent windowing, not necessarily for psychoacoustic reasons, but still, I can't wait!


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## JohnM

New beta builds with frequency dependent window options in the IR windows dialog (defaults in Analysis preferences) and revised psychoacoustic smoothing are now available for Windows and Mac.


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## ajinfla

Thanks again John


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## jtalden

JohnM,
Thanks for this feature! While I don't have particular use for it in SPL analysis, It sure works great for phase analysis. See below.


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## Niick

From the SysTune manual, referring to their TFC (frequency dependent) window

"For each frequency data point there is a different window length, namely for each doubling of the frequency the window length is cut in half. For example, the window length at 1 kHz is twice as long as the length at 2 kHz but only half as long as at 500 Hz. The actual window for each frequency is a Tukey 50% window. " 

Can you give us a description of the time/frequency relationship used in the beta version you mentioned. I haven't been able to test it out yet, but I will real soon.


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## JohnM

The window width varies continuously with frequency, width is per the user selection (e.g. 15 cycles is 15 ms at 1 kHz, 1.5 ms at 10 kHz, 150 ms at 100 Hz etc). The window shape is Gaussian.

Edit: Here is the relevant part of the help file: 

_In addition to the left and right windows a frequency-dependent Gaussian window can be applied. This is a window whose width varies inversely with frequency, getting progressively narrower as frequency increases. The width of the window can be specified as a number of cycles or an octave fraction. If the width is in cycles then the width (between the half amplitude points of the window) at any frequency will be the number of cycles times the period of that frequency, for example a 15 cycle window will have a width at 1 kHz of 15 * (1/1000) = 0.015 s or 15 ms. The corresponding octave fraction has an effect similar to applying a smoothing of the same octave fraction, except the variable window excludes progressively more of the late arriving sound as frequency increases rather than just averaging it out - this has similarities with the way the ear increasingly picks out the direct sound from the speaker at higher frequencies.

If *Add frequency dependent window* is selected the window is applied after first applying the selected left and right windows. The FDW is centred on the window reference time - for best results this should be at the peak of the impulse._


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## AudiocRaver

Wow, John, great work! Thanks so much.

I see that the Psychoacoustical smoothing must be applied to a curve that has full original data resolution, not one with 1/96th oct data (or averaged, or any amount of previous smoothing, etc.), as with the previous beta. Could we get a brief explanation of how that smoothing is done, please?


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## JohnM

AudiocRaver said:


> I see that the Psychoacoustical smoothing must be applied to a curve that has full original data resolution, not one with 1/96th oct data (or averaged, or any amount of previous smoothing, etc.), as with the previous beta. Could we get a brief explanation of how that smoothing is done, please?


Could you elaborate on that, please? Previous smoothing settings shouldn't make any difference. Log spaced data will give slightly different results as some averaging has already happened at HF to convert the data to log spaced. The PSY smoothing is 1/3 octave below 100 Hz, 1/6 octave above 1 kHz. It calculates a cubic mean (cube root of average of the cubed values) to place more emphasis on peaks.


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## AudiocRaver

JohnM said:


> Could you elaborate on that, please? Previous smoothing settings shouldn't make any difference. Log spaced data will give slightly different results as some averaging has already happened at HF to convert the data to log spaced. The PSY smoothing is 1/3 octave below 100 Hz, 1/6 octave above 1 kHz. It calculates a cubic mean (cube root of average of the cubed values) to place more emphasis on peaks.











Red plot is a regular measurement with Psychoacoustic smoothing added.

Purple plot is first plot with no smoothing, then click the Average the Responses button to make a "copy," (do it all the time, super convenient, realizing it contains 1/96th oct data and no phase), then Psychoacoustic smoothing added. Difference is small, as you say, but 1 dB above 10 kHz (still pretty small).


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## JohnM

That will be due to the lack of phase data I think, causing the measurement to be treated a little differently. I will take a look at it tonight. If you convert to log by Apply Windows with convert to log allowed you will find the results are much closer.


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## JohnM

I've fixed the handling of measurements without phase data for the next build.


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## AudiocRaver

Thanks, John. I know it is not what the "Average the Responses" button was designed for, but it is such a convenient "copy" function, I use it all the time.:bigsmile:


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## JohnM

Try this version.


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## AudiocRaver

Looks great, thank you so much.:T

A couple of picky things:

Ctrl-Alt-0 does not work any more to return to "unsmoothed" (toggling via the same key combo works)
Ctrl-Alt-Y does not work to apply Psych smoothing to a curve that results from the "Average the Responses" function, nor does the toggle of Ctrl-Alt-Y


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## JohnM

The Ctrl+Shift+0 shortcut often gets intercepted by Windows before reaching the application (see the help on keyboard shortcuts). Up to Win 8 the workaround in the help of reassigning the language bar hot keys fixed that, but it doesn't seem to fix it for Win 8.1.

I've fixed Ctrl+Shift+Y not working for measurements that do not have an impulse response, that will be in the next build.


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## AudiocRaver

Super. I never missed the C-S-0 before, just noticed it while checking out the new functions. Thanks again, John.


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## isabido

HI John! thanks thanks very thanks for FDW. 

If I want to apply the auto EQ on FWD smoothing as you would?

I have no option in the EQ section to apply.

Thank you and I hope you will understand me.


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## JohnM

Auto EQ is applied to the response with whatever windows you have selected, so if you apply FDW to the measurement (selected in the IR Windows dialog) that windowed response is is what auto EQ will attempt to match to the target. However, the EQ predicted trace in the beta version does not include any selected FDW after filters have been applied, so the measurement and prediction will not correspond very well. It is better not to use FDW with auto EQ in this version.


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## isabido

Thanks John. 

I look forward to applying FDW in autoEQ !.

I imagine you've read this great Denis Sobreign DRC-FIR documentation.

Maybe you can help yourself to inspriracion.

Uli (acuorate) and Denis (drc-fir) shared principles in their origins.


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## JohnM

Here are Windows and OS X beta versions that allow EQ target match with responses that have frequency-dependent windows applied. Note that FDW and smoothing can be combined, smoothing is recommended if applying EQ above a couple of hundred Hz.


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## isabido

Amazing John, you can not do faster.

Curiously, I have normally apply a manual filter (HS12 -4db aprox) to follow the defined target

http://i.imgur.com/zVJSkeU.png

I take this opportunity to give me your opinion about the parameters that apply to the AutoEQ.


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## JohnM

There are lots of very wide filters because your target level should really be about 6 dB higher - most of the filters are being used to achieve a sort of volume control above 200 Hz or so. If there is a sub then better to increase its volume setting so you don't have to pull everything else down to allow the LF to be higher.


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## isabido

Thanks for the suggestion John, the target is automatically adjusted, I will try 6dB up manually.

No sub, it's just a YAMAHA HS7, if I apply this target, The sound becomes anemic.

http://i.imgur.com/NPtNQn4.png


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## MaximalC

JohnM said:


> The PSY smoothing is 1/3 octave below 100 Hz, 1/6 octave above 1 kHz. It calculates a cubic mean (cube root of average of the cubed values) to place more emphasis on peaks.


Thanks for the new update, John, especially for the long awaited FDW. I have a question regarding the change to variable smoothing behavior, however. I believe the previous incarnation of variable smoothing decreased the octave resolution as frequency increased, however this version does the opposite, with only 1/3 octave resolution for the lowest frequencies (which were previously not smoothed at all). What was the reason for the change? Or am I misunderstanding how one of the two was employed? It just seemed like a pretty big departure to me and I was curious as to what makes the new implementation superior.


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## JohnM

Variable smoothing hasn't changed. Psychoacoustic smoothing is new and is to give more of an indication of how the response would be perceived. It shouldn't be used on responses that are used for EQ target match.


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## MaximalC

JohnM said:


> Variable smoothing hasn't changed. Psychoacoustic smoothing is new and is to give more of an indication of how the response would be perceived. It shouldn't be used on responses that are used for EQ target match.


Thank you for the clarification.


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## AudiocRaver

Thanks for all the changes, John, really nice.

Old news, I'm sure, but I just noticed that the RTA FFT Length and the Pink PN Sequence Length now track automatically, giving an accurate, alias-free FFT measurement. Nice touch.


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## Mitchco

Hi John, I finally got around to checking this out. 

Here is an REW measure of my right speaker at the listening position. I wanted to see how close the measured response matches the target response. I applied the new psychoacoustic smoothing to the measurement and overlaid my preferred target response. It matches almost perfectly:











Measurement signal path is REW digital output routed to JRiver ASIO (digital) line input using Lynx Hilo internal mixer, through JRiver’s 64 bit Convolution engine, hosting FIR filters containing 3 way digital XO, time alignment, amplitude and excess phase correction, out to 6 channels of Hilo DAC to 6 amps and speakers, right speaker to mic at listening positon, mic preamp, to ADC Hilo converter to REW digital input. 

It is amazing to me the level of precision between the target and measured response given the number of transfer functions in the signal path  With this functionality, I can take multiple measures of my right speaker, like I did around this 6' x 2' grid area, where my couch is, to see what the response looks like across a large sweet spot:










Thanks again!

Cheers, Mitch


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## natehansen66

Mitch, is this just with the Pys smoothing or are you using the freq dependent window as well?


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## Mitchco

Nate, that was with Psy smoothing only. My mistake, I should have FDW'd as well, which I have here:










I messed up somewhere in the process as these measures are from 1 1/2 years ago...

I am changing one of the XO points in my system which will give me an opportunity to re-measure/compare again. Will post the results.


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## natehansen66

Cool thanks


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## Mitchco

After investigation, the woofers in my 3-way time aligned system were offset by a few samples relative to the mids and highs. Fixing that plus linearizing the drivers led to a better response:


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## natehansen66

What's the window length and how many cycles for the fdw?


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## Mitchco

Default values used for both window length and FDW.


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## natehansen66

Does anyone know of any literature that goes into our perception of sound and these "sliding" windows we're using? It makes sense to me, but whats optimum? Or to put that a different way what window length vs. frequency best corresponds with the way we perceive sound in a room? REW seems to default to 15 cycles, is there anything behind that number to explain the choice there?

What's interesting is in my setup the floor suckout is at 400hz. With the right window at 500ms, and the FDW at 15 cycles the suckout is visible in the FR on the left speaker but not the right. Change the FDW to 5 cycles and the suckout is apparent in both speakers. Obviously there's other reflections "filling" the floor bounce with the longer FDW window for the right speaker but not the left(lots of asymmetry in this room), but what is correct here?

I have links to a few AES papers somewhere, but before I go and spend $20 a piece on several papers I'm wondering if anyone has any suggestions? I'll see if I can find those links....


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## 3ll3d00d

There is some commentary on this in http://www.amazon.co.uk/Auditory-Neuroscience-Making-Sense-Sound/dp/0262518023 which starts with talking about how the basilar membrane works, how it can be modelled as a set of gamma tone filter banks and how those filters have progressively shorter impulse responses as frequency rises. I am not aware of any aes papers on the subject (though no doubt they exist).


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## natehansen66

Thanks for the link. Now that you mention it I seem to recall Geddes posted a chart showing gamma-tone impulses and their duration not too long ago.


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## 3ll3d00d

natehansen66 said:


> Thanks for the link. Now that you mention it I seem to recall Geddes posted a chart showing gamma-tone impulses and their duration not too long ago.


it's something like this 

http://www.mathworks.com/matlabcent.../submissions/32212/versions/18/screenshot.jpg

which is, AIUI, basically like what FDW does

as to the choice of window length (cycles), 15 is the default in acourate IIRC for magnitude correction but 6 is commonly recommended as a good starting point for the excess phase correction. This is an example of how it depends on you're trying to do. I've experimenting with a variable cycle length to try to get a view into some LF issues, I find that playing around with it just gives you more data to try and understand what is going on. There was a suggestion once around using decay times to guide your choice of cycle count but that's the only objective suggestion I've seen.


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## natehansen66

I was just about to post that same link 

Geddes said he calculated the gammatone filter at 70hz to be about 60ms and 50ms at 100hz. If we were to go by these numbers then it seems a variable fdw would be necessary. This is basically what I've done in the past...........just use different windows for different freq ranges. The nice thing about the fdw is I can fairly easily do low crossovers (80-300hz) with measurements taken at the lp. Due to our perception at these lower freqs the resolution is "close enough".

Here's the papers that I was talking about, cited by speakerdave aka David Smith:

A New Psychoacoustically More Correct Way of Measuring Loudspeaker Frequency Responses
Preprints 1871 (F-4) and 1963 (G-4)
Jorma Salmi

A Perceptual Criterion for Loudspeaker Evaluation
James M. Kates
JAES Vol. 32, No. 12, Dec 1984

and: Samuel Bridges
Effect of Direct Sound on Perceived Frequency Response of a Sound System
AES preprint 1644 (H-6)

I haven't checked these out for myself yet but I'll probably pick up the Salmi paper.


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## 3ll3d00d

natehansen66 said:


> Geddes said he calculated the gammatone filter at 70hz to be about 60ms and 50ms at 100hz. If we were to go by these numbers then it seems a variable fdw would be necessary.


Thanks for the links. 

FWIW acourate has a variable fdw, you can set the left and right and low and high cycles independently. IIRC low is something like 20Hz and high 20kHz, it then interpolates between the two points.


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## AudiocRaver

natehansen66 said:


> ...what window length vs. frequency best corresponds with the way we perceive sound in a room? REW seems to default to 15 cycles, is there anything behind that number to explain the choice there...


It is not my intention to start an argument, just food for thought...

I am thoroughly convinced that, although the "norms" given from the research are certainly "normal" and accurate and therefore useful in understanding a baseline, that _some_ of these perceptual patterns are subject to exposure and experience, and can be trained to discriminate with finer resolution well into the areas where they are thought not to be able to. This does no mean that such an individual is exceptional or weird or is "wired funny," only that he has _paid attention_ and learned to hear things outside the norm. FWIW.


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## natehansen66

AudiocRaver said:


> It is not my intention to start an argument, just food for thought...
> 
> I am thoroughly convinced that, although the "norms" given from the research are certainly "normal" and accurate and therefore useful in understanding a baseline, that some of these perceptual patterns are subject to exposure and experience, and can be trained to discriminate with finer resolution well into the areas where they are thought not to be able to. This does no mean that such an individual is exceptional or weird or is "wired funny," only that he has paid attention and learned to hear things outside the norm. FWIW.


It's clear to me that highly trained listeners can "hear" things that the average Joe can't, but I'm not so sure that's outside of the current understanding of how we hear. There seems to be a lot of grey area in psychoacoustics and it seems much of that is interpreted differently by many. I seek to find the interpretation that correlates objective data with my subjective impression......when these things jive I feel I'm on the right track.


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## AudiocRaver

natehansen66 said:


> I seek to find the interpretation that correlates objective data with my subjective impression......when these things jive I feel I'm on the right track.


Well said.


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## Mark Fuller

It would be great if the RTA could run in 31 band mode for use with that common graphic equalizer design.


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## lkwrnflknf4lk

I think FDWs only make sense after exporting minimum phase versions of the meassurements. any delayed frequency in the original meassurement will obviously be (partly) missing in the plot.
here is the same meassurement, 15 cycles, no smoothing: green is original, red is minimumphase export, re-imported


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## lkwrnflknf4lk

3ll3d00d said:


> Some measurements of my L speaker (which is a fairly small sealed speaker, MK MP150). Mic is at the usual MLP about 3m away, mic pre is an RME FireFace 800, mic is a CSL calibrated EMM6, same levels used for both sweeps & same length sweeps used (approximately... 512k in REW is 10.9s and it was a 10s long sweep in Acourate) & same freq range (10-24000Hz).
> 
> Two wavs attached, one is the acourate sweep itself (in 24 bit PCM form for import into REW) and the other is the same after a 15 cycle FDW applied.
> 
> Comparison of REW variable smoothing and acourate 15 cycle FDW below
> 
> View attachment 94442


Here is MP exported reimported 15 cycles


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## lkwrnflknf4lk

bobkatz said:


> 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


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