Anyone try the QSC HPR152i waveguide w/Radian 475 Be CD? - Page 2 - AVS Forum | Home Theater Discussions And Reviews
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post #31 of 41 Old 04-20-2017, 12:50 PM
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Ryan, dd you see any disadvantage using the larger format CD and EQ to the top to make it flat? Possibly issues off axis? Did the Be extend higher without EQ?
The Be did not extend higher, it played cleaner at higher frequencies. I passively shaped the response to be close to flat from 300hz and up. Maybe a 2db down tilt. The iwatas are quite directional though. So the overall sound was very laid back yet maintained a lot of detail. Difficult to pull off, very difficult to pull off with a tweeter that played down to 600hz. Therefore I did not find it to be an issue to eq the top flat. The Ti phragms were ok but not nearly as good. Slightly grating and just not to the same fidelity as the Be.

I have frequency response measurements somewhere. I didnt bother to post polars and CSD and all that. I keep that stuff mostly to myself now.

Ryan
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post #32 of 41 Old 04-21-2017, 01:31 AM
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A few notes:

Typically, Be drivers show less output at the top because of the lack of break-up to artificially boost it.

CSD is essentially derived directly from un-smoothed frequency response data.
FWIW, I believe "flat enough" for best sound may be practically impossible to achieve passively. I can't emphasize enough how much of a change even 0.25 dB is if it's over a broad enough range (like 1/3rd octave or more). In my latest EQ config, I've been tweaking in 0.05 dB adjustments.

I would love to hear the benefits of Be, but unless I hear that Be driver in an optimized active system, it won't stand a chance, IMO. Now I'm sure everyone thinks I'm crazy. I'll point out that the extremely flat frequency response is the likely reason the JBL M2 speaker is so widely renowned, despite its lack of Be CD.

Last edited by awediophile; 04-21-2017 at 02:28 AM. Reason: fix paragraphs
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post #33 of 41 Old 04-21-2017, 06:57 AM
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Originally Posted by awediophile View Post
A few notes:

Typically, Be drivers show less output at the top because of the lack of break-up to artificially boost it.

CSD is essentially derived directly from un-smoothed frequency response data.
FWIW, I believe "flat enough" for best sound may be practically impossible to achieve passively. I can't emphasize enough how much of a change even 0.25 dB is if it's over a broad enough range (like 1/3rd octave or more). In my latest EQ config, I've been tweaking in 0.05 dB adjustments.

I would love to hear the benefits of Be, but unless I hear that Be driver in an optimized active system, it won't stand a chance, IMO. Now I'm sure everyone thinks I'm crazy. I'll point out that the extremely flat frequency response is the likely reason the JBL M2 speaker is so widely renowned, despite its lack of Be CD.
Unless Im not understanding you quite right, that isnt how CSD works.

Where are you getting the idea Be has less top end output? That wasnt my experience except at the break up peaks. My experience basically found the Be had more output yet sounded quiter due to less distortion, which was a more pleasing sound.

If you are tuning at 0.05db increments, you are suffering from placebo that you can hear that. I agree that 0.25db is audible, for instance across the entire treble range. But not 0.05db. And especially not in narrow bandwidths.

The OP plans to use active. So as Im sure you are very proud of your setup, you arent making sense. Active is nice and all, but you seem to think it fixes everything. Eq'ing break up helps, and Ive agreed with you on that point up until now, but it doesnt eliminate it. Be goes a long ways to pushing break up much higher. On a 1.75" phragm I would not doubt its past 20khz. So ya, I do think it "stands a chance" against your speaker.

Ryan
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post #34 of 41 Old 04-21-2017, 05:18 PM
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Unless Im not understanding you quite right, that isnt how CSD works.
Assuming you have unsmoothed frequency magnitude + phase response data, the CSD and any other analysis of *linear* characteristics can be derived from that data. Furthermore, transducers are almost always minimum phase, which means that the phase response can be derived from the magnitude response data alone.

That transducers are usually minimum phase is how I first discovered that break-up can be non-linear. I was seeing apparent non-minimum phase phenomena in my impulse response measurements, which led me on a crazy chase to try to figure out which piece of digital electronic equipment was introducing this flaw. I failed to find anything wrong. Then I tried running sweeps with different FFT lengths, and low and behold, the frequency response changed substantially at the problem area. The non-linearity was introducing error into the sine sweep measurement that gave the appearance of non-minimum phase behavior where there wasn't any.

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Originally Posted by tuxedocivic View Post
Where are you getting the idea Be has less top end output? That wasnt my experience except at the break up peaks. My experience basically found the Be had more output yet sounded quiter due to less distortion, which was a more pleasing sound.
Sorry. I misunderstood. Although I also vaguely recall seeing a comparison of Be vs. non-Be Radian measurements in which the Be measurements had less top end output too. My point is that this isn't really a flaw when it arises due to a lack of break-up.

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If you are tuning at 0.05db increments, you are suffering from placebo that you can hear that. I agree that 0.25db is audible, for instance across the entire treble range. But not 0.05db. And especially not in narrow bandwidths.
I'm quite certain this is not placebo. I am able to switch EQ changes in a fraction of a second while sitting and listening at the MLP to avoid loss of audio memory. The change with increments of 0.05 dB are subtle. Changes of 0.1 dB are not.

Mind you, this is with a response that's already very flat. My first run configuration that I settled on about 6 months ago used measurements from a single location, the MLP. However, my SEOS-15 horns have relative off-axis variations on the order of 1 dB, so I was likely within ~1 dB of ideal. I recently revisited this and discovered there was a lot of performance gain still to be had. I'm now relying on measurements on-axis and at the MLP (about 30 deg). I suspect I'm within 0.25 dB or so of ideal now. Next time I'll probably try to use even more measurements and a more sophisticated analysis do determine what the speaker is doing in the room for an even tighter calibration.

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The OP plans to use active. So as Im sure you are very proud of your setup, you arent making sense. Active is nice and all, but you seem to think it fixes everything. Eq'ing break up helps, and Ive agreed with you on that point up until now, but it doesnt eliminate it. Be goes a long ways to pushing break up much higher. On a 1.75" phragm I would not doubt its past 20khz. So ya, I do think it "stands a chance" against your speaker.
Active doesn't fix anything on its own. It's only a prerequisite for optimal sound quality. Getting the response shape right is the hard part. As I've recently learned, uou can't just shape for flat response at the listening position, either. Though this will get you within 1 dB or so with an SEOS horn.

And I still agree that EQ cannot eliminate break-up. However, in some circumstances it can mitigate the problem enough that other considerations take precedence.

What are the negative consequences of breakup? They are 1) resonance peak(s) and/or dips; 2) potentially non-linear response; and 3) polar response aberrations. Consequence #1 is fixable with EQ but only if consequence #2 does not arise during operation and only if the EQ takes into account response at multiple locations, which typically differs substantially due to #3. This is why I argued that both the linearity and polar response measurements are so critical. Even in the best case, the EQ won't fix the polar response variation, so of course there's room for improvement using Be. I'm just arguing that it won't be noticed if bigger issues remain uncorrected.

Consider that almost all drivers exhibit resonances other than those from break-up, and these also sound bad if not EQed. Consider too that throat mismatch, beaming, and horn diffraction can cause unwanted polar response variation. And as noted above, the SEOS horn is not perfect. It (my SEOS-15 at least) exhibits ~1 dB relative variations. Most other horns are probably worse. The major challenges when optimizing high frequency EQ are: (1) Find and eliminate all resonances without creating new ones by erroneously correcting peaks and dips that arise from diffraction or other phenomena that are spatially varying. (2) Shape the broader response to be as flat as possible within the listening window, ideally within 0.1 dB on average, excepting a top-end roll-off consistent with air absorption over some distance.

I really shouldn't be surprised that you question my claims here. I certainly would knowing what I knew before I went down this path. I've reached my conclusions after many many hours of iterative EQ adjustment, listening, measurement, and analysis. It's taken me a long time even to figure out how to analyze an impulse response measurement to determine what's audibly relevant, and I'm still learning.
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post #35 of 41 Old 04-21-2017, 06:38 PM - Thread Starter
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I'm not sure if this helps....

Truextent has some testing done using the 4" Be diaphragm vs the aluminum one from the Radian 950. They posted the following waterfall showing no hash and the freq response showing the break-up on the 4" is over 20k which supports what Ryan heard which was that very diaphragm in a JBL 2446, a very clean presentation. I think anyone looking to explore Be benefits should first listen to one in a system. I think it clarifies the 'why' and 'worth'. I agree measured performance is king but in this case, impulse response, waterfall, and break-up all support a clean and detailed presentation. I can describe the Mona Lisa to you using all sorts of testing descriptions and measurements but something would be missing if you never saw it in person.



https://materion.com/~/media/Files/P..._Vs_Radian.pdf

I tried finding a water fall plot for the 4550 and B&C 250 but no luck.
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post #36 of 41 Old 04-21-2017, 06:56 PM - Thread Starter
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Quote:
Originally Posted by awediophile View Post

I would love to hear the benefits of Be, but unless I hear that Be driver in an optimized active system, it won't stand a chance, IMO.
You are starting the investigation with too strong an anti-Be position which is too far from objective for me. You dismiss break-up location, lack of stored energy, and reduced mass as contributing to improved transient response with no data to support it other than an opinion. I think if you were objective and did tests first and draw one of three conclusions it would lend itself to discovery. For example, you start from a belief that the break-up is not higher in frequency and you dismiss the break-up on the dna-360 as benign due to dsp adjustment. That's not objective and you are also setting the stage to dismiss the benefit should Be be found to actually have a higher than 20k break-up by saying your's is near perfect with dsp.

Try getting to the first one, (I can respect the second one):

1-There is no benefit to Be supported by testing
2-There is a benefit but it is not worth it cost wise
3-The benefits are there and I'm switching.

I know near nothing compared to Ryan but I know this Radian sounds far better than the 4550 CD sounded with the best I could coax from it. Sold. Money is irrelevant to me.

Thank you all for sharing/posting
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post #37 of 41 Old 04-22-2017, 05:23 AM
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Quote:
Originally Posted by SpinMonster View Post
You are starting the investigation with too strong an anti-Be position which is too far from objective for me. You dismiss break-up location, lack of stored energy, and reduced mass as contributing to improved transient response with no data to support it other than an opinion. I think if you were objective and did tests first and draw one of three conclusions it would lend itself to discovery. For example, you start from a belief that the break-up is not higher in frequency and you dismiss the break-up on the dna-360 as benign due to dsp adjustment. That's not objective and you are also setting the stage to dismiss the benefit should Be be found to actually have a higher than 20k break-up by saying your's is near perfect with dsp.
Break-up locations certainly matters. Break-up at 17 kHz is way less audible than break-up at 10 kHz. Breakup above 20 kHz is better than break-up at 17 kHz. However, break-up severity matters too. If break-up is severe enough, it can have audible consequences even above 20 kHz. I've also heard this among the thousands of different EQ configurations I've cycled through. It sounds very weird because you don't hear the ultrasound, you just hear the consequences of it masking the rest of the content. It can mask all the way down to the deep bass. Crazy stuff.

The other concerns you named and that I highlighted are actually addressed indirectly by frequency response. In audio, people talk about frequency response vs. transient response as if they were two distinct things, but as long as your system is behaving linearly, they are related via a mathematical transform. Impulse (transient) response and frequency response are just two different ways of looking at the same information. Likewise, a waterfall or CSD is just another mathematical transform involving this same information.

What this means is that if you EQ frequency response to be extended and flat, you improve the transient response and CSD too. I don't need any evidence for this. It is a mathematical fact as long as the system is behaving linearly. Note, I keep using the word linear. This is very important and is why I did my linearity tests. If the system is not behaving linearly, then the math doesn't work. Indeed, a sine sweep measurement that pushes a driver into significant non-linearity will yield a bad result.

The other issue, of course is the aberration in polar response. Essentially, you will measure a different frequency response and CSD at each location. So if you correct the frequency response at one location, it won't correct it at the others. However, when you correct a narrow resonant peak as in break-up, you tend to create notches off-axis. Those notches are not nearly as audible as the peak was, especially if your room boundaries do a good job of dispersing high frequency energy throughout the room. (Diffusers for the win here.)

You can't completely cure break-up with DSP, but with careful analysis and skill, you can turn it into a very minor problem. In fact, after EQ my 17 kHz break-up peak is just a region that beams a bit more narrowly than the surrounding frequencies. That's it, so long as I don't ever push the break-up into non-linearity. The funny thing is that if you use a dome tweeter or a CD with a bigger exit, you will probably see a lot more beaming in the UHF even if it uses Be.

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Originally Posted by SpinMonster View Post
Try getting to the first one, (I can respect the second one):

1-There is no benefit to Be supported by testing
2-There is a benefit but it is not worth it cost wise
3-The benefits are there and I'm switching.

I know near nothing compared to Ryan but I know this Radian sounds far better than the 4550 CD sounded with the best I could coax from it. Sold. Money is irrelevant to me.

Thank you all for sharing/posting
Of course there is benefit to Be, but how much? With speaker design as with most engineering tasks, there are *always* trade-offs. You forgot option 4:
The benefits are there, but I won't switch unless (1) It works with a 950 Hz XO without having to move to a larger model that beams more (and breaks-up lower, as size tends to do); (2) It matches the throat well so as not to create worse dispersion characteristics than the EQed DNA-360 break-up does, and (3) The resonances in the other parts (surround and whatnot) do not exhibit any non-linearity of more severe dispersion problems than my EQed DNA-360 break-up does.
I have no bias against Be. I just insist on taking the whole package into consideration. (Why do I have to keep repeating myself here?) If the above criteria are not met, then I have no interest in switching. Only measurements of the kind I linked to can verify that these criteria are met. From the data I've seen, the 1" exit Radian drivers don't handle 950 Hz in SEOS well, and they exhibit more pronounced peaks and dips at the top end than comparable drivers. Are these peaks and dips due to resonance of other parts like the surround? Or are they there because of throat mismatch?

I do realize that other design approaches will lead to considerations. I know of one person who used passive components to tame the break-up in the BMS 4550, so it is do-able. However, this may be too expensive or too complicated in the design. Or maybe it's an active design but the EQ capabilities are limited or its not possible to do the measurements required to fix this problem using EQ.

Nevertheless, It is my impression that *you are biased* in favor of Be. You have made as much clear in that you refuse to consider anything other than a Be driver. I sincerely hope the Radian 475Be works well in your design. But you should ask yourself: "How much am I willing to give up in order to enjoy the benefits of using Be?" If the driver has other shortcomings or it just doesn't work well in the particular design, are you willing to change the design? Give up on Be? Or accept the other shortcomings?

****

Anyway, I am still posting here because the perfectionist in me is still interested in Be. I am very curious to see for myself (in measurements) what the Radian 475 Be does in an SEOS-15 horn. If I could be convinced that it satisfies my above criteria, then I would most certainly switch. From the anecdotal data I've seen so far though, I kind of doubt it.
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post #38 of 41 Old 04-22-2017, 07:38 AM
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Originally Posted by awediophile View Post
Assuming you have unsmoothed frequency magnitude + phase response data, the CSD and any other analysis of *linear* characteristics can be derived from that data. Furthermore, transducers are almost always minimum phase, which means that the phase response can be derived from the magnitude response data alone.

That transducers are usually minimum phase is how I first discovered that break-up can be non-linear. I was seeing apparent non-minimum phase phenomena in my impulse response measurements, which led me on a crazy chase to try to figure out which piece of digital electronic equipment was introducing this flaw. I failed to find anything wrong. Then I tried running sweeps with different FFT lengths, and low and behold, the frequency response changed substantially at the problem area. The non-linearity was introducing error into the sine sweep measurement that gave the appearance of non-minimum phase behavior where there wasn't any.
Assuming you have the frequency response you have the impulse response. But why talk about CSD as a function of FR when it's actually a function of IR. CSD is a series of slices from the impulse response plotted in a 3 axis graph. Which is why I said that's not how CSD works. You aren't wrong that it's all related, but it's kind of like saying an apple is basically a part of a tree. Well, yes it is but it's a piece of fruit. Which is why your original statement didn't make sense to me. Making FR dead flat does not make CSD razor sharp.

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I'm quite certain this is not placebo. I am able to switch EQ changes in a fraction of a second while sitting and listening at the MLP to avoid loss of audio memory. The change with increments of 0.05 dB are subtle. Changes of 0.1 dB are not.

Mind you, this is with a response that's already very flat. My first run configuration that I settled on about 6 months ago used measurements from a single location, the MLP. However, my SEOS-15 horns have relative off-axis variations on the order of 1 dB, so I was likely within ~1 dB of ideal. I recently revisited this and discovered there was a lot of performance gain still to be had. I'm now relying on measurements on-axis and at the MLP (about 30 deg). I suspect I'm within 0.25 dB or so of ideal now. Next time I'll probably try to use even more measurements and a more sophisticated analysis do determine what the speaker is doing in the room for an even tighter calibration.
It still could be placebo. After all, you know what change you've made, therefore you have a preconception that such adjustment should make such a difference. Have a friend make the changes, tell you he changed it without, etc. If you can still get it right then I'm impressed. 0.1db can be audible to me over a range of an entire decade. Very subtle though, and passive can easily do that. When you are talking about 0.1db difference only active can handle I get suspect

Also, in my experience, and based on research, you are far better off getting your response right using anechoic measurements. You are talking about 0.05db changes based on a room measurements where changing you head position 6" can have a bigger impact. Heck, sometime I leave the mic in the same spot and two back to back sweeps have 0.25db difference, even in a dead silent room. You're talking about changes the measurement setup really can't even quantify right.

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I really shouldn't be surprised that you question my claims here. I certainly would knowing what I knew before I went down this path. I've reached my conclusions after many many hours of iterative EQ adjustment, listening, measurement, and analysis. It's taken me a long time even to figure out how to analyze an impulse response measurement to determine what's audibly relevant, and I'm still learning.
I question your claims because I've been there and did not find it to be audible. You aren't more enlightened based on your hours obsessing over a SEOS 15 design. Like I said, 0.25db over a decade is obvious. But you sound like you're talking about pretty narrow band stuff (breakup). Heck, research doesn't support your claims. And I haven't even mentioned the aberrations in the recording that you can't compensate for.

Sorry if I'm nit picking. I just called out someone in another thread for doing that and here I am doing it. But if you are going to claim Be is a waste of time until you've eq'ed your speaker to 0.05db then we should discuss it. I found Be to be well worth the money. It doesn't change the low end capability so you should reconsider that point of your argument. The Radian 475 may have less low end than your CD choice, but that separate from Be. Which is why I actually backed you up earlier, because that was my concern going with the Radian, despite me wanting Be.

Ryan
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post #39 of 41 Old 04-22-2017, 05:40 PM - Thread Starter
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Is the TAD TD2001 low end extension limited?
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Assuming you have the frequency response you have the impulse response. But why talk about CSD as a function of FR when it's actually a function of IR. CSD is a series of slices from the impulse response plotted in a 3 axis graph. Which is why I said that's not how CSD works. You aren't wrong that it's all related, but it's kind of like saying an apple is basically a part of a tree. Well, yes it is but it's a piece of fruit. Which is why your original statement didn't make sense to me. Making FR dead flat does not make CSD razor sharp.
Slicing the impulse response is a convenient method for generating CSD but methods that act directly on frequency response are possible too. Impulse response and frequency response are just two different ways of representing the same thing. A tree and an apple is the wrong analogy. The actual "thing" is a complete characterization of the behavior of a single-input, single-output linear dynamic system. The thing can be represented as frequency response, impulse response, or one of many time-frequency representations. However, some time-frequency transformations discard some information and cannot be reversed. Likewise, frequency response smoothing discards information. You cannot construct a CSD from smoothed frequency response data.

If you make un-smoothed frequency response flat with zero phase, then you will make the CSD pristine. If the response is minimum phase, as is typically the case with a raw driver response, then any minimum phase EQ you use to make frequency magnitude response flat will also zero the phase. The ringing from break-up can be completely eliminated at a single spatial location (or far-field angle) this way. The trouble with break-up is that (1) it can behave non-linear; and (2) you'll get a different frequency response and CSD at different angles, and you can only eliminate it at one location or reduce its impact "on average", over multiple spatial angles. The question of how to weight the data is interesting, important, and not clearly answered. In a listening room with early reflections, a listener will hear sound produced from the speaker at different angles, so an averaging approach may be better even if the application calls for optimization for a single seat only.

Anyway, I like looking at frequency data more than impulse response data. Frequency data can be presented on a logarithmic axis, which is the perceptually relevant scale to use. This can't be done in a meaningful way with the impulse response, and information from the highest frequencies dominates the visual presentation of the data. When looking at frequency data, the narrower the peak or dip, the greater the time required for it to manifest. Tall narrow minimum phase peaks are high Q and ring a lot. Knock down those peaks down with some EQ and the ringing will be reduced too. What the minimum phase EQ can't correct is the reflected sounds, unless they are indistinguishable from the direct sound in the frequency range of interest.

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It still could be placebo. After all, you know what change you've made, therefore you have a preconception that such adjustment should make such a difference. Have a friend make the changes, tell you he changed it without, etc. If you can still get it right then I'm impressed. 0.1db can be audible to me over a range of an entire decade. Very subtle though, and passive can easily do that. When you are talking about 0.1db difference only active can handle I get suspect

Also, in my experience, and based on research, you are far better off getting your response right using anechoic measurements. You are talking about 0.05db changes based on a room measurements where changing you head position 6" can have a bigger impact. Heck, sometime I leave the mic in the same spot and two back to back sweeps have 0.25db difference, even in a dead silent room. You're talking about changes the measurement setup really can't even quantify right.
Hmm it could be placebo, but probably not. Because a lot of times I change something and I don't hear it. Indeed, not all 0.1 dB changes are equally audible. I believe it has to do with masking thresholds. The change is likely to be audible if some content suddenly becomes masked or unmasked as a consequence. Now, if the thing you are changing is far out of balance, than small changes won't really be audible. If you are boosting or attenuating something that's completely masked or something that's completely masking something else, then you don't hear any change.

There are undoubtedly advantages to using anechoic measurements to do calibration. But I am doing it the "hard" way on purpose, because I am very interested in the subject of room calibration. I also believe that even in the high frequencies, an in-room calibration may be able to give superior results to an anechoic one, even though existing room EQ technology hasn't really gotten this right yet. The challenge is to figure out a good psychoacoustic model for analyzing impulse response and to design a system around it that is flexible enough to meet the needs of both single seat and multi-seat applications

My approach is based on modeling a high quality anechoic speaker in what I imagine a high quality mastering room to be like. I seem to be able to get results that work great for most music, but movies are a total mess. The more recent home mixes these days are decent, but I end up basically re-EQing stuff that is likely either theatrical original or just poorly re-mixed/re-mastered home material. It's possible that I'm more tolerant of variation in music just because I accept that some tonal imbalance is introduced on purpose for style.

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Originally Posted by tuxedocivic View Post
I question your claims because I've been there and did not find it to be audible. You aren't more enlightened based on your hours obsessing over a SEOS 15 design. Like I said, 0.25db over a decade is obvious. But you sound like you're talking about pretty narrow band stuff (breakup). Heck, research doesn't support your claims. And I haven't even mentioned the aberrations in the recording that you can't compensate for.

Sorry if I'm nit picking. I just called out someone in another thread for doing that and here I am doing it. But if you are going to claim Be is a waste of time until you've eq'ed your speaker to 0.05db then we should discuss it. I found Be to be well worth the money. It doesn't change the low end capability so you should reconsider that point of your argument. The Radian 475 may have less low end than your CD choice, but that separate from Be. Which is why I actually backed you up earlier, because that was my concern going with the Radian, despite me wanting Be.
I'm not sweating 0.05 dB for anything narrower than 1/3rd octave. For bands as wide as 1/3rd octave or wider, I definitely do perceive very small changes, particularly in the vicinity of masking thresholds.

Your point about variation between content is interesting. I absolutely do have to listen to a wide variety of content to subjectively evaluate my configurations. However, the closer I get to an ideal response, the better everything sounds on average. I have a few thoughts on why this is true. Note that these apply to music more to movies:

1) The negative effects of flaws in the production vs. playback system are largely additive. If your system has imbalances like most systems do, you may occasionally find content that was produced in a very similar environment that will sound better on your system than it would on an ideally neutral system, but this is a lot less likely than the converse.

2) Skilled mix and master engineers are aware of the flaws and shortcomings of their systems and are able to compensate for them. These flaws do impact the ability of an engineer to make adjustments to the sound that may be needed, but the good engineers know when to doubt what they hear and avoid doing harm to the sound.

3) When dealing with difficult areas like bass, engineers rely on reference material to make judgments. Ironically, the behavior of mixing to match established reference material has been described by Floyd Toole of NRC and Harman as being part of a "circle of confusion", even though it seems to work out better in practice for music compared to how movies are produced.

In general, I believe a lot of sound passes through the production chain with little to no alteration. That's usually for the best because it means that the recording was high quality in the first place.

Anyway, this discussion may be swaying off topic. I am still posting here because I am legitimately interested in Be. Even though I've cured at least 90% of the break-up problem in my DNA-360, I am left to wonder if it could be better. Of course, I probably still haven't done all I can with the 360 alone. I'm quite certain that my methods have room for improvement.

What I'd really like is if I could get a Be diaphragm for my DNA-360 or even something of that size that plays even lower like the BA-750, but I reckon this is wishful thinking.
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FWIW I thought I remembered seeing throat angle measurements for Radian drivers a few years ago. I'm looking at a post on DiyA that claims the 475 has an exit angle of 5.5°. I would guess that's the half angle. Shouldn't be too difficult to pull the diagram and figure it out I would think?

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